Release 0.11.0

Merge branch 'master' into release/current
DOCS: update client API changes for release
2015-09-23 01:13:22 +02:00 · 2015-09-23 01:12:42 +02:00 · 2015-09-23 01:12:15 +02:00 · 2015-09-23 01:07:52 +02:00 · 2015-09-22 21:54:43 +02:00 · 2015-09-22 11:46:57 +02:00
352 changed files with 14133 additions and 15400 deletions
@@ -47,7 +47,7 @@ notifications:
  email: false
  irc:
    channels:
-      - "irc.freenode.org#mpv-player-dev"
+      - "irc.freenode.org#mpv-devel"
    on_success: change
    on_failure: always

@@ -2,9 +2,15 @@ mpv is a fork of mplayer2, which is a fork of MPlayer.

 mpv as a whole is licensed as GPL version 2 or later (see LICENSE). Most source
 files are GPLv2+, but some files are available under a more liberal license,
-such as LGPLv2+, BSD, MIT, ISC, and possibly others. Look at the copyright
+such as LGPLv2.1+, BSD, MIT, ISC, and possibly others. Look at the copyright
 header of each source file, and grep the sources for "Copyright" if you need
-to know details. Files without Copyright notice are licensed as LGPLv2+.
+to know details. Files without Copyright notice are licensed as LGPLv2.1+.
+
+All new contributions must be LGPLv2.1+ licensed, or if the changes are done on
+GPL code, must come with the implicit agreement that the project can relicense
+the code to LGPLv2.1+ at a later point without asking the contributor. (This
+is a safeguard for making potential relicensing of the project to LGPLv2.1+
+easier.) Using a more liberal license compatible to LGPLv2.1+ is also ok.

 For information about authors and contributors, consult the git log, which
 contains the complete SVN and CVS history as well.
@@ -17,3 +23,5 @@ MPlayer as a whole is licensed under GPLv2 (incompatible to GPLv3!), because
 some files are licensed to GPLv2 (and _not_ any later version of the license).
 In particular, this affects the file libmpdemux/demux_ty_osd.c. It is disabled
 under mplayer2, and has been removed from mpv.
+
+"v2.1+" in this context means "version 2.1 or later".
@@ -4,9 +4,7 @@ Introduction
 This file lists all changes that can cause compatibility issues when using
 mpv through the client API (libmpv and ``client.h``). Since the client API
 interfaces to input handling (commands, properties) as well as command line
-options, this list is interesting for other uses of mpv, such as the Lua
-scripting interface, key bindings in ``input.rst``, or plain command line
-usage.
+options, you should also look at ``interface-changes.rst``.

 Normally, changes to the C API that are incompatible to previous iterations
 receive a major version bump (i.e. the first version number is increased),
@@ -20,21 +18,38 @@ The version number is the same as used for MPV_CLIENT_API_VERSION (see

 Also, read the section ``Compatibility`` in ``client.h``.

+Options, commands, properties
+=============================
+
+Changes to these are not listed here, but in ``interface-changes.rst``. (Before
+client API version 1.17, they were listed here partially.)
+
+This listing includes changes to the bare C API and behavior only, not what
+you can access with them.
+
 API changes
 ===========

 ::

- 1.17   - add MPV_FORMAT_BYTE_ARRAY
- --- mpv 0.9.0 will be released ---
+ --- mpv 0.11.0 ---
+ 1.19   - add "GL_MP_D3D_interfaces" pseudo extension to make it possible to
+          use DXVA2 in OpenGL fullscreen mode in some situations
+        - mpv_request_log_messages() now accepts "terminal-default" as parameter
+ 1.18   - add MPV_END_FILE_REASON_REDIRECT, and change behavior of
+          MPV_EVENT_END_FILE accordingly
+        - a bunch of interface-changes.rst changes
+ 1.17   - mpv_initialize() now blocks SIGPIPE (details see client.h)
+ --- mpv 0.9.0 ---
 1.16   - add mpv_opengl_cb_report_flip()
        - introduce mpv_opengl_cb_draw() and deprecate mpv_opengl_cb_render()
+        - add MPV_FORMAT_BYTE_ARRAY
 1.15   - mpv_initialize() will now load config files. This requires setting
          the "config" and "config-dir" options. In particular, it will load
          mpv.conf.
        - minor backwards-compatible change to the "seek" and "screenshot"
          commands (new flag syntax, old additional args deprecated)
- --- mpv 0.8.0 is released ---
+ --- mpv 0.8.0 ---
 1.14   - add mpv_wait_async_requests()
        - the --msg-level option changes its native type from a flat string to
          a key-value list (setting/reading the option as string still works)
@@ -47,7 +62,7 @@ API changes
          its own and mpv's OpenGL rendering
          Warning: this API is not stable yet - anything in opengl_cb.h might
                   be changed in completely incompatible ways in minor API bumps
- --- mpv 0.7.0 is released ---
+ --- mpv 0.7.0 ---
 1.10   - deprecate/disable everything directly related to script_dispatch
          (most likely affects nobody)
 1.9    - add enum mpv_end_file_reason for mpv_event_end_file.reason
@@ -70,7 +85,7 @@ API changes
 1.6    - modify "core-idle" property behavior
        - MPV_EVENT_LOG_MESSAGE now always sends complete lines
        - introduce numeric log levels (mpv_log_level)
- --- mpv 0.6.0 is released ---
+ --- mpv 0.6.0 ---
 1.5    - change in X11 and "--wid" behavior again. The previous change didn't
          work as expected, and now the behavior can be explicitly controlled
          with the "input-x11-keyboard" option. This is only a temporary
@@ -79,7 +94,7 @@ API changes
          although the old option name still works.
 1.4    - subtle change in X11 and "--wid" behavior
          (this change was added to 0.5.2, and broke some things, see #1090)
- --- mpv 0.5.0 is released ---
+ --- mpv 0.5.0 ---
 1.3    - add MPV_MAKE_VERSION()
 1.2    - remove "stream-time-pos" property (no replacement)
 1.1    - remap dvdnav:// to dvd://
@@ -91,6 +106,6 @@ API changes
          insignificant is now significant
        - add "cache-free" and "cache-used" properties
        - OSX: the "coreaudio" AO spdif code is split into a separate AO
- --- mpv 0.4.0 is released ---
+ --- mpv 0.4.0 ---
 1.0    - the API is declared stable

@@ -24,6 +24,11 @@ qml_direct
 Alternative example, which typically avoids a FBO indirection. Might be
 slightly faster, but is less flexible and harder to use.

+sdl
+---
+
+Show how to embed the mpv OpenGL renderer in SDL.
+
 simple
 ------

@@ -0,0 +1,304 @@
+// Plays a video from the command line in an opengl view in its own window.
+
+// Build with: clang -o cocoa-openglcb cocoa-openglcb.m `pkg-config --libs --cflags mpv` -framework Cocoa -framework OpenGL
+
+#import <mpv/client.h>
+#import <mpv/opengl_cb.h>
+
+#import <stdio.h>
+#import <stdlib.h>
+#import <OpenGL/gl.h>
+
+#import <Cocoa/Cocoa.h>
+
+
+static inline void check_error(int status)
+{
+    if (status < 0) {
+        printf("mpv API error: %s\n", mpv_error_string(status));
+        exit(1);
+    }
+}
+
+static void *get_proc_address(void *ctx, const char *name)
+{
+    CFStringRef symbolName = CFStringCreateWithCString(kCFAllocatorDefault, name, kCFStringEncodingASCII);
+    void *addr = CFBundleGetFunctionPointerForName(CFBundleGetBundleWithIdentifier(CFSTR("com.apple.opengl")), symbolName);
+    CFRelease(symbolName);
+    return addr;
+}
+
+static void glupdate(void *ctx);
+
+@interface MpvClientOGLView : NSOpenGLView
+@property mpv_opengl_cb_context *mpvGL;
+- (instancetype)initWithFrame:(NSRect)frame;
+- (void)drawRect;
+- (void)fillBlack;
+@end
+
+@implementation MpvClientOGLView
+- (instancetype)initWithFrame:(NSRect)frame
+{
+    // make sure the pixel format is double buffered so we can use
+    // [[self openGLContext] flushBuffer].
+    NSOpenGLPixelFormatAttribute attributes[] = {
+        NSOpenGLPFADoubleBuffer,
+        0
+    };
+    self = [super initWithFrame:frame
+                    pixelFormat:[[NSOpenGLPixelFormat alloc] initWithAttributes:attributes]];
+
+    if (self) {
+        [self setAutoresizingMask:NSViewWidthSizable|NSViewHeightSizable];
+        // swap on vsyncs
+        GLint swapInt = 1;
+        [[self openGLContext] setValues:&swapInt forParameter:NSOpenGLCPSwapInterval];
+        [[self openGLContext] makeCurrentContext];
+        self.mpvGL = nil;
+    }
+    return self;
+}
+
+- (void)fillBlack
+{
+    glClearColor(0, 0, 0, 0);
+    glClear(GL_COLOR_BUFFER_BIT);
+}
+
+- (void)drawRect
+{
+    if (self.mpvGL)
+        mpv_opengl_cb_draw(self.mpvGL, 0, self.bounds.size.width, -self.bounds.size.height);
+    else
+        [self fillBlack];
+    [[self openGLContext] flushBuffer];
+}
+
+- (void)drawRect:(NSRect)dirtyRect
+{
+    [self drawRect];
+}
+@end
+
+@interface CocoaWindow : NSWindow
+@property(retain, readonly) MpvClientOGLView *glView;
+@property(retain, readonly) NSButton *pauseButton;
+@end
+
+@implementation CocoaWindow
+- (BOOL)canBecomeMainWindow { return YES; }
+- (BOOL)canBecomeKeyWindow { return YES; }
+- (void)initOGLView {
+    NSRect bounds = [[self contentView] bounds];
+    // window coordinate origin is bottom left
+    NSRect glFrame = NSMakeRect(bounds.origin.x, bounds.origin.y + 30, bounds.size.width, bounds.size.height - 30);
+    _glView = [[MpvClientOGLView alloc] initWithFrame:glFrame];
+    [self.contentView addSubview:_glView];
+
+    NSRect buttonFrame = NSMakeRect(bounds.origin.x, bounds.origin.y, 60, 30);
+    _pauseButton = [[NSButton alloc] initWithFrame:buttonFrame];
+    _pauseButton.buttonType = NSToggleButton;
+    // button target has to be the delegate (it holds the mpv context
+    // pointer), so that's set later.
+    _pauseButton.action = @selector(togglePause:);
+    _pauseButton.title = @"Pause";
+    _pauseButton.alternateTitle = @"Play";
+    [self.contentView addSubview:_pauseButton];
+}
+@end
+
+@interface AppDelegate : NSObject <NSApplicationDelegate>
+{
+    mpv_handle *mpv;
+    dispatch_queue_t queue;
+    CocoaWindow *window;
+}
+@end
+
+static void wakeup(void *);
+
+@implementation AppDelegate
+
+- (void)createWindow {
+
+    int mask = NSTitledWindowMask|NSClosableWindowMask|
+               NSMiniaturizableWindowMask|NSResizableWindowMask;
+
+    window = [[CocoaWindow alloc]
+        initWithContentRect:NSMakeRect(0, 0, 1280, 720)
+                  styleMask:mask
+                    backing:NSBackingStoreBuffered
+                      defer:NO];
+
+    // force a minimum size to stop opengl from exploding.
+    [window setMinSize:NSMakeSize(200, 200)];
+    [window initOGLView];
+    [window setTitle:@"cocoa-openglcb example"];
+    [window makeMainWindow];
+    [window makeKeyAndOrderFront:nil];
+
+    NSMenu *m = [[NSMenu alloc] initWithTitle:@"AMainMenu"];
+    NSMenuItem *item = [m addItemWithTitle:@"Apple" action:nil keyEquivalent:@""];
+    NSMenu *sm = [[NSMenu alloc] initWithTitle:@"Apple"];
+    [m setSubmenu:sm forItem:item];
+    [sm addItemWithTitle: @"quit" action:@selector(terminate:) keyEquivalent:@"q"];
+    [NSApp setMenu:m];
+    [NSApp activateIgnoringOtherApps:YES];
+}
+
+- (void) applicationDidFinishLaunching:(NSNotification *)notification {
+    [NSApp setActivationPolicy:NSApplicationActivationPolicyRegular];
+    atexit_b(^{
+        // Because activation policy has just been set to behave like a real
+        // application, that policy must be reset on exit to prevent, among
+        // other things, the menubar created here from remaining on screen.
+        [NSApp setActivationPolicy:NSApplicationActivationPolicyProhibited];
+    });
+
+    // Read filename
+    NSArray *args = [NSProcessInfo processInfo].arguments;
+    if (args.count < 2) {
+        NSLog(@"Expected filename on command line");
+        exit(1);
+    }
+    NSString *filename = args[1];
+
+    [self createWindow];
+    window.pauseButton.target = self;
+
+    mpv = mpv_create();
+    if (!mpv) {
+        printf("failed creating context\n");
+        exit(1);
+    }
+
+    check_error(mpv_set_option_string(mpv, "input-media-keys", "yes"));
+    // request important errors
+    check_error(mpv_request_log_messages(mpv, "warn"));
+
+    check_error(mpv_initialize(mpv));
+    check_error(mpv_set_option_string(mpv, "vo", "opengl-cb"));
+    mpv_opengl_cb_context *mpvGL = mpv_get_sub_api(mpv, MPV_SUB_API_OPENGL_CB);
+    if (!mpvGL) {
+        puts("libmpv does not have the opengl-cb sub-API.");
+        exit(1);
+    }
+    // pass the mpvGL context to our view
+    window.glView.mpvGL = mpvGL;
+    int r = mpv_opengl_cb_init_gl(mpvGL, NULL, get_proc_address, NULL);
+    if (r < 0) {
+        puts("gl init has failed.");
+        exit(1);
+    }
+    mpv_opengl_cb_set_update_callback(mpvGL, glupdate, (__bridge void *)window.glView);
+
+    // Deal with MPV in the background.
+    queue = dispatch_queue_create("mpv", DISPATCH_QUEUE_SERIAL);
+    dispatch_async(queue, ^{
+        // Register to be woken up whenever mpv generates new events.
+        mpv_set_wakeup_callback(mpv, wakeup, (__bridge void *)self);
+        // Load the indicated file
+        const char *cmd[] = {"loadfile", filename.UTF8String, NULL};
+        check_error(mpv_command(mpv, cmd));
+    });
+}
+
+static void glupdate(void *ctx)
+{
+    MpvClientOGLView *glView = (__bridge MpvClientOGLView *)ctx;
+    // I'm still not sure what the best way to handle this is, but this
+    // works.
+    dispatch_async(dispatch_get_main_queue(), ^{
+        [glView drawRect];
+    });
+}
+
+- (void) handleEvent:(mpv_event *)event
+{
+    switch (event->event_id) {
+    case MPV_EVENT_SHUTDOWN: {
+        mpv_detach_destroy(mpv);
+        mpv_opengl_cb_uninit_gl(window.glView.mpvGL);
+        mpv = NULL;
+        printf("event: shutdown\n");
+        break;
+    }
+
+    case MPV_EVENT_LOG_MESSAGE: {
+        struct mpv_event_log_message *msg = (struct mpv_event_log_message *)event->data;
+        printf("[%s] %s: %s", msg->prefix, msg->level, msg->text);
+    }
+
+    default:
+        printf("event: %s\n", mpv_event_name(event->event_id));
+    }
+}
+
+- (void)togglePause:(NSButton *)button
+{
+    if (mpv) {
+        switch (button.state) {
+            case NSOffState:
+                {
+                    int pause = 0;
+                    mpv_set_property(mpv, "pause", MPV_FORMAT_FLAG, &pause);
+                }
+                break;
+            case NSOnState:
+                {
+                    int pause = 1;
+                    mpv_set_property(mpv, "pause", MPV_FORMAT_FLAG, &pause);
+                }
+                break;
+            default:
+                NSLog(@"This should never happen.");
+        }
+    }
+}
+
+- (void) readEvents
+{
+    dispatch_async(queue, ^{
+        while (mpv) {
+            mpv_event *event = mpv_wait_event(mpv, 0);
+            if (event->event_id == MPV_EVENT_NONE)
+                break;
+            [self handleEvent:event];
+        }
+    });
+}
+
+static void wakeup(void *context)
+{
+    AppDelegate *a = (__bridge AppDelegate *) context;
+    [a readEvents];
+}
+
+// quit when the window is closed.
+- (BOOL)applicationShouldTerminateAfterLastWindowClosed:(NSApplication *)app
+{
+    return YES;
+}
+
+- (NSApplicationTerminateReply)applicationShouldTerminate:(NSApplication *)sender
+{
+    NSLog(@"Terminating.");
+    const char *args[] = {"quit", NULL};
+    mpv_command(mpv, args);
+    [window.glView clearGLContext];
+    return NSTerminateNow;
+}
+
+@end
+
+// Delete this if you already have a main.m.
+int main(int argc, const char * argv[]) {
+    @autoreleasepool {
+        NSApplication *app = [NSApplication sharedApplication];
+        AppDelegate *delegate = [AppDelegate new];
+        app.delegate = delegate;
+        [app run];
+    }
+    return 0;
+}
@@ -1,9 +1,7 @@
-// Plays a video from the command line in a window provided by mpv.
-// You likely want to play the video in your own window instead,
-// but that's not quite ready yet.
-// You may need a basic Info.plist and MainMenu.xib to make this work.
+// Plays a video from the command line in a view provided by the client
+// application.

-// Build with: clang -o cocoabasic cocoabasic.m `pkg-config --libs --cflags mpv`
+// Build with: clang -o cocoabasic cocoabasic.m `pkg-config --libs --cflags mpv` -framework cocoa

 #include <mpv/client.h>

@@ -207,5 +205,5 @@ int main(int argc, const char * argv[]) {
        app.delegate = delegate;
        [app run];
    }
-    return EXIT_SUCCESS;
+    return 0;
 }
@@ -3,6 +3,7 @@ QT += qml quick
 HEADERS += main.h
 SOURCES += main.cpp

+QT_CONFIG -= no-pkg-config
 CONFIG += link_pkgconfig debug
 PKGCONFIG += mpv

@@ -3,6 +3,7 @@ QT += qml quick
 HEADERS += main.h
 SOURCES += main.cpp

+QT_CONFIG -= no-pkg-config
 CONFIG += link_pkgconfig debug
 PKGCONFIG += mpv

@@ -5,6 +5,7 @@ greaterThan(QT_MAJOR_VERSION, 4): QT += widgets
 TARGET = qtexample
 TEMPLATE = app

+QT_CONFIG -= no-pkg-config
 CONFIG += link_pkgconfig debug
 PKGCONFIG += mpv

@@ -0,0 +1,158 @@
+// Build with: gcc -o main main.c `pkg-config --libs --cflags mpv sdl2`
+
+#include <stddef.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <SDL.h>
+
+#include <mpv/client.h>
+#include <mpv/opengl_cb.h>
+
+static Uint32 wakeup_on_mpv_redraw, wakeup_on_mpv_events;
+
+static void die(const char *msg)
+{
+    fprintf(stderr, "%s\n", msg);
+    exit(1);
+}
+
+static void *get_proc_address_mpv(void *fn_ctx, const char *name)
+{
+    return SDL_GL_GetProcAddress(name);
+}
+
+static void on_mpv_events(void *ctx)
+{
+    SDL_Event event = {.type = wakeup_on_mpv_events};
+    SDL_PushEvent(&event);
+}
+
+static void on_mpv_redraw(void *ctx)
+{
+    SDL_Event event = {.type = wakeup_on_mpv_redraw};
+    SDL_PushEvent(&event);
+}
+
+int main(int argc, char *argv[])
+{
+    if (argc != 2)
+        die("pass a single media file as argument");
+
+    mpv_handle *mpv = mpv_create();
+    if (!mpv)
+        die("context init failed");
+
+    // Some minor options can only be set before mpv_initialize().
+    if (mpv_initialize(mpv) < 0)
+        die("mpv init failed");
+
+    if (SDL_Init(SDL_INIT_VIDEO) < 0)
+        die("SDL init failed");
+
+    SDL_Window *window =
+        SDL_CreateWindow("hi", SDL_WINDOWPOS_CENTERED, SDL_WINDOWPOS_CENTERED,
+                         1000, 500, SDL_WINDOW_OPENGL | SDL_WINDOW_SHOWN |
+                                    SDL_WINDOW_RESIZABLE);
+    if (!window)
+        die("failed to create SDL window");
+
+    // The OpenGL API is somewhat separate from the normal mpv API. This only
+    // returns NULL if no OpenGL support is compiled.
+    mpv_opengl_cb_context *mpv_gl = mpv_get_sub_api(mpv, MPV_SUB_API_OPENGL_CB);
+    if (!mpv_gl)
+        die("failed to create mpv GL API handle");
+
+    SDL_GLContext glcontext = SDL_GL_CreateContext(window);
+    if (!glcontext)
+        die("failed to create SDL GL context");
+
+    // This makes mpv use the currently set GL context. It will use the callback
+    // to resolve GL builtin functions, as well as extensions.
+    if (mpv_opengl_cb_init_gl(mpv_gl, NULL, get_proc_address_mpv, NULL) < 0)
+        die("failed to initialize mpv GL context");
+
+    // Actually using the opengl_cb state has to be explicitly requested.
+    // Otherwise, mpv will create a separate platform window.
+    if (mpv_set_option_string(mpv, "vo", "opengl-cb") < 0)
+        die("failed to set VO");
+
+    // We use events for thread-safe notification of the SDL main loop.
+    // Generally, the wakeup callbacks (set further below) should do as least
+    // work as possible, and merely wake up another thread to do actual work.
+    // On SDL, waking up the mainloop is the ideal course of action. SDL's
+    // SDL_PushEvent() is thread-safe, so we use that.
+    wakeup_on_mpv_redraw = SDL_RegisterEvents(1);
+    wakeup_on_mpv_events = SDL_RegisterEvents(1);
+    if (wakeup_on_mpv_redraw == (Uint32)-1 || wakeup_on_mpv_events == (Uint32)-1)
+        die("could not register events");
+
+    // When normal mpv events are available.
+    mpv_set_wakeup_callback(mpv, on_mpv_events, NULL);
+
+    // When a new frame should be drawn with mpv_opengl_cb_draw().
+    // (Separate from the normal event handling mechanism for the sake of
+    //  users which run OpenGL on a different thread.)
+    mpv_opengl_cb_set_update_callback(mpv_gl, on_mpv_redraw, NULL);
+
+    // Play this file. Note that this starts playback asynchronously.
+    const char *cmd[] = {"loadfile", argv[1], NULL};
+    mpv_command(mpv, cmd);
+
+    while (1) {
+        SDL_Event event;
+        if (SDL_WaitEvent(&event) != 1)
+            die("event loop error");
+        int redraw = 0;
+        switch (event.type) {
+        case SDL_QUIT:
+            goto done;
+        case SDL_WINDOWEVENT:
+            if (event.window.event == SDL_WINDOWEVENT_EXPOSED)
+                redraw = 1;
+            break;
+        case SDL_KEYDOWN:
+            if (event.key.keysym.sym == SDLK_SPACE)
+                mpv_command_string(mpv, "cycle pause");
+            break;
+        default:
+            // Happens when a new video frame should be rendered, or if the
+            // current frame has to be redrawn e.g. due to OSD changes.
+            if (event.type == wakeup_on_mpv_redraw)
+                redraw = 1;
+            // Happens when at least 1 new event is in the mpv event queue.
+            if (event.type == wakeup_on_mpv_events) {
+                // Handle all remaining mpv events.
+                while (1) {
+                    mpv_event *mp_event = mpv_wait_event(mpv, 0);
+                    if (mp_event->event_id == MPV_EVENT_NONE)
+                        break;
+                    printf("event: %s\n", mpv_event_name(mp_event->event_id));
+                }
+            }
+        }
+        if (redraw) {
+            int w, h;
+            SDL_GetWindowSize(window, &w, &h);
+            // Note:
+            // - The 0 is the FBO to use; 0 is the default framebuffer (i.e.
+            //   render to the window directly.
+            // - The negative height tells mpv to flip the coordinate system.
+            // - If you do not want the video to cover the whole screen, or want
+            //   to apply any form of fancy transformation, you will have to
+            //   render to a FBO.
+            // - See opengl_cb.h on what OpenGL environment mpv expects, and
+            //   other API details.
+            mpv_opengl_cb_draw(mpv_gl, 0, w, -h);
+            SDL_GL_SwapWindow(window);
+        }
+    }
+done:
+
+    // Destroy the GL renderer and all of the GL objects it allocated. If video
+    // is still running, the video track will be deselected.
+    mpv_opengl_cb_uninit_gl(mpv_gl);
+
+    mpv_terminate_destroy(mpv);
+    return 0;
+}
@@ -134,10 +134,10 @@ pacman -S git pkg-config python3 mingw-w64-x86_64-gcc

 # Install the most important MinGW-w64 dependencies. libass, libbluray and
 # lcms2 are also pulled in as dependencies of ffmpeg.
-pacman -S mingw-w64-x86_64-ffmpeg mingw-w64-x86_64-libjpeg-turbo mingw-w64-x86_64-lua
+pacman -S mingw-w64-x86_64-ffmpeg mingw-w64-x86_64-libjpeg-turbo mingw-w64-x86_64-lua51

 # Install additional (optional) dependencies
-pacman -S mingw-w64-x86_64-libdvdnav mingw-w64-x86_64-mpg123 mingw-w64-x86_64-libguess
+pacman -S mingw-w64-x86_64-libdvdnav mingw-w64-x86_64-libguess
 ```

 For a 32-bit build, install ``mingw-w64-i686-*`` packages instead.
@@ -4,8 +4,8 @@ How to contribute
 General
 -------

-The main contact for mpv development is IRC, specifically #mpv-player
-and #mpv-player-dev on Freenode.
+The main contact for mpv development is IRC, specifically #mpv
+and #mpv-devel on Freenode.

 Sending patches
 ---------------
@@ -54,6 +54,9 @@ Sending patches
  change in the same commit.
 - If you add a new command line option, document it in options.rst. If you
  add a new input property, document it in input.rst.
+- All new code must be LGPLv2.1+ licensed, or come with the implicit agreement
+  that it will be relicensed to LGPLv2.1+ later (see ``Copyright`` in the
+  repository root directory).

 Code formatting
 ---------------
@@ -0,0 +1,90 @@
+Introduction
+============
+
+mpv provides access to its internal via the following means:
+
+- options
+- commands
+- properties
+- events
+
+All of these are important for interfacing both with end users and API users
+(which include Lua scripts, libmpv, and the JSON IPC). As such, they constitute
+a large part of the user interface and APIs.
+
+This document lists changes to them. New changes are added to the top.
+
+Interface changes
+=================
+
+::
+
+ --- mpv 0.11.0 ---
+    - add "af-metadata" property
+ --- mpv 0.10.0 ---
+    - add --video-aspect-method option
+    - add --playlist-pos option
+    - add --video-sync* options
+      "display-sync-active" property
+      "vo-missed-frame-count" property
+      "audio-speed-correction" and "video-speed-correction" properties
+    - remove --demuxer-readahead-packets and --demuxer-readahead-bytes
+      add --demuxer-max-packets and --demuxer-max-bytes
+      (the new options are not replacement and have very different semantics)
+    - change "video-aspect" property: always settable, even if no video is
+      running; always return the override - if no override is set, return
+      the video's aspect ratio
+    - remove disc-nav (DVD, BD) related properties and commands
+    - add "option-info/<name>/set-locally" property
+    - add --cache-backbuffer; change --cache-default default to 75MB
+      the new total cache size is the sum of backbuffer and the cache size
+      specified by --cache-default or --cache
+    - add ``track-list/N/audio-channels`` property
+    - change --screenshot-tag-colorspace default value
+    - add --stretch-image-subs-to-screen
+    - add "playlist/N/title" property
+    - add --video-stereo-mode=no to disable auto-conversions
+    - add --force-seekable, and change default seekability in some cases
+    - add vf yadif/vavpp/vdpaupp interlaced-only suboptions
+      Also, the option is enabled by default (Except vf_yadif, which has
+      it enabled only if it's inserted by the deinterlace property.)
+    - add --hwdec-preload
+    - add ao coreaudio exclusive suboption
+    - add ``track-list/N/forced`` property
+    - add audio-params/channel-count and ``audio-params-out/channel-count props.
+    - add af volume replaygain-fallback suboption
+    - add video-params/stereo-in property
+    - add "keypress", "keydown", and "keyup" commands
+    - deprecate --ad-spdif-dtshd and enabling passthrough via --ad
+      add --audio-spdif as replacement
+    - remove "get_property" command
+    - remove --slave-broken
+    - add vo opengl custom shader suboptions (source-shader, scale-shader,
+      pre-shaders, post-shaders)
+    - completely change how the hwdec properties work:
+        - "hwdec" now reflects the --hwdec option
+        - "hwdec-detected" does partially what the old "hwdec" property did
+          (and also, "detected-hwdec" is removed)
+        - "hwdec-active" is added
+    - add protocol-list property
+    - deprecate audio-samplerate and audio-channels properties
+      (audio-params sub-properties are the replacement)
+    - add audio-params and audio-out-params properties
+    - deprecate "audio-format" property, replaced with "audio-codec-name"
+    - deprecate --media-title, replaced with --force-media-title
+    - deprecate "length" property, replaced with "duration"
+    - change volume property:
+        - the value 100 is now always "unchanged volume" - with softvol, the
+          range is 0 to --softvol-max, without it is 0-100
+        - the minimum value of --softvol-max is raised to 100
+    - remove vo opengl npot suboption
+    - add relative seeking by percentage to "seek" command
+    - add playlist_shuffle command
+    - add --force-window=immediate
+    - add ao coreaudio change-physical-format suboption
+    - remove vo opengl icc-cache suboption, add icc-cache-dir suboption
+    - add --screenshot-directory
+    - add --screenshot-high-bit-depth
+    - add --screenshot-jpeg-source-chroma
+    - default action for "rescan_external_files" command changes
+ --- mpv 0.9.0 ---
@@ -11,6 +11,10 @@ syntax is:

    To get a full list of available audio filters, see ``--af=help``.

+    Also, keep in mind that most actual filters are available via the ``lavfi``
+    wrapper, which gives you access to most of libavfilter's filters. This
+    includes all filters that have been ported from MPlayer to libavfilter.
+
 You can also set defaults for each filter. The defaults are applied before the
 normal filter parameters.

@@ -59,6 +63,13 @@ Available filters are:
        (If you just want to set defaults for this filter that will be used
        even by automatically inserted lavrresample instances, you should
        prefer setting them with ``--af-defaults=lavrresample:...``.)
+    ``normalize=<yes|no>``
+        Whether to normalize when remixing channel layouts (default: yes). This
+        is e.g. applied when downmixing surround audio to stereo. The advantage
+        is that this guarantees that no clipping can happen. Unfortunately,
+        this can also lead to too low volume levels. Whether you enable or
+        disable this is essentially a matter of taste, but the default uses
+        the safer choice.
    ``o=<string>``
        Set AVOptions on the SwrContext or AVAudioResampleContext. These should
        be documented by FFmpeg or Libav.
@@ -96,59 +107,6 @@ Available filters are:
        If the input channel number is less than ``<minch>``, the filter will
        detach itself (default: 3).

-``sweep[=speed]``
-    Produces a sine sweep.
-
-    ``<0.0-1.0>``
-        Sine function delta, use very low values to hear the sweep.
-
-``sinesuppress[=freq:decay]``
-    Remove a sine at the specified frequency. Useful to get rid of the 50/60 Hz
-    noise on low quality audio equipment. It only works on mono input.
-
-    ``<freq>``
-        The frequency of the sine which should be removed (in Hz) (default:
-        50)
-    ``<decay>``
-        Controls the adaptivity (a larger value will make the filter adapt to
-        amplitude and phase changes quicker, a smaller value will make the
-        adaptation slower) (default: 0.0001). Reasonable values are around
-        0.001.
-
-``bs2b[=option1:option2:...]``
-    Bauer stereophonic to binaural transformation using libbs2b. Improves the
-    headphone listening experience by making the sound similar to that from
-    loudspeakers, allowing each ear to hear both channels and taking into
-    account the distance difference and the head shadowing effect. It is
-    applicable only to 2-channel audio.
-
-    ``fcut=<300-1000>``
-        Set cut frequency in Hz.
-    ``feed=<10-150>``
-        Set feed level for low frequencies in 0.1*dB.
-    ``profile=<value>``
-        Several profiles are available for convenience:
-
-        :default: will be used if nothing else was specified (fcut=700,
-                  feed=45)
-        :cmoy:    Chu Moy circuit implementation (fcut=700, feed=60)
-        :jmeier:  Jan Meier circuit implementation (fcut=650, feed=95)
-
-    If ``fcut`` or ``feed`` options are specified together with a profile, they
-    will be applied on top of the selected profile.
-
-``hrtf[=flag]``
-    Head-related transfer function: Converts multichannel audio to 2-channel
-    output for headphones, preserving the spatiality of the sound.
-
-    ==== ===================================
-    Flag Meaning
-    ==== ===================================
-    m    matrix decoding of the rear channel
-    s    2-channel matrix decoding
-    0    no matrix decoding (default)
-    ==== ===================================
-
 ``equalizer=g1:g2:g3:...:g10``
    10 octave band graphic equalizer, implemented using 10 IIR band-pass
    filters. This means that it works regardless of what type of audio is
@@ -263,13 +221,6 @@ Available filters are:
    used to do conversion itself, unlike this one which lets the filter system
    handle the conversion.

-``convert24``
-    Filter for internal use only. Converts between 24-bit and 32-bit sample
-    formats.
-
-``convertsign``
-    Filter for internal use only. Converts between signed/unsigned formats.
-
 ``volume[=<volumedb>[:...]]``
    Implements software volume control. Use this filter with caution since it
    can reduce the signal to noise ratio of the sound. In most cases it is
@@ -294,6 +245,10 @@ Available filters are:
    ``replaygain-clip=yes|no``
        Prevent clipping caused by replaygain by automatically lowering the
        gain (default). Use ``replaygain-clip=no`` to disable this.
+    ``replaygain-fallback``
+        Gain in dB to apply if the file has no replay gain tags. This option
+        is always applied if the replaygain logic is somehow inactive. If this
+        is applied, no other replaygain options are applied.
    ``softclip``
        Turns soft clipping on. Soft-clipping can make the
        sound more smooth if very high volume levels are used. Enable this
@@ -357,64 +312,6 @@ Available filters are:
        ``mpv '--af=format=channels=5.1' '--audio-channels=5.1'`` would always force
        remixing audio to 5.1 and output it like this.

-``sub[=fc:ch]``
-    Adds a subwoofer channel to the audio stream. The audio data used for
-    creating the subwoofer channel is an average of the sound in channel 0 and
-    channel 1. The resulting sound is then low-pass filtered by a 4th order
-    Butterworth filter with a default cutoff frequency of 60Hz and added to a
-    separate channel in the audio stream.
-
-    .. warning::
-
-        Disable this filter when you are playing media with an LFE channel
-        (e.g. 5.1 surround sound), otherwise this filter will disrupt the sound
-        to the subwoofer.
-
-    ``<fc>``
-        cutoff frequency in Hz for the low-pass filter (20 Hz to 300 Hz)
-        (default: 60 Hz) For the best result try setting the cutoff frequency
-        as low as possible. This will improve the stereo or surround sound
-        experience.
-    ``<ch>``
-        Determines the channel number in which to insert the sub-channel
-        audio. Channel number can be between 0 and 7 (default: 5). Observe
-        that the number of channels will automatically be increased to <ch> if
-        necessary.
-
-    .. admonition:: Example
-
-        ``mpv --af=sub=100:4 --audio-channels=5 media.avi``
-            Would add a subwoofer channel with a cutoff frequency of 100 Hz to
-            output channel 4.
-
-``center``
-    Creates a center channel from the front channels. May currently be low
-    quality as it does not implement a high-pass filter for proper extraction
-    yet, but averages and halves the channels instead.
-
-    ``<ch>``
-        Determines the channel number in which to insert the center channel.
-        Channel number can be between 0 and 7 (default: 5). Observe that the
-        number of channels will automatically be increased to ``<ch>`` if
-        necessary.
-
-``surround[=delay]``
-    Decoder for matrix encoded surround sound like Dolby Surround. Some files
-    with 2-channel audio actually contain matrix encoded surround sound.
-
-    ``<delay>``
-        delay time in ms for the rear speakers (0 to 1000) (default: 20) This
-        delay should be set as follows: If d1 is the distance from the
-        listening position to the front speakers and d2 is the distance from
-        the listening position to the rear speakers, then the delay should be
-        set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2.
-
-    .. admonition:: Example
-
-        ``mpv --af=surround=15 --audio-channels=4 media.avi``
-            Would add surround sound decoding with 15 ms delay for the sound to
-            the rear speakers.
-
 ``delay[=[ch1,ch2,...]]``
    Delays the sound to the loudspeakers such that the sound from the
    different channels arrives at the listening position simultaneously. It is
@@ -443,36 +340,6 @@ Available filters are:
            Would delay front left and right by 10.5 ms, the two rear channels
            and the subwoofer by 0 ms and the center channel by 7 ms.

-``export=mmapped_file:nsamples]``
-    Exports the incoming signal to other processes using memory mapping
-    (``mmap()``). Memory mapped areas contain a header::
-
-        int nch                      /* number of channels */
-        int size                     /* buffer size */
-        unsigned long long counter   /* Used to keep sync, updated every time
-                                        new data is exported. */
-
-    The rest is payload (non-interleaved) 16-bit data.
-
-    ``<mmapped_file>``
-        File to map data to (required)
-    ``<nsamples>``
-        number of samples per channel (default: 512).
-
-    .. admonition:: Example
-
-        ``mpv --af=export=/tmp/mpv-af_export:1024 media.avi``
-            Would export 1024 samples per channel to ``/tmp/mpv-af_export``.
-
-``extrastereo[=mul]``
-    (Linearly) increases the difference between left and right channels which
-    adds some sort of "live" effect to playback.
-
-    ``<mul>``
-        Sets the difference coefficient (default: 2.5). 0.0 means mono sound
-        (average of both channels), with 1.0 sound will be unchanged, with
-        -1.0 left and right channels will be swapped.
-
 ``drc[=method:target]``
    Applies dynamic range compression. This maximizes the volume by compressing
    the audio signal's dynamic range. (Formerly called ``volnorm``.)
@@ -496,45 +363,6 @@ Available filters are:
        This filter can cause distortion with audio signals that have a very
        large dynamic range.

-``ladspa=file:label:[<control0>,<control1>,...]``
-    Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. This
-    filter is reentrant, so multiple LADSPA plugins can be used at once.
-
-    ``<file>``
-        Specifies the LADSPA plugin library file.
-
-        .. note::
-
-            See also the note about the ``LADSPA_PATH`` variable in the
-            `ENVIRONMENT VARIABLES`_ section.
-    ``<label>``
-        Specifies the filter within the library. Some libraries contain only
-        one filter, but others contain many of them. Entering 'help' here
-        will list all available filters within the specified library, which
-        eliminates the use of 'listplugins' from the LADSPA SDK.
-    ``[<control0>,<control1>,...]``
-        Controls are zero or more ``,`` separated floating point values that
-        determine the behavior of the loaded plugin (for example delay,
-        threshold or gain).
-        In verbose mode (add ``-v`` to the mpv command line), all
-        available controls and their valid ranges are printed. This eliminates
-        the use of 'analyseplugin' from the LADSPA SDK.
-        Note that ``,`` is already used by the option parser to separate
-        filters, so you must quote the list of values with ``[...]`` or
-        similar.
-
-    .. admonition:: Example
-
-        ``mpv --af=ladspa='/usr/lib/ladspa/delay.so':delay_5s:[0.5,0.2] media.avi``
-            Does something.
-
-``karaoke``
-    Simple voice removal filter exploiting the fact that voice is usually
-    recorded with mono gear and later 'center' mixed onto the final audio
-    stream. Beware that this filter will turn your signal into mono. Works
-    well for 2 channel tracks; do not bother trying it on anything but 2
-    channel stereo.
-
 ``scaletempo[=option1:option2:...]``
    Scales audio tempo without altering pitch, optionally synced to playback
    speed (default).
@@ -55,7 +55,7 @@ Available audio output drivers are:
        Allow output of non-interleaved formats (if the audio decoder uses
        this format). Currently disabled by default, because some popular
        ALSA plugins are utterly broken with non-interleaved formats.
-    ``ingore-chmap``
+    ``ignore-chmap``
        Don't read or set the channel map of the ALSA device - only request the
        required number of channels, and then pass the audio as-is to it. This
        option most likely should not be used. It can be useful for debugging,
@@ -162,12 +162,23 @@ Available audio output drivers are:
    Automatically redirects to ``coreaudio_exclusive`` when playing compressed
    formats.

+    ``change-physical-format=<yes|no>``
+        Change the physical format to one similar to the requested audio format
+        (default: no). This has the advantage that multichannel audio output
+        will actually work. The disadvantage is that it will change the
+        system-wide audio settings. This is equivalent to changing the ``Format``
+        setting in the ``Audio Devices`` dialog in the ``Audio MIDI Setup``
+        utility. Note that this does not effect the selected speaker setup.
+
+    ``exclusive``
+        Use exclusive mode access. This merely redirects to
+        ``coreaudio_exclusive``, but should be preferred over using that AO
+        directly.
+
 ``coreaudio_exclusive`` (Mac OS X only)
    Native Mac OS X audio output driver using direct device access and
    exclusive mode (bypasses the sound server).

-    Supports only compressed formats (AC3 and DTS).
-
 ``openal``
    Experimental OpenAL audio output driver

@@ -260,6 +271,10 @@ Available audio output drivers are:
    ``broken-delay``
        Simulate broken audio drivers, which don't report latency correctly.

+    ``channel-layouts``
+        If not empty, this is a ``,`` separated list of channel layouts the
+        AO allows. This can be used to test channel layout selection.
+
 ``pcm``
    Raw PCM/WAVE file writer audio output

@@ -111,7 +111,7 @@ You can encode files from one format/codec to another using this facility.

    .. admonition:: Examples

-        ``"--ovc=mpeg4 --oacopts=qscale=5"``
+        ``"--ovc=mpeg4 --ovcopts=qscale=5"``
            selects constant quantizer scale 5 for MPEG-4 encoding.

        ``"--ovc=libx264 --ovcopts=crf=23"``
@@ -65,11 +65,11 @@ C-style escaping can be used.

 You can bind multiple commands to one key. For example:

-| a show_text "command 1" ; show_text "command 2"
+| a show-text "command 1" ; show-text "command 2"

 It's also possible to bind a command to a sequence of keys:

-| a-b-c show_text "command run after a, b, c have been pressed"
+| a-b-c show-text "command run after a, b, c have been pressed"

 (This is not shown in the general command syntax.)

@@ -86,7 +86,7 @@ List of Input Commands
    disabling default bindings, without disabling all bindings with
    ``--no-input-default-bindings``.

-``seek <seconds> [relative|absolute|absolute-percent|exact|keyframes]``
+``seek <seconds> [relative|absolute|absolute-percent|relative-percent|exact|keyframes]``
    Change the playback position. By default, seeks by a relative amount of
    seconds.

@@ -98,6 +98,8 @@ List of Input Commands
        Seek to a given time.
    absolute-percent
        Seek to a given percent position.
+    relative-percent
+        Seek relative to current position in percent.
    keyframes
        Always restart playback at keyframe boundaries (fast).
    exact
@@ -112,25 +114,25 @@ List of Input Commands
    3rd parameter (essentially using a space instead of ``+``). The 3rd
    parameter is still parsed, but is considered deprecated.

-``revert_seek [mode]``
+``revert-seek [mode]``
    Undoes the ``seek`` command, and some other commands that seek (but not
    necessarily all of them). Calling this command once will jump to the
    playback position before the seek. Calling it a second time undoes the
-    ``revert_seek`` command itself. This only works within a single file.
+    ``revert-seek`` command itself. This only works within a single file.

    The first argument is optional, and can change the behavior:

    mark
-        Mark the current time position. The next normal ``revert_seek`` command
+        Mark the current time position. The next normal ``revert-seek`` command
        will seek back to this point, no matter how many seeks happened since
        last time.

    Using it without any arguments gives you the default behavior.

-``frame_step``
+``frame-step``
    Play one frame, then pause. Does nothing with audio-only playback.

-``frame_back_step``
+``frame-back-step``
    Go back by one frame, then pause. Note that this can be very slow (it tries
    to be precise, not fast), and sometimes fails to behave as expected. How
    well this works depends on whether precise seeking works correctly (e.g.
@@ -180,7 +182,7 @@ List of Input Commands
        frame was dropped. This flag can be combined with the other flags,
        e.g. ``video+each-frame``.

-``screenshot_to_file "<filename>" [subtitles|video|window]``
+``screenshot-to-file "<filename>" [subtitles|video|window]``
    Take a screenshot and save it to a given file. The format of the file will
    be guessed by the extension (and ``--screenshot-format`` is ignored - the
    behavior when the extension is missing or unknown is arbitrary).
@@ -192,7 +194,7 @@ List of Input Commands
    Like all input command parameters, the filename is subject to property
    expansion as described in `Property Expansion`_.

-``playlist_next [weak|force]``
+``playlist-next [weak|force]``
    Go to the next entry on the playlist.

    weak (default)
@@ -200,7 +202,7 @@ List of Input Commands
    force
        Terminate playback if there are no more files on the playlist.

-``playlist_prev [weak|force]``
+``playlist-prev [weak|force]``
    Go to the previous entry on the playlist.

    weak (default)
@@ -230,22 +232,26 @@ List of Input Commands
 ``loadlist "<playlist>" [replace|append]``
    Load the given playlist file (like ``--playlist``).

-``playlist_clear``
+``playlist-clear``
    Clear the playlist, except the currently played file.

-``playlist_remove current|<index>``
+``playlist-remove current|<index>``
    Remove the playlist entry at the given index. Index values start counting
    with 0. The special value ``current`` removes the current entry. Note that
    removing the current entry also stops playback and starts playing the next
    entry.

-``playlist_move <index1> <index2>``
+``playlist-move <index1> <index2>``
    Move the playlist entry at index1, so that it takes the place of the
    entry index2. (Paradoxically, the moved playlist entry will not have
    the index value index2 after moving if index1 was lower than index2,
    because index2 refers to the target entry, not the index the entry
    will have after moving.)

+``playlist-shuffle``
+    Shuffle the playlist. This is similar to what is done on start if the
+    ``--shuffle`` option is used.
+
 ``run "command" "arg1" "arg2" ...``
    Run the given command. Unlike in MPlayer/mplayer2 and earlier versions of
    mpv (0.2.x and older), this doesn't call the shell. Instead, the command
@@ -271,12 +277,12 @@ List of Input Commands
 ``quit [<code>]``
    Exit the player. If an argument is given, it's used as process exit code.

-``quit_watch_later [<code>]``
+``quit-watch-later [<code>]``
    Exit player, and store current playback position. Playing that file later
    will seek to the previous position on start. The (optional) argument is
    exactly as in the ``quit`` command.

-``sub_add "<file>" [<flags> [<title> [<lang>]]]``
+``sub-add "<file>" [<flags> [<title> [<lang>]]]``
    Load the given subtitle file. It is selected as current subtitle after
    loading.

@@ -303,27 +309,27 @@ List of Input Commands
    The ``lang`` argument sets the track language, and can also influence
    stream selection with ``flags`` set to ``auto``.

-``sub_remove [<id>]``
+``sub-remove [<id>]``
    Remove the given subtitle track. If the ``id`` argument is missing, remove
    the current track. (Works on external subtitle files only.)

-``sub_reload [<id>]``
+``sub-reload [<id>]``
    Reload the given subtitle tracks. If the ``id`` argument is missing, reload
    the current track. (Works on external subtitle files only.)

    This works by unloading and re-adding the subtitle track.

-``sub_step <skip>``
+``sub-step <skip>``
    Change subtitle timing such, that the subtitle event after the next
    ``<skip>`` subtitle events is displayed. ``<skip>`` can be negative to step
    backwards.

-``sub_seek <skip>``
+``sub-seek <skip>``
    Seek to the next (skip set to 1) or the previous (skip set to -1) subtitle.
-    This is similar to ``sub_step``, except that it seeks video and audio
+    This is similar to ``sub-step``, except that it seeks video and audio
    instead of adjusting the subtitle delay.

-    Like with ``sub_step``, this works with external text subtitles only. For
+    Like with ``sub-step``, this works with external text subtitles only. For
    embedded text subtitles (like with Matroska), this works only with subtitle
    events that have already been displayed.

@@ -331,11 +337,11 @@ List of Input Commands
    Toggle OSD level. If ``<level>`` is specified, set the OSD mode
    (see ``--osd-level`` for valid values).

-``print_text "<string>"``
+``print-text "<string>"``
    Print text to stdout. The string can contain properties (see
    `Property Expansion`_).

-``show_text "<string>" [<duration>|- [<level>]]``
+``show-text "<string>" [<duration>|- [<level>]]``
    Show text on the OSD. The string can contain properties, which are expanded
    as described in `Property Expansion`_. This can be used to show playback
    time, filename, and so on.
@@ -347,24 +353,12 @@ List of Input Commands
    <level>
        The minimum OSD level to show the text at (see ``--osd-level``).

-``show_progress``
+``show-progress``
    Show the progress bar, the elapsed time and the total duration of the file
    on the OSD.

-``discnav "<command>"``
-    Send a menu control command to the DVD/BD menu implementation. The following
-    commands are defined: ``up``, ``down``, ``left``, ``right``,
-    ``menu`` (request to enter menu), ``prev`` (previous screen),
-    ``select`` (activate current button), ``mouse`` (the mouse was clicked),
-    ``mouse_move`` (the mouse cursor changed position).
-
-    ``mouse_move`` will use the current mouse position.
-
-    Note that while the menu is active, the input section ``discnav-menu`` will
-    be enabled, so different key bindings can be mapped for menu mode.
-
-``write_watch_later_config``
-    Write the resume config file that the ``quit_watch_later`` command writes,
+``write-watch-later-config``
+    Write the resume config file that the ``quit-watch-later`` command writes,
    but continue playback normally.

 ``stop``
@@ -389,30 +383,47 @@ List of Input Commands
    <double>
        The mouse event represents double-click.

-``audio_add "<file>" [<flags> [<title> [<lang>]]]``
-    Load the given audio file. See ``sub_add`` command.
+``keypress <key_name>``
+    Send a key event through mpv's input handler, triggering whatever
+    behavior is configured to that key. ``key_name`` uses the ``input.conf``
+    naming scheme for keys and modifiers. Useful for the client API: key events
+    can be sent to libmpv to handle internally.

-``audio_remove [<id>]``
-    Remove the given audio track. See ``sub_remove`` command.
+``keydown <key_name>``
+    Similar to ``keypress``, but sets the ``KEYDOWN`` flag so that if the key is
+    bound to a repeatable command, it will be run repeatedly with mpv's key
+    repeat timing until the ``keyup`` command is called.

-``audio_reload [<id>]``
-    Reload the given audio tracks. See ``sub_reload`` command.
+``keyup [<key_name>]``
+    Set the ``KEYUP`` flag, stopping any repeated behavior that had been
+    triggered. ``key_name`` is optional. If ``key_name`` is not given or is an
+    empty string, ``KEYUP`` will be set on all keys. Otherwise, ``KEYUP`` will
+    only be set on the key specified by ``key_name``.

-``rescan_external_files [<mode>]``
+``audio-add "<file>" [<flags> [<title> [<lang>]]]``
+    Load the given audio file. See ``sub-add`` command.
+
+``audio-remove [<id>]``
+    Remove the given audio track. See ``sub-remove`` command.
+
+``audio-reload [<id>]``
+    Reload the given audio tracks. See ``sub-reload`` command.
+
+``rescan-external-files [<mode>]``
    Rescan external files according to the current ``--sub-auto`` and
    ``--audio-file-auto`` settings. This can be used to auto-load external
    files *after* the file was loaded.

    The ``mode`` argument is one of the following:

-    <keep-selection> (default)
-        Do not change current track selections.
-
-    <reselect>
-        Select the default audio and video streams, which typically selects
+    <reselect> (default)
+        Select the default audio and subtitle streams, which typically selects
        external files with highest preference. (The implementation is not
        perfect, and could be improved on request.)

+    <keep-selection>
+        Do not change current track selections.
+

 Input Commands that are Possibly Subject to Change
 --------------------------------------------------
@@ -455,7 +466,7 @@ Input Commands that are Possibly Subject to Change

    The ``vf`` command shows the list of requested filters on the OSD after
    changing the filter chain. This is roughly equivalent to
-    ``show_text ${vf}``. Note that auto-inserted filters for format conversion
+    ``show-text ${vf}``. Note that auto-inserted filters for format conversion
    are not shown on the list, only what was requested by the user.

    Normally, the commands will check whether the video chain is recreated
@@ -470,7 +481,7 @@ Input Commands that are Possibly Subject to Change
        - ``b vf set ""`` remove all video filters on ``b``
        - ``c vf toggle lavfi=gradfun`` toggle debanding on ``c``

-``cycle_values ["!reverse"] <property> "<value1>" "<value2>" ...``
+``cycle-values ["!reverse"] <property> "<value1>" "<value2>" ...``
    Cycle through a list of values. Each invocation of the command will set the
    given property to the next value in the list. The command maintains an
    internal counter which value to pick next, and which is initially 0. It is
@@ -488,7 +499,7 @@ Input Commands that are Possibly Subject to Change
    Note that there is a static limit of (as of this writing) 10 arguments
    (this limit could be raised on demand).

-``enable_section "<section>" [default|exclusive]``
+``enable-section "<section>" [flags]``
    Enable all key bindings in the named input section.

    The enabled input sections form a stack. Bindings in sections on the top of
@@ -497,14 +508,45 @@ Input Commands that are Possibly Subject to Change
    implicitly removed beforehand. (A section cannot be on the stack more than
    once.)

-    If ``exclusive`` is specified as second argument, all sections below the
-    newly enabled section are disabled. They will be re-enabled as soon as
-    all exclusive sections above them are removed.
+    The ``flags`` parameter can be a combination (separated by ``+``) of the
+    following flags:

-``disable_section "<section>"``
-    Disable the named input section. Undoes ``enable_section``.
+    <exclusive>
+        All sections enabled before the newly enabled section are disabled.
+        They will be re-enabled as soon as all exclusive sections above them
+        are removed. In other words, the new section shadows all previous
+        sections.
+    <allow-hide-cursor>
+        This feature can't be used through the public API.
+    <allow-vo-dragging>
+        Same.

-``overlay_add <id> <x> <y> "<file>" <offset> "<fmt>" <w> <h> <stride>``
+``disable-section "<section>"``
+    Disable the named input section. Undoes ``enable-section``.
+
+``define-section "<section>" "<contents>" [default|forced]``
+    Create a named input section, or replace the contents of an already existing
+    input section. The ``contents`` parameter uses the same syntax as the
+    ``input.conf`` file (except that using the section syntax in it is not
+    allowed), including the need to separate bindings with a newline character.
+
+    If the ``contents`` parameter is an empty string, the section is removed.
+
+    The section with the name ``default`` is the normal input section.
+
+    In general, input sections have to be enabled with the ``enable-section``
+    command, or they are ignored.
+
+    The last parameter has the following meaning:
+
+    <default> (also used if parameter omitted)
+        Use a key binding defined by this section only if the user hasn't
+        already bound this key to a command.
+    <forced>
+        Always bind a key. (The input section that was made active most recently
+        wins if there are ambiguities.)
+
+``overlay-add <id> <x> <y> "<file>" <offset> "<fmt>" <w> <h> <stride>``
    Add an OSD overlay sourced from raw data. This might be useful for scripts
    and applications controlling mpv, and which want to display things on top
    of the video window.
@@ -520,7 +562,7 @@ Input Commands that are Possibly Subject to Change
    ``id`` is an integer between 0 and 63 identifying the overlay element. The
    ID can be used to add multiple overlay parts, update a part by using this
    command with an already existing ID, or to remove a part with
-    ``overlay_remove``. Using a previously unused ID will add a new overlay,
+    ``overlay-remove``. Using a previously unused ID will add a new overlay,
    while reusing an ID will update it. (Future directions: there should be
    something to ensure different programs wanting to create overlays don't
    conflict with each others, should that ever be needed.)
@@ -534,11 +576,11 @@ Input Commands that are Possibly Subject to Change
    vdpau), so no actual copying is involved. Truncating the source file while
    the overlay is active will crash the player. You shouldn't change the data
    while the overlay is active, because the data is essentially accessed at
-    random points. Instead, call ``overlay_add`` again (preferably with a
+    random points. Instead, call ``overlay-add`` again (preferably with a
    different memory region to prevent tearing).

    It is also possible to pass a raw memory address for use as bitmap memory
-    by passing a memory address as integer prefixed with a ``&`` character.
+    by passing a memory address as integer prefixed with an ``&`` character.
    Passing the wrong thing here will crash the player. This mode might be
    useful for use with libmpv. The ``offset`` parameter is simply added to the
    memory address (since mpv 0.8.0, ignored before).
@@ -578,22 +620,22 @@ Input Commands that are Possibly Subject to Change
        an overlay's memory at random times whenever it feels the need to do
        so, for example when redrawing the screen.

-``overlay_remove <id>``
-    Remove an overlay added with ``overlay_add`` and the same ID. Does nothing
+``overlay-remove <id>``
+    Remove an overlay added with ``overlay-add`` and the same ID. Does nothing
    if no overlay with this ID exists.

-``script_message "<arg1>" "<arg2>" ...``
+``script-message "<arg1>" "<arg2>" ...``
    Send a message to all clients, and pass it the following list of arguments.
    What this message means, how many arguments it takes, and what the arguments
    mean is fully up to the receiver and the sender. Every client receives the
    message, so be careful about name clashes (or use ``script_message_to``).

-``script_message_to "<target>" "<arg1>" "<arg2>" ...``
+``script-message-to "<target>" "<arg1>" "<arg2>" ...``
    Same as ``script_message``, but send it only to the client named
    ``<target>``. Each client (scripts etc.) has a unique name. For example,
    Lua scripts can get their name via ``mp.get_script_name()``.

-``script_binding "<name>"``
+``script-binding "<name>"``
    Invoke a script-provided key binding. This can be used to remap key
    bindings provided by external Lua scripts.

@@ -614,24 +656,24 @@ Input Commands that are Possibly Subject to Change
    tracked). The second letter whether the event originates from the mouse,
    either ``m`` (mouse button) or ``-`` (something else).

-``ab_loop``
+``ab-loop``
    Cycle through A-B loop states. The first command will set the ``A`` point
    (the ``ab-loop-a`` property); the second the ``B`` point, and the third
    will clear both points.

-``vo_cmdline "<args>"``
+``vo-cmdline "<args>"``
    Reset the sub-option of the current VO. Currently works with ``opengl``
    (including ``opengl-hq``). The argument is the sub-option string usually
    passed to the VO on the command line. Not all sub-options can be set, but
    those which can will be reset even if they don't appear in the argument.
    This command might be changed or removed in the future.

-``drop_buffers``
+``drop-buffers``
    Drop audio/video/demuxer buffers, and restart from fresh. Might help with
    unseekable streams that are going out of sync.
    This command might be changed or removed in the future.

-``screenshot_raw [subtitles|video|window]``
+``screenshot-raw [subtitles|video|window]``
    Return a screenshot in memory. This can be used only through the client
    API. The MPV_FORMAT_NODE_MAP returned by this command has the ``w``, ``h``,
    ``stride`` fields set to obvious contents. A ``format`` field is set to
@@ -640,14 +682,14 @@ Input Commands that are Possibly Subject to Change
    field is of type MPV_FORMAT_BYTE_ARRAY with the actual image data. The image
    is freed as soon as the result node is freed.

-Undocumented commands: ``tv_last_channel`` (TV/DVB only),
-``get_property`` (deprecated), ``ao_reload`` (experimental/internal).
+Undocumented commands: ``tv-last-channel`` (TV/DVB only),
+``ao-reload`` (experimental/internal).

 Hooks
 ~~~~~

 Hooks are synchronous events between player core and a script or similar. This
-applies to the Lua scripting interface and the client API and only. Normally,
+applies to client API (including the Lua scripting interface). Normally,
 events are supposed to be asynchronous, and the hook API provides an awkward
 and obscure way to handle events that require stricter coordination. There are
 no API stability guarantees made. Not following the protocol exactly can make
@@ -656,7 +698,7 @@ the player freeze randomly. Basically, nobody should use this API.
 There are two special commands involved. Also, the client must listen for
 client messages (``MPV_EVENT_CLIENT_MESSAGE`` in the C API).

-``hook_add <hook-name> <id> <priority>``
+``hook-add <hook-name> <id> <priority>``
    Subscribe to the hook identified by the first argument (basically, the
    name of event). The ``id`` argument is an arbitrary integer chosen by the
    user. ``priority`` is used to sort all hook handlers globally across all
@@ -681,9 +723,9 @@ client messages (``MPV_EVENT_CLIENT_MESSAGE`` in the C API).
    typically be stopped.

    When the client is done, it must continue the core's hook execution by
-    running the ``hook_ack`` command.
+    running the ``hook-ack`` command.

-``hook_ack <string>``
+``hook-ack <string>``
    Run the next hook in the global chain of hooks. The argument is the 3rd
    argument of the client message that starts hook execution for the
    current client.
@@ -756,7 +798,7 @@ Properties

 Properties are used to set mpv options during runtime, or to query arbitrary
 information. They can be manipulated with the ``set``/``add``/``cycle``
-commands, and retrieved with ``show_text``, or anything else that uses property
+commands, and retrieved with ``show-text``, or anything else that uses property
 expansion. (See `Property Expansion`_.)

 The property name is annotated with RW to indicate whether the property is
@@ -784,6 +826,16 @@ Property list
 ``speed`` (RW)
    See ``--speed``.

+``audio-speed-correction``, ``video-speed-correction``
+    Factor multiplied with ``speed`` at which the player attempts to play the
+    file. Usually it's exactly 1. (Display sync mode will make this useful.)
+
+    OSD formatting will display it in the form of ``+1.23456%``, with the number
+    being ``(raw - 1) * 100`` for the given raw property value.
+
+``display-sync-active``
+    Return whether ``--video-sync=display`` is actually active.
+
 ``filename``
    Currently played file, with path stripped. If this is an URL, try to undo
    percent encoding as well. (The result is not necessarily correct, but
@@ -836,11 +888,14 @@ Property list
 ``stream-end``
    Raw end position in bytes in source stream.

-``length``
-    Length of the current file in seconds. If the length is unknown, the
+``duration``
+    Duration of the current file in seconds. If the duration is unknown, the
    property is unavailable. Note that the file duration is not always exactly
    known, so this is an estimate.

+    This replaces the ``length`` property, which was deprecated after the
+    mpv 0.9 release. (The semantics are the same.)
+
 ``avsync``
    Last A/V synchronization difference. Unavailable if audio or video is
    disabled.
@@ -876,10 +931,12 @@ Property list
    always exactly known, so this is an estimate.

 ``playtime-remaining``
-    ``time-remaining`` scaled by the the current ``speed``.
+    ``time-remaining`` scaled by the current ``speed``.

-``playback-time``
-    Return the playback time, which is the time difference between start PTS and current PTS.
+``playback-time`` (RW)
+    The playback time, which is the time relative to playback start. (This can
+    be different from the ``time-pos`` property if the file does not start at
+    position ``0``, in which case ``time-pos`` is the source timestamp.)

 ``chapter`` (RW)
    Current chapter number. The number of the first chapter is 0.
@@ -922,16 +979,6 @@ Property list
    Current BD/DVD title number. Writing works only for ``dvdnav://`` and
    ``bd://`` (and aliases for these).

-``disc-menu-active``
-    Return ``yes`` if the BD/DVD menu is active, or ``no`` on normal video
-    playback. The property is unavailable when playing something that is not
-    a BD or DVD. Use the ``discnav menu`` command to actually enter or leave
-    menu mode.
-
-``disc-mouse-on-button``
-    Return ``yes`` when the mouse cursor is located on a button, or ``no``
-    when cursor is outside of any button for disc navigation.
-
 ``chapters``
    Number of chapters.

@@ -1044,6 +1091,9 @@ Property list
    An example of these kind of metadata are the cropping parameters
    added by ``--vf=lavfi=cropdetect``.

+``af-metadata/<filter-label>``
+    Equivalent to ``vf-metadata/<filter-label>``, but for audio filters.
+
 ``pause`` (RW)
    Pause status. This is usually ``yes`` or ``no``. See ``--pause``.

@@ -1064,11 +1114,13 @@ Property list
    Network cache fill state (0-100.0).

 ``cache-size`` (RW)
-    Total network cache size in KB. This is similar to ``--cache``. This allows
+    Network cache size in KB. This is similar to ``--cache``. This allows
    to set the cache size at runtime. Currently, it's not possible to enable
    or disable the cache at runtime using this property, just to resize an
    existing cache.

+    This does not include the backbuffer size (changed after mpv 0.10.0).
+
    Note that this tries to keep the cache contents as far as possible. To make
    this easier, the cache resizing code will allocate the new cache while the
    old cache is still allocated.
@@ -1092,7 +1144,7 @@ Property list

 ``demuxer-cache-time``
    Approximate time of video buffered in the demuxer, in seconds. Same as
-    ``demuxer-cache-duration`` but returns the last timestamp of bufferred
+    ``demuxer-cache-duration`` but returns the last timestamp of buffered
    data in demuxer.

 ``demuxer-cache-idle``
@@ -1126,7 +1178,7 @@ Property list
    See ``--hr-seek``.

 ``volume`` (RW)
-    Current volume (0-100).
+    Current volume (see ``--volume`` for details).

 ``mute`` (RW)
    Current mute status (``yes``/``no``).
@@ -1134,18 +1186,52 @@ Property list
 ``audio-delay`` (RW)
    See ``--audio-delay``.

-``audio-format``
-    Audio format as string.
-
 ``audio-codec``
    Audio codec selected for decoding.

-``audio-samplerate``
-    Audio samplerate.
+``audio-codec-name``
+    Audio codec.

-``audio-channels``
-    Number of audio channels. The OSD value of this property is actually the
-    channel layout, while the raw value returns the number of channels only.
+``audio-params``
+    Audio format as output by the audio decoder.
+    This has a number of sub-properties:
+
+    ``audio-params/format``
+        The sample format as string. This uses the same names as used in other
+        places of mpv.
+
+    ``audio-params/samplerate``
+        Samplerate.
+
+    ``audio-params/channels``
+        The channel layout as a string. This is similar to what the
+        ``--audio-channels`` accepts.
+
+    ``audio-params/hr-channels``
+        As ``channels``, but instead of the possibly cryptic actual layout
+        sent to the audio device, return a hopefully more human readable form.
+        (Usually only ``audio-out-params/hr-channels`` makes sense.)
+
+    ``audio-params/channel-count``
+        Number of audio channels. This is redundant to the ``channels`` field
+        described above.
+
+    When querying the property with the client API using ``MPV_FORMAT_NODE``,
+    or with Lua ``mp.get_property_native``, this will return a mpv_node with
+    the following contents:
+
+    ::
+
+        MPV_FORMAT_NODE_MAP
+            "format"            MPV_FORMAT_STRING
+            "samplerate"        MPV_FORMAT_INT64
+            "channels"          MPV_FORMAT_STRING
+            "channel-count"     MPV_FORMAT_INT64
+            "hr-channels"       MPV_FORMAT_STRING
+
+``audio-out-params``
+    Same as ``audio-params``, but the format of the data written to the audio
+    API.

 ``aid`` (RW)
    Current audio track (similar to ``--aid``).
@@ -1156,7 +1242,7 @@ Property list
 ``balance`` (RW)
    Audio channel balance. (The implementation of this feature is rather odd.
    It doesn't change the volumes of each channel, but instead sets up a pan
-    matrix to mix the the left and right channels.)
+    matrix to mix the left and right channels.)

 ``fullscreen`` (RW)
    See ``--fullscreen``.
@@ -1167,17 +1253,18 @@ Property list
 ``field-dominance`` (RW)
    See ``--field-dominance``

-``colormatrix`` (RW)
-    See ``--colormatrix``.
+``colormatrix`` (R)
+    Redirects to ``video-params/colormatrix``. This parameter (as well as
+    similar ones) can be overridden with the ``format`` video filter.

-``colormatrix-input-range`` (RW)
-    See ``--colormatrix-input-range``.
+``colormatrix-input-range`` (R)
+    See ``colormatrix``.

-``colormatrix-output-range`` (RW)
-    See ``--colormatrix-output-range``.
+``colormatrix-output-range`` (R)
+    See ``colormatrix``.

-``colormatrix-primaries`` (RW)
-    See ``--colormatrix-primaries``.
+``colormatrix-primaries`` (R)
+    See ``colormatrix``.

 ``ontop`` (RW)
    See ``--ontop``.
@@ -1207,25 +1294,29 @@ Property list
    See ``--hue``.

 ``hwdec`` (RW)
-    Return the current hardware decoder that is used. This uses the same values
-    as the ``--hwdec`` option. If software decoding is active, this returns
-    ``no``. You can write this property. Then the ``--hwdec`` option is set to
-    the new value, and video decoding will be reinitialized (internally, the
-    player will perform a seek to refresh the video properly).
+    Reflects the ``--hwdec`` option.

-    Note that you don't know the success of the operation immediately after
-    writing this property. It happens with a delay as video is reinitialized.
+    Writing to it may change the currently used hardware decoder, if possible.
+    (Internally, the player may reinitialize the decoder, and will perform a
+    seek to refresh the video properly.) You can watch the other hwdec
+    properties to see whether this was successful.

-``detected-hwdec``
-    Return the current hardware decoder that was detected and opened. Returns
-    the same values as ``hwdec``.
+    Unlike in mpv 0.9.x and before, this does not return the currently active
+    hardware decoder.

-    This is known only once the VO has opened (and possibly later). With some
-    VOs (like ``opengl``), this is never known in advance, but only when the
-    decoder attempted to create the hw decoder successfully. Also, hw decoders
-    with ``-copy`` suffix are returned only while hw decoding is active (and
-    unset afterwards). All this reflects how detecting hw decoders are
-    detected and used internally in mpv.
+``hwdec-active``
+    Return ``yes`` or ``no``, depending on whether any type of hardware decoding
+    is actually in use.
+
+``hwdec-detected``
+    If software decoding is active, this returns the hardware decoder in use.
+    Otherwise, it returns either ``no``, or if applicable, the currently loaded
+    hardware decoding API. This is known only once the VO has opened (and
+    possibly later). With some VOs (like ``opengl``), this is never known in
+    advance, but only when the decoder attempted to create the hw decoder
+    successfully. Also, hw decoders with ``-copy`` suffix will return ``no``
+    while no video is being decoded. All this reflects how detecting hw decoders
+    are detected and used internally in mpv.

 ``panscan`` (RW)
    See ``--panscan``.
@@ -1289,26 +1380,29 @@ Property list
    ``video-params/rotate``
        Intended display rotation in degrees (clockwise).

+    ``video-params/stereo-in``
+        Source file stereo 3D mode. (See ``--video-stereo-mode`` option.)
+
    When querying the property with the client API using ``MPV_FORMAT_NODE``,
    or with Lua ``mp.get_property_native``, this will return a mpv_node with
    the following contents:

    ::

-        MPV_FORMAT_NODE_ARRAY
-            MPV_FORMAT_NODE_MAP (for each track)
-                "pixelformat"       MPV_FORMAT_STRING
-                "w"                 MPV_FORMAT_INT64
-                "h"                 MPV_FORMAT_INT64
-                "dw"                MPV_FORMAT_INT64
-                "dh"                MPV_FORMAT_INT64
-                "aspect"            MPV_FORMAT_DOUBLE
-                "par"               MPV_FORMAT_DOUBLE
-                "colormatrix"       MPV_FORMAT_STRING
-                "colorlevels"       MPV_FORMAT_STRING
-                "primaries"         MPV_FORMAT_STRING
-                "chroma-location"   MPV_FORMAT_STRING
-                "rotate"            MPV_FORMAT_INT64
+        MPV_FORMAT_NODE_MAP
+            "pixelformat"       MPV_FORMAT_STRING
+            "w"                 MPV_FORMAT_INT64
+            "h"                 MPV_FORMAT_INT64
+            "dw"                MPV_FORMAT_INT64
+            "dh"                MPV_FORMAT_INT64
+            "aspect"            MPV_FORMAT_DOUBLE
+            "par"               MPV_FORMAT_DOUBLE
+            "colormatrix"       MPV_FORMAT_STRING
+            "colorlevels"       MPV_FORMAT_STRING
+            "primaries"         MPV_FORMAT_STRING
+            "chroma-location"   MPV_FORMAT_STRING
+            "rotate"            MPV_FORMAT_INT64
+            "stereo-in"         MPV_FORMAT_STRING

 ``dwidth``, ``dheight``
    Video display size. This is the video size after filters and aspect scaling
@@ -1462,6 +1556,11 @@ Property list
        been unloaded yet; in this case, ``current`` refers to the new
        selection. (Since mpv 0.7.0.)

+    ``playlist/N/title``
+        Name of the Nth entry. Only available if the playlist file contains
+        such fields, and only if mpv's parser supports it for the given
+        playlist format.
+
    When querying the property with the client API using ``MPV_FORMAT_NODE``,
    or with Lua ``mp.get_property_native``, this will return a mpv_node with
    the following contents:
@@ -1473,6 +1572,7 @@ Property list
                "filename"  MPV_FORMAT_STRING
                "current"   MPV_FORMAT_FLAG (might be missing; since mpv 0.7.0)
                "playing"   MPV_FORMAT_FLAG (same)
+                "title"     MPV_FORMAT_STRING (optional)

 ``track-list``
    List of audio/video/sub tracks, current entry marked. Currently, the raw
@@ -1500,6 +1600,10 @@ Property list
    ``track-list/N/lang``
        Track language as identified by the file. Not always available.

+    ``track-list/N/audio-channels``
+        For audio tracks, the number of audio channels in the audio stream.
+        Not always accurate (depends on container hints). Not always available.
+
    ``track-list/N/albumart``
        ``yes`` if this is a video track that consists of a single picture,
        ``no`` or unavailable otherwise. This is used for video tracks that are
@@ -1509,6 +1613,10 @@ Property list
        ``yes`` if the track has the default flag set in the file, ``no``
        otherwise.

+    ``track-list/N/forced``
+        ``yes`` if the track has the forced flag set in the file, ``no``
+        otherwise.
+
    ``track-list/N/codec``
        The codec name used by this track, for example ``h264``. Unavailable
        in some rare cases.
@@ -1544,8 +1652,10 @@ Property list
                "src-id"            MPV_FORMAT_INT64
                "title"             MPV_FORMAT_STRING
                "lang"              MPV_FORMAT_STRING
+                "audio-channels"    MPV_FORMAT_INT64
                "albumart"          MPV_FORMAT_FLAG
                "default"           MPV_FORMAT_FLAG
+                "forced"            MPV_FORMAT_FLAG
                "external"          MPV_FORMAT_FLAG
                "external-filename" MPV_FORMAT_STRING
                "codec"             MPV_FORMAT_STRING
@@ -1624,7 +1734,7 @@ Property list

 ``osd-sym-cc``
    Inserts the current OSD symbol as opaque OSD control code (cc). This makes
-    sense only with the ``show_text`` command or options which set OSD messages.
+    sense only with the ``show-text`` command or options which set OSD messages.
    The control code is implementation specific and is useless for anything else.

 ``osd-ass-cc``
@@ -1632,13 +1742,13 @@ Property list
    ``${osd-ass-cc/1}`` enables it again. By default, ASS sequences are
    escaped to avoid accidental formatting, and this property can disable
    this behavior. Note that the properties return an opaque OSD control
-    code, which only makes sense for the ``show_text`` command or options
+    code, which only makes sense for the ``show-text`` command or options
    which set OSD messages.

    .. admonition:: Example

        - ``--osd-status-msg='This is ${osd-ass-cc/0}{\\b1}bold text'``
-        - ``show_text "This is ${osd-ass-cc/0}{\b1}bold text"``
+        - ``show-text "This is ${osd-ass-cc/0}{\b1}bold text"``

    Any ASS override tags as understood by libass can be used.

@@ -1739,6 +1849,12 @@ Property list
    Return the working directory of the mpv process. Can be useful for JSON IPC
    users, because the command line player usually works with relative paths.

+``protocol-list``
+    List of protocol prefixes potentially recognized by the player. They are
+    returned without trailing ``://`` suffix (which is still always required).
+    In some cases, the protocol will not actually be supported (consider
+    ``https`` if ffmpeg is not compiled with TLS support).
+
 ``mpv-version``
    Return the mpv version/copyright string. Depending on how the binary was
    built, it might contain either a release version, or just a git hash.
@@ -1778,11 +1894,16 @@ Property list
        For many complex types, this isn't very accurate.

    ``option-info/<name>/set-from-commandline``
-
        Return ``yes`` if the option was set from the mpv command line,
        ``no`` otherwise. What this is set to if the option is e.g. changed
        at runtime is left undefined (meaning it could change in the future).

+    ``option-info/<name>/set-locally``
+        Return ``yes`` if the option was set per-file. This is the case with
+        automatically loaded profiles, file-dir configs, and other cases. It
+        means the option value will be restored to the value before playback
+        start when playback ends.
+
    ``option-info/<name>/default-value``
        The default value of the option. May not always be available.

@@ -1805,11 +1926,14 @@ Property Expansion
 ------------------

 All string arguments to input commands as well as certain options (like
-``--term-playing-msg``) are subject to property expansion.
+``--term-playing-msg``) are subject to property expansion. Note that property
+expansion does not work in places where e.g. numeric parameters are expected.
+(For example, the ``add`` command does not do property expansion. The ``set``
+command is an exception and not a general rule.)

 .. admonition:: Example for input.conf

-    ``i show_text "Filename: ${filename}"``
+    ``i show-text "Filename: ${filename}"``
        shows the filename of the current file when pressing the ``i`` key

 Within ``input.conf``, property expansion can be inhibited by putting the
@@ -79,6 +79,24 @@ mpv will also send events to clients with JSON messages of the following form:
 where ``event_name`` is the name of the event. Additional event-specific fields
 can also be present. See `List of events`_ for a list of all supported events.

+Because events can occur at any time, it may be difficult at times to determine
+which response goes with which command. Commands may optionally include a
+``request_id`` which, if provided in the command request, will be copied
+verbatim into the response. mpv does not intrepret the ``request_id`` in any
+way; it is solely for the use of the requester.
+
+For example, this request:
+
+::
+
+    { "command": ["get_property", "time-pos"], "request_id": 100 }
+
+Would generate this response:
+
+::
+
+    { "error": "success", "data": 1.468135, "request_id": 100 }
+
 All commands, replies, and events are separated from each other with a line
 break character (``\n``).

@@ -73,6 +73,10 @@ The ``mp`` module is preloaded, although it can be loaded manually with
    Run the given command. This is similar to the commands used in input.conf.
    See `List of Input Commands`_.

+    By default, this will show something on the OSD (depending on the command),
+    as if it was used in ``input.conf``. See `Input Command Prefixes`_ how
+    to influence OSD usage per command.
+
    Returns ``true`` on success, or ``nil, error`` on error.

 ``mp.commandv(arg1, arg2, ...)``
@@ -94,6 +98,9 @@ The ``mp`` module is preloaded, although it can be loaded manually with
    the ``expand-properties`` prefix, or the ``mp.get_property`` family of
    functions.

+    Unlike ``mp.command``, this will not use OSD by default either (except
+    for some OSD-specific commands).
+
 ``mp.command_native(table [,def])``
    Similar to ``mp.commandv``, but pass the argument list as table. This has
    the advantage that in at least some cases, arguments can be passed as
@@ -512,7 +519,7 @@ Example implementation::
        optionC = true,
    }
    read_options(options, "myscript")
-    print(option.optionA)
+    print(options.optionA)


 The config file will be stored in ``lua-settings/identifier.conf`` in mpv's user
@@ -623,6 +630,10 @@ strictly part of the guaranteed API.
            On Windows, ``killed`` is only returned when the process has been
            killed by mpv as a result of ``cancellable`` being set to ``true``.

+        ``killed_by_us``
+            Set to ``true`` if the process has been killed by mpv as a result
+            of ``cancellable`` being set to ``true``.
+
    In all cases, ``mp.resume_all()`` is implicitly called.

 ``utils.parse_json(str [, trail])``
@@ -683,6 +694,33 @@ List of events
    Happens after a file was unloaded. Typically, the player will load the
    next file right away, or quit if this was the last file.

+    The event has the ``reason`` field, which takes one of these values:
+
+    ``eof``
+        The file has ended. This can (but doesn't have to) include
+        incomplete files or broken network connections under
+        circumstances.
+
+    ``stop``
+        Playback was ended by a command.
+
+    ``quit``
+        Playback was ended by sending the quit command.
+
+    ``error``
+        An error happened. In this case, an ``error`` field is present with
+        the error string.
+
+    ``redirect``
+        Happens with playlists and similar. Details see
+        ``MPV_END_FILE_REASON_REDIRECT`` in the C API.
+
+    ``unknown``
+        Unknown. Normally doesn't happen, unless the Lua API is out of sync
+        with the C API. (Likewise, it could happen that your script gets
+        reason strings that did not exist yet at the time your script was
+        written.)
+
 ``file-loaded``
    Happens after a file was loaded and begins playback.

@@ -116,11 +116,7 @@ o (also P)
    Show progression bar, elapsed time and total duration on the OSD.

 O
-    Toggle OSD states: none / seek / seek + timer / seek + timer + total time.
-
-d
-    Toggle frame dropping states: none / skip display / skip decoding (see
-    ``--framedrop``).
+    Toggle OSD states between normal and playback time/duration.

 v
    Toggle subtitle visibility.
@@ -134,6 +130,9 @@ x and z
 l
    Set/clear A-B loop points. See ``ab_loop`` command for details.

+L
+    Toggle infinite looping.
+
 Ctrl + and Ctrl -
    Adjust audio delay by +/- 0.1 seconds.

@@ -156,6 +155,10 @@ S
    Take a screenshot, without subtitles. (Whether this works depends on VO
    driver support.)

+Ctrl s
+    Take a screenshot, as the window shows it (with subtitles, OSD, and scaled
+    video).
+
 I
    Show filename on the OSD.

@@ -168,7 +171,7 @@ Shift+PGUP and Shift+PGDWN
    Seek backward or forward by 10 minutes. (This used to be mapped to
    PGUP/PGDWN without Shift.)

-D
+d
    Activate/deactivate deinterlacer.

 A
@@ -273,7 +276,7 @@ spaces or characters like ``,`` or ``:``, you need to quote them:
    ``mpv '--vo=opengl:icc-profile="file with spaces.icc",xv'``

 Shells may actually strip some quotes from the string passed to the commandline,
-so the example quotes the string twice, ensuring that mpv recieves the ``"``
+so the example quotes the string twice, ensuring that mpv receives the ``"``
 quotes.

 The ``[...]`` form of quotes wraps everything between ``[`` and ``]``. It's
@@ -302,7 +305,7 @@ Suboptions passed to the client API are also subject to escaping. Using
 command line (but without shell processing of the string). Some options
 support passing values in a more structured way instead of flat strings, and
 can avoid the suboption parsing mess. For example, ``--vf`` supports
-``MPV_FORMAT_NODE``, which let's you pass suboptions as a nested data structure
+``MPV_FORMAT_NODE``, which lets you pass suboptions as a nested data structure
 of maps and arrays. (``--vo`` supports this in the same way, although this
 fact is undocumented.)

@@ -327,7 +330,17 @@ additionally wrapped in the fixed-length syntax, e.g. ``%n%string_of_length_n``
 Some mpv options interpret paths starting with ``~``. Currently, the prefix
 ``~~/`` expands to the mpv configuration directory (usually ``~/.config/mpv/``).
 ``~/`` expands to the user's home directory. (The trailing ``/`` is always
-required.)
+required.) There are the following paths as well:
+
+================ ===============================================================
+Name             Meaning
+================ ===============================================================
+``~~home/``      same as ``~~/``
+``~~global/``    the global config path, if available (not on win32)
+``~~osxbundle/`` the OSX bundle resource path (OSX only)
+``~~desktop/``   the path to the desktop (win32, OSX)
+================ ===============================================================
+

 Per-File Options
 ----------------
@@ -378,7 +391,8 @@ Location and Syntax
 You can put all of the options in configuration files which will be read every
 time mpv is run. The system-wide configuration file 'mpv.conf' is in your
 configuration directory (e.g. ``/etc/mpv`` or ``/usr/local/etc/mpv``), the
-user-specific one is ``~/.config/mpv/mpv.conf``.
+user-specific one is ``~/.config/mpv/mpv.conf``. For details and platform
+specifics see the `FILES`_ section.
 User-specific options override system-wide options and options given on the
 command line override either. The syntax of the configuration files is
 ``option=<value>``; everything after a *#* is considered a comment. Options
@@ -402,7 +416,7 @@ but option values still need to be quoted as a whole if it contains certain
 characters like spaces. A config entry can be quoted with ``"`` and ``'``,
 as well as with the fixed-length syntax (``%n%``) mentioned before. This is like
 passing the exact contents of the quoted string as command line option. C-style
-escapes are currently _not_ interpreted on this level, although some options to
+escapes are currently _not_ interpreted on this level, although some options do
 this manually. (This is a mess and should probably be changed at some point.)

 Putting Command Line Options into the Configuration File
@@ -512,13 +526,21 @@ listed.
  if there is audio "missing", or not enough frames can be dropped. Usually
  this will indicate a problem. (``total-avsync-change`` property.)
 - Encoding state in ``{...}``, only shown in encoding mode.
+- Display sync state. If display sync is active (``display-sync-active``
+  property), this shows ``DS: +0.02598%``, where the number is the speed change
+  factor applied to audio to achieve sync to display, expressed in percent
+  deviation from 1.0 (``audio-speed-correction`` property). In sync modes which
+  don't resample, this will always be ``+0.00000%``.
+- Missed frames, e.g. ``Missed: 4``. (``vo-missed-frame-count`` property.) Shows
+  up in display sync mode only. This is incremented each time a frame took
+  longer to display than intended.
 - Dropped frames, e.g. ``Dropped: 4``. Shows up only if the count is not 0. Can
  grow if the video framerate is higher than that of the display, or if video
  rendering is too slow. Also can be incremented on "hiccups" and when the video
  frame couldn't be displayed on time. (``vo-drop-frame-count`` property.)
  If the decoder drops frames, the number of decoder-dropped frames is appended
  to the display as well, e.g.: ``Dropped: 4/34``. This happens only if
-  decoder-framedropping is enabled with the ``--framedrop`` options.
+  decoder frame dropping is enabled with the ``--framedrop`` options.
  (``drop-frame-count`` property.)
 - Cache state, e.g. ``Cache:  2s+134KB``. Visible if the stream cache is enabled.
  The first value shows the amount of video buffered in the demuxer in seconds,
@@ -549,13 +571,9 @@ PROTOCOLS
    you must mount the ISO file as filesystem, and point ``--bluray-device``
    to the mounted directory directly.

-``bdnav://[title][/device]``
-    Play a Blu-Ray disc, with navigation features enabled. This feature is
-    permanently experimental.
-
 ``dvd://[title|[starttitle]-endtitle][/device]`` ``--dvd-device=PATH``
-    Play a DVD. If you want dvdnav menus, use ``dvd://menu``. If no title
-    is given, the longest title is auto-selected.
+    Play a DVD. DVD menus are not supported. If no title is given, the longest
+    title is auto-selected.

    ``dvdnav://`` is an old alias for ``dvd://`` and does exactly the same
    thing.
@@ -600,6 +618,11 @@ PROTOCOLS
    ``PATH`` itself should start with a third ``/`` to make the path an
    absolute path.

+``fd://123``
+    Read data from the given UNIX FD (for example 123). This is similar to
+    piping data to stdin via ``-``, but can use an arbitrary file descriptor.
+    Will not work correctly on MS Windows.
+
 ``edl://[edl specification as in edl-mpv.rst]``
    Stitch together parts of multiple files and play them.

@@ -634,6 +657,7 @@ the ``pseudo-gui`` profile being predefined with the following contents:
    terminal=no
    force-window=yes
    idle=once
+    screenshot-directory=~~desktop/

 This follows the mpv config file format. To customize pseudo-GUI mode, you can
 put your own ``pseudo-gui`` profile into your ``mpv.conf``. This profile will
@@ -788,6 +812,8 @@ If errors happen, the following exit codes can be returned:
        immediately after initialization.
    :3: There were some files that could be played, and some files which
        couldn't (using the definition of success from above).
+    :4: Quit due to a signal, Ctrl+c in a VO window (by default), or from the
+        default quit key bindings in encoding mode.

 Note that quitting the player manually will always lead to exit code 0,
 overriding the exit code that would be returned normally. Also, the ``quit``
@@ -796,6 +822,8 @@ input command can take an exit code: in this case, that exit code is returned.
 FILES
 =====

+For Windows-specifics, see `FILES ON WINDOWS`_ section.
+
 ``/usr/local/etc/mpv/mpv.conf``
    mpv system-wide settings (depends on ``--prefix`` passed to configure - mpv
    in default configuration will use ``/usr/local/etc/mpv/`` as config
@@ -857,6 +885,14 @@ Other config files (such as ``input.conf``) are in the same directory. See the
 The environment variable ``$MPV_HOME`` completely overrides these, like on
 UNIX.

+If a directory named ``portable_config`` next to the mpv.exe exists, all
+config will be loaded from this directory only. Watch later config files are
+written to this directory as well. (This exists on Windows only and is redundant
+with ``$MPV_HOME``. However, since Windows is very scripting unfriendly, a
+wrapper script just setting ``$MPV_HOME``, like you could do it on other
+systems, won't work. ``portable_config`` is provided for convenience to get
+around this restriction.)
+
 Config files located in the same directory as ``mpv.exe`` are loaded with
 lower priority. Some config files are loaded only once, which means that
 e.g. of 2 ``input.conf`` files located in two config directories, only the
@@ -129,6 +129,17 @@ Playback Control
    Specify which chapter to start playing at. Optionally specify which
    chapter to end playing at. Also see ``--start``.

+``--playlist-pos=<no|index>``
+    Set which file on the internal playlist to start playback with. The index
+    is an integer, with 0 meaning the first file. The value ``no`` means that
+    the selection of the entry to play is left to the playback resume mechanism
+    (default). If an entry with the given index doesn't exist, the behavior is
+    unspecified and might change in future mpv versions. The same applies if
+    the playlist contains further playlists (don't expect any reasonable
+    behavior). Passing a playlist file to mpv should work with this option,
+    though. E.g. ``mpv playlist.m3u --playlist-pos=123`` will work as expected,
+    as long as ``playlist.m3u`` does not link to further playlists.
+
 ``--playlist=<filename>``
    Play files according to a playlist file (Supports some common formats. If
    no format is detected, it will be treated as list of files, separated by
@@ -348,7 +359,7 @@ Program Behavior
 ``--idle=<no|yes|once>``
    Makes mpv wait idly instead of quitting when there is no file to play.
    Mostly useful in slave mode, where mpv can be controlled through input
-    commands (see also ``--slave-broken``).
+    commands.

    ``once`` will only idle at start and let the player close once the
    first playlist has finished playing back.
@@ -457,14 +468,14 @@ Program Behavior
    (Default: ``best``)

 ``--ytdl-raw-options=<key>=<value>[,<key>=<value>[,...]]``
-    Pass arbitraty options to youtube-dl. Parameter and argument should be
+    Pass arbitrary options to youtube-dl. Parameter and argument should be
    passed as a key-value pair. Options without argument must include ``=``.
-    
+
    There is no sanity checking so it's possible to break things (i.e.
    passing invalid parameters to youtube-dl).

    .. admonition:: Example
- 
+
        ``--ytdl-raw-options=username=user,password=pass``
        ``--ytdl-raw-options=force-ipv6=``

@@ -498,6 +509,10 @@ Video
    Do not play video. With some demuxers this may not work. In those cases
    you can try ``--vo=null`` instead.

+    mpv will try to download the audio only if media is streamed with
+    youtube-dl, because it saves bandwidth. This is done by setting the ytdl_format
+    to "bestaudio/best" in the ytdl_hook.lua script.
+
 ``--untimed``
    Do not sleep when outputting video frames. Useful for benchmarks when used
    with ``--no-audio.``
@@ -524,7 +539,7 @@ Video
        Old, decoder-based framedrop mode. (This is the same as ``--framedrop=yes``
        in mpv 0.5.x and before.) This tells the decoder to skip frames (unless
        they are needed to decode future frames). May help with slow systems,
-        but can produce unwatchably choppy output, or even freeze the display
+        but can produce unwatchable choppy output, or even freeze the display
        completely. Not recommended.
        The ``--vd-lavc-framedrop`` option controls what frames to drop.
    <decoder+vo>
@@ -555,6 +570,7 @@ Video
    :vaapi:     requires ``--vo=opengl`` or ``--vo=vaapi`` (Linux with Intel GPUs only)
    :vaapi-copy: copies video back into system RAM (Linux with Intel GPUs only)
    :vda:       requires ``--vo=opengl`` (OS X only)
+    :videotoolbox: requires ``--vo=opengl`` (newer OS X only)
    :dxva2-copy: copies video back to system RAM (Windows only)
    :rpi:      requires ``--vo=rpi`` (Raspberry Pi only - default if available)

@@ -575,6 +591,22 @@ Video
        codecs. See ``--hwdec-codecs`` to enable hardware decoding for more
        codecs.

+``--hwdec-preload=<api>``
+    This is useful for the ``opengl`` and ``opengl-cb`` VOs for creating the
+    hardware decoding OpenGL interop context, but without actually enabling
+    hardware decoding itself (like ``--hwdec`` does).
+
+    If set to ``no`` (default), the ``--hwdec`` option is used.
+
+    For ``opengl``, if set, do not create the interop context on demand, but
+    when the VO is created.
+
+    For ``opengl-cb``, if set, load the interop context as soon as the OpenGL
+    context is created. Since ``opengl-cb`` has no on-demand loading, this
+    allows enabling hardware decoding at runtime at all, without having to
+    to temporarily set the ``hwdec`` option just during OpenGL context
+    initialization with ``mpv_opengl_cb_init_gl()``.
+
 ``--panscan=<0.0-1.0>``
    Enables pan-and-scan functionality (cropping the sides of e.g. a 16:9
    video to make it fit a 4:3 display without black bands). The range
@@ -601,6 +633,23 @@ Video
    Ignore aspect ratio information from video file and assume the video has
    square pixels. See also ``--video-aspect``.

+``--video-aspect-method=<hybrid|bitstream|container>``
+    This sets the default video aspect determination method (if the aspect is
+    _not_ overridden by the user with ``--video-aspect`` or others).
+
+    :hybrid:    Prefer the container aspect ratio. If the bitstream aspect
+                switches mid-stream, switch to preferring the bitstream aspect.
+                This is the default behavior in mpv and mplayer2.
+    :container: Strictly prefer the container aspect ratio. This is apparently
+                the default behavior with VLC, at least with Matroska.
+    :bitstream: Strictly prefer the bitstream aspect ratio, unless the bitstream
+                aspect ratio is not set. This is apparently the default behavior
+                with XBMC/kodi, at least with Matroska.
+
+    Normally you should not set this. Try the ``container`` and ``bitstream``
+    choices if you encounter video that has the wrong aspect ratio in mpv,
+    but seems to be correct in other players.
+
 ``--video-unscaled``
    Disable scaling of the video. If the window is larger than the video,
    black bars are added. Otherwise, the video is cropped. The video still
@@ -635,17 +684,17 @@ Video
    which means the value ``0`` would rotate the video according to the
    rotation metadata.)

-``--video-stereo-mode=<mode>``
+``--video-stereo-mode=<no|mode>``
    Set the stereo 3D output mode (default: ``mono``). This is done by inserting
    the ``stereo3d`` conversion filter.

+    The pseudo-mode ``no`` disables automatic conversion completely.
+
    The mode ``mono`` is an alias to ``ml``, which refers to the left frame in
    2D. This is the default, which means mpv will try to show 3D movies in 2D,
    instead of the mangled 3D image not intended for consumption (such as
    showing the left and right frame side by side, etc.).

-    The pseudo-mode ``none`` disables automatic conversion completely.
-
    Use ``--video-stereo-mode=help`` to list all available modes. Check with
    the ``stereo3d`` filter documentation to see what the names mean. Note that
    some names refer to modes not supported by ``stereo3d`` - these modes can
@@ -712,6 +761,10 @@ Video
    :top:     top field first
    :bottom:  bottom field first

+    .. note::
+
+        Setting either ``top`` or ``bottom`` will flag all frames as interlaced.
+
 ``--frames=<number>``
    Play/convert only first ``<number>`` video frames, then quit.

@@ -729,7 +782,7 @@ Video
    You can get the list of allowed codecs with ``mpv --vd=help``. Remove the
    prefix, e.g. instead of ``lavc:h264`` use ``h264``.

-    By default this is set to ``h264,vc1,wmv3``. Note that the hardware
+    By default this is set to ``h264,vc1,wmv3,hevc``. Note that the hardware
    acceleration special codecs like ``h264_vdpau`` are not relevant anymore,
    and in fact have been removed from Libav in this form.

@@ -804,10 +857,11 @@ Video
    Set framedropping mode used with ``--framedrop`` (see skiploopfilter for
    available skip values).

-``--vd-lavc-threads=<0-16>``
+``--vd-lavc-threads=<N>``
    Number of threads to use for decoding. Whether threading is actually
-    supported depends on codec. 0 means autodetect number of cores on the
-    machine and use that, up to the maximum of 16 (default: 0).
+    supported depends on codec (default: 0). 0 means autodetect number of cores
+    on the machine and use that, up to the maximum of 16. You can set more than
+    16 threads manually.



@@ -834,7 +888,7 @@ Audio
    Note that many AOs have a ``device`` sub-option, which overrides the
    device selection of this option (but not the audio output selection).
    Likewise, forcing an AO with ``--ao`` will override the audio output
-    selection of ``--audio-device`` (but not the device selecton).
+    selection of ``--audio-device`` (but not the device selection).

    Currently not implemented for most AOs.

@@ -851,6 +905,25 @@ Audio
    ``--af-clr`` exist to modify a previously specified list, but you
    should not need these for typical use.

+``--audio-spdif=<codecs>``
+    List of codecs for which compressed audio passthrough should be used. This
+    works for both classic S/PDIF and HDMI.
+
+    Possible codecs are ``ac3``, ``dts``, ``dts-hd``. Multiple codecs can be
+    specified by separating them with ``,``. ``dts`` refers to low bitrate DTS
+    core, while ``dts-hd`` refers to DTS MA (receiver and OS support varies).
+    You should only use either ``dts`` or ``dts-hd`` (if both are specified,
+    and ``dts`` comes first, only ``dts`` will be used).
+
+    In general, all codecs in the ``spdif`` family listed with ``--ad=help``
+    are supported in theory.
+
+    .. admonition:: Warning
+
+        There is not much reason to use this. HDMI supports uncompressed
+        multichannel PCM, and mpv supports lossless DTS-HD decoding via
+        FFmpeg's libdcadec wrapper.
+
 ``--ad=<[+|-]family1:(*|decoder1),[+|-]family2:(*|decoder2),...[-]>``
    Specify a priority list of audio decoders to be used, according to their
    family and decoder name. Entries like ``family:*`` prioritize all decoders
@@ -878,9 +951,21 @@ Audio
        ``--ad=help``
            List all available decoders.

-``--volume=<-1-100>``
-    Set the startup volume. A value of -1 (the default) will not change the
-    volume. See also ``--softvol``.
+    .. admonition:: Warning
+
+        Enabling compressed audio passthrough (AC3 and DTS via SPDIF/HDMI) with
+        this option is deprecated. Use ``--audio-spdif`` instead.
+
+``--volume=<value>``
+    Set the startup volume. 0 means silence, 100 means no volume reduction or
+    amplification. A value of -1 (the default) will not change the volume. See
+    also ``--softvol``.
+
+    .. note::
+
+        This was changed after the mpv 0.9 release. Before that, 100 actually
+        meant maximum volume. At the same time, the volume scale was made cubic,
+        so the old values won't match up with the new ones anyway.

 ``--audio-delay=<sec>``
    Audio delay in seconds (positive or negative float value). Positive values
@@ -942,17 +1027,12 @@ Audio
    welcome. A full list of AVOptions can be found in the FFmpeg manual.

 ``--ad-spdif-dtshd=<yes|no>``, ``--dtshd``, ``--no-dtshd``
-    When using DTS pass-through, output any DTS-HD track as-is.
-    With ``ad-spdif-dtshd=no`` (the default), only the DTS Core parts will be
-    output.
+    If DTS is passed through, use DTS-HD.

-    DTS-HD tracks can be sent over HDMI but not over the original
-    coax/TOSLINK S/PDIF system.
+    .. admonition:: Warning

-    Some receivers don't accept DTS core-only when ``--ad-spdif-dtshd=yes`` is
-    used, even though they accept DTS-HD.
-
-    ``--dtshd`` and ``--no-dtshd`` are deprecated aliases.
+        This and enabling passthrough via ``--ad`` are deprecated in favor of
+        using ``--audio-spdif=dts-hd``.

 ``--audio-channels=<number|layout>``
    Request a channel layout for audio output (default: auto). This  will ask
@@ -981,6 +1061,14 @@ Audio
    If the channel layout of the media file (i.e. the decoder) and the AO's
    channel layout don't match, mpv will attempt to insert a conversion filter.

+    .. admonition:: Warning
+
+        Using ``auto`` can cause issues when using audio over HDMI. The OS will
+        typically report all channel layouts that _can_ go over HDMI, even if
+        the receiver does not support them. If a receiver gets an unsupported
+        channel layout, random things can happen, such as dropping the
+        additional channels, or adding noise.
+
 ``--audio-display=<no|attachment>``
    Setting this option to ``attachment`` (default) will display image
    attachments (e.g. album cover art) when playing audio files. It will
@@ -1049,19 +1137,9 @@ Audio
    their start timestamps differ, and then video timing is gradually adjusted
    if necessary to reach correct synchronization later.

-``--softvol-max=<10.0-10000.0>``
-    Set the maximum amplification level in percent (default: 200). A value of
-    200 will allow you to adjust the volume up to a maximum of double the
-    current level. With values below 100 the initial volume (which is 100%)
-    will be above the maximum, which e.g. the OSD cannot display correctly.
-
-    .. admonition:: Note
-
-        The maximum value of ``--volume`` as well as the ``volume`` property
-        is always 100. Likewise, the volume OSD bar always goes from 0 to 100.
-        This means that with ``--softvol-max=200``, ``--volume=100`` sets
-        maximum amplification, i.e. amplify by 200%. The default volume (no
-        change in volume) will be ``50`` in this case.
+``--softvol-max=<100.0-1000.0>``
+    Set the maximum amplification level in percent (default: 130). A value of
+    130 will allow you to adjust the volume up to about double the normal level.

 ``--audio-file-auto=<no|exact|fuzzy|all>``, ``--no-audio-file-auto``
    Load additional audio files matching the video filename. The parameter
@@ -1136,7 +1214,7 @@ Subtitles
    the top of the screen) alongside the normal subtitle, and provides a way
    to render two subtitles at once.

-    there are some caveats associated with this feature. For example, bitmap
+    There are some caveats associated with this feature. For example, bitmap
    subtitles will always be rendered in their usual position, so selecting a
    bitmap subtitle as secondary subtitle will result in overlapping subtitles.
    Secondary subtitles are never shown on the terminal if video is disabled.
@@ -1352,6 +1430,18 @@ Subtitles

    Disabled by default.

+``--stretch-image-subs-to-screen=<yes|no>``
+    Stretch DVD and other image subtitles to the screen, ignoring the video
+    margins. This has a similar effect as ``--sub-use-margins`` for text
+    subtitles, except that the text itself will be stretched, not only just
+    repositioned. (At least in general it is unavoidable, as an image bitmap
+    can in theory consist of a single bitmap covering the whole screen, and
+    the player won't know where exactly the text parts are located.)
+
+    This option does not display subtitles correctly. Use with care.
+
+    Disabled by default.
+
 ``--sub-ass``, ``--no-sub-ass``
    Render ASS subtitles natively (enabled by default).

@@ -1381,10 +1471,11 @@ Subtitles

 ``--sub-codepage=<codepage>``
    If your system supports ``iconv(3)``, you can use this option to specify
-    the subtitle codepage. By default, ENCA will be used to guess the charset.
-    If mpv is not compiled with ENCA, ``UTF-8:UTF-8-BROKEN`` is the default,
-    which means it will try to use UTF-8, otherwise the ``UTF-8-BROKEN``
-    pseudo codepage (see below).
+    the subtitle codepage. By default, uchardet will be used to guess the
+    charset. If mpv is not compiled with uchardet, enca will be used.
+    If mpv is compiled with neither uchardet nor enca, ``UTF-8:UTF-8-BROKEN``
+    is the default, which means it will try to use UTF-8, otherwise the
+    ``UTF-8-BROKEN`` pseudo codepage (see below).

    The default value for this option is ``auto``, whose actual effect depends
    on whether ENCA is compiled.
@@ -1434,6 +1525,12 @@ Subtitles
    mode. Use ``--sub-codepage=guess:help`` to get a list of
    languages subject to the same caveat as with ENCA above.

+    If the player was compiled with uchardet support you can use it with:
+
+    ``--sub-codepage=uchardet``
+
+    This mode doesn't take language or fallback codepage.
+
 ``--sub-fix-timing``, ``--no-sub-fix-timing``
    By default, external text subtitles are preprocessed to remove minor gaps
    or overlaps between subtitles (if the difference is smaller than 200 ms,
@@ -1586,7 +1683,7 @@ Window
    file.mkv normally, then fail to open ``/dev/null``, then exit). (In
    mpv 0.8.0, ``always`` was introduced, which restores the old behavior.)

-``--force-window``
+``--force-window=<yes|no|immediate>``
    Create a video output window even if there is no video. This can be useful
    when pretending that mpv is a GUI application. Currently, the window
    always has the size 640x480, and is subject to ``--geometry``,
@@ -1598,7 +1695,9 @@ Window
        window placement still works if the video size is different from the
        ``--force-window`` default window size). This can be a problem if
        initialization doesn't work perfectly, such as when opening URLs with
-        bad network connection, or opening broken video files.
+        bad network connection, or opening broken video files. The ``immediate``
+        mode can be used to create the window always on program start, but this
+        may cause other issues.

 ``--ontop``
    Makes the player window stay on top of other windows.
@@ -1849,7 +1948,7 @@ Window
    ``intptr_t``. mpv will create its own window, and set the wid window as
    parent, like with X11.

-    On OSX/Cocoa. the ID is interpreted as ``NSView*``. Pass it as value cast
+    On OSX/Cocoa, the ID is interpreted as ``NSView*``. Pass it as value cast
    to ``intptr_t``. mpv will creates its own sub-view. Because OSX does not
    support window embedding of foreign processes, this works only with libmpv,
    and will crash when used from the command line.
@@ -1943,7 +2042,7 @@ Disc Devices
    (Never) accept imperfect data reconstruction.

 ``--cdda-cdtext=<yes|no>``
-    Print CD text. This is disabled by default, because it ruins perfomance
+    Print CD text. This is disabled by default, because it ruins performance
    with CD-ROM drives for unknown reasons.

 ``--dvd-speed=<speed>``
@@ -2074,7 +2173,7 @@ Demuxer
    seeks only.

    You can use the ``--demuxer-mkv-subtitle-preroll-secs`` option to specify
-    how mach data the demuxer should pre-read at most in order to find subtitle
+    how much data the demuxer should pre-read at most in order to find subtitle
    packets that may overlap. Setting this to 0 will effectively disable this
    preroll mechanism. Setting a very large value can make seeking very slow,
    and an extremely large value would completely reread the entire file from
@@ -2099,15 +2198,20 @@ Demuxer
 ``--demuxer-mkv-subtitle-preroll-secs=<value>``
    See ``--demuxer-mkv-subtitle-preroll``.

-``--demuxer-mkv-probe-video-duration``
+``--demuxer-mkv-probe-video-duration=<yes|no|full>``
    When opening the file, seek to the end of it, and check what timestamp the
    last video packet has, and report that as file duration. This is strictly
    for compatibility with Haali only. In this mode, it's possible that opening
    will be slower (especially when playing over http), or that behavior with
    broken files is much worse. So don't use this option.

+    The ``yes`` mode merely uses the index and reads a small number of blocks
+    from the end of the file. The ``full`` mode actually traverses the entire
+    file and can make a reliable estimate even without an index present (such
+    as partial files).
+
 ``--demuxer-mkv-fix-timestamps=<yes|no>``
-    Fix rounded Matroska timestamps (enabled by default). Matroska usually
+    Fix rounded Matroska timestamps (disabled by default). Matroska usually
    stores timestamps rounded to milliseconds. This means timestamps jitter
    by some amount around the intended timestamp. mpv can correct the timestamps
    based on the framerate value stored in the file: the timestamp is rounded
@@ -2140,7 +2244,7 @@ Demuxer
        Play a raw YUV sample::

            mpv sample-720x576.yuv --demuxer=rawvideo \
-            --demuxer-rawvideo=w=720:h=576
+            --demuxer-rawvideo-w=720 --demuxer-rawvideo-h=576

 ``--demuxer-rawvideo-format=<value>``
    Color space (fourcc) in hex or string for ``--demuxer=rawvideo``
@@ -2158,6 +2262,20 @@ Demuxer
 ``--demuxer-rawvideo-size=<value>``
    Frame size in bytes when using ``--demuxer=rawvideo``.

+``--demuxer-max-packets=<packets>``, ``--demuxer-max-bytes=<bytes>``
+    This controls how much the demuxer is allowed to buffer ahead. The demuxer
+    will normally try to read ahead as much as necessary, or as much is
+    requested with ``--demuxer-readahead-secs``. The ``--demuxer-max-...``
+    options can be used to restrict the maximum readahead. This limits excessive
+    readahead in case of broken files or desynced playback. The demuxer will
+    stop reading additional packets as soon as one of the limits is reached.
+    (The limits still can be slightly overstepped due to technical reasons.)
+
+    Set these limits highher if you get a packet queue overflow warning, and
+    you think normal playback would be possible with a larger packet queue.
+
+    See ``--list-options`` for defaults and value range.
+
 ``--demuxer-thread=<yes|no>``
    Run the demuxer in a separate thread, and let it prefetch a certain amount
    of packets (default: yes). Having this enabled may lead to smoother
@@ -2176,21 +2294,11 @@ Demuxer
    (This value tends to be fuzzy, because many file formats don't store linear
    timestamps.)

-``--demuxer-readahead-packets=<packets>``
-    If ``--demuxer-thread`` is enabled, this controls how much the demuxer
-    should buffer ahead. As long as the number of packets in the packet queue
-    doesn't exceed ``--demuxer-readahead-packets``, and the total number of
-    bytes doesn't exceed ``--demuxer-readahead-bytes``, the thread keeps
-    reading ahead.
-
-    Note that if you set these options near the maximum, you might get a
-    packet queue overflow warning.
-
-    See ``--list-options`` for defaults and value range.
-
-``--demuxer-readahead-bytes=<bytes>``
-    See ``--demuxer-readahead-packets``.
-
+``--force-seekable=<yes|no>``
+    If the player thinks that the media is not seekable (e.g. playing from a
+    pipe, or it's a http stream with a server that doesn't support range
+    requests), seeking will be disabled. This option can forcibly enable it.
+    For seeks within the cache, there's a good chance of success.

 Input
 -----
@@ -2247,8 +2355,6 @@ Input
    This can also specify a direct file descriptor with ``fd://N`` (UNIX only).
    In this case, JSON replies will be written if the FD is writable.

-    See also ``--slave-broken``.
-
    .. note::

        When the given file is a FIFO mpv opens both ends, so you can do several
@@ -2562,14 +2668,19 @@ Screenshot

    Note that not all formats are supported.

-    Default: ``yes``.
+    Default: ``no``.
+
+``--screenshot-high-bit-depth=<yes|no>``
+    If possible, write screenshots with a bit depth similar to the source
+    video (default: yes). This is interesting in particular for PNG, as this
+    sometimes triggers writing 16 bit PNGs with huge file sizes.

 ``--screenshot-template=<template>``
    Specify the filename template used to save screenshots. The template
    specifies the filename without file extension, and can contain format
    specifiers, which will be substituted when taking a screenshot.
-    By default the template is ``shot%n``, which results in filenames like
-    ``shot0012.png`` for example.
+    By default the template is ``mpv-shot%n``, which results in filenames like
+    ``mpv-shot0012.png`` for example.

    The template can start with a relative or absolute path, in order to
    specify a directory location where screenshots should be saved.
@@ -2612,7 +2723,7 @@ Screenshot

        .. note::

-            This is a simple way for getting unique per-frame timestamps. Frame
+            This is a simple way for getting unique per-frame timestamps. (Frame
            numbers would be more intuitive, but are not easily implementable
            because container formats usually use time stamps for identifying
            frames.)
@@ -2651,9 +2762,26 @@ Screenshot
    ``%%``
        Replaced with the ``%`` character itself.

+``--screenshot-directory=<path>``
+    Store screenshots in this directory. This path is joined with the filename
+    generated by ``--screenshot-template``. If the template filename is already
+    absolute, the directory is ignored.
+
+    If the directory does not exist, it is created on the first screenshot. If
+    it is not a directory, an error is generated when trying to write a
+    screenshot.
+
+    This option is not set by default, and thus will write screenshots to the
+    directory from which mpv was started. In pseudo-gui mode
+    (see `PSEUDO GUI MODE`_), this is set to the desktop.
+
 ``--screenshot-jpeg-quality=<0-100>``
    Set the JPEG quality level. Higher means better quality. The default is 90.

+``--screenshot-jpeg-source-chroma=<yes|no>``
+    Write JPEG files with the same chroma subsampling as the video
+    (default: yes). If disabled, the libjpeg default is used.
+
 ``--screenshot-png-compression=<0-9>``
    Set the PNG compression level. Higher means better compression. This will
    affect the file size of the written screenshot file and the time it takes
@@ -2984,13 +3112,13 @@ Cache
    seeking, such as MP4.

    Note that half the cache size will be used to allow fast seeking back. This
-    is also the reason why a full cache is usually reported as 50% full. The
-    cache fill display does not include the part of the cache reserved for
-    seeking back. Likewise, when starting a file the cache will be at 100%,
-    because no space is reserved for seeking back yet.
+    is also the reason why a full cache is usually not reported as 100% full.
+    The cache fill display does not include the part of the cache reserved for
+    seeking back. The actual maximum percentage will usually be the ratio
+    between readahead and backbuffer sizes.

 ``--cache-default=<kBytes|no>``
-    Set the size of the cache in kilobytes (default: 150000 KB). Using ``no``
+    Set the size of the cache in kilobytes (default: 75000 KB). Using ``no``
    will not automatically enable the cache e.g. when playing from a network
    stream. Note that using ``--cache`` will always override this option.

@@ -3010,6 +3138,12 @@ Cache
    on the situation, either of these might be slower than the other method.
    This option allows control over this.

+``--cache-backbuffer=<kBytes>``
+    Size of the cache back buffer (default: 75000 KB). This will add to the total
+    cache size, and reserved the amount for seeking back. The reserved amount
+    will not be used for readahead, and instead preserves already read data to
+    enable fast seeking back.
+
 ``--cache-file=<TMP|path>``
    Create a cache file on the filesystem.

@@ -3124,7 +3258,7 @@ Network
    network transport when playing ``rtsp://...`` URLs. The value ``lavf``
    leaves the decision to libavformat.

-``--hls-bitrate=<no|min|max>``
+``--hls-bitrate=<no|min|max|<rate>>``
    If HLS streams are played, this option controls what streams are selected
    by default. The option allows the following parameters:

@@ -3133,6 +3267,9 @@ Network
    :min:       Pick the streams with the lowest bitrate.
    :max:       Same, but highest bitrate. (Default.)

+    Additionally, if the option is a number, the stream with the highest rate
+    equal or below the option value is selected.
+
    The bitrate as used is sent by the server, and there's no guarantee it's
    actually meaningful.

@@ -3258,6 +3395,72 @@ Miscellaneous
    out. This delay in reaction time to sudden A/V offsets should be the only
    side-effect of turning this option on, for all sound drivers.

+``--video-sync=<audio|...>``
+    How the player synchronizes audio and video.
+
+    The modes starting with ``display-`` try to output video frames completely
+    synchronously to the display, using the detected display vertical refresh
+    rate as a hint how fast frames will be displayed on average. These modes
+    change video speed slightly to match the display. See ``--video-sync-...``
+    options for fine tuning. The robustness of this mode is further reduced by
+    making a some idealized assumptions, which may not always apply in reality.
+    Behavior can depend on the VO and the system's video and audio drivers.
+    Media files must use constant framerate. Section-wise VFR might work as well
+    with some container formats (but not e.g. mkv). If the sync code detects
+    severe A/V desync, or the framerate cannot be detected, the player
+    automatically reverts to ``audio`` mode for some time or permanently.
+
+    The modes with ``desync`` in their names do not attempt to keep audio/video
+    in sync. They will slowly (or quickly) desync, until e.g. the next seek
+    happens. These modes are meant for testing, not serious use.
+
+    :audio:             Time video frames to audio. This is the most robust
+                        mode, because the player doesn't have to assume anything
+                        about how the display behaves. The disadvantage is that
+                        it can lead to occasional frame drops or repeats. If
+                        audio is disabled, this uses the system clock. This is
+                        the default mode.
+    :display-resample:  Resample audio to match the video. This mode will also
+                        try to adjust audio speed to compensate for other drift.
+                        (This means it will play the audio at a different speed
+                        every once in a while to reduce the A/V difference.)
+    :display-resample-vdrop:  Resample audio to match the video. Drop video
+                        frames to compensate for drift.
+    :display-resample-desync: Like the previous mode, but no A/V compensation.
+    :display-vdrop:     Drop or repeat video frames to compensate desyncing
+                        video. (Although it should have the same effects as
+                        ``audio``, the implementation is very different.)
+    :display-desync:    Sync video to display, and let audio play on its own.
+    :desync:            Sync video according to system clock, and let audio play
+                        on its own.
+
+``--video-sync-max-video-change=<value>``
+    Maximum speed difference in percent that is applied to video with
+    ``--video-sync=display-...`` (default: 1). Display sync mode will be
+    disabled if the monitor and video refresh way do not match within the
+    given range. It tries multiples as well: playing 30 fps video on a 60 Hz
+    screen will duplicate every second frame. Playing 24 fps video on a 60 Hz
+    screen will play video in a 2-3-2-3-... pattern.
+
+    The default settings are not loose enough to speed up 23.976 fps video to
+    25 fps. We consider the pitch change too extreme to allow this behavior
+    by default. Set this option to a value of ``5`` to enable it.
+
+    Note that in the ``--video-sync=display-resample`` mode, audio speed will
+    additionally be changed by a small amount if necessary for A/V sync. See
+    ``--video-sync-max-audio-change``.
+
+``--video-sync-max-audio-change=<value>``
+    Maximum *additional* speed difference in percent that is applied to audio
+    with ``--video-sync=display-...`` (default: 0.125). Normally, the player
+    play the audio at the speed of the video. But if the difference between
+    audio and video position is too high, e.g. due to drift or other timing
+    errors, it will attempt to speed up or slow down audio by this additional
+    factor. Too low values could lead to video frame dropping or repeating if
+    the A/V desync cannot be compensated, too high values could lead to chaotic
+    frame dropping due to the audio "overshooting" and skipping multiple video
+    frames before the sync logic can react.
+
 ``--mf-fps=<value>``
    Framerate used when decoding from multiple PNG or JPEG files with ``mf://``
    (default: 1).
@@ -3305,7 +3508,7 @@ Miscellaneous
    you should not need to change this option.

    :decoder: Use decoder reordering functionality. Unlike in classic MPlayer
-              and mplayer2, this includes a dTS fallback. (Default.)
+              and mplayer2, this includes a DTS fallback. (Default.)
    :sort:    Maintain a buffer of unused pts values and use the lowest value
              for the frame.
    :auto:    Try to pick a working mode from the ones above automatically.
@@ -3313,25 +3516,7 @@ Miscellaneous
    You can also try to use ``--no-correct-pts`` for files with completely
    broken timestamps.

-``--media-title=<string>``
+``--force-media-title=<string>``
    Force the contents of the ``media-title`` property to this value. Useful
    for scripts which want to set a title, without overriding the user's
    setting in ``--title``.
-
-``--slave-broken``
-    Switches on the old slave mode. This is for testing only, and incompatible
-    to the removed ``--slave`` switch.
-
-    .. attention::
-        Changes incompatible to slave mode applications have been made. In
-        particular, the status line output was changed, which is used by some
-        applications to determine the current playback position. This switch
-        has been renamed to prevent these applications from working with this
-        version of mpv, because it would lead to buggy and confusing behavior
-        only. Moreover, the slave mode protocol is so horribly bad that it
-        should not be used for new programs, nor should existing programs
-        attempt to adapt to the changed output and use the ``--slave-broken``
-        switch. Instead, a new, saner protocol should be developed (and will be,
-        if there is enough interest).
-
-        This affects most third-party GUI frontends.
@@ -5,7 +5,7 @@ The On Screen Controller (short: OSC) is a minimal GUI integrated with mpv to
 offer basic mouse-controllability. It is intended to make interaction easier
 for new users and to enable precise and direct seeking.

-The OSC is enabled by default if mpv was compiled with lua support. It can be
+The OSC is enabled by default if mpv was compiled with Lua support. It can be
 disabled entirely using the ``--osc=no`` option.

 Using the OSC
@@ -169,34 +169,35 @@ Configurable Options

 ``showwindowed``
    | Default: yes
-    | Show OSC when windowed?
+    | Enable the OSC when windowed

 ``showfullscreen``
    | Default: yes
-    | Show OSC when fullscreen?
+    | Enable the OSC when fullscreen

 ``scalewindowed``
-    | Default: 1
-    | Scaling of the controller when windowed
+    | Default: 1.0
+    | Scale factor of the OSC when windowed

 ``scalefullscreen``
-    | Default: 1
-    | Scaling of the controller when fullscreen
+    | Default: 1.0
+    | Scale factor of the OSC when fullscreen

 ``scaleforcedwindow``
-    | Default: 2
-    | Scaling of the controller when rendered on a forced (dummy) window
+    | Default: 2.0
+    | Scale factor of the OSC when rendered on a forced (dummy) window

 ``vidscale``
    | Default: yes
-    | Scale the controller with the video?
+    | Scale the OSC with the video
+    | ``no`` tries to keep the OSC size constant as much as the window size allows

 ``valign``
    | Default: 0.8
    | Vertical alignment, -1 (top) to 1 (bottom)

 ``halign``
-    | Default: 0
+    | Default: 0.0
    | Horizontal alignment, -1 (left) to 1 (right)

 ``boxalpha``
@@ -234,6 +235,14 @@ Configurable Options
    | Default: slider
    | Sets the style of the seekbar, slider (diamond marker) or bar (fill)

+``timetotal``
+    | Default: no
+    | Show total time instead of time remaining
+
+``timems``
+    | Default: no
+    | Display timecodes with milliseconds
+
 Script Commands
 ~~~~~~~~~~~~~~~

@@ -17,6 +17,10 @@ normal filter parameters.

    To get a full list of available video filters, see ``--vf=help``.

+    Also, keep in mind that most actual filters are available via the ``lavfi``
+    wrapper, which gives you access to most of libavfilter's filters. This
+    includes all filters that have been ported from MPlayer to libavfilter.
+
 Video filters are managed in lists. There are a few commands to manage the
 filter list.

@@ -97,7 +101,7 @@ Available filters are:

    All parameters are optional.

-    ``<w>,<h>``
+    ``<w>:<h>``
        scaled width/height (default: original width/height)

        :0:      scaled d_width/d_height
@@ -152,7 +156,7 @@ Available filters are:
    number (1.33). Alternatively, you may specify the exact display width and
    height desired. Note that this filter does *not* do any scaling itself; it
    just affects what later scalers (software or hardware) will do when
-    auto-scaling to correct aspect.
+    auto-scaling to the correct aspect.

    ``<w>,<h>``
        New display width and height.
@@ -198,7 +202,7 @@ Available filters are:
    ``<aspect>``
        Force an aspect ratio.

-``format[=fmt[:outfmt]]``
+``format=fmt=<value>:colormatrix=<value>:...``
    Restricts the color space for the next filter without doing any conversion.
    Use together with the scale filter for a real conversion.

@@ -409,41 +413,6 @@ Available filters are:
            ``'--vf=lavfi=yadif:o="threads=2,thread_type=slice"'``
                forces a specific threading configuration.

-``noise[=<strength>[:averaged][:pattern][:temporal][:uniform][:hq]``
-    Adds noise.
-
-    ``strength``
-        Set the noise for all components. If you want different strength
-        values for luma and chroma, use libavfilter's noise filter directly
-        (using ``--vf=lavfi=[noise=...]``), or tell the libavfilter developers
-        to stop being stupid.
-
-    ``averaged``
-        averaged temporal noise (smoother, but a lot slower)
-
-    ``pattern``
-        mix random noise with a (semi)regular pattern
-
-    ``temporal``
-        temporal noise (noise pattern changes between frames)
-
-    ``uniform``
-        uniform noise (Gaussian otherwise)
-
-``hqdn3d[=luma_spatial:chroma_spatial:luma_tmp:chroma_tmp]``
-    This filter aims to reduce image noise producing smooth images and making
-    still images really still (This should enhance compressibility.).
-
-    ``<luma_spatial>``
-        spatial luma strength (default: 4)
-    ``<chroma_spatial>``
-        spatial chroma strength (default: 3)
-    ``<luma_tmp>``
-        luma temporal strength (default: 6)
-    ``<chroma_tmp>``
-        chroma temporal strength (default:
-        ``luma_tmp*chroma_spatial/luma_spatial``)
-
 ``eq[=gamma:contrast:brightness:saturation:rg:gg:bg:weight]``
    Software equalizer that uses lookup tables (slow), allowing gamma correction
    in addition to simple brightness and contrast adjustment. The parameters are
@@ -470,23 +439,6 @@ Available filters are:
        and just plain white. A value of 0.0 turns the gamma correction all
        the way down while 1.0 leaves it at its full strength (default: 1.0).

-``unsharp[=lx:ly:la:cx:cy:ca]``
-    unsharp mask / Gaussian blur
-
-    ``l`` is for the luma component, ``c`` for the chroma component. ``x``/``y``
-    is the filter size. ``a`` is the amount.
-
-    ``lx``, ``ly``, ``cx``, ``cy``
-        width and height of the matrix, odd sized in both directions (min =
-        3:3, max = 13:11 or 11:13, usually something between 3:3 and 7:7)
-
-    ``la``, ``ca``
-        Relative amount of sharpness/blur to add to the image (a sane range
-        should be -1.5-1.5).
-
-        :<0: blur
-        :>0: sharpen
-
 ``pullup[=jl:jr:jt:jb:sb:mp]``
    Pulldown reversal (inverse telecine) filter, capable of handling mixed
    hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps progressive
@@ -518,7 +470,7 @@ Available filters are:
        video. The main purpose of setting ``mp`` to a chroma plane is to reduce
        CPU load and make pullup usable in realtime on slow machines.

-``yadif=[mode[:enabled=yes|no]]``
+``yadif=[mode:interlaced-only]``
    Yet another deinterlacing filter

    ``<mode>``
@@ -527,10 +479,10 @@ Available filters are:
        :frame-nospatial: Like ``frame`` but skips spatial interlacing check.
        :field-nospatial: Like ``field`` but skips spatial interlacing check.

-    ``<enabled>``
-        :yes: Filter is active (default).
-        :no:  Filter is not active, but can be activated with the ``D`` key
-              (or any other key that toggles the ``deinterlace`` property).
+    ``<interlaced-only>``
+        :no:  Deinterlace all frames (default).
+        :yes: Only deinterlace frames marked as interlaced (default if this
+              filter is inserted via ``deinterlace`` property).

    This filter, is automatically inserted when using the ``D`` key (or any
    other key that toggles the ``deinterlace`` property or when using the
@@ -545,22 +497,6 @@ Available filters are:
    when inserting yadif with ``--vf``, so using the above methods is
    recommended.

-``delogo[=x:y:w:h:t:show]``
-    Suppresses a TV station logo by a simple interpolation of the surrounding
-    pixels. Just set a rectangle covering the logo and watch it disappear (and
-    sometimes something even uglier appear - your mileage may vary).
-
-    ``<x>,<y>``
-        top left corner of the logo
-    ``<w>,<h>``
-        width and height of the cleared rectangle
-    ``<t>``
-        Thickness of the fuzzy edge of the rectangle (added to ``w`` and
-        ``h``). When set to -1, a green rectangle is drawn on the screen to
-        simplify finding the right ``x``,``y``,``w``,``h`` parameters.
-    ``show``
-        Draw a rectangle showing the area defined by x/y/w/h.
-
 ``sub=[=bottom-margin:top-margin]``
    Moves subtitle rendering to an arbitrary point in the filter chain, or force
    subtitle rendering in the video filter as opposed to using video output OSD
@@ -669,6 +605,8 @@ Available filters are:
    Loads an external library to filter the image. The library interface
    is the ``vf_dlopen`` interface specified using ``libmpcodecs/vf_dlopen.h``.

+    .. warning:: This filter is deprecated.
+
    ``dll=<library>``
        Specify the library to load. This may require a full file system path
        in some cases. This argument is required.
@@ -778,7 +716,7 @@ Available filters are:
 ``vapoursynth-lazy``
    The same as ``vapoursynth``, but doesn't load Python scripts. Instead, a
    custom backend using Lua and the raw VapourSynth API is used. The syntax
-    is completely different, and absolutely no conveniencve features are
+    is completely different, and absolutely no convenience features are
    provided. There's no type checking either, and you can trigger crashes.

    .. admonition:: Example:
@@ -820,6 +758,10 @@ Available filters are:
            depends on the GPU hardware, the GPU drivers, driver bugs, and
            mpv bugs.

+    ``<interlaced-only>``
+        :no:  Deinterlace all frames.
+        :yes: Only deinterlace frames marked as interlaced (default).
+
 ``vdpaupp``
    VDPAU video post processing. Works with ``--vo=vdpau`` and ``--vo=opengl``
    only. This filter is automatically inserted if deinterlacing is requested
@@ -864,12 +806,21 @@ Available filters are:
    ``pullup``
        Try to apply inverse telecine, needs motion adaptive temporal
        deinterlacing.
+    ``interlaced-only=<yes|no>``
+        If ``yes`` (default), only deinterlace frames marked as interlaced.
    ``hqscaling=<0-9>``
        0
            Use default VDPAU scaling (default).
        1-9
            Apply high quality VDPAU scaling (needs capable hardware).

+``vdpaurb``
+    VDPAU video read back. Works with ``--vo=vdpau`` and ``--vo=opengl`` only.
+    This filter will read back frames decoded by VDPAU so that other filters,
+    which are not normally compatible with VDPAU, can be used like normal.
+    This filter must be specified before ``vdpaupp`` in the filter chain if
+    ``vdpaupp`` is used.
+
 ``buffer=<num>``
    Buffer ``<num>`` frames in the filter chain. This filter is probably pretty
    useless, except for debugging. (Note that this won't help smoothing out
@@ -69,11 +69,11 @@ Available video output drivers are:
    ``no-colorkey``
        Disables color-keying.

-``x11`` (X11 only)
-    Shared memory video output driver without hardware acceleration that works
-    whenever X11 is present.
-
-    .. note:: This is a fallback only, and should not be normally used.
+    ``buffers=<number>``
+        Number of image buffers to use for the internal ringbuffer (default: 2).
+        Increasing this will use more memory, but might help with the X server
+        not responding quickly enough if video FPS is close to or higher than
+        the display refresh rate.

 ``vdpau`` (X11 only)
    Uses the VDPAU interface to display and optionally also decode video.
@@ -290,6 +290,26 @@ Available video output drivers are:
    color space conversion and chroma upsampling is generally in the hand of
    the hardware decoder APIs.

+    ``opengl`` makes use of FBOs by default. Sometimes you can achieve better
+    quality or performance by changing the ``fbo-format`` suboption to
+    ``rgb16f``, ``rgb32f`` or ``rgb``. Known problems include Mesa/Intel not
+    accepting ``rgb16``, Mesa sometimes not being compiled with float texture
+    support, and some OS X setups being very slow with ``rgb16`` but fast
+    with ``rgb32f``. If you have problems, you can also try passing the
+    ``dumb-mode=yes`` sub-option.
+
+    ``dumb-mode=<yes|no>``
+        This mode is extremely restricted, and will disable most extended
+        OpenGL features. This includes high quality scalers and custom
+        shaders!
+
+        It is intended for hardware that does not support FBOs (including GLES,
+        which supports it insufficiently), or to get some more performance out
+        of bad or old hardware.
+
+        This mode is forced automatically if needed, and this option is mostly
+        useful for debugging.
+
    ``scale=<filter>``

        ``bilinear``
@@ -336,6 +356,9 @@ Available video output drivers are:
            exchange for adding some blur. This filter is good at temporal
            interpolation, and also known as "smoothmotion" (see ``tscale``).

+        ``custom``
+            A user-defined custom shader (see ``scale-shader``).
+
        There are some more filters, but most are not as useful. For a complete
        list, pass ``help`` as value, e.g.::

@@ -452,6 +475,11 @@ Available video output drivers are:
        Unfortunately, this can lead to flicker on LCD displays, since these
        have a high reaction time.

+    ``temporal-dither-period=<1-128>``
+        Determines how often the dithering pattern is updated when
+        ``temporal-dither`` is in use. 1 (the default) will update on every
+        video frame, 2 on every other frame, etc.
+
    ``debug``
        Check for OpenGL errors, i.e. call ``glGetError()``. Also request a
        debug OpenGL context (which does nothing with current graphics drivers
@@ -494,10 +522,15 @@ Available video output drivers are:
        for ``tscale`` are separable convolution filters (use ``tscale=help``
        to get a list). The default is ``oversample``.

-        Note that the maximum supported filter radius is currently 3, and that
-        using filters with larger radius may introduce issues when pausing or
-        framestepping, proportional to the radius used. It is recommended to
-        stick to a radius of 1 or 2.
+        Note that the maximum supported filter radius is currently 3, due to
+        limitations in the number of video textures that can be loaded
+        simultaneously.
+
+    ``tscale-clamp``
+        Clamp the ``tscale`` filter kernel's value range to [0-1]. This reduces
+        excessive ringing artifacts in the temporal domain (which typically
+        manifest themselves as short flashes or fringes of black, mostly
+        around moving edges) in exchange for potentially adding more blur.

    ``dscale-radius``, ``cscale-radius``, ``tscale-radius``, etc.
        Set filter parameters for ``dscale``, ``cscale`` and ``tscale``,
@@ -506,10 +539,8 @@ Available video output drivers are:
        See the corresponding options for ``scale``.

    ``linear-scaling``
-        Scale in linear light. This is automatically enabled if
-        ``target-prim``, ``target-trc``, ``icc-profile`` or
-        ``sigmoid-upscaling`` is set. It should only be used with a
-        ``fbo-format`` that has at least 16 bit precision.
+        Scale in linear light. It should only be used with a ``fbo-format``
+        that has at least 16 bit precision.

    ``fancy-downscaling``
        When using convolution based filters, extend the filter size
@@ -519,9 +550,92 @@ Available video output drivers are:
        feature doesn't work correctly with different scale factors in
        different directions.

+    ``pre-shaders=<files>``, ``post-shaders=<files>``, ``scale-shader=<file>``
+        Custom GLSL fragment shaders.
+
+        pre-shaders (list)
+            These get applied after conversion to RGB and before linearization
+            and upscaling. Operates on non-linear RGB (same as input). This is
+            the best place to put things like sharpen filters.
+        scale-shader
+            This gets used instead of scale/cscale when those options are set
+            to ``custom``. The colorspace it operates on depends on the values
+            of ``linear-scaling`` and ``sigmoid-upscaling``, so no assumptions
+            should be made here.
+        post-shaders (list)
+            These get applied after upscaling and subtitle blending (when
+            ``blend-subtitles`` is enabled), but before color management.
+            Operates on linear RGB if ``linear-scaling`` is in effect,
+            otherwise non-linear RGB. This is the best place for colorspace
+            transformations (eg. saturation mapping).
+
+        These files must define a function with the following signature::
+
+            vec4 sample(sampler2D tex, vec2 pos, vec2 tex_size)
+
+        The meanings of the parameters are as follows:
+
+        sampler2D tex
+            The source texture for the shader.
+        vec2 pos
+            The position to be sampled, in coordinate space [0-1].
+        vec2 tex_size
+            The size of the texture, in pixels. This may differ from image_size,
+            eg. for subsampled content or for post-shaders.
+
+        In addition to these parameters, the following uniforms are also
+        globally available:
+
+        float random
+            A random number in the range [0-1], different per frame.
+        int frame
+            A simple count of frames rendered, increases by one per frame and
+            never resets (regardless of seeks).
+        vec2 image_size
+            The size in pixels of the input image.
+
+        For example, a shader that inverts the colors could look like this::
+
+            vec4 sample(sampler2D tex, vec2 pos, vec2 tex_size)
+            {
+                vec4 color = texture(tex, pos);
+                return vec4(1.0 - color.rgb, color.a);
+            }
+
+    ``deband``
+        Enable the debanding algorithm. This greatly reduces the amount of
+        visible banding, blocking and other quantization artifacts, at the
+        expensive of very slightly blurring some of the finest details. In
+        practice, it's virtually always an improvement - the only reason to
+        disable it would be for performance.
+
+    ``deband-iterations=<1..16>``
+        The number of debanding steps to perform per sample. Each step reduces
+        a bit more banding, but takes time to compute. Note that the strength
+        of each step falls off very quickly, so high numbers are practically
+        useless. (Default 4)
+
+        If the performance hit of debanding is too great, you can reduce this
+        to 2 or 1 with marginal visual quality loss.
+
+    ``deband-threshold=<0..4096>``
+        The debanding filter's cut-off threshold. Higher numbers increase the
+        debanding strength dramatically but progressively diminish image
+        details. (Default 64)
+
+    ``deband-range=<1..64>``
+        The debanding filter's initial radius. The radius increases linearly
+        for each iteration. A higher radius will find more gradients, but
+        a lower radius will smooth more aggressively. (Default 8)
+
+    ``deband-grain=<0..4096>``
+        Add some extra noise to the image. This significantly helps cover up
+        remaining quantization artifacts. Higher numbers add more noise.
+        (Default 48)
+
    ``sigmoid-upscaling``
        When upscaling, use a sigmoidal color transform to avoid emphasizing
-        ringing artifacts. This also enables ``linear-scaling``.
+        ringing artifacts. This also implies ``linear-scaling``.

    ``sigmoid-center``
        The center of the sigmoid curve used for ``sigmoid-upscaling``, must
@@ -531,10 +645,6 @@ Available video output drivers are:
        The slope of the sigmoid curve used for ``sigmoid-upscaling``, must
        be a float between 1.0 and 20.0. Defaults to 6.5 if not specified.

-    ``no-npot``
-        Force use of power-of-2 texture sizes. For debugging only.
-        Borders will be distorted due to filtering.
-
    ``glfinish``
        Call ``glFinish()`` before and after swapping buffers (default: disabled).
        Slower, but might help getting better results when doing framedropping.
@@ -555,7 +665,8 @@ Available video output drivers are:
        full screen).
        This may help getting more consistent frame intervals, especially with
        high-fps clips - which might also reduce dropped frames. Typically a
-        value of 1 should be enough since full screen may bypass the DWM.
+        value of ``windowed`` should be enough since full screen may bypass the
+        DWM.

        Windows only.

@@ -573,20 +684,22 @@ Available video output drivers are:
            Cocoa/OS X
        win
            Win32/WGL
-        x11, x11es
-            X11/GLX (the ``es`` variant forces GLES)
+        x11
+            X11/GLX
        wayland
            Wayland/EGL
-        x11egl, x11egles
-            X11/EGL (the ``es`` variant forces GLES)
+        x11egl
+            X11/EGL
+
+    ``es``
+        Force or prefer GLES2/3 over desktop OpenGL, if supported.

    ``fbo-format=<fmt>``
        Selects the internal format of textures used for FBOs. The format can
-        influence performance and quality of the video output. (FBOs are not
-        always used, and typically only when using extended scalers.)
+        influence performance and quality of the video output.
        ``fmt`` can be one of: rgb, rgba, rgb8, rgb10, rgb10_a2, rgb16, rgb16f,
        rgb32f, rgba12, rgba16, rgba16f, rgba32f.
-        Default: rgba.
+        Default: rgba16.

    ``gamma=<0.1..2.0>``
        Set a gamma value (default: 1.0). If gamma is adjusted in other ways
@@ -660,8 +773,7 @@ Available video output drivers are:
    ``icc-profile=<file>``
        Load an ICC profile and use it to transform linear RGB to screen output.
        Needs LittleCMS 2 support compiled in. This option overrides the
-        ``target-prim`` and ``target-trc`` options. It also enables
-        ``linear-scaling``.
+        ``target-prim``, ``target-trc`` and ``icc-profile-auto`` options.

    ``icc-profile-auto``
        Automatically select the ICC display profile currently specified by
@@ -669,11 +781,14 @@ Available video output drivers are:

        NOTE: Only implemented on OS X and X11

-    ``icc-cache=<file>``
-        Store and load the 3D LUT created from the ICC profile in this file.
+    ``icc-cache-dir=<dirname>``
+        Store and load the 3D LUTs created from the ICC profile in this directory.
        This can be used to speed up loading, since LittleCMS 2 can take a while
-        to create the 3D LUT. Note that this file contains an uncompressed LUT.
-        Its size depends on the ``3dlut-size``, and can be very big.
+        to create a 3D LUT. Note that these files contain uncompressed LUTs.
+        Their size depends on the ``3dlut-size``, and can be very big.
+
+        NOTE: This is not cleaned automatically, so old, unused cache files
+        may stick around indefinitely.

    ``icc-intent=<value>``
        Specifies the ICC intent used for the color transformation (when using
@@ -697,8 +812,8 @@ Available video output drivers are:
        Blend subtitles directly onto upscaled video frames, before
        interpolation and/or color management (default: no). Enabling this
        causes subtitles to be affected by ``icc-profile``, ``target-prim``,
-        ``target-trc``, ``interpolation``, ``gamma`` and ``linear-scaling``.
-        It also increases subtitle performance when using ``interpolation``.
+        ``target-trc``, ``interpolation``, ``gamma`` and ``post-shader``. It
+        also increases subtitle performance when using ``interpolation``.

        The downside of enabling this is that it restricts subtitles to the
        visible portion of the video, so you can't have subtitles exist in the
@@ -744,18 +859,11 @@ Available video output drivers are:

    This is equivalent to::

-        --vo=opengl:scale=spline36:cscale=spline36:dscale=mitchell:dither-depth=auto:fbo-format=rgba16:fancy-downscaling:sigmoid-upscaling
+        --vo=opengl:scale=spline36:cscale=spline36:dscale=mitchell:dither-depth=auto:fancy-downscaling:sigmoid-upscaling:pbo:deband

    Note that some cheaper LCDs do dithering that gravely interferes with
    ``opengl``'s dithering. Disabling dithering with ``dither-depth=no`` helps.

-    Unlike ``opengl``, ``opengl-hq`` makes use of FBOs by default. Sometimes you
-    can achieve better quality or performance by changing the ``fbo-format``
-    suboption to ``rgb16f``, ``rgb32f`` or ``rgb``. Known problems include
-    Mesa/Intel not accepting ``rgb16``, Mesa sometimes not being compiled with
-    float texture support, and some OS X setups being very slow with ``rgb16``
-    but fast with ``rgb32f``.
-
 ``sdl``
    SDL 2.0+ Render video output driver, depending on system with or without
    hardware acceleration. Should work on all platforms supported by SDL 2.0.
@@ -816,6 +924,12 @@ Available video output drivers are:
 ``null``
    Produces no video output. Useful for benchmarking.

+    Usually, it's better to disable video with ``--no-video`` instead.
+
+    ``fps=<value>``
+        Simulate display FPS. This artificially limits how many frames the
+        VO accepts per second.
+
 ``caca``
    Color ASCII art video output driver that works on a text console.

@@ -895,13 +1009,15 @@ Available video output drivers are:
    ``frame-queue-size=<1..100>``
        The maximum count of frames which the frame queue can hold (default: 1)

-    ``frame-drop-mode=<pop|clear>``
+    ``frame-drop-mode=<pop|clear|block>``
        Select the behavior when the frame queue is full.

        pop
-            Drop the oldest frame in the frame queue. (default)
+            Drop the oldest frame in the frame queue.
        clear
            Drop all frames in the frame queue.
+        block
+            Wait for a short time, behave like ``clear`` on timeout. (default)

    This also supports many of the suboptions the ``opengl`` VO has. Run
    ``mpv --vo=opengl-cb:help`` for a list.
@@ -921,6 +1037,11 @@ Available video output drivers are:
        selected layer, to handle the window background and OSD. Actual video
        rendering will happen on the layer above the selected layer.

+    ``background=<yes|no>``
+        Whether to render a black background behind the video (default: no).
+        Normally it's better to kill the console framebuffer instead, which
+        gives better performance.
+
 ``drm`` (Direct Rendering Manager)
    Video output driver using Kernel Mode Setting / Direct Rendering Manager.
    Does not support hardware acceleration. Should be used when one doesn't
@@ -933,3 +1054,7 @@ Available video output drivers are:
    ``devpath=<filename>``
        Path to graphic card device.
        (default: /dev/dri/card0)
+
+    ``mode=<number>``
+        Mode ID to use (resolution, bit depth and frame rate).
+        (default: 0)
@@ -49,8 +49,10 @@ Input
 * Classic LIRC support was removed. Install remotes as input devices instead.
  This way they will send X11 key events to the mpv window, which can be bound
  using the normal ``input.conf``.
+  Also see: http://github.com/mpv-player/mpv/wiki/IR-remotes
 * Joystick support was removed. It was considered useless and was the cause
  of some problems (e.g. a laptop's accelerator being recognized as joystick).
+* Support for relative seeking by percentage.

 Audio
 ~~~~~
@@ -92,6 +94,8 @@ Video

 * Image subtitles (DVDs etc.) are rendered in color and use more correct
  positioning (color for image subs can be disabled with ``--sub-gray``).
+* Support for the X11 video output is removed, since it was considered
+  deprecated. SDL video output can still be used as a fallback.

 OSD and terminal
 ~~~~~~~~~~~~~~~~
@@ -275,7 +279,7 @@ Command Line Switches
    ``-xineramascreen``         ``--screen`` (different values)
    ``-xy W``                   ``--autofit=W``
    ``-zoom``                   Inverse available as ``--video-unscaled``
-    ``dvdnav://``               ``dvdnav://menu``
+    ``dvdnav://``               Removed.
    ``dvd://1``                 ``dvd://0`` (0-based offset)
    =========================== ========================================

@@ -395,7 +399,8 @@ Slave mode

  Assuming the system supports ``/dev/stdin``.

-  (The option was readded in 0.5.1 and sets exactly these options.)
+  (The option was added back in 0.5.1 and sets exactly these options. It was
+  removed in 0.10.x again.)

 * A JSON RPC protocol giving access to the client API is also supported. See
  `JSON IPC`_ for more information.
@@ -1,64 +1,106 @@
 Release Policy
 ==============

-Every few months, a new release is cut off of the master branch and is assigned
-a 0.X.0 version number.
+Once or twice a month, a new release is cut off of the master branch and is
+assigned a 0.X.Y version number, where X is incremented each time a release
+contains breaking changes, such as changed options or added/removed features,
+and Y is incremented if a release contains only bugfixes and other minor
+changes.

-As part of the maintenance process, minor releases are made, which are assigned
-0.X.Y version numbers. Minor releases contain bug fixes only. They never merge
-the master branch, and no features are added to it. Only the latest release is
-maintained.
+There is only one release branch that keeps track of the latest version and
+will not be maintained separately.

-The goal of releases is to provide stability and an unchanged base for the sake
-of Linux distributions. If you want the newest features, just use the master
-branch, which is stable most of the time, except sometimes, when it's not.
+The goal of releases is to provide Linux distributions with something to
+package. If you want the newest features, just use the master branch.
+We try our best to keep it deployable at all times.

 Releases other than the latest release are unsupported and unmaintained.

 Release procedure
 -----------------

- Create branch release/0.X or cherry-pick commits to the relevant branch.
+- Merge master into branch release/current.

- Create and/or update the ``RELEASE_NOTES`` file.
+- Create and/or update the `RELEASE_NOTES` file.

- Create and/or update the ``VERSION`` file.
+- Create and/or update the `VERSION` file.

- Update ``DOCS/client-api-changes.rst`` (on major releases).
+- Update `DOCS/client-api-changes.rst` (on major releases).

 - Create tag v0.X.Y.

 - Push branch and tag to GitHub.

- Create a new GitHub release using the content of ``RELEASE_NOTES`` related to
+- Create a new GitHub release using the content of `RELEASE_NOTES` related to
  the new version.

 Release notes template
 ----------------------

-Here is a template that can be used for writing the ``RELEASE_NOTES`` file.
+Here is a template that can be used for writing the `RELEASE_NOTES` file.

 ```markdown
 Release 0.X.Y
 =============

-Changes
-------
+Features
+--------

- List of changes.
+New
+~~~

-Bug fixes
---------
+- List of new features

- List of bug fixes.
+Removed
+~~~~~~~

-New features
------------
+- List of removed features

- List of new features.
+Deprecated
+~~~~~~~~~~
+
+- List of deprecated features
+
+Behavior
+--------
+
+- List of user-visible changes in behavior
+
+Options and Commands
+--------------------
+
+Added
+~~~~~
+
+- List of added options and commands
+
+Changed
+~~~~~~~
+
+- List of changed options and commands
+
+Renamed
+~~~~~~~
+
+- List of renamed options and commands
+
+Deprecated
+~~~~~~~~~~
+
+- List of deprecated options and commands
+
+Removed
+~~~~~~~
+
+- List of removed options and commands
+
+Fixes and Minor Enhancements
+----------------------------
+
+- List of fixes and minor enhancements

 This listing is not complete. There are many more bug fixes and changes. The
-complete change log can be viewed by running ``git log <start>..<end>`` in
+complete change log can be viewed by running `git log <start>..<end>` in
 the git repository.
 ```

@@ -66,6 +108,6 @@ Note that the "Release 0.X.Y" title should be removed when creating a new GitHub
 release.

 When creating a new point release its changes should be added on top of the
-``RELEASE_NOTES`` file (with the appropriate title) so that all the changes in
-the current 0.X branch will be included. This way the ``RELEASE_NOTES`` file
+`RELEASE_NOTES` file (with the appropriate title) so that all the changes in
+the current 0.X branch will be included. This way the `RELEASE_NOTES` file
 can be used by distributors as changelog for point releases too.
@@ -75,8 +75,7 @@ talloc.h & talloc.c:
    allocation call will never return NULL.

    One very useful feature of talloc is fast tracking of memory leaks. ("Fast"
-    as in it doesn't require valgrind.) You can enable it by passing the option
-    --leak-report as first parameter, or better, setting the
+    as in it doesn't require valgrind.) You can enable it by setting the
    MPV_LEAK_REPORT environment variable to "1":
        export MPV_LEAK_REPORT=1
    This will list all unfree'd allocations on exit.
@@ -1,7 +1,7 @@
 waf build system overview
 =========================

-mpv's new build system is based on waf and it should completly replace the
+mpv's new build system is based on waf and it should completely replace the
 custom ./configure + Makefile based system inherited from MPlayer.

 Goals and the choice of waf
@@ -65,7 +65,7 @@ This defines a feature called ``vdpau`` which can be enabled or disabled by
 the users with configure flags (that's the meaning of ``--``). This feature
 depends on another feature whose name is ``x11``, and the autodetection check
 consists of running ``pkg-config`` and looking for ``vdpau`` with version
-``>= 0.2``. If the check succeds a ``#define HAVE_VDPAU 1`` will be added to
+``>= 0.2``. If the check succeeds a ``#define HAVE_VDPAU 1`` will be added to
 ``config.h``, if not ``#define HAVE_VDPAU 0`` will be added.

 The defines names are automatically prepended with ``HAVE_``, capitalized and
@@ -57,9 +57,6 @@ To build the software you can use `./waf build`: the result of the compilation
 will be located in `build/mpv`. You can use `./waf install` to install mpv
 to the *prefix* after it is compiled.

-NOTE: Using the old build system (with `./old-configure`) should still work,
-but will be removed in a future version of mpv.
-
 Essential dependencies (incomplete list):

 - gcc or clang
@@ -174,11 +171,9 @@ list of changes is located [here][mplayer-changes].
 Most activity happens on the IRC channel and the github issue tracker. The
 mailing lists are mostly unused.

- - **Github issue tracker**: [issue tracker][issue-tracker] (report bugs here)
- - **User IRC Channel**: `#mpv-player` on `irc.freenode.net`
- - **Developer IRC Channel**: `#mpv-player-dev` on `irc.freenode.net`
- - **Users Mailing List**: `mpv-users@googlegroups.com` ([Archive / Subscribe][mpv-users]).
- - **Devel Mailing List**: `mpv-devel@googlegroups.com` ([Archive / Subscribe][mpv-devel])
+ - **GitHub issue tracker**: [issue tracker][issue-tracker] (report bugs here)
+ - **User IRC Channel**: `#mpv` on `irc.freenode.net`
+ - **Developer IRC Channel**: `#mpv-devel` on `irc.freenode.net`

 To contact the `mpv` team in private write to `mpv-team@googlegroups.com`. Use
 only if discretion is required.
@@ -187,8 +182,6 @@ only if discretion is required.
 [mpv-build]: https://github.com/mpv-player/mpv-build
 [homebrew-mpv]: https://github.com/mpv-player/homebrew-mpv
 [issue-tracker]:  https://github.com/mpv-player/mpv/issues
-[mpv-users]: https://groups.google.com/forum/?hl=en#!forum/mpv-users
-[mpv-devel]: https://groups.google.com/forum/?hl=en#!forum/mpv-devel
 [ffmpeg_vs_libav]: https://github.com/mpv-player/mpv/wiki/FFmpeg-versus-Libav
 [release-policy]: https://github.com/mpv-player/mpv/blob/master/DOCS/release-policy.md
 [windows_compilation]: https://github.com/mpv-player/mpv/blob/master/DOCS/compile-windows.md
@@ -47,7 +47,7 @@ judge()
        progressive=$((progressive + progressive1))
        undetermined=$((undetermined + undetermined1))
    done <<EOF
-$(testfun "$@")
+$(testfun "$@" | sed 's/:/: /g')
 EOF

    interlaced=$((bff + tff))
@@ -95,18 +95,18 @@ case "$verdict" in
        exit 0
        ;;
    interlaced-tff)
-        judge "$@" --vf-clr --vf-pre=pullup --field-dominance=top
+        judge "$@" --vf-clr --vf-pre=lavfi=\[setfield=tff\],pullup
        case "$verdict" in
            progressive)
                [ -n "$ILDETECT_DRY_RUN" ] || \
-                    $ILDETECT_MPV "$@" --vf-pre=pullup --field-dominance=top
+                    $ILDETECT_MPV "$@" --vf-pre=lavfi=\[setfield=tff\],pullup
                r=$?
                [ $r -eq 0 ] || exit $((r | 16))
                exit 1
                ;;
            *)
                [ -n "$ILDETECT_DRY_RUN" ] || \
-                    $ILDETECT_MPV "$@" --vf-pre=yadif --field-dominance=top
+                    $ILDETECT_MPV "$@" --vf-pre=lavfi=\[setfield=tff\],yadif
                r=$?
                [ $r -eq 0 ] || exit $((r | 16))
                exit 2
@@ -114,18 +114,18 @@ case "$verdict" in
        esac
        ;;
    interlaced-bff)
-        judge "$@" --vf-clr --vf-pre=pullup --field-dominance=bottom
+        judge "$@" --vf-clr --vf-pre=lavfi=\[setfield=bff\],pullup
        case "$verdict" in
            progressive)
                [ -n "$ILDETECT_DRY_RUN" ] || \
-                    $ILDETECT_MPV "$@" --vf-pre=pullup --field-dominance=bottom
+                    $ILDETECT_MPV "$@" --vf-pre=lavfi=\[setfield=bff\],pullup
                r=$?
                [ $r -eq 0 ] || exit $((r | 16))
                exit 1
                ;;
            *)
                [ -n "$ILDETECT_DRY_RUN" ] || \
-                    $ILDETECT_MPV "$@" --vf-pre=yadif --field-dominance=bottom
+                    $ILDETECT_MPV "$@" --vf-pre=lavfi=\[setfield=bff\],yadif
                r=$?
                [ $r -eq 0 ] || exit $((r | 16))
                exit 2
@@ -84,7 +84,7 @@ sub elem_by_hexid {
 # used by elem when setting the 'valname' key
 use constant TYPE_MAP => {
    uint    => 'uint64_t',
-    str     => 'struct bstr',
+    str     => 'char *',
    binary  => 'struct bstr',
    ebml_id => 'uint32_t',
    float   => 'double',
@@ -2,21 +2,16 @@
 -- appropriate deinterlacing filter based on a short section of the
 -- currently playing video.
 --
-- It registers the key-binding ctrl+d, which when pressed, inserts
-- the filter vf=lavfi=idet. After 4 second, it examines the results
-- to determine whether the content progressive or interlaced and the
-- interlacing field dominance. It immediately sets the latter within
-- mpv.
+-- It registers the key-binding ctrl+d, which when pressed, inserts the filters
+-- ``vf=lavfi=idet,pullup,vf=lavfi=idet``. After 4 seconds, it removes these
+-- filters and decides whether the content is progressive, interlaced, or
+-- telecined and the interlacing field dominance.
 --
-- If the content is judged not to be progressive, it may be either
-- interlaced or telecined. To determine which one, the video is
-- seeked back to the detection starting point with the "pullup"
-- filter inserted. After another four seconds, if idet thinks that
-- the pulled up viedo is "progressive", then pullup is the correct
-- filter and this script leaves it in places while removing the idet
-- filter. Otherwise, the content plain interlaced and mpv's deinterlace
-- property is set. This will usually insert the yadif deinterlacing filter.
+-- Based on this information, it may set mpv's ``deinterlace`` property (which
+-- usually inserts the yadif filter), or insert the ``pullup`` filter if the
+-- content is telecined.  It also sets mpv's ``field-dominance`` property.
 --
+-- OPTIONS:
 -- The default detection time may be overridden by adding
 --
 -- --script-opts=autodeint.detect_seconds=<number of seconds>
@@ -37,6 +32,7 @@ require "mp.msg"
 script_name = mp.get_script_name()
 detect_label = string.format("%s-detect", script_name)
 pullup_label = string.format("%s", script_name)
+ivtc_detect_label = string.format("%s-ivtc-detect", script_name)

 -- number of seconds to gather cropdetect data
 detect_seconds = tonumber(mp.get_opt(string.format("%s.detect_seconds", script_name)))
@@ -69,30 +65,29 @@ function start_detect()
    mp.set_property("deinterlace","no")
    del_filter_if_present(pullup_label)

-    percent_pos = mp.get_property("percent-pos")
-
    -- insert the detection filter
-    local cmd = string.format('vf add @%s:lavfi=graph="idet"', detect_label)
+    local cmd = string.format('vf add @%s:lavfi=graph="idet",@%s:pullup,@%s:lavfi=graph="idet"',
+                              detect_label, pullup_label, ivtc_detect_label)
    if not mp.command(cmd) then
-        mp.msg.error("failed to insert detection filter")
+        mp.msg.error("failed to insert detection filters")
        return
    end

    -- wait to gather data
-    timer = mp.add_timeout(detect_seconds, judge_field_dominance)
+    timer = mp.add_timeout(detect_seconds, select_filter)
 end

 function stop_detect()
    del_filter_if_present(detect_label)
+    del_filter_if_present(ivtc_detect_label)
    timer = nil
-    mp.set_property("playback-pos", percent_pos)
 end

 progressive, interlaced_tff, interlaced_bff, interlaced = 0, 1, 2, 3, 4

-function judge()
+function judge(label)
    -- get the metadata
-    local result = mp.get_property_native(string.format("vf-metadata/%s", detect_label))
+    local result = mp.get_property_native(string.format("vf-metadata/%s", label))
    num_tff          = tonumber(result["lavfi.idet.multiple.tff"])
    num_bff          = tonumber(result["lavfi.idet.multiple.bff"])
    num_progressive  = tonumber(result["lavfi.idet.multiple.progressive"])
@@ -100,10 +95,10 @@ function judge()
    num_interlaced   = num_tff + num_bff
    num_determined   = num_interlaced + num_progressive

-    mp.msg.verbose("progressive    = "..num_progressive)
-    mp.msg.verbose("interlaced-tff = "..num_tff)
-    mp.msg.verbose("interlaced-bff = "..num_bff) 
-    mp.msg.verbose("undetermined   = "..num_undetermined)
+    mp.msg.verbose(label.." progressive    = "..num_progressive)
+    mp.msg.verbose(label.." interlaced-tff = "..num_tff)
+    mp.msg.verbose(label.." interlaced-bff = "..num_bff)
+    mp.msg.verbose(label.." undetermined   = "..num_undetermined)

    if num_determined < num_undetermined then
        mp.msg.warn("majority undetermined frames")
@@ -119,11 +114,12 @@ function judge()
    end
 end

-function judge_field_dominance()
-    verdict = judge()
+function select_filter()
+    -- handle the first detection filter results
+    verdict = judge(detect_label)
    if verdict == progressive then
-        stop_detect()
        mp.msg.info("progressive: doing nothing")
+        stop_detect()
        return
    elseif verdict == interlaced_tff then
        mp.set_property("field-dominance", "top")
@@ -133,21 +129,8 @@ function judge_field_dominance()
        mp.set_property("field-dominance", "auto")
    end

-    local cmd = string.format("vf pre @%s:pullup", pullup_label)
-    if not mp.command(cmd) then
-        mp.msg.error("failed to insert pullup filter")
-        stop_detect()
-        return
-    end
-
-    -- redo the detection with the pullup filter to see if it's telecined
-    mp.set_property("percent-pos", percent_pos)
-    timer = mp.add_timeout(detect_seconds, judge_ivtc)
-end
-
-function judge_ivtc()
-    verdict = judge()
-
+    -- handle the ivtc detection filter results
+    verdict = judge(ivtc_detect_label)
    if verdict == progressive then
        mp.msg.info(string.format("telecinied with %s field dominance: using pullup", mp.get_property("field-dominance")))
        stop_detect()
@@ -157,8 +140,6 @@ function judge_ivtc()
        mp.set_property("deinterlace","yes")
        stop_detect()
    end
-
-    return
 end

 mp.add_key_binding("ctrl+d", script_name, start_detect)
@@ -4,8 +4,8 @@
 -- alphabetically, and adds entries before and after the current file to
 -- the internal playlist. (It stops if the it would add an already existing
 -- playlist entry at the same position - this makes it "stable".)
-- Add at most 5 * 2 files when starting a file (before + after).
-MAXENTRIES = 5
+-- Add at most 5000 * 2 files when starting a file (before + after).
+MAXENTRIES = 5000

 function Set (t)
    local set = {}
@@ -47,6 +47,13 @@ function find_and_add_entries()
    if #dir == 0 then
        return
    end
+    local pl_count = mp.get_property_number("playlist-count", 1)
+    if (pl_count > 1 and autoload == nil) or
+       (pl_count == 1 and EXTENSIONS[string.lower(get_extension(filename))] == nil) then
+        return
+    else
+        autoload = true
+    end

    local files = mputils.readdir(dir, "files")
    if files == nil then
@@ -4,7 +4,7 @@
 -- parameters. See the entry for "drc" under the "AUDIO FILTERS"
 -- section of the man page for a complete description of the filter.
 --
-- This script registers the key-binding "\" to toggle the filter between 
+-- This script registers the key-binding "\" to toggle the filter between
 --
 -- * off
 -- * method=1 (single-sample smoothing)
@@ -92,7 +92,7 @@ function drc_quieter_handler()
    drc_scale_target(0.5)
 end

-- toggle between off, method 1 and method 2 
+-- toggle between off, method 1 and method 2
 mp.add_key_binding("\\", "drc_toggle_method", drc_toggle_method_handler)
 -- increase or decrease target volume
 mp.add_key_binding("ctrl+9", "drc_quieter", drc_quieter_handler)
@@ -1,5 +1,7 @@
 -- Rebuild the terminal status line as a lua script
 -- Be aware that this will require more cpu power!
+-- Also, this is based on a rather old version of the
+-- builtin mpv status line.

 -- Add a string to the status line
 function atsl(s)
@@ -28,7 +30,7 @@ function update_status_line()
    atsl(mp.get_property_osd("time-pos"))

    atsl(" / ");
-    atsl(mp.get_property_osd("length"));
+    atsl(mp.get_property_osd("duration"));

    atsl(" (")
    atsl(mp.get_property_osd("percent-pos", -1))
@@ -1,34 +0,0 @@
--sets the startime of a youtube video as specified in the "t=HHhMMmSSs" part of the url
--NOTE: This might become obsolete once youtube-dl adds the functionality
-
-local msg = require 'mp.msg'
-
-function youtube_starttime()
-  url = mp.get_property("path", "")
-  start = 0
-
-  if string.find(url, "youtu%.?be") and
-    ((url:find("http://") == 1) or (url:find("https://") == 1)) then
-      time = string.match(url, "[#&%?]t=%d*h?%d*m?%d+s?m?h?")
-      --the time-string can start with #, & or ? followed by t= and the timing parameters
-      --at least one number needs to be present after t=, followed by h, m, s or nothing (>implies s)
-
-      if time then
-        for pos in string.gmatch(time,"%d+%a?") do
-          if string.match(pos,"%d+h") then            --find out multiplier for
-            multiplier = 60*60                        --hours
-          elseif string.match(pos,"%d+m") then
-            multiplier = 60                           --minutes
-          else multiplier = 1 end                     --seconds
-
-          start = start + (string.match(pos,"%d+") * multiplier)
-        end
-
-        msg.info("parsed '" .. time .. "' into '" .. start .. "' seconds")
-      end
-
-      mp.set_property("file-local-options/start",start)
-  end
-end
-
-mp.add_hook("on_load", 50, youtube_starttime)
@@ -0,0 +1,74 @@
+--  zones.lua: mpv script for handling commands depending on where the mouse pointer is at,
+--             mostly for mouse wheel handling, by configuring it via input.conf, e.g.:
+--
+--  Ported from avih's ( https://github.com/avih ) zones.js
+--
+--  Vertical positions can be top, middle, bottom or "*" to represent the whole column.
+--  Horizontal positions can be left, middle, bottom or "*" to represent the whole row.
+--  "default" will be the fallback command to be used if no command is assigned to that area.
+--
+--  input.conf example of use:
+-- # wheel up/down with mouse
+-- MOUSE_BTN3 script_message_to zones commands "middle-right: add brightness  1" "*-left: add volume  5" "default: seek  10"
+-- MOUSE_BTN4 script_message_to zones commands "middle-right: add brightness -1" "*-left: add volume -5"  "default: seek -10"
+
+local ZONE_THRESH_PERCENTAGE = 20;
+-- sides get 20% each, mid gets 60%, same vertically
+
+local msg = mp.msg
+
+function getMouseZone()
+    -- returns the mouse zone as two strings [top/middle/bottom], [left/middle/right], e.g. "middle", "right"
+
+    local screenW, screenH = mp.get_osd_resolution()
+    local mouseX, mouseY   = mp.get_mouse_pos()
+
+    local threshY = screenH * ZONE_THRESH_PERCENTAGE / 100
+    local threshX = screenW * ZONE_THRESH_PERCENTAGE / 100
+
+    local yZone = (mouseY < threshY) and "top"  or (mouseY < (screenH - threshY)) and "middle" or "bottom"
+    local xZone = (mouseX < threshX) and "left" or (mouseX < (screenW - threshX)) and "middle" or "right"
+
+    return yZone, xZone
+end
+
+function main (...)
+    local arg={...}
+    msg.debug('commands: \n\t'..table.concat(arg,'\n\t'))
+
+    local keyY, keyX = getMouseZone()
+    msg.debug("mouse at: " .. keyY .. '-' .. keyX)
+
+    local fallback = nil
+
+    for i, v in ipairs(arg) do
+        cmdY = v:match("^([%w%*]+)%-?[%w%*]*:")
+        cmdX = v:match("^[%w%*]*%-([%w%*]+)%s*:")
+        cmd  = v:match("^[%S]-%s*:%s*(.+)")
+        msg.debug('cmdY: '..tostring(cmdY))
+        msg.debug('cmdX: '..tostring(cmdX))
+        msg.debug('cmd : '..tostring(cmd))
+
+        if (cmdY == keyY and cmdX == keyX) then
+            msg.verbose("running cmd: "..cmd)
+            mp.command(cmd)
+            return
+        elseif  (cmdY == "*"  and cmdX == keyX) or
+                (cmdY == keyY and cmdX == "*") then
+            msg.verbose("running cmd: "..cmd)
+            mp.command(cmd)
+            return
+        elseif cmdY == "default" then
+            fallback = cmd
+        end
+    end
+    if fallback ~= nil then
+        msg.verbose("running cmd: "..fallback)
+        mp.command(fallback)
+        return
+    else
+        msg.debug("no command assigned for "..keyY .. '-' .. keyX)
+        return
+    end
+end
+mp.register_script_message("commands", main)
@@ -4,6 +4,13 @@
 # Cleanups all over the place (c) 2001 pl
 # Rewritten for mpv in 2014.
 #
+#
+# Warning: this is not officially supported. Use on your own risk. Might require
+#          modifications when used on systems other than Linux. Contributors
+#          are not expected to update these files when making changes to the
+#          normal/preferred waf build system.
+#
+#
 # This configure script is *not* autoconf-based and has different semantics.
 # It attempts to autodetect all settings and options where possible. It is
 # possible to override autodetection with the --enable-option/--disable-option
@@ -176,6 +183,7 @@ options_state_machine() {
    opt_yes_no _dvdread     "libdvdread"
    opt_yes_no _dvdnav      "libdvdnav"
    opt_yes_no _enca        "ENCA charset oracle library"
+    opt_yes_no _uchardet    "uchardet charset detection library"
    opt_yes_no _libass      "subtitle rendering with libass"
    opt_yes_no _libavdevice "libavdevice demuxers"
    opt_yes_no _libavfilter "libavfilter"
@@ -183,8 +191,6 @@ options_state_machine() {
    opt_yes_no _libcdio     "libcdio support"
    opt_yes_no _librubberband "librubberband support"
    opt_yes_no _ffmpeg      "skip FFmpeg/Libav autodetection"
-    opt_yes_no _ladspa      "LADSPA plugin support"
-    opt_yes_no _libbs2b     "libbs2b audio filter support"
    opt_yes_no _libavresample "libavresample (preferred over libswresample)"
    opt_yes_no _libswresample "libswresample"
    opt_yes_no _caca        "CACA  video output"
@@ -208,6 +214,7 @@ options_state_machine() {
    opt_yes_no _lua         "Lua scripting"
    opt_yes_no _vapoursynth "VapourSynth filter bridge (Python)"
    opt_yes_no _vapoursynth_lazy "VapourSynth filter bridge (Lua)"
+    opt_yes_no _libarchive  "libarchive"
    opt_yes_no _encoding    "encoding functionality" yes
    opt_yes_no _build_man   "building manpage"
 }
@@ -376,7 +383,7 @@ addcflags() { cflag_check "$@" && OURCFLAGS="$OURCFLAGS $@" ; }
 OURCFLAGS="-std=c99 -Wall $_opt"

 addcflags -g -g3 -ggdb
-addcflags -Wundef -Wmissing-prototypes -Wshadow -Wno-switch -Wparentheses -Wpointer-arith -Wredundant-decls -Wno-pointer-sign -Werror=implicit-function-declaration -Wno-error=deprecated-declarations -Wno-error=unused-function
+addcflags -Wundef -Wmissing-prototypes -Wshadow -Wno-switch -Wparentheses -Wpointer-arith -Wno-redundant-decls -Wno-pointer-sign -Werror=implicit-function-declaration -Wno-error=deprecated-declarations -Wno-error=unused-function
 # clang
 addcflags -Wno-logical-op-parentheses -fcolor-diagnostics -Wno-tautological-compare -Wno-tautological-constant-out-of-range-compare
 # extra
@@ -506,9 +513,12 @@ check_statement_libs "compiler support for __sync built-ins" $_sync SYNC_BUILTIN
    stdint.h 'int64_t test = 0; test = __sync_add_and_fetch(&test, 1)'
 _sync=$(defretval)

+_any_atomic=yes
 if test "$_atomic" = no && test "$_sync" = no && test "$_stdatomic" = no ; then
  echo "your compiler must support either stdatomic.h, or __atomic, or __sync built-ins."
+  _any_atomic=no
 fi
+define_yes_no $_any_atomic HAVE_ATOMICS

 check_compile "iconv" $_iconv ICONV waftools/fragments/iconv.c " " "-liconv" "-liconv $_ld_dl"
 _iconv=$(defretval)
@@ -709,7 +719,7 @@ check_pkg_config "OpenAL" $_openal OPENAL 'openal >= 1.13'

 check_pkg_config "ALSA audio" $_alsa ALSA 'alsa >= 1.0.9'

-check_pkg_config "Blu-ray support" $_bluray LIBBLURAY 'libbluray >= 0.2.1'
+check_pkg_config "Blu-ray support" $_bluray LIBBLURAY 'libbluray >= 0.3.0'

 check_pkg_config "dvdread" $_dvdread DVDREAD 'dvdread >= 4.1.0'

@@ -732,15 +742,11 @@ echo "LIBASS_OSD = $_libass" >> $CONFIG_MAK
 echo "DUMMY_OSD = $_dummy_osd" >> $CONFIG_MAK

 check_pkg_config "ENCA" $_enca ENCA 'enca'
+check_pkg_config "uchardet" $_uchardet UCHARDET 'uchardet'

 check_pkg_config "zlib" auto ZLIB 'zlib'
 test $(defretval) = no && die "Unable to find development files for zlib."

-test "$_dl" = no && _ladspa=no
-check_statement_libs "LADSPA plugin support" $_ladspa LADSPA ladspa.h 'LADSPA_Descriptor ld = {0}'
-
-check_pkg_config "libbs2b audio filter support" $_libbs2b LIBBS2B 'libbs2b'
-
 check_pkg_config "LCMS2 support" $_lcms2 LCMS2 'lcms2 >= 2.6'

 check_pkg_config "FFmpeg/Libav" $_ffmpeg FFMPEG \
@@ -786,6 +792,18 @@ api_statement_check \
    libavutil/frame.h \
    'enum AVFrameSideDataType type = AV_FRAME_DATA_SKIP_SAMPLES'

+api_statement_check \
+    "libavutil av_version_info()" \
+    HAVE_AV_VERSION_INFO \
+    libavutil/avutil.h \
+    'const char *x = av_version_info()'
+
+api_statement_check \
+    "libavutil new pixdesc fields" \
+    HAVE_AV_NEW_PIXDESC \
+    libavutil/pixdesc.h \
+    'AVComponentDescriptor d; int x = d.depth'
+
 check_pkg_config "libavfilter" $_libavfilter LIBAVFILTER 'libavfilter >= 5.0.0'

 check_pkg_config "libavdevice" $_libavdevice LIBAVDEVICE 'libavdevice >= 55.0.0'
@@ -817,9 +835,9 @@ test_lua "lua >= 5.1.0 lua < 5.2.0"
 test_lua "lua5.1 >= 5.1.0" # debian
 test_lua "lua51 >= 5.1.0" # OpenBSD
 test_lua "luajit >= 2.0.0"
-test_lua "lua >= 5.2.0"
 test_lua "lua5.2 >= 5.2.0" # debian
 test_lua "lua52 >= 5.2.0" # OpenBSD
+test_lua "lua >= 5.2.0"

 test "$_lua" != yes && check_yes_no no LUA

@@ -841,6 +859,8 @@ if test "$_vapoursynth" = no && test "$_vapoursynth_lazy" = no ; then
 fi
 check_trivial "VapourSynth core" $_vapoursynth_core VAPOURSYNTH_CORE

+check_pkg_config "libarchive support" $_libarchive LIBARCHIVE 'libarchive >= 3.0.0'
+
 check_trivial "encoding" $_encoding ENCODING

 # needs dlopen on unix
@@ -858,7 +878,7 @@ CFLAGS="$CFLAGS -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE64_SOURCE
 # in-tree builds.

 if test ! -f Makefile ; then
-    ln -s old-makefile Makefile
+    ln -s TOOLS/old-makefile Makefile
 fi

 cat > old_build/config.mak << EOF
@@ -905,7 +925,7 @@ cat > $TMPC << EOF
 /* we didn't bother to add actual config checks for this, or they are
   for platforms not supported by this configure script */
 #define HAVE_BSD_FSTATFS 0
-#define HAVE_LINUX_FSTATFS 0
+#define HAVE_LINUX_FSTATFS 1
 #define HAVE_VDA_AV_VDA_ALLOC_CONTEXT 0
 #define HAVE_VDA_HWACCEL 0
 #define HAVE_VDA_LIBAVCODEC_REFCOUNTING 0
@@ -936,6 +956,9 @@ cat > $TMPC << EOF
 #define HAVE_RPI_GLES 0
 #define HAVE_AV_PIX_FMT_MMAL 0
 #define HAVE_DRM 0
+#define HAVE_VIDEOTOOLBOX_HWACCEL 0
+#define HAVE_VIDEOTOOLBOX_GL 0
+#define HAVE_VIDEOTOOLBOX_VDA_GL 0

 #ifdef __OpenBSD__
 #define DEFAULT_CDROM_DEVICE "/dev/rcd0c"
@@ -35,13 +35,11 @@ SOURCES-$(DVDREAD)              += stream/stream_dvd.c \
 SOURCES-$(DVDNAV)               += stream/stream_dvdnav.c \
                                   stream/stream_dvd_common.c

-SOURCES-$(LADSPA)               += audio/filter/af_ladspa.c
 SOURCES-$(RUBBERBAND)           += audio/filter/af_rubberband.c
 SOURCES-$(LIBASS)               += sub/ass_mp.c sub/sd_ass.c \
                                   demux/demux_libass.c

 SOURCES-$(LIBBLURAY)            += stream/stream_bluray.c
-SOURCES-$(LIBBS2B)              += audio/filter/af_bs2b.c

 SOURCES-$(LIBSMBCLIENT)         += stream/stream_smb.c

@@ -58,19 +56,24 @@ SOURCES-$(LIBASS_OSD)           += sub/osd_libass.c
 SOURCES-$(ALSA)                 += audio/out/ao_alsa.c
 SOURCES-$(CACA)                 += video/out/vo_caca.c
 SOURCES-$(SDL2)                 += audio/out/ao_sdl.c video/out/vo_sdl.c
-SOURCES-$(GL)                   += video/out/gl_common.c video/out/gl_osd.c \
-                                   video/out/vo_opengl.c video/out/gl_lcms.c \
-                                   video/out/gl_video.c video/out/dither.c \
-                                   video/out/gl_hwdec.c video/out/gl_utils.c \
+SOURCES-$(GL)                   += video/out/opengl/common.c \
+                                   video/out/opengl/osd.c \
+                                   video/out/opengl/lcms.c \
+                                   video/out/opengl/video.c \
+                                   video/out/opengl/video_shaders.c \
+                                   video/out/dither.c \
+                                   video/out/opengl/hwdec.c \
+                                   video/out/opengl/utils.c \
+                                   video/out/vo_opengl.c \
                                   video/out/vo_opengl_cb.c

 SOURCES-$(ENCODING)             += video/out/vo_lavc.c audio/out/ao_lavc.c \
                                   common/encode_lavc.c

-SOURCES-$(GL_X11)               += video/out/x11_common.c video/out/gl_x11.c
-SOURCES-$(EGL_X11)              += video/out/x11_common.c video/out/gl_x11egl.c
+SOURCES-$(GL_X11)               += video/out/x11_common.c video/out/opengl/x11.c
+SOURCES-$(EGL_X11)              += video/out/x11_common.c video/out/opengl/x11egl.c
 SOURCES-$(GL_WAYLAND)           += video/out/wayland_common.c \
-                                   video/out/gl_wayland.c
+                                   video/out/opengl/wayland.c

 SOURCES-$(JACK)                 += audio/out/ao_jack.c
 SOURCES-$(OPENAL)               += audio/out/ao_openal.c
@@ -80,34 +83,32 @@ SOURCES-$(RSOUND)               += audio/out/ao_rsound.c
 SOURCES-$(SNDIO)                += audio/out/ao_sndio.c
 SOURCES-$(VDPAU)                += video/vdpau.c video/vdpau_mixer.c \
                                   video/out/vo_vdpau.c video/decode/vdpau.c \
-                                   video/filter/vf_vdpaupp.c
-SOURCES-$(VDPAU_GL_X11)         += video/out/gl_hwdec_vdpau.c
+                                   video/filter/vf_vdpaupp.c \
+                                   video/filter/vf_vdpaurb.c
+SOURCES-$(VDPAU_GL_X11)         += video/out/opengl/hwdec_vdpau.c
 SOURCES-$(VAAPI)                += video/out/vo_vaapi.c \
                                   video/decode/vaapi.c \
                                   video/vaapi.c
 SOURCES-$(VAAPI_VPP)            += video/filter/vf_vavpp.c
-SOURCES-$(VAAPI_GLX)            += video/out/gl_hwdec_vaglx.c
+SOURCES-$(VAAPI_GLX)            += video/out/opengl/hwdec_vaglx.c

-SOURCES-$(X11)                  += video/out/vo_x11.c video/out/x11_common.c
-SOURCES-$(XV)                   += video/out/vo_xv.c
+SOURCES-$(XV)                   += video/out/x11_common.c video/out/vo_xv.c
 SOURCES-$(WAYLAND)              += video/out/vo_wayland.c \
                                   video/out/wayland_common.c \
                                   video/out/wayland/buffer.c \
                                   video/out/wayland/memfile.c

 SOURCES-$(LIBAVFILTER)          += video/filter/vf_lavfi.c \
-                                   video/filter/vf_delogo.c \
                                   video/filter/vf_gradfun.c \
-                                   video/filter/vf_hqdn3d.c \
-                                   video/filter/vf_noise.c \
                                   video/filter/vf_pullup.c \
                                   video/filter/vf_rotate.c \
-                                   video/filter/vf_unsharp.c \
                                   video/filter/vf_yadif.c \
                                   audio/filter/af_lavfi.c

 SOURCES-$(LUA)                  += player/lua.c
 SOURCES-$(VAPOURSYNTH_CORE)     += video/filter/vf_vapoursynth.c
+SOURCES-$(LIBARCHIVE)           += demux/demux_libarchive.c \
+                                   stream/stream_libarchive.c
 SOURCES-$(DLOPEN)               += video/filter/vf_dlopen.c

 SOURCES = audio/audio.c \
@@ -121,31 +122,17 @@ SOURCES = audio/audio.c \
          audio/decode/ad_spdif.c      \
          audio/decode/dec_audio.c \
          audio/filter/af.c \
-          audio/filter/af_center.c \
          audio/filter/af_channels.c \
-          audio/filter/af_convert24.c \
-          audio/filter/af_convertsignendian.c \
          audio/filter/af_delay.c \
-          audio/filter/af_dummy.c \
          audio/filter/af_equalizer.c \
-          audio/filter/af_export.c \
-          audio/filter/af_extrastereo.c \
          audio/filter/af_format.c \
-          audio/filter/af_hrtf.c \
-          audio/filter/af_karaoke.c \
          audio/filter/af_lavcac3enc.c \
          audio/filter/af_lavrresample.c \
          audio/filter/af_pan.c \
          audio/filter/af_scaletempo.c \
-          audio/filter/af_sinesuppress.c \
-          audio/filter/af_sub.c \
-          audio/filter/af_surround.c \
-          audio/filter/af_sweep.c \
          audio/filter/af_drc.c \
          audio/filter/af_volume.c \
-          audio/filter/filter.c \
          audio/filter/tools.c \
-          audio/filter/window.c \
          audio/out/ao.c \
          audio/out/ao_null.c \
          audio/out/ao_pcm.c \
@@ -160,6 +147,7 @@ SOURCES = audio/audio.c \
          common/tags.c \
          common/version.c \
          demux/codec_tags.c \
+          demux/cue.c \
          demux/demux.c \
          demux/demux_edl.c \
          demux/demux_cue.c \
@@ -195,6 +183,7 @@ SOURCES = audio/audio.c \
          options/parse_configfile.c \
          options/path.c \
          osdep/io.c \
+          osdep/path-unix.c \
          osdep/semaphore_osx.c \
          osdep/subprocess.c \
          osdep/subprocess-posix.c \
@@ -206,7 +195,7 @@ SOURCES = audio/audio.c \
          player/client.c \
          player/configfiles.c \
          player/command.c \
-          player/discnav.c \
+          player/external_files.c \
          player/loadfile.c \
          player/main.c \
          player/misc.c \
@@ -231,7 +220,6 @@ SOURCES = audio/audio.c \
          stream/stream_rar.c \
          sub/dec_sub.c \
          sub/draw_bmp.c \
-          sub/find_subfiles.c \
          sub/img_convert.c \
          sub/osd.c \
          sub/sd_lavc.c \
@@ -334,7 +322,7 @@ all: $(ALL_TARGETS)
 %.o: %.c
 	$(CC) $(DEPFLAGS) $(CFLAGS) -c -o $@ $<

-mpv: $(OBJECTS) player/main-fn-unix.o
+mpv: $(OBJECTS) osdep/main-fn-unix.o
 	$(CC) -o $@ $^ $(EXTRALIBS)

 input/input.c: input/input_conf.h
@@ -381,8 +369,8 @@ config.mak: configure
 	@echo "####### Please run ./configure again - it's changed! #######"
 	@echo "############################################################"

-version.h .version: version.sh
-	./$<
+old_build/version.h .version: version.sh
+	./version.sh --versionh=old_build/version.h

 # Force version.sh to run to potentially regenerate version.h
 -include .version
@@ -393,7 +381,7 @@ version.h .version: version.sh

 ###### dependency declarations / specific CFLAGS ######

-common/version.c: version.h
+common/version.c: old_build/version.h

 DOCS/man/mpv.1: DOCS/man/af.rst \
                   DOCS/man/ao.rst \
@@ -464,7 +452,7 @@ clean:
 	-$(RM) $(call ADD_ALL_DIRS,/*.o /*.a /*.ho /*~)
 	-$(RM) mpv
 	-$(RM) DOCS/man/*/mpv.1
-	-$(RM) version.h
+	-$(RM) old_build/version.h version.h
 	-$(RM) input/input_conf.h
 	-$(RM) video/out/vdpau_template.c
 	-$(RM) demux/ebml_types.h demux/ebml_defs.c
@@ -40,7 +40,7 @@ def apply_plist_template(plist_file, version):
        print (line.rstrip().replace('${VERSION}', version))

 def main():
-    version = sh("./version.sh --print").strip()
+    version = sh("./version.sh").strip()

    usage = "usage: %prog [options] arg"
    parser = OptionParser(usage)
@@ -1,2 +1 @@
-screenshot-template=~/Desktop/shot%n
 profile=pseudo-gui
@@ -5,7 +5,10 @@ import re

 filename = sys.argv[1]

-event_regex = re.compile(".*")
+events = ".*"
+if len(sys.argv) > 2:
+    events = sys.argv[2]
+event_regex = re.compile(events)

 """
 This script is meant to display stats written by mpv --dump-stats=filename.
@@ -26,8 +29,8 @@ Currently, the following event types are supported:
    'end' <name>                end of the named event
    'value' <float> <name>      a normal value (as opposed to event)
    'event-timed' <ts> <name>   singular event at the given timestamp
-    'value-timed' <ts> <float> <name>
-                                a value for an event at the given timestamp
+    'value-timed' <ts> <float> <name>       a value for an event at the given timestamp
+    'range-timed' <ts1> <ts2> <name>        like start/end, but explicit times
    <name>                      singular event (same as 'signal')

 """
@@ -91,6 +94,15 @@ for line in [line.split("#")[0].strip() for line in open(filename, "r")]:
        val = int(val) / SCALE - G.start
        e = get_event(name, "event-signal")
        e.vals.append((val, 1))
+    elif event.startswith("range-timed "):
+        _, ts1, ts2, name = event.split(" ", 3)
+        ts1 = int(ts1) / SCALE - G.start
+        ts2 = int(ts2) / SCALE - G.start
+        e = get_event(name, "event")
+        e.vals.append((ts1, 0))
+        e.vals.append((ts1, 1))
+        e.vals.append((ts2, 1))
+        e.vals.append((ts2, 0))
    elif event.startswith("value-timed "):
        _, tsval, val, name = event.split(" ", 3)
        tsval = int(tsval) / SCALE - G.start
@@ -141,7 +141,7 @@ sub parse_main_opts {
    my ($cmd, $regex) = @_;

    my @list;
-    my @lines = split /\n/, `"$mpv" --no-config $cmd`;
+    my @lines = call_mpv($cmd);

    foreach my $line (@lines) {
        my ($name, $desc) = ($line =~ /^$regex/) or next;
@@ -206,7 +206,7 @@ sub parse_opts {
    my ($cmd, $regex) = @_;

    my @list;
-    my @lines = split /\n/, `"$mpv" --no-config $cmd`;
+    my @lines = call_mpv($cmd);

    foreach my $line (@lines) {
        if ($line !~ /^$regex/) {
@@ -226,3 +226,14 @@ sub parse_opts {

    return @list;
 }
+
+sub call_mpv {
+    my ($cmd) = @_;
+    my $output = `"$mpv" --no-config $cmd`;
+    if ($? == -1) {
+        die "Could not run mpv: $!";
+    } elsif ((my $exit_code = $? >> 8) != 0) {
+        die "mpv returned $exit_code with output:\n$output";
+    }
+    return split /\n/, $output;
+}
@@ -0,0 +1 @@
+0.11.0
@@ -34,8 +34,8 @@ static void update_redundant_info(struct mp_audio *mpa)
    assert(mp_chmap_is_empty(&mpa->channels) ||
           mp_chmap_is_valid(&mpa->channels));
    mpa->nch = mpa->channels.num;
-    mpa->bps = af_fmt2bps(mpa->format);
-    if (AF_FORMAT_IS_PLANAR(mpa->format)) {
+    mpa->bps = af_fmt_to_bytes(mpa->format);
+    if (af_fmt_is_planar(mpa->format)) {
        mpa->spf = 1;
        mpa->num_planes = mpa->nch;
        mpa->sstride = mpa->bps;
@@ -58,13 +58,6 @@ void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels)
    update_redundant_info(mpa);
 }

-// Use old MPlayer/ALSA channel layout.
-void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels)
-{
-    mp_chmap_from_channels_alsa(&mpa->channels, num_channels);
-    update_redundant_info(mpa);
-}
-
 void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap)
 {
    mpa->channels = *chmap;
@@ -93,15 +86,19 @@ bool mp_audio_config_valid(const struct mp_audio *mpa)

 char *mp_audio_config_to_str_buf(char *buf, size_t buf_sz, struct mp_audio *mpa)
 {
+    char ch[128];
+    mp_chmap_to_str_buf(ch, sizeof(ch), &mpa->channels);
+    char *hr_ch = mp_chmap_to_str_hr(&mpa->channels);
+    if (strcmp(hr_ch, ch) != 0)
+        mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch);
    snprintf(buf, buf_sz, "%dHz %s %dch %s", mpa->rate,
-             mp_chmap_to_str(&mpa->channels), mpa->channels.num,
-             af_fmt_to_str(mpa->format));
+             ch, mpa->channels.num, af_fmt_to_str(mpa->format));
    return buf;
 }

 void mp_audio_force_interleaved_format(struct mp_audio *mpa)
 {
-    if (AF_FORMAT_IS_PLANAR(mpa->format))
+    if (af_fmt_is_planar(mpa->format))
        mp_audio_set_format(mpa, af_fmt_from_planar(mpa->format));
 }

@@ -159,8 +156,8 @@ void mp_audio_realloc(struct mp_audio *mpa, int samples)
        if (!mpa->allocated[n] || size != mpa->allocated[n]->size) {
            if (av_buffer_realloc(&mpa->allocated[n], size) < 0)
                abort(); // OOM
-            mpa->planes[n] = mpa->allocated[n]->data;
        }
+        mpa->planes[n] = mpa->allocated[n]->data;
    }
    for (int n = mpa->num_planes; n < MP_NUM_CHANNELS; n++) {
        av_buffer_unref(&mpa->allocated[n]);
@@ -191,8 +188,18 @@ int mp_audio_get_allocated_size(struct mp_audio *mpa)
 {
    int size = 0;
    for (int n = 0; n < mpa->num_planes; n++) {
-        int s = mpa->allocated[n] ? mpa->allocated[n]->size / mpa->sstride : 0;
-        size = n == 0 ? s : MPMIN(size, s);
+        for (int i = 0; i < MP_NUM_CHANNELS && mpa->allocated[i]; i++) {
+            uint8_t *start = mpa->allocated[i]->data;
+            uint8_t *end = start + mpa->allocated[i]->size;
+            uint8_t *plane = mpa->planes[n];
+            if (plane >= start && plane < end) {
+                int s = MPMIN((end - plane) / mpa->sstride, INT_MAX);
+                size = n == 0 ? s : MPMIN(size, s);
+                goto next;
+            }
+        }
+        return 0; // plane is not covered by any buffer
+    next: ;
    }
    return size;
 }
@@ -35,15 +35,17 @@ struct mp_audio {
    int nch;            // number of channels (redundant with chmap)
    int spf;            // sub-samples per sample on each plane
    int num_planes;     // number of planes
-    int bps;            // size of sub-samples (af_fmt2bps(format))
+    int bps;            // size of sub-samples (af_fmt_to_bytes(format))

-    // private
+    // --- private
+    // These do not necessarily map directly to planes[]. They can have
+    // different order or count. There shouldn't be more buffers than planes.
+    // If allocated[n] is NULL, allocated[n+1] must also be NULL.
    struct AVBufferRef *allocated[MP_NUM_CHANNELS];
 };

 void mp_audio_set_format(struct mp_audio *mpa, int format);
 void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels);
-void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels);
 void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap);
 void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src);
 bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b);
@@ -18,6 +18,8 @@
 #include <stdlib.h>
 #include <assert.h>

+#include <libavutil/common.h>
+
 #include "common/common.h"
 #include "common/msg.h"
 #include "chmap.h"
@@ -50,6 +52,7 @@ static const char *const speaker_names[MP_SPEAKER_ID_COUNT][2] = {
    [MP_SPEAKER_ID_SDL]         = {"sdl",  "surround direct left"},
    [MP_SPEAKER_ID_SDR]         = {"sdr",  "surround direct right"},
    [MP_SPEAKER_ID_LFE2]        = {"lfe2", "low frequency 2"},
+    [MP_SPEAKER_ID_NA]          = {"na",   "not available"},
 };

 // Names taken from libavutil/channel_layout.c (Not accessible by API.)
@@ -67,6 +70,7 @@ static const char *const std_layout_names[][2] = {
    {"quad",            "fl-fr-bl-br"},
    {"quad(side)",      "fl-fr-sl-sr"},
    {"3.1",             "fl-fr-fc-lfe"},
+    {"3.1(back)",       "fl-fr-lfe-bc"}, // not in lavc
    {"5.0",             "fl-fr-fc-bl-br"},
    {"5.0(alsa)",       "fl-fr-bl-br-fc"}, // not in lavc
    {"5.0(side)",       "fl-fr-fc-sl-sr"},
@@ -80,9 +84,11 @@ static const char *const std_layout_names[][2] = {
    {"hexagonal",       "fl-fr-fc-bl-br-bc"},
    {"6.1",             "fl-fr-fc-lfe-bc-sl-sr"},
    {"6.1(back)",       "fl-fr-fc-lfe-bl-br-bc"}, // lavc calls this "6.1" too
+    {"6.1(top)",        "fl-fr-fc-lfe-bl-br-tc"}, // not in lavc
    {"6.1(front)",      "fl-fr-lfe-flc-frc-sl-sr"},
    {"7.0",             "fl-fr-fc-bl-br-sl-sr"},
    {"7.0(front)",      "fl-fr-fc-flc-frc-sl-sr"},
+    {"7.0(rear)",       "fl-fr-fc-bl-br-sdl-sdr"}, // not in lavc
    {"7.1",             "fl-fr-fc-lfe-bl-br-sl-sr"},
    {"7.1(alsa)",       "fl-fr-bl-br-fc-lfe-sl-sr"}, // not in lavc
    {"7.1(wide)",       "fl-fr-fc-lfe-bl-br-flc-frc"},
@@ -94,7 +100,7 @@ static const char *const std_layout_names[][2] = {
    {0}
 };

-static const struct mp_chmap default_layouts[MP_NUM_CHANNELS + 1] = {
+static const struct mp_chmap default_layouts[] = {
    {0},                                        // empty
    MP_CHMAP_INIT_MONO,                         // mono
    MP_CHMAP2(FL, FR),                          // stereo
@@ -127,7 +133,8 @@ bool mp_chmap_is_valid(const struct mp_chmap *src)
        int sp = src->speaker[n];
        if (sp >= MP_SPEAKER_ID_COUNT || mapped[sp])
            return false;
-        mapped[sp] = true;
+        if (sp != MP_SPEAKER_ID_NA)
+            mapped[sp] = true;
    }
    return src->num > 0;
 }
@@ -142,12 +149,10 @@ bool mp_chmap_is_empty(const struct mp_chmap *src)
 bool mp_chmap_is_unknown(const struct mp_chmap *src)
 {
    for (int n = 0; n < src->num; n++) {
-        int speaker = src->speaker[n];
-        if (speaker >= MP_SPEAKER_ID_UNKNOWN0 &&
-            speaker <= MP_SPEAKER_ID_UNKNOWN_LAST)
-            return true;
+        if (src->speaker[n] != MP_SPEAKER_ID_NA)
+            return false;
    }
-    return false;
+    return mp_chmap_is_valid(src);
 }

 // Note: empty channel maps compare as equal. Invalid ones can equal too.
@@ -171,15 +176,6 @@ bool mp_chmap_equals_reordered(const struct mp_chmap *a, const struct mp_chmap *
    return mp_chmap_equals(&t1, &t2);
 }

-bool mp_chmap_is_compatible(const struct mp_chmap *a, const struct mp_chmap *b)
-{
-    if (mp_chmap_equals(a, b))
-        return true;
-    if (a->num == b->num && (mp_chmap_is_unknown(a) || mp_chmap_is_unknown(b)))
-        return true;
-    return false;
-}
-
 bool mp_chmap_is_stereo(const struct mp_chmap *src)
 {
    static const struct mp_chmap stereo = MP_CHMAP_INIT_STEREO;
@@ -199,16 +195,36 @@ void mp_chmap_reorder_norm(struct mp_chmap *map)
    qsort(arr, map->num, 1, comp_uint8);
 }

+// Remove silent (NA) channels, if any.
+void mp_chmap_remove_na(struct mp_chmap *map)
+{
+    struct mp_chmap new = {0};
+    for (int n = 0; n < map->num; n++) {
+        int sp = map->speaker[n];
+        if (sp != MP_SPEAKER_ID_NA)
+            new.speaker[new.num++] = map->speaker[n];
+    }
+    *map = new;
+}
+
+// Add silent (NA) channels to map until map->num >= num.
+void mp_chmap_fill_na(struct mp_chmap *map, int num)
+{
+    assert(num <= MP_NUM_CHANNELS);
+    while (map->num < num)
+        map->speaker[map->num++] = MP_SPEAKER_ID_NA;
+}
+
 // Set *dst to a standard layout with the given number of channels.
 // If the number of channels is invalid, an invalid map is set, and
 // mp_chmap_is_valid(dst) will return false.
 void mp_chmap_from_channels(struct mp_chmap *dst, int num_channels)
 {
-    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
-        *dst = (struct mp_chmap) {0};
-    } else {
+    *dst = (struct mp_chmap) {0};
+    if (num_channels >= 0 && num_channels < MP_ARRAY_SIZE(default_layouts))
        *dst = default_layouts[num_channels];
-    }
+    if (!dst->num)
+        mp_chmap_set_unknown(dst, num_channels);
 }

 // Try to do what mplayer/mplayer2/mpv did before channel layouts were
@@ -227,6 +243,8 @@ void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels)
 // Set *dst to an unknown layout for the given numbers of channels.
 // If the number of channels is invalid, an invalid map is set, and
 // mp_chmap_is_valid(dst) will return false.
+// A mp_chmap with all entries set to NA is treated specially in some
+// contexts (watch out for mp_chmap_is_unknown()).
 void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels)
 {
    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
@@ -234,7 +252,7 @@ void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels)
    } else {
        dst->num = num_channels;
        for (int n = 0; n < dst->num; n++)
-            dst->speaker[n] = MP_SPEAKER_ID_UNKNOWN0 + n;
+            dst->speaker[n] = MP_SPEAKER_ID_NA;
    }
 }

@@ -277,6 +295,7 @@ void mp_chmap_remove_useless_channels(struct mp_chmap *map,
 }

 // Return the ffmpeg/libav channel layout as in <libavutil/channel_layout.h>.
+// Speakers not representable by ffmpeg/libav are dropped.
 // Warning: this ignores the order of the channels, and will return a channel
 //          mask even if the order is different from libavcodec's.
 uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src)
@@ -286,8 +305,10 @@ uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src)
    if (mp_chmap_is_unknown(&t))
        mp_chmap_from_channels(&t, t.num);
    uint64_t mask = 0;
-    for (int n = 0; n < t.num; n++)
-        mask |= 1ULL << t.speaker[n];
+    for (int n = 0; n < t.num; n++) {
+        if (t.speaker[n] < 64) // ignore MP_SPEAKER_ID_NA etc.
+            mask |= 1ULL << t.speaker[n];
+    }
    return mask;
 }

@@ -348,56 +369,43 @@ void mp_chmap_reorder_to_lavc(struct mp_chmap *map)

 // Get reordering array for from->to reordering. from->to must have the same set
 // of speakers (i.e. same number and speaker IDs, just different order). Then,
-// for each speaker n, dst[n] will be set such that:
-//      to->speaker[dst[n]] = from->speaker[n]
-// (dst[n] gives the source channel for destination channel n)
-void mp_chmap_get_reorder(int dst[MP_NUM_CHANNELS], const struct mp_chmap *from,
+// for each speaker n, src[n] will be set such that:
+//      to->speaker[n] = from->speaker[src[n]]
+// (src[n] gives the source channel for destination channel n)
+// If *from and *to don't contain the same set of speakers, then the above
+// invariant is not guaranteed. Instead, src[n] can be set to -1 if the channel
+// at to->speaker[n] is unmapped.
+void mp_chmap_get_reorder(int src[MP_NUM_CHANNELS], const struct mp_chmap *from,
                          const struct mp_chmap *to)
 {
-    assert(from->num == to->num);
+    for (int n = 0; n < MP_NUM_CHANNELS; n++)
+        src[n] = -1;
+
    if (mp_chmap_is_unknown(from) || mp_chmap_is_unknown(to)) {
-        for (int n = 0; n < from->num; n++)
-            dst[n] = n;
+        for (int n = 0; n < to->num; n++)
+            src[n] = n < from->num ? n : -1;
        return;
    }
-    // Same set of speakers required
-    assert(mp_chmap_equals_reordered(from, to));
-    for (int n = 0; n < from->num; n++) {
-        int src = from->speaker[n];
-        dst[n] = -1;
-        for (int i = 0; i < to->num; i++) {
-            if (src == to->speaker[i]) {
-                dst[n] = i;
+
+    for (int n = 0; n < to->num; n++) {
+        for (int i = 0; i < from->num; i++) {
+            if (to->speaker[n] == from->speaker[i]) {
+                src[n] = i;
                break;
            }
        }
-        assert(dst[n] != -1);
    }
-    for (int n = 0; n < from->num; n++)
-        assert(to->speaker[dst[n]] == from->speaker[n]);
+
+    for (int n = 0; n < to->num; n++)
+        assert(src[n] < 0 || (to->speaker[n] == from->speaker[src[n]]));
 }

-// Performs the difference between a and b, and store it in diff. If b has
-// channels that do not appear in a, those will not appear in the difference.
-// To get to those the argument ordering in the function call has to be
-// inverted. For the same reason, the diff with a superset will return no
-// speakers.
-void mp_chmap_diff(const struct mp_chmap *a, const struct mp_chmap *b,
-                   struct mp_chmap *diff)
+// Return the number of channels only in a.
+int mp_chmap_diffn(const struct mp_chmap *a, const struct mp_chmap *b)
 {
    uint64_t a_mask = mp_chmap_to_lavc_unchecked(a);
    uint64_t b_mask = mp_chmap_to_lavc_unchecked(b);
-    mp_chmap_from_lavc(diff, (a_mask ^ b_mask) & a_mask);
-}
-
-// Checks whether a contains all the speakers in b
-bool mp_chmap_contains(const struct mp_chmap *a, const struct mp_chmap *b)
-{
-    struct mp_chmap d1;
-    struct mp_chmap d2;
-    mp_chmap_diff(a, b, &d1);
-    mp_chmap_diff(b, a, &d2);
-    return a->num >= b->num && d1.num >= 0 && d2.num == 0;
+    return av_popcount64((a_mask ^ b_mask) & a_mask);
 }

 // Returns something like "fl-fr-fc". If there's a standard layout in lavc
@@ -500,6 +508,25 @@ bool mp_chmap_from_str(struct mp_chmap *dst, bstr src)
    return true;
 }

+// Output a human readable "canonical" channel map string. Converting this from
+// a string back to a channel map can yield a different map, but the string
+// looks nicer. E.g. "fc-fl-fr-na" becomes "3.0".
+char *mp_chmap_to_str_hr_buf(char *buf, size_t buf_size, const struct mp_chmap *src)
+{
+    struct mp_chmap map = *src;
+    mp_chmap_remove_na(&map);
+    for (int n = 0; std_layout_names[n][0]; n++) {
+        struct mp_chmap s;
+        if (mp_chmap_from_str(&s, bstr0(std_layout_names[n][0])) &&
+            mp_chmap_equals_reordered(&s, &map))
+        {
+            map = s;
+            break;
+        }
+    }
+    return mp_chmap_to_str_buf(buf, buf_size, &map);
+}
+
 void mp_chmap_print_help(struct mp_log *log)
 {
    mp_info(log, "Speakers:\n");
@@ -47,7 +47,7 @@ enum mp_speaker_id {
    MP_SPEAKER_ID_TBL,          // TOP_BACK_LEFT
    MP_SPEAKER_ID_TBC,          // TOP_BACK_CENTER
    MP_SPEAKER_ID_TBR,          // TOP_BACK_RIGHT
-     // Inofficial/libav* extensions
+     // Unofficial/libav* extensions
    MP_SPEAKER_ID_DL = 29,      // STEREO_LEFT (stereo downmix special speakers)
    MP_SPEAKER_ID_DR,           // STEREO_RIGHT
    MP_SPEAKER_ID_WL,           // WIDE_LEFT
@@ -56,10 +56,12 @@ enum mp_speaker_id {
    MP_SPEAKER_ID_SDR,          // SURROUND_DIRECT_RIGHT
    MP_SPEAKER_ID_LFE2,         // LOW_FREQUENCY_2

-    // Special mpv-specific speaker entries reserved for channels which have no
-    // known meaning.
-    MP_SPEAKER_ID_UNKNOWN0 = 64,
-    MP_SPEAKER_ID_UNKNOWN_LAST = MP_SPEAKER_ID_UNKNOWN0 + MP_NUM_CHANNELS - 1,
+    // Speaker IDs >= 64 are not representable in WAVEFORMATEXTENSIBLE or libav*.
+
+    // "Silent" channels. These are sometimes used to insert padding for
+    // unused channels. Unlike other speaker types, multiple of these can
+    // occur in a single mp_chmap.
+    MP_SPEAKER_ID_NA = 64,

    // Including the unassigned IDs in between. This is not a valid ID anymore,
    // but is still within uint8_t.
@@ -98,10 +100,11 @@ bool mp_chmap_is_empty(const struct mp_chmap *src);
 bool mp_chmap_is_unknown(const struct mp_chmap *src);
 bool mp_chmap_equals(const struct mp_chmap *a, const struct mp_chmap *b);
 bool mp_chmap_equals_reordered(const struct mp_chmap *a, const struct mp_chmap *b);
-bool mp_chmap_is_compatible(const struct mp_chmap *a, const struct mp_chmap *b);
 bool mp_chmap_is_stereo(const struct mp_chmap *src);

 void mp_chmap_reorder_norm(struct mp_chmap *map);
+void mp_chmap_remove_na(struct mp_chmap *map);
+void mp_chmap_fill_na(struct mp_chmap *map, int num);

 void mp_chmap_from_channels(struct mp_chmap *dst, int num_channels);
 void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels);
@@ -117,17 +120,17 @@ void mp_chmap_from_lavc(struct mp_chmap *dst, uint64_t src);
 bool mp_chmap_is_lavc(const struct mp_chmap *src);
 void mp_chmap_reorder_to_lavc(struct mp_chmap *map);

-void mp_chmap_get_reorder(int dst[MP_NUM_CHANNELS], const struct mp_chmap *from,
+void mp_chmap_get_reorder(int src[MP_NUM_CHANNELS], const struct mp_chmap *from,
                          const struct mp_chmap *to);

-void mp_chmap_diff(const struct mp_chmap *a, const struct mp_chmap *b,
-                   struct mp_chmap *diff);
-
-bool mp_chmap_contains(const struct mp_chmap *a, const struct mp_chmap *b);
+int mp_chmap_diffn(const struct mp_chmap *a, const struct mp_chmap *b);

 char *mp_chmap_to_str_buf(char *buf, size_t buf_size, const struct mp_chmap *src);
 #define mp_chmap_to_str(m) mp_chmap_to_str_buf((char[64]){0}, 64, (m))

+char *mp_chmap_to_str_hr_buf(char *buf, size_t buf_size, const struct mp_chmap *src);
+#define mp_chmap_to_str_hr(m) mp_chmap_to_str_hr_buf((char[128]){0}, 128, (m))
+
 bool mp_chmap_from_str(struct mp_chmap *dst, bstr src);

 struct mp_log;
@@ -58,6 +58,29 @@ static bool replace_speakers(struct mp_chmap *map, struct mp_chmap list[2])
    return false;
 }

+// These go strictly from the first to the second entry and always use the
+// full layout (possibly reordered and/or padding channels added).
+static const struct mp_chmap preferred_remix[][2] = {
+    // mono can be perfectly played as stereo
+    { MP_CHMAP_INIT_MONO, MP_CHMAP_INIT_STEREO },
+};
+
+// Conversion from src to dst is explicitly encouraged and should be preferred
+// over "mathematical" upmixes or downmixes (which minimize lost channels).
+static bool test_preferred_remix(const struct mp_chmap *src,
+                                 const struct mp_chmap *dst)
+{
+    struct mp_chmap src_p = *src, dst_p = *dst;
+    mp_chmap_remove_na(&src_p);
+    mp_chmap_remove_na(&dst_p);
+    for (int n = 0; n < MP_ARRAY_SIZE(preferred_remix); n++) {
+        if (mp_chmap_equals_reordered(&src_p, &preferred_remix[n][0]) &&
+            mp_chmap_equals_reordered(&dst_p, &preferred_remix[n][1]))
+            return true;
+    }
+    return false;
+}
+
 // Allow all channel layouts that can be expressed with mp_chmap.
 // (By default, all layouts are rejected.)
 void mp_chmap_sel_add_any(struct mp_chmap_sel *s)
@@ -176,6 +199,10 @@ bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map)
            return true;
        }
    }
+
+    if (mp_chmap_sel_fallback(s, map))
+        return true;
+
    for (int i = 0; i < MP_ARRAY_SIZE(speaker_replacements); i++) {
        struct mp_chmap  t = *map;
        struct mp_chmap *r = (struct mp_chmap *)speaker_replacements[i];
@@ -185,9 +212,6 @@ bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map)
        }
    }

-    if (mp_chmap_sel_fallback(s, map))
-        return true;
-
    // Fallback to mono/stereo as last resort
    *map = (struct mp_chmap) MP_CHMAP_INIT_STEREO;
    if (test_layout(s, map))
@@ -199,64 +223,95 @@ bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map)
    return false;
 }

-#define UPMIX_IDX 0
-#define DOWNMIX_IDX 1
-#define FALLBACK_IDX 2
-
-static bool test_fallbacks(struct mp_chmap *a, struct mp_chmap *b,
-                           int best_diffs[2], struct mp_chmap best[2])
+// Like mp_chmap_diffn(), but find the minimum difference with all possible
+// speaker replacements considered.
+static int mp_chmap_diffn_r(const struct mp_chmap *a, const struct mp_chmap *b)
 {
-    struct mp_chmap diff1, diff2;
+    int mindiff = INT_MAX;

-    mp_chmap_diff(a, b, &diff1);
-    if (mp_chmap_contains(a, b) && best_diffs[UPMIX_IDX] > diff1.num) {
-        best[UPMIX_IDX] = *a;
-        best_diffs[UPMIX_IDX] = diff1.num;
-        return true;
+    for (int i = -1; i < (int)MP_ARRAY_SIZE(speaker_replacements); i++) {
+        struct mp_chmap ar = *a;
+        if (i >= 0) {
+            struct mp_chmap *r = (struct mp_chmap *)speaker_replacements[i];
+            if (!replace_speakers(&ar, r))
+                continue;
+        }
+        int d = mp_chmap_diffn(&ar, b);
+        if (d < mindiff)
+            mindiff = d;
    }

-    mp_chmap_diff(b, a, &diff2);
-    if (mp_chmap_contains(b, a) && best_diffs[DOWNMIX_IDX] > diff2.num) {
-        best[DOWNMIX_IDX] = *a;
-        best_diffs[DOWNMIX_IDX] = diff2.num;
-        return true;
-    }
-
-    if (diff1.num > 0 && best_diffs[FALLBACK_IDX] > diff1.num) {
-        best[FALLBACK_IDX] = *a;
-        best_diffs[FALLBACK_IDX] = diff1.num;
-        return true;
-    }
-
-    if (diff2.num > 0 && best_diffs[FALLBACK_IDX] > diff2.num) {
-        best[FALLBACK_IDX] = *a;
-        best_diffs[FALLBACK_IDX] = diff2.num;
-        return true;
-    }
-
-    return false;
-}
-
-// Determine which channel map to fallback to given a source channel map. It
-// uses the following heuristic:
-// 1) If mono is required always prefer stereo to a multichannel upmix.
-// 2) Search for an upmix that is an exact superset of the required chmap.
-// 3) Search for a downmix that is the exact subset of the required chmap.
-// 4) Search for either an upmix or downmix that is the closest (minimum
-//    difference of speakers) to the required chmap.
-bool mp_chmap_sel_fallback(const struct mp_chmap_sel *s, struct mp_chmap *map)
-{
-    // special case: if possible always fallback mono to stereo (instead of
-    // looking for a multichannel upmix)
+    // Special-case: we consider stereo a replacement for mono. (This is not
+    // true in the other direction. Also, fl-fr is generally not a replacement
+    // for fc. Thus it's not part of the speaker replacement list.)
    struct mp_chmap mono   = MP_CHMAP_INIT_MONO;
    struct mp_chmap stereo = MP_CHMAP_INIT_STEREO;
-    if (mp_chmap_equals(&mono, map) && test_layout(s, &stereo)) {
-        *map = stereo;
-        return true;
-    }
+    if (mp_chmap_equals(&mono, b) && mp_chmap_equals(&stereo, a))
+        mindiff = 0;

-    int best_diffs[] = { INT_MAX, INT_MAX, INT_MAX };
-    struct mp_chmap best[] = { {0}, {0}, {0} };
+    return mindiff;
+}
+
+// Decide whether we should prefer old or new for the requested layout.
+// Return true if new should be used, false if old should be used.
+// If old is empty, always return new (initial case).
+static bool mp_chmap_is_better(struct mp_chmap *req, struct mp_chmap *old,
+                               struct mp_chmap *new)
+{
+    // Initial case
+    if (!old->num)
+        return true;
+
+    // Exact pick - this also ensures that the best layout is chosen if the
+    // layouts are the same, but with different order of channels.
+    if (mp_chmap_equals(req, old))
+        return false;
+    if (mp_chmap_equals(req, new))
+        return true;
+
+    // If there's no exact match, strictly do a preferred conversion.
+    bool old_pref = test_preferred_remix(req, old);
+    bool new_pref = test_preferred_remix(req, new);
+    if (old_pref && !new_pref)
+        return false;
+    if (!old_pref && new_pref)
+        return true;
+
+    int old_lost_r = mp_chmap_diffn_r(req, old); // num. channels only in req
+    int new_lost_r = mp_chmap_diffn_r(req, new);
+
+    // Imperfect upmix (no real superset) - minimize lost channels
+    if (new_lost_r != old_lost_r)
+        return new_lost_r < old_lost_r;
+
+    int old_lost = mp_chmap_diffn(req, old);
+    int new_lost = mp_chmap_diffn(req, new);
+
+    // If the situation is equal with replaced speakers, but one of them loses
+    // less if no replacements are performed, pick the better one, even if it
+    // means an upmix. This prefers exact supersets over inexact equivalents.
+    if (new_lost != old_lost)
+        return new_lost < old_lost;
+
+    struct mp_chmap old_p = *old, new_p = *new;
+    mp_chmap_remove_na(&old_p);
+    mp_chmap_remove_na(&new_p);
+
+    // Some kind of upmix. If it's perfect, prefer the smaller one. Even if not,
+    // both have equal loss, so also prefer the smaller one.
+    // Drop padding channels (NA) for the sake of this check, as the number of
+    // padding channels isn't really meaningful.
+    if (new_p.num != old_p.num)
+        return new_p.num < old_p.num;
+
+    // Again, with physical channels (minimizes number of NA channels).
+    return new->num < old->num;
+}
+
+// Determine which channel map to fallback to given a source channel map.
+bool mp_chmap_sel_fallback(const struct mp_chmap_sel *s, struct mp_chmap *map)
+{
+    struct mp_chmap best = {0};

    for (int n = 0; n < s->num_chmaps; n++) {
        struct mp_chmap e = s->chmaps[n];
@@ -264,25 +319,13 @@ bool mp_chmap_sel_fallback(const struct mp_chmap_sel *s, struct mp_chmap *map)
        if (mp_chmap_is_unknown(&e))
            continue;

-        if (test_fallbacks(&e, map, best_diffs, best))
-            continue;
-
-        // in case we didn't match any fallback retry after replacing speakers
-        for (int i = 0; i < MP_ARRAY_SIZE(speaker_replacements); i++) {
-            struct mp_chmap  t = e;
-            struct mp_chmap *r = (struct mp_chmap *)speaker_replacements[i];
-            if (replace_speakers(&t, r)) {
-                if (test_fallbacks(&t, map, best_diffs, best))
-                    continue;
-            }
-        }
+        if (mp_chmap_is_better(map, &best, &e))
+            best = e;
    }

-    for (int i = UPMIX_IDX; i < MP_ARRAY_SIZE(best); i++) {
-        if (best_diffs[i] < INT_MAX) {
-            *map = best[i];
-            return true;
-        }
+    if (best.num) {
+        *map = best;
+        return true;
    }

    return false;
@@ -116,7 +116,7 @@ static int init(struct dec_audio *da, const char *decoder)

    mp_set_avopts(da->log, lavc_context, opts->avopts);

-    lavc_context->codec_tag = sh->format;
+    lavc_context->codec_tag = sh->codec_tag;
    lavc_context->sample_rate = sh_audio->samplerate;
    lavc_context->bit_rate = sh_audio->bitrate;
    lavc_context->block_align = sh_audio->block_align;
@@ -126,11 +126,7 @@ static int init(struct dec_audio *da, const char *decoder)
        lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);

    // demux_mkv
-    if (sh_audio->codecdata_len && sh_audio->codecdata &&
-            !lavc_context->extradata) {
-        mp_lavc_set_extradata(lavc_context, sh_audio->codecdata,
-                              sh_audio->codecdata_len);
-    }
+    mp_lavc_set_extradata(lavc_context, sh->extradata, sh->extradata_size);

    if (sh->lav_headers)
        mp_copy_lav_codec_headers(lavc_context, sh->lav_headers);
@@ -34,10 +34,12 @@

 struct spdifContext {
    struct mp_log   *log;
+    enum AVCodecID   codec_id;
    AVFormatContext *lavf_ctx;
    int              out_buffer_len;
    uint8_t          out_buffer[OUTBUF_SIZE];
    bool             need_close;
+    bool             use_dts_hd;
    struct mp_audio  fmt;
 };

@@ -76,17 +78,78 @@ static int init(struct dec_audio *da, const char *decoder)
    struct spdifContext *spdif_ctx = talloc_zero(NULL, struct spdifContext);
    da->priv = spdif_ctx;
    spdif_ctx->log = da->log;
+    spdif_ctx->use_dts_hd = da->opts->dtshd;
+
+    if (strcmp(decoder, "dts-hd") == 0) {
+        decoder = "dts";
+        spdif_ctx->use_dts_hd = true;
+    }
+
+    spdif_ctx->codec_id = mp_codec_to_av_codec_id(decoder);
+    return spdif_ctx->codec_id != AV_CODEC_ID_NONE;
+}
+
+static int determine_codec_profile(struct dec_audio *da, AVPacket *pkt)
+{
+    struct spdifContext *spdif_ctx = da->priv;
+    int profile = FF_PROFILE_UNKNOWN;
+    AVCodecContext *ctx = NULL;
+    AVFrame *frame = NULL;
+
+    AVCodec *codec = avcodec_find_decoder(spdif_ctx->codec_id);
+    if (!codec)
+        goto done;
+
+    frame = av_frame_alloc();
+    if (!frame)
+        goto done;
+
+    ctx = avcodec_alloc_context3(codec);
+    if (!ctx)
+        goto done;
+
+    if (avcodec_open2(ctx, codec, NULL) < 0) {
+        av_free(ctx); // don't attempt to avcodec_close() an unopened ctx
+        ctx = NULL;
+        goto done;
+    }
+
+    int got_frame = 0;
+    if (avcodec_decode_audio4(ctx, frame, &got_frame, pkt) < 1 || !got_frame)
+        goto done;
+
+    profile = ctx->profile;
+
+done:
+    av_frame_free(&frame);
+    if (ctx)
+        avcodec_close(ctx);
+    avcodec_free_context(&ctx);
+
+    if (profile == FF_PROFILE_UNKNOWN)
+        MP_WARN(da, "Failed to parse codec profile.\n");
+
+    return profile;
+}
+
+static int init_filter(struct dec_audio *da, AVPacket *pkt)
+{
+    struct spdifContext *spdif_ctx = da->priv;
+
+    int profile = FF_PROFILE_UNKNOWN;
+    if (spdif_ctx->codec_id == AV_CODEC_ID_DTS)
+        profile = determine_codec_profile(da, pkt);

    AVFormatContext *lavf_ctx  = avformat_alloc_context();
    if (!lavf_ctx)
        goto fail;

+    spdif_ctx->lavf_ctx = lavf_ctx;
+
    lavf_ctx->oformat = av_guess_format("spdif", NULL, NULL);
    if (!lavf_ctx->oformat)
        goto fail;

-    spdif_ctx->lavf_ctx = lavf_ctx;
-
    void *buffer = av_mallocz(OUTBUF_SIZE);
    if (!buffer)
        abort();
@@ -106,14 +169,14 @@ static int init(struct dec_audio *da, const char *decoder)
    if (!stream)
        goto fail;

-    stream->codec->codec_id = mp_codec_to_av_codec_id(decoder);
+    stream->codec->codec_id = spdif_ctx->codec_id;

    AVDictionary *format_opts = NULL;

    int num_channels = 0;
    int sample_format = 0;
    int samplerate = 0;
-    switch (stream->codec->codec_id) {
+    switch (spdif_ctx->codec_id) {
    case AV_CODEC_ID_AAC:
        sample_format                   = AF_FORMAT_S_AAC;
        samplerate                      = 48000;
@@ -124,18 +187,21 @@ static int init(struct dec_audio *da, const char *decoder)
        samplerate                      = 48000;
        num_channels                    = 2;
        break;
-    case AV_CODEC_ID_DTS:
-        if (da->opts->dtshd) {
+    case AV_CODEC_ID_DTS: {
+        bool is_hd = profile == FF_PROFILE_DTS_HD_HRA ||
+                     profile == FF_PROFILE_DTS_HD_MA;
+        if (spdif_ctx->use_dts_hd && is_hd) {
            av_dict_set(&format_opts, "dtshd_rate", "768000", 0); // 4*192000
-            sample_format                   = AF_FORMAT_S_DTSHD;
-            samplerate                      = 192000;
-            num_channels                    = 2*4;
+            sample_format               = AF_FORMAT_S_DTSHD;
+            samplerate                  = 192000;
+            num_channels                = 2*4;
        } else {
-            sample_format                   = AF_FORMAT_S_DTS;
-            samplerate                      = 48000;
-            num_channels                    = 2;
+            sample_format               = AF_FORMAT_S_DTS;
+            samplerate                  = 48000;
+            num_channels                = 2;
        }
        break;
+    }
    case AV_CODEC_ID_EAC3:
        sample_format                   = AF_FORMAT_S_EAC3;
        samplerate                      = 192000;
@@ -167,17 +233,16 @@ static int init(struct dec_audio *da, const char *decoder)

    spdif_ctx->need_close = true;

-    return 1;
+    return 0;

 fail:
    uninit(da);
-    return 0;
+    return -1;
 }

 static int decode_packet(struct dec_audio *da, struct mp_audio **out)
 {
    struct spdifContext *spdif_ctx = da->priv;
-    AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;

    spdif_ctx->out_buffer_len  = 0;

@@ -195,9 +260,13 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
        da->pts        = mpkt->pts;
        da->pts_offset = 0;
    }
-    int ret = av_write_frame(lavf_ctx, &pkt);
+    if (!spdif_ctx->lavf_ctx) {
+        if (init_filter(da, &pkt) < 0)
+            return AD_ERR;
+    }
+    int ret = av_write_frame(spdif_ctx->lavf_ctx, &pkt);
    talloc_free(mpkt);
-    avio_flush(lavf_ctx->pb);
+    avio_flush(spdif_ctx->lavf_ctx->pb);
    if (ret < 0)
        return AD_ERR;

@@ -235,6 +304,8 @@ static void add_decoders(struct mp_decoder_list *list)
                           "libavformat/spdifenc audio pass-through decoder");
        }
    }
+    mp_add_decoder(list, "spdif", "dts", "dts-hd",
+                   "libavformat/spdifenc audio pass-through decoder");
 }

 const struct ad_functions ad_spdif = {
@@ -52,6 +52,7 @@ static const struct ad_functions * const ad_drivers[] = {

 static void uninit_decoder(struct dec_audio *d_audio)
 {
+    audio_reset_decoding(d_audio);
    if (d_audio->ad_driver) {
        MP_VERBOSE(d_audio, "Uninit audio decoder.\n");
        d_audio->ad_driver->uninit(d_audio);
@@ -59,6 +60,8 @@ static void uninit_decoder(struct dec_audio *d_audio)
    d_audio->ad_driver = NULL;
    talloc_free(d_audio->priv);
    d_audio->priv = NULL;
+    d_audio->afilter->initialized = -1;
+    d_audio->decode_format = (struct mp_audio){0};
 }

 static int init_audio_codec(struct dec_audio *d_audio, const char *decoder)
@@ -81,11 +84,21 @@ struct mp_decoder_list *audio_decoder_list(void)
    return list;
 }

-static struct mp_decoder_list *audio_select_decoders(const char *codec,
-                                                     char *selection)
+static struct mp_decoder_list *audio_select_decoders(struct dec_audio *d_audio)
 {
+    struct MPOpts *opts = d_audio->opts;
+    const char *codec = d_audio->header->codec;
+
    struct mp_decoder_list *list = audio_decoder_list();
-    struct mp_decoder_list *new = mp_select_decoders(list, codec, selection);
+    struct mp_decoder_list *new =
+        mp_select_decoders(list, codec, opts->audio_decoders);
+    if (d_audio->spdif_passthrough) {
+        struct mp_decoder_list *spdif =
+            mp_select_decoder_list(list, codec, "spdif", opts->audio_spdif);
+        mp_append_decoders(spdif, new);
+        talloc_free(new);
+        new = spdif;
+    }
    talloc_free(list);
    return new;
 }
@@ -99,14 +112,13 @@ static const struct ad_functions *find_driver(const char *name)
    return NULL;
 }

-int audio_init_best_codec(struct dec_audio *d_audio, char *audio_decoders)
+int audio_init_best_codec(struct dec_audio *d_audio)
 {
+    uninit_decoder(d_audio);
    assert(!d_audio->ad_driver);
-    audio_reset_decoding(d_audio);

    struct mp_decoder_entry *decoder = NULL;
-    struct mp_decoder_list *list =
-        audio_select_decoders(d_audio->header->codec, audio_decoders);
+    struct mp_decoder_list *list = audio_select_decoders(d_audio);

    mp_print_decoders(d_audio->log, MSGL_V, "Codec list:", list);

@@ -145,8 +157,8 @@ void audio_uninit(struct dec_audio *d_audio)
    if (!d_audio)
        return;
    MP_VERBOSE(d_audio, "Uninit audio filters...\n");
-    af_destroy(d_audio->afilter);
    uninit_decoder(d_audio);
+    af_destroy(d_audio->afilter);
    talloc_free(d_audio->waiting);
    talloc_free(d_audio);
 }
@@ -30,6 +30,7 @@ struct dec_audio {
    struct mp_log *log;
    struct MPOpts *opts;
    struct mpv_global *global;
+    bool spdif_passthrough;
    const struct ad_functions *ad_driver;
    struct sh_stream *header;
    struct af_stream *afilter;
@@ -57,7 +58,7 @@ enum {
 };

 struct mp_decoder_list *audio_decoder_list(void);
-int audio_init_best_codec(struct dec_audio *d_audio, char *audio_decoders);
+int audio_init_best_codec(struct dec_audio *d_audio);
 int audio_decode(struct dec_audio *d_audio, struct mp_audio_buffer *outbuf,
                 int minsamples);
 int initial_audio_decode(struct dec_audio *d_audio);
@@ -31,70 +31,37 @@
 #include "af.h"

 // Static list of filters
-extern const struct af_info af_info_dummy;
 extern const struct af_info af_info_delay;
 extern const struct af_info af_info_channels;
 extern const struct af_info af_info_format;
-extern const struct af_info af_info_force;
 extern const struct af_info af_info_volume;
 extern const struct af_info af_info_equalizer;
 extern const struct af_info af_info_pan;
-extern const struct af_info af_info_surround;
-extern const struct af_info af_info_sub;
-extern const struct af_info af_info_export;
 extern const struct af_info af_info_drc;
-extern const struct af_info af_info_extrastereo;
 extern const struct af_info af_info_lavcac3enc;
 extern const struct af_info af_info_lavrresample;
-extern const struct af_info af_info_sweep;
-extern const struct af_info af_info_hrtf;
-extern const struct af_info af_info_ladspa;
-extern const struct af_info af_info_center;
-extern const struct af_info af_info_sinesuppress;
-extern const struct af_info af_info_karaoke;
 extern const struct af_info af_info_scaletempo;
 extern const struct af_info af_info_bs2b;
 extern const struct af_info af_info_lavfi;
-extern const struct af_info af_info_convert24;
-extern const struct af_info af_info_convertsignendian;
 extern const struct af_info af_info_rubberband;

 static const struct af_info *const filter_list[] = {
-    &af_info_dummy,
    &af_info_delay,
    &af_info_channels,
    &af_info_format,
    &af_info_volume,
    &af_info_equalizer,
    &af_info_pan,
-    &af_info_surround,
-    &af_info_sub,
-    &af_info_export,
    &af_info_drc,
-    &af_info_extrastereo,
    &af_info_lavcac3enc,
    &af_info_lavrresample,
-    &af_info_sweep,
-    &af_info_hrtf,
-#if HAVE_LADSPA
-    &af_info_ladspa,
-#endif
 #if HAVE_RUBBERBAND
    &af_info_rubberband,
 #endif
-    &af_info_center,
-    &af_info_sinesuppress,
-    &af_info_karaoke,
    &af_info_scaletempo,
-#if HAVE_LIBBS2B
-    &af_info_bs2b,
-#endif
 #if HAVE_LIBAVFILTER
    &af_info_lavfi,
 #endif
-    // Must come last, because they're fallback format conversion filter
-    &af_info_convert24,
-    &af_info_convertsignendian,
    NULL
 };

@@ -192,24 +159,11 @@ static struct af_instance *af_create(struct af_stream *s, char *name,
        MP_ERR(s, "Couldn't find audio filter '%s'.\n", name);
        return NULL;
    }
-    const struct af_info *info = desc.p;
-    /* Make sure that the filter is not already in the list if it is
-       non-reentrant */
-    if (info->flags & AF_FLAGS_NOT_REENTRANT) {
-        for (struct af_instance *cur = s->first; cur; cur = cur->next) {
-            if (cur->info == info) {
-                MP_ERR(s, "There can only be one "
-                       "instance of the filter '%s' in each stream\n", name);
-                return NULL;
-            }
-        }
-    }
-
    MP_VERBOSE(s, "Adding filter %s \n", name);

    struct af_instance *af = talloc_zero(NULL, struct af_instance);
    *af = (struct af_instance) {
-        .info = info,
+        .info = desc.p,
        .data = talloc_zero(af, struct mp_audio),
        .log = mp_log_new(af, s->log, name),
        .replaygain_data = s->replaygain_data,
@@ -334,89 +288,9 @@ static void af_print_filter_chain(struct af_stream *s, struct af_instance *at,
    MP_MSG(s, msg_level, "  [ao] %s\n", mp_audio_config_to_str(&s->output));
 }

-static int af_count_filters(struct af_stream *s)
-{
-    int count = 0;
-    for (struct af_instance *af = s->first; af; af = af->next)
-        count++;
-    return count;
-}
-
-// Finds the first conversion filter on the way from srcfmt to dstfmt.
-// Conversions form a DAG: each node is a format/filter pair, and possible
-// conversions are edges. We search the DAG for the shortest path.
-// Some cases visit the same filter multiple times, but with different formats
-// (like u24le->s8), so one node per format or filter separate is not enough.
-// Returns the filter and dest. format for the first conversion step.
-// (So we know what conversion filter with what format to insert next.)
-static char *af_find_conversion_filter(int srcfmt, int *dstfmt)
-{
-#define MAX_NODES (64 * 32)
-    int num_fmt = 0, num_filt = 0;
-    for (int n = 0; filter_list[n]; n++)
-        num_filt = n + 1;
-    for (int n = 0; af_fmtstr_table[n].format; n++)
-        num_fmt = n + 1;
-
-    int num_nodes = num_fmt * num_filt;
-    assert(num_nodes < MAX_NODES);
-
-    bool visited[MAX_NODES] = {0};
-    unsigned char distance[MAX_NODES];
-    short previous[MAX_NODES] = {0};
-    for (int n = 0; n < num_nodes; n++) {
-        distance[n] = 255;
-        if (af_fmtstr_table[n % num_fmt].format == srcfmt)
-            distance[n] = 0;
-    }
-
-    while (1) {
-        int next = -1;
-        for (int n = 0; n < num_nodes; n++) {
-            if (!visited[n] && (next < 0 || (distance[n] < distance[next])))
-                next = n;
-        }
-        if (next < 0 || distance[next] == 255)
-            return NULL;
-        visited[next] = true;
-
-        int fmt = next % num_fmt;
-        if (af_fmtstr_table[fmt].format == *dstfmt) {
-            // Best match found
-            for (int cur = next; cur >= 0; cur = previous[cur] - 1) {
-                if (distance[cur] == 1) {
-                    *dstfmt = af_fmtstr_table[cur % num_fmt].format;
-                    return (char *)filter_list[cur / num_fmt]->name;
-                }
-            }
-            return NULL;
-        }
-
-        for (int n = 0; filter_list[n]; n++) {
-            const struct af_info *af = filter_list[n];
-            if (!af->test_conversion)
-                continue;
-            for (int i = 0; af_fmtstr_table[i].format; i++) {
-                if (i != fmt && af->test_conversion(af_fmtstr_table[fmt].format,
-                                                    af_fmtstr_table[i].format))
-                {
-                    int other = n * num_fmt + i;
-                    int ndist = distance[next] + 1;
-                    if (ndist < distance[other]) {
-                        distance[other] = ndist;
-                        previous[other] = next + 1;
-                    }
-                }
-            }
-        }
-    }
-    assert(0);
-#undef MAX_NODES
-}
-
 static bool af_is_conversion_filter(struct af_instance *af)
 {
-    return af && af->info->test_conversion != NULL;
+    return af && strcmp(af->info->name, "lavrresample") == 0;
 }

 // in is what af can take as input - insert a conversion filter if the actual
@@ -436,21 +310,24 @@ static int af_fix_format_conversion(struct af_stream *s,
    if (actual.format == in.format)
        return AF_FALSE;
    int dstfmt = in.format;
-    char *filter = af_find_conversion_filter(actual.format, &dstfmt);
-    if (!filter)
+    char *filter = "lavrresample";
+    if (!af_lavrresample_test_conversion(actual.format, dstfmt))
        return AF_ERROR;
    if (strcmp(filter, prev->info->name) == 0) {
        if (prev->control(prev, AF_CONTROL_SET_FORMAT, &dstfmt) == AF_OK) {
            *p_af = prev;
            return AF_OK;
        }
+        return AF_ERROR;
    }
    struct af_instance *new = af_prepend(s, af, filter, NULL);
    if (new == NULL)
        return AF_ERROR;
    new->auto_inserted = true;
-    if (AF_OK != (rv = new->control(new, AF_CONTROL_SET_FORMAT, &dstfmt)))
+    if (AF_OK != (rv = new->control(new, AF_CONTROL_SET_FORMAT, &dstfmt))) {
+        af_remove(s, new);
        return rv;
+    }
    *p_af = new;
    return AF_OK;
 }
@@ -528,8 +405,8 @@ static int af_reinit(struct af_stream *s)
    // Start with the second filter, as the first filter is the special input
    // filter which needs no initialization.
    struct af_instance *af = s->first->next;
-    // Up to 7 retries per filter (channel, rate, 5x format conversions)
-    int max_retry = af_count_filters(s) * 7;
+    // Up to 4 retries per filter (channel, rate, format conversions)
+    int max_retry = 4;
    int retry = 0;
    while (af) {
        if (retry >= max_retry)
@@ -575,9 +452,9 @@ static int af_reinit(struct af_stream *s)
            int fmt_in1 = af->prev->data->format;
            int fmt_in2 = in.format;
            if (af_fmt_is_valid(fmt_in1) && af_fmt_is_valid(fmt_in2)) {
-                bool spd1 = AF_FORMAT_IS_IEC61937(fmt_in1);
-                bool spd2 = AF_FORMAT_IS_IEC61937(fmt_in2);
-                if (spd1 != spd2) {
+                bool spd1 = af_fmt_is_spdif(fmt_in1);
+                bool spd2 = af_fmt_is_spdif(fmt_in2);
+                if (spd1 != spd2 && af->next) {
                    MP_WARN(af, "Filter %s apparently cannot be used due to "
                                "spdif passthrough - removing it.\n",
                                af->info->name);
@@ -601,6 +478,8 @@ static int af_reinit(struct af_stream *s)
                   af->info->name, rv);
            goto error;
        }
+        if (af && !af->auto_inserted)
+            retry = 0;
    }

    /* Set previously unset fields in s->output to those of the filter chain
@@ -733,7 +612,7 @@ struct af_instance *af_add(struct af_stream *s, char *name, char *label,
        return NULL;
    new->label = talloc_strdup(new, label);

-    // Reinitalize the filter list
+    // Reinitialize the filter list
    if (af_reinit(s) != AF_OK) {
        af_remove_by_label(s, label);
        return NULL;
@@ -803,6 +682,18 @@ void af_control_all(struct af_stream *s, int cmd, void *arg)
        af->control(af, cmd, arg);
 }

+int af_control_by_label(struct af_stream *s, int cmd, void *arg, bstr label)
+{
+    char *label_str = bstrdup0(NULL, label);
+    struct af_instance *cur = af_find_by_label(s, label_str);
+    talloc_free(label_str);
+    if (cur) {
+        return cur->control ? cur->control(cur, cmd, arg) : CONTROL_NA;
+    } else {
+        return CONTROL_UNKNOWN;
+    }
+}
+
 // Used by filters to add a filtered frame to the output queue.
 // Ownership of frame is transferred from caller to the filter chain.
 void af_add_output_frame(struct af_instance *af, struct mp_audio *frame)
@@ -832,6 +723,16 @@ static struct mp_audio *af_dequeue_output_frame(struct af_instance *af)
    return res;
 }

+static void read_remaining(struct af_instance *af)
+{
+    int num_frames;
+    do {
+        num_frames = af->num_out_queued;
+        if (!af->filter_out || af->filter_out(af) < 0)
+            break;
+    } while (num_frames != af->num_out_queued);
+}
+
 static int af_do_filter(struct af_instance *af, struct mp_audio *frame)
 {
    if (frame)
@@ -871,6 +772,7 @@ int af_output_frame(struct af_stream *s, bool eof)
            // Flush remaining frames on EOF, but do that only if the previous
            // filters have been flushed (i.e. they have no more output).
            if (eof && !last) {
+                read_remaining(cur);
                int r = af_do_filter(cur, NULL);
                if (r < 0)
                    return r;
@@ -27,6 +27,7 @@
 #include "audio/chmap.h"
 #include "audio/audio.h"
 #include "common/msg.h"
+#include "common/common.h"

 struct af_instance;
 struct mpv_global;
@@ -34,10 +35,6 @@ struct mpv_global;
 // Number of channels
 #define AF_NCH MP_NUM_CHANNELS

-// Flags used for defining the behavior of an audio filter
-#define AF_FLAGS_REENTRANT      0x00000000
-#define AF_FLAGS_NOT_REENTRANT  0x00000001
-
 // Flags for af->filter()
 #define AF_FILTER_FLAG_EOF 1

@@ -46,9 +43,7 @@ struct mpv_global;
 struct af_info {
    const char *info;
    const char *name;
-    const int flags;
    int (*open)(struct af_instance *vf);
-    bool (*test_conversion)(int src_format, int dst_format);
    int priv_size;
    const void *priv_defaults;
    const struct m_option *options;
@@ -105,13 +100,12 @@ struct af_stream {
 };

 // Return values
-#define AF_DETACH   2
-#define AF_OK       1
-#define AF_TRUE     1
-#define AF_FALSE    0
-#define AF_UNKNOWN -1
-#define AF_ERROR   -2
-#define AF_FATAL   -3
+#define AF_DETACH   (CONTROL_OK + 1)
+#define AF_OK       CONTROL_OK
+#define AF_TRUE     CONTROL_TRUE
+#define AF_FALSE    CONTROL_FALSE
+#define AF_UNKNOWN  CONTROL_UNKNOWN
+#define AF_ERROR    CONTROL_ERROR

 // Parameters for af_control_*
 enum af_control {
@@ -128,6 +122,7 @@ enum af_control {
    AF_CONTROL_GET_PAN_BALANCE,
    AF_CONTROL_SET_PLAYBACK_SPEED,
    AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE,
+    AF_CONTROL_GET_METADATA,
 };

 // Argument for AF_CONTROL_SET_PAN_LEVEL
@@ -146,6 +141,7 @@ int af_remove_by_label(struct af_stream *s, char *label);
 struct af_instance *af_find_by_label(struct af_stream *s, char *label);
 struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg);
 void af_control_all(struct af_stream *s, int cmd, void *arg);
+int af_control_by_label(struct af_stream *s, int cmd, void *arg, bstr label);
 void af_seek_reset(struct af_stream *s);

 void af_add_output_frame(struct af_instance *af, struct mp_audio *frame);
@@ -158,8 +154,9 @@ double af_calc_delay(struct af_stream *s);

 int af_test_output(struct af_instance *af, struct mp_audio *out);

-int af_from_dB(int n, float *in, float *out, float k, float mi, float ma);
 int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma);
 float af_softclip(float a);

+bool af_lavrresample_test_conversion(int src_format, int dst_format);
+
 #endif /* MPLAYER_AF_H */
@@ -1,172 +0,0 @@
-/*
- * The Bauer stereophonic-to-binaural DSP using bs2b library:
- * http://bs2b.sourceforge.net/
- *
- * Copyright (c) 2009 Andrew Savchenko
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <bs2b.h>
-#include <inttypes.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "af.h"
-#include "options/m_option.h"
-
-/// Internal specific data of the filter
-struct af_bs2b {
-    int fcut;           ///< cut frequency in Hz
-    int feed;           ///< feed level for low frequencies in 0.1*dB
-    int profile  ;      ///< profile (available crossfeed presets)
-    t_bs2bdp filter;    ///< instance of a library filter
-};
-
-#define DEF_FILTER(fmt, name) \
-static int filter_##name(struct af_instance *af, struct mp_audio *data) \
-{ \
-    if (!data) \
-        return 0; \
-    if (af_make_writeable(af, data) < 0) { \
-        talloc_free(data); \
-        return -1; \
-    } \
-    bs2b_cross_feed_##name(((struct af_bs2b*)(af->priv))->filter, \
-        (data->planes[0]), data->samples); \
-    af_add_output_frame(af, data); \
-    return 0; \
-}
-
-#define GET_FILTER(fmt, name) \
-    case AF_FORMAT_##fmt: return filter_##name;
-
-#define FILTERS \
-    FILTER(FLOAT,   f)      \
-    FILTER(S32,     s32)    \
-    FILTER(U32,     u32)    \
-    FILTER(S24,     s24)    \
-    FILTER(U24,     u24)    \
-    FILTER(S16,     s16)    \
-    FILTER(U16,     u16)    \
-    FILTER(S8,      s8)     \
-    FILTER(U8,      u8)
-
-#define FILTER DEF_FILTER
-FILTERS
-#undef FILTER
-
-typedef int (*filter)(struct af_instance *af, struct mp_audio *d);
-static filter get_filter(int fmt)
-{
-    switch (fmt) {
-#define FILTER GET_FILTER
-FILTERS
-#undef FILTER
-    default: return NULL;
-    }
-}
-
-/// Initialization and runtime control
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    struct af_bs2b *s = af->priv;
-
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        int format;
-        // Sanity check
-        if (!arg) return AF_ERROR;
-
-        format           = ((struct mp_audio*)arg)->format;
-        af->data->rate   = ((struct mp_audio*)arg)->rate;
-        mp_audio_set_num_channels(af->data, 2);     // bs2b is useful only for 2ch audio
-        mp_audio_set_format(af->data, format);
-
-        /* check for formats supported by libbs2b
-           and assign corresponding handlers */
-        af->filter_frame = get_filter(format);
-        if (!af->filter_frame) {
-            af->filter_frame = filter_f;
-            mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-        }
-
-        // bs2b have srate limits, try to resample if needed
-        if (af->data->rate > BS2B_MAXSRATE || af->data->rate < BS2B_MINSRATE) {
-            af->data->rate = BS2B_DEFAULT_SRATE;
-            MP_WARN(af, "Requested sample rate %d Hz is out of bounds [%d..%d] Hz.\n"
-                   "Trying to resample to %d Hz.\n",
-                   af->data->rate, BS2B_MINSRATE, BS2B_MAXSRATE, BS2B_DEFAULT_SRATE);
-        }
-        bs2b_set_srate(s->filter, (long)af->data->rate);
-        MP_VERBOSE(af, "using format %s\n",
-               af_fmt_to_str(af->data->format));
-
-        return af_test_output(af,(struct mp_audio*)arg);
-    }
-    }
-    return AF_UNKNOWN;
-}
-
-/// Deallocate memory and close library
-static void uninit(struct af_instance *af)
-{
-    struct af_bs2b *s = af->priv;
-    if (s->filter)
-        bs2b_close(s->filter);
-}
-
-/// Allocate memory, set function pointers and init library
-static int af_open(struct af_instance *af)
-{
-    struct af_bs2b *s = af->priv;
-    af->control = control;
-    af->uninit  = uninit;
-
-    // NULL means failed initialization
-    if (!(s->filter = bs2b_open())) {
-        return AF_ERROR;
-    }
-
-    if (s->profile)
-        bs2b_set_level(s->filter, s->profile);
-    // set fcut and feed only if specified, otherwise defaults will be used
-    if (s->fcut)
-        bs2b_set_level_fcut(s->filter, s->fcut);
-    if (s->feed)
-        bs2b_set_level_feed(s->filter, s->feed);
-    return AF_OK;
-}
-
-#define OPT_BASE_STRUCT struct af_bs2b
-
-/// Description of this filter
-const struct af_info af_info_bs2b = {
-    .info = "Bauer stereophonic-to-binaural audio filter",
-    .name = "bs2b",
-    .open = af_open,
-    .priv_size = sizeof(struct af_bs2b),
-    .options = (const struct m_option[]) {
-        OPT_INTRANGE("fcut", fcut, 0, BS2B_MINFCUT, BS2B_MAXFCUT),
-        OPT_INTRANGE("feed", feed, 0, BS2B_MINFEED, BS2B_MAXFEED),
-        OPT_CHOICE("profile", profile, 0,
-                   ({"unset", 0},
-                    {"default", BS2B_DEFAULT_CLEVEL},
-                    {"cmoy", BS2B_CMOY_CLEVEL},
-                    {"jmeier", BS2B_JMEIER_CLEVEL})),
-        {0}
-    },
-};
@@ -1,104 +0,0 @@
-/*
- * This filter adds a center channel to the audio stream by
- * averaging the left and right channel.
- * There are two runtime controls one for setting which channel
- * to insert the center-audio into called AF_CONTROL_SUB_CH.
- *
- * FIXME: implement a high-pass filter for better results.
- *
- * copyright (c) 2005 Alex Beregszaszi
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "common/common.h"
-#include "af.h"
-
-// Data for specific instances of this filter
-typedef struct af_center_s
-{
-  int ch;               // Channel number which to insert the filtered data
-}af_center_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  af_center_t* s   = af->priv;
-
-  switch(cmd){
-  case AF_CONTROL_REINIT:{
-    // Sanity check
-    if(!arg) return AF_ERROR;
-
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
-    mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-
-    return af_test_output(af,(struct mp_audio*)arg);
-  }
-  }
-  return AF_UNKNOWN;
-}
-
-static int filter_frame(struct af_instance* af, struct mp_audio* data)
-{
-  if (!data)
-    return 0;
-  if (af_make_writeable(af, data) < 0) {
-    talloc_free(data);
-    return -1;
-  }
-  struct mp_audio*    c   = data;        // Current working data
-  af_center_t*  s   = af->priv; // Setup for this instance
-  float*        a   = c->planes[0];      // Audio data
-  int           nch = c->nch;    // Number of channels
-  int           len = c->samples*c->nch;         // Number of samples in current audio block
-  int           ch  = s->ch;     // Channel in which to insert the center audio
-  register int  i;
-
-  // Run filter
-  for(i=0;i<len;i+=nch){
-    // Average left and right
-    a[i+ch] = (a[i]/2) + (a[i+1]/2);
-  }
-
-  af_add_output_frame(af, data);
-  return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
-  af->control=control;
-  af->filter_frame = filter_frame;
-  return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_center_t
-const struct af_info af_info_center = {
-    .info = "Audio filter for adding a center channel",
-    .name = "center",
-    .flags = AF_FLAGS_NOT_REENTRANT,
-    .open = af_open,
-    .priv_size = sizeof(af_center_t),
-    .options = (const struct m_option[]) {
-        OPT_INTRANGE("channel", ch, 0, 0, AF_NCH - 1, OPTDEF_INT(1)),
-        {0}
-    },
-};
@@ -1,122 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdlib.h>
-#include <assert.h>
-
-#include "audio/format.h"
-#include "af.h"
-#include "osdep/endian.h"
-
-static bool test_conversion(int src_format, int dst_format)
-{
-    return (src_format == AF_FORMAT_U24 && dst_format == AF_FORMAT_U32) ||
-           (src_format == AF_FORMAT_S24 && dst_format == AF_FORMAT_S32) ||
-           (src_format == AF_FORMAT_U32 && dst_format == AF_FORMAT_U24) ||
-           (src_format == AF_FORMAT_S32 && dst_format == AF_FORMAT_S24);
-}
-
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-        struct mp_audio *out = af->data;
-
-        if (!test_conversion(in->format, out->format))
-            return AF_DETACH;
-
-        if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_24BIT) {
-            mp_audio_set_format(out, af_fmt_change_bits(in->format, 32));
-        } else if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_32BIT) {
-            mp_audio_set_format(out, af_fmt_change_bits(in->format, 24));
-        } else {
-            abort();
-        }
-
-        out->rate = in->rate;
-        mp_audio_set_channels(out, &in->channels);
-
-        assert(test_conversion(in->format, out->format));
-
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
-    }
-    case AF_CONTROL_SET_FORMAT: {
-        mp_audio_set_format(af->data, *(int*)arg);
-        return AF_OK;
-    }
-    }
-    return AF_UNKNOWN;
-}
-
-// The LSB is always ignored.
-#if BYTE_ORDER == BIG_ENDIAN
-#define SHIFT(x) ((3-(x))*8)
-#else
-#define SHIFT(x) (((x)+1)*8)
-#endif
-
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
-    if (!data)
-        return 0;
-    struct mp_audio *out =
-        mp_audio_pool_get(af->out_pool, af->data, data->samples);
-    if (!out) {
-        talloc_free(data);
-        return -1;
-    }
-    mp_audio_copy_attributes(out, data);
-
-    size_t len = mp_audio_psize(data) / data->bps;
-    if (data->bps == 4) {
-        for (int s = 0; s < len; s++) {
-            uint32_t val = *((uint32_t *)data->planes[0] + s);
-            uint8_t *ptr = (uint8_t *)out->planes[0] + s * 3;
-            ptr[0] = val >> SHIFT(0);
-            ptr[1] = val >> SHIFT(1);
-            ptr[2] = val >> SHIFT(2);
-        }
-    } else {
-        for (int s = 0; s < len; s++) {
-            uint8_t *ptr = (uint8_t *)data->planes[0] + s * 3;
-            uint32_t val = ptr[0] << SHIFT(0)
-                         | ptr[1] << SHIFT(1)
-                         | ptr[2] << SHIFT(2);
-            *((uint32_t *)out->planes[0] + s) = val;
-        }
-    }
-
-    talloc_free(data);
-    af_add_output_frame(af, out);
-    return 0;
-}
-
-static int af_open(struct af_instance *af)
-{
-    af->control = control;
-    af->filter_frame = filter;
-    return AF_OK;
-}
-
-const struct af_info af_info_convert24 = {
-    .info = "Convert between 24 and 32 bit sample format",
-    .name = "convert24",
-    .open = af_open,
-    .test_conversion = test_conversion,
-};
@@ -1,107 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdlib.h>
-#include <assert.h>
-
-#include "af.h"
-#include "audio/format.h"
-#include "osdep/endian.h"
-
-static bool test_conversion(int src_format, int dst_format)
-{
-    if ((src_format & AF_FORMAT_PLANAR) ||
-        (dst_format & AF_FORMAT_PLANAR))
-        return false;
-    if (((src_format & ~AF_FORMAT_SIGN_MASK) ==
-         (dst_format & ~AF_FORMAT_SIGN_MASK)) &&
-        ((src_format & AF_FORMAT_TYPE_MASK) == AF_FORMAT_I))
-        return true;
-    return false;
-}
-
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-        struct mp_audio *out = af->data;
-
-        if (!test_conversion(in->format, out->format))
-            return AF_DETACH;
-
-        out->rate = in->rate;
-        mp_audio_set_channels(out, &in->channels);
-
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
-    }
-    case AF_CONTROL_SET_FORMAT: {
-        mp_audio_set_format(af->data, *(int*)arg);
-        return AF_OK;
-    }
-    }
-    return AF_UNKNOWN;
-}
-
-static void si2us(void *data, int len, int bps)
-{
-    ptrdiff_t i = -(len * bps);
-    uint8_t *p = &((uint8_t *)data)[len * bps];
-    if (BYTE_ORDER == LITTLE_ENDIAN && bps > 1)
-        p += bps - 1;
-    if (len <= 0)
-        return;
-    do {
-        p[i] ^= 0x80;
-    } while (i += bps);
-}
-
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
-    if (!data)
-        return 0;
-    if (af_make_writeable(af, data) < 0) {
-        talloc_free(data);
-        return -1;
-    }
-
-    int infmt = data->format;
-    int outfmt = af->data->format;
-    size_t len = data->samples * data->nch;
-
-    if ((infmt & AF_FORMAT_SIGN_MASK) != (outfmt & AF_FORMAT_SIGN_MASK))
-        si2us(data->planes[0], len, data->bps);
-
-    mp_audio_set_format(data, outfmt);
-    af_add_output_frame(af, data);
-    return 0;
-}
-
-static int af_open(struct af_instance *af)
-{
-    af->control = control;
-    af->filter_frame = filter;
-    return AF_OK;
-}
-
-const struct af_info af_info_convertsignendian = {
-    .info = "Convert between sample format sign",
-    .name = "convertsign",
-    .open = af_open,
-    .test_conversion = test_conversion,
-};
@@ -131,7 +131,7 @@ static void method1_int16(af_drc_t *s, struct mp_audio *c)
    data[i] = tmp;
  }

-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
  newavg = s->mul * curavg;

  // Stores computed values for future smoothing
@@ -168,7 +168,7 @@ static void method1_float(af_drc_t *s, struct mp_audio *c)
  for (i = 0; i < len; i++)
    data[i] *= s->mul;

-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
  newavg = s->mul * curavg;

  // Stores computed values for future smoothing
@@ -216,7 +216,7 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c)
    data[i] = tmp;
  }

-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
  newavg = s->mul * curavg;

  // Stores computed values for future smoothing
@@ -262,7 +262,7 @@ static void method2_float(af_drc_t *s, struct mp_audio *c)
  for (i = 0; i < len; i++)
    data[i] *= s->mul;

-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
  newavg = s->mul * curavg;

  // Stores computed values for future smoothing
@@ -324,7 +324,6 @@ static int af_open(struct af_instance* af){
 const struct af_info af_info_drc = {
    .info = "Dynamic range compression filter",
    .name = "drc",
-    .flags = AF_FLAGS_NOT_REENTRANT,
    .open = af_open,
    .priv_size = sizeof(af_drc_t),
    .options = (const struct m_option[]) {
@@ -1,61 +0,0 @@
-/*
- * The name speaks for itself. This filter is a dummy and will
- * not blow up regardless of what you do with it.
- *
- * Original author: Anders
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "af.h"
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  switch(cmd){
-  case AF_CONTROL_REINIT: ;
-    *af->data = *(struct mp_audio*)arg;
-    MP_VERBOSE(af, "Was reinitialized: %iHz/%ich/%s\n",
-        af->data->rate,af->data->nch,af_fmt_to_str(af->data->format));
-    return AF_OK;
-  }
-  return AF_UNKNOWN;
-}
-
-// Filter data through filter
-static int filter(struct af_instance* af, struct mp_audio* data)
-{
-    af_add_output_frame(af, data);
-    return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
-  af->control=control;
-  af->filter_frame=filter;
-  return AF_OK;
-}
-
-// Description of this filter
-const struct af_info af_info_dummy = {
-    .info = "dummy",
-    .name = "dummy",
-    .open = af_open,
-};
@@ -203,7 +203,6 @@ static int af_open(struct af_instance* af){
 const struct af_info af_info_equalizer = {
  .info = "Equalizer audio filter",
  .name = "equalizer",
-  .flags = AF_FLAGS_NOT_REENTRANT,
  .open = af_open,
  .priv_size = sizeof(af_equalizer_t),
  .options = (const struct m_option[]) {
@@ -1,237 +0,0 @@
-/*
- * This audio filter exports the incoming signal to other processes
- * using memory mapping. The memory mapped area contains a header:
- *   int nch,
- *   int size,
- *   unsigned long long counter (updated every time the  contents of
- *                               the area changes),
- * the rest is payload (non-interleaved).
- *
- * Authors: Anders; Gustavo Sverzut Barbieri <gustavo.barbieri@ic.unicamp.br>
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-#include <unistd.h>
-#include "config.h"
-
-#include <sys/types.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-
-#include "osdep/io.h"
-
-#include "talloc.h"
-#include "af.h"
-#include "options/path.h"
-
-#define DEF_SZ 512 // default buffer size (in samples)
-#define SHARED_FILE "mpv-af_export" /* default file name
-                                           (relative to ~/.mpv/ */
-
-#define SIZE_HEADER (2 * sizeof(int) + sizeof(unsigned long long))
-
-// Data for specific instances of this filter
-typedef struct af_export_s
-{
-  unsigned long long  count; // Used for sync
-  void* buf[AF_NCH];    // Buffers for storing the data before it is exported
-  int   sz;             // Size of buffer in samples
-  int   wi;             // Write index
-  int   fd;             // File descriptor to shared memory area
-  char* filename;       // File to export data
-  uint8_t *mmap_area;   // MMap shared area
-} af_export_t;
-
-
-/* Initialization and runtime control
-   af audio filter instance
-   cmd control command
-   arg argument
-*/
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  af_export_t* s = af->priv;
-  switch (cmd){
-  case AF_CONTROL_REINIT:{
-    int i=0;
-    int mapsize;
-
-    // Free previous buffers
-    free(s->buf[0]);
-
-    // unmap previous area
-    if(s->mmap_area)
-      munmap(s->mmap_area, SIZE_HEADER + (af->data->bps*s->sz*af->data->nch));
-    // close previous file descriptor
-    if(s->fd)
-      close(s->fd);
-
-    // Accept only int16_t as input format (which sucks)
-    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-    mp_audio_set_format(af->data, AF_FORMAT_S16);
-
-    // Allocate new buffers (as one continuous block)
-    s->buf[0] = calloc(s->sz*af->data->nch, af->data->bps);
-    if(NULL == s->buf[0]) {
-      MP_FATAL(af, "Out of memory\n");
-      return AF_ERROR;
-    }
-    for(i = 1; i < af->data->nch; i++)
-      s->buf[i] = (uint8_t *)s->buf[0] + i*s->sz*af->data->bps;
-
-    if (!s->filename) {
-        MP_FATAL(af, "No filename set.\n");
-        return AF_ERROR;
-    }
-
-    // Init memory mapping
-    s->fd = open(s->filename, O_RDWR | O_CREAT | O_TRUNC | O_CLOEXEC, 0640);
-    MP_INFO(af, "Exporting to file: %s\n", s->filename);
-    if(s->fd < 0) {
-      MP_FATAL(af, "Could not open/create file: %s\n",
-             s->filename);
-      return AF_ERROR;
-    }
-
-    // header + buffer
-    mapsize = (SIZE_HEADER + (af->data->bps * s->sz * af->data->nch));
-
-    // grow file to needed size
-    for(i = 0; i < mapsize; i++){
-      char null = 0;
-      write(s->fd, (void*) &null, 1);
-    }
-
-    // mmap size
-    s->mmap_area = mmap(0, mapsize, PROT_READ|PROT_WRITE,MAP_SHARED, s->fd, 0);
-    if(s->mmap_area == NULL)
-      MP_FATAL(af, "Could not mmap file %s\n", s->filename);
-    MP_INFO(af, "Memory mapped to file: %s (%p)\n",
-           s->filename, s->mmap_area);
-
-    // Initialize header
-    *((int*)s->mmap_area) = af->data->nch;
-    *((int*)s->mmap_area + 1) = s->sz * af->data->bps * af->data->nch;
-    msync(s->mmap_area, mapsize, MS_ASYNC);
-
-    // Use test_output to return FALSE if necessary
-    return af_test_output(af, (struct mp_audio*)arg);
-  }
-  }
-  return AF_UNKNOWN;
-}
-
-/* Free allocated memory and clean up other stuff too.
-   af audio filter instance
-*/
-static void uninit( struct af_instance* af )
-{
-    af_export_t* s = af->priv;
-
-    free(s->buf[0]);
-
-    // Free mmaped area
-    if(s->mmap_area)
-      munmap(s->mmap_area, sizeof(af_export_t));
-
-    if(s->fd > -1)
-      close(s->fd);
-}
-
-/* Filter data through filter
-   af audio filter instance
-   data audio data
-*/
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
-  if (!data)
-    return 0;
-  struct mp_audio*      c   = data;          // Current working data
-  af_export_t*  s   = af->priv;     // Setup for this instance
-  int16_t*      a   = c->planes[0];          // Incomming sound
-  int           nch = c->nch;        // Number of channels
-  int           len = c->samples*c->nch; // Number of sample in data chunk
-  int           sz  = s->sz;         // buffer size (in samples)
-  int           flag = 0;            // Set to 1 if buffer is filled
-
-  int           ch, i;
-
-  // Fill all buffers
-  for(ch = 0; ch < nch; ch++){
-    int         wi = s->wi;      // Reset write index
-    int16_t*    b  = s->buf[ch]; // Current buffer
-
-    // Copy data to export buffers
-    for(i = ch; i < len; i += nch){
-      b[wi++] = a[i];
-      if(wi >= sz){ // Don't write outside the end of the buffer
-        flag = 1;
-        break;
-      }
-    }
-    s->wi = wi % s->sz;
-  }
-
-  // Export buffer to mmaped area
-  if(flag){
-    // update buffer in mapped area
-    memcpy(s->mmap_area + SIZE_HEADER, s->buf[0], sz * c->bps * nch);
-    s->count++; // increment counter (to sync)
-    memcpy(s->mmap_area + SIZE_HEADER - sizeof(s->count),
-           &(s->count), sizeof(s->count));
-  }
-
-  af_add_output_frame(af, data);
-  return 0;
-}
-
-/* Allocate memory and set function pointers
-   af audio filter instance
-   returns AF_OK or AF_ERROR
-*/
-static int af_open( struct af_instance* af )
-{
-  af->control = control;
-  af->uninit  = uninit;
-  af->filter_frame = filter;
-  af_export_t *priv = af->priv;
-
-  if (!priv->filename || !priv->filename[0]) {
-      MP_FATAL(af, "no export filename given");
-      return AF_ERROR;
-  }
-
-  return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_export_t
-const struct af_info af_info_export = {
-    .info = "Sound export filter",
-    .name = "export",
-    .open = af_open,
-    .priv_size = sizeof(af_export_t),
-    .options = (const struct m_option[]) {
-        OPT_STRING("filename", filename, 0),
-        OPT_INTRANGE("buffersamples", sz, 0, 1, 2048, OPTDEF_INT(DEF_SZ)),
-        {0}
-    },
-};
@@ -1,132 +0,0 @@
-/*
- * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <inttypes.h>
-#include <math.h>
-#include <limits.h>
-
-#include "common/common.h"
-#include "af.h"
-
-// Data for specific instances of this filter
-typedef struct af_extrastereo_s
-{
-    float mul;
-}af_extrastereo_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  switch(cmd){
-  case AF_CONTROL_REINIT:{
-    // Sanity check
-    if(!arg) return AF_ERROR;
-
-    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-    mp_audio_force_interleaved_format(af->data);
-    mp_audio_set_num_channels(af->data, 2);
-    if (af->data->format != AF_FORMAT_FLOAT)
-        mp_audio_set_format(af->data, AF_FORMAT_S16);
-
-    return af_test_output(af,(struct mp_audio*)arg);
-  }
-  }
-  return AF_UNKNOWN;
-}
-
-// Filter data through filter
-static void play_s16(af_extrastereo_t *s, struct mp_audio* data)
-{
-  register int i = 0;
-  int16_t *a = (int16_t*)data->planes[0];       // Audio data
-  int len = data->samples*data->nch;            // Number of samples
-  int avg, l, r;
-
-  for (i = 0; i < len; i+=2)
-  {
-    avg = (a[i] + a[i + 1]) / 2;
-
-    l = avg + (int)(s->mul * (a[i] - avg));
-    r = avg + (int)(s->mul * (a[i + 1] - avg));
-
-    a[i] = MPCLAMP(l, SHRT_MIN, SHRT_MAX);
-    a[i + 1] = MPCLAMP(r, SHRT_MIN, SHRT_MAX);
-  }
-}
-
-static void play_float(af_extrastereo_t *s, struct mp_audio* data)
-{
-  register int i = 0;
-  float *a = (float*)data->planes[0];   // Audio data
-  int len = data->samples * data->nch;  // Number of samples
-  float avg, l, r;
-
-  for (i = 0; i < len; i+=2)
-  {
-    avg = (a[i] + a[i + 1]) / 2;
-
-    l = avg + (s->mul * (a[i] - avg));
-    r = avg + (s->mul * (a[i + 1] - avg));
-
-    a[i] = af_softclip(l);
-    a[i + 1] = af_softclip(r);
-  }
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *data)
-{
-    if (!data)
-        return 0;
-    if (af_make_writeable(af, data) < 0) {
-        talloc_free(data);
-        return -1;
-    }
-    if (data->format == AF_FORMAT_FLOAT) {
-        play_float(af->priv, data);
-    } else {
-        play_s16(af->priv, data);
-    }
-    af_add_output_frame(af, data);
-    return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
-  af->control=control;
-  af->filter_frame = filter_frame;
-
-  return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_extrastereo_t
-const struct af_info af_info_extrastereo = {
-    .info = "Increase difference between audio channels",
-    .name = "extrastereo",
-    .flags = AF_FLAGS_NOT_REENTRANT,
-    .open = af_open,
-    .priv_size = sizeof(af_extrastereo_t),
-    .options = (const struct m_option[]) {
-        OPT_FLOAT("mul", mul, 0, OPTDEF_FLOAT(2.5)),
-        {0}
-    },
-};
@@ -1,670 +0,0 @@
-/*
- * Experimental audio filter that mixes 5.1 and 5.1 with matrix
- * encoded rear channels into headphone signal using FIR filtering
- * with HRTF.
- *
- * Author: ylai
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-//#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-
-#include <math.h>
-#include <libavutil/common.h>
-
-#include "af.h"
-#include "dsp.h"
-
-/* HRTF filter coefficients and adjustable parameters */
-#include "af_hrtf.h"
-
-typedef struct af_hrtf_s {
-    /* Lengths */
-    int dlbuflen, hrflen, basslen;
-    /* L, C, R, Ls, Rs channels */
-    float *lf, *rf, *lr, *rr, *cf, *cr;
-    const float *cf_ir, *af_ir, *of_ir, *ar_ir, *or_ir, *cr_ir;
-    int cf_o, af_o, of_o, ar_o, or_o, cr_o;
-    /* Bass */
-    float *ba_l, *ba_r;
-    float *ba_ir;
-    /* Whether to matrix decode the rear center channel */
-    int matrix_mode;
-    /* How to decode the input:
-       0 = 5/5+1 channels
-       1 = 2 channels
-       2 = matrix encoded 2 channels */
-    int decode_mode;
-    /* Full wave rectified (FWR) amplitudes and gain used to steer the
-       active matrix decoding of front channels (variable names
-       lpr/lmr means Lt + Rt, Lt - Rt) */
-    float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
-    float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
-    /* Matrix input decoding require special FWR buffer, since the
-       decoding is done in place. */
-    float *fwrbuf_l, *fwrbuf_r, *fwrbuf_lr, *fwrbuf_rr;
-    /* Rear channel delay buffer for matrix decoding */
-    float *rear_dlbuf;
-    /* Full wave rectified amplitude and gain used to steer the active
-       matrix decoding of center rear channel */
-    float lr_fwr, rr_fwr, lrprr_fwr, lrmrr_fwr;
-    float adapt_lr_gain, adapt_rr_gain;
-    float adapt_lrprr_gain, adapt_lrmrr_gain;
-    /* Cyclic position on the ring buffer */
-    int cyc_pos;
-    int print_flag;
-    int mode;
-} af_hrtf_t;
-
-/* Convolution on a ring buffer
- *    nx:       length of the ring buffer
- *    nk:       length of the convolution kernel
- *    sx:       ring buffer
- *    sk:       convolution kernel
- *    offset:   offset on the ring buffer, can be
- */
-static float conv(const int nx, const int nk, const float *sx, const float *sk,
-                  const int offset)
-{
-    /* k = reminder of offset / nx */
-    int k = offset >= 0 ? offset % nx : nx + (offset % nx);
-
-    if(nk + k <= nx)
-        return af_filter_fir(nk, sx + k, sk);
-    else
-        return af_filter_fir(nk + k - nx, sx, sk + nx - k) +
-            af_filter_fir(nx - k, sx + k, sk);
-}
-
-/* Detect when the impulse response starts (significantly) */
-static int pulse_detect(const float *sx)
-{
-    /* nmax must be the reference impulse response length (128) minus
-       s->hrflen */
-    const int nmax = 128 - HRTFFILTLEN;
-    const float thresh = IRTHRESH;
-    int i;
-
-    for(i = 0; i < nmax; i++)
-        if(fabs(sx[i]) > thresh)
-            return i;
-    return 0;
-}
-
-/* Fuzzy matrix coefficient transfer function to "lock" the matrix on
-   a effectively passive mode if the gain is approximately 1 */
-static inline float passive_lock(float x)
-{
-   const float x1 = x - 1;
-   const float ax1s = fabs(x - 1) * (1.0 / MATAGCLOCK);
-
-   return x1 - x1 / (1 + ax1s * ax1s) + 1;
-}
-
-/* Unified active matrix decoder for 2 channel matrix encoded surround
-   sources */
-static inline void matrix_decode(short *in, const int k, const int il,
-                          const int ir, const int decode_rear,
-                          const int dlbuflen,
-                          float l_fwr, float r_fwr,
-                          float lpr_fwr, float lmr_fwr,
-                          float *adapt_l_gain, float *adapt_r_gain,
-                          float *adapt_lpr_gain, float *adapt_lmr_gain,
-                          float *lf, float *rf, float *lr,
-                          float *rr, float *cf)
-{
-   const int kr = (k + MATREARDELAY) % dlbuflen;
-   float l_gain = (l_fwr + r_fwr) /
-      (1 + l_fwr + l_fwr);
-   float r_gain = (l_fwr + r_fwr) /
-      (1 + r_fwr + r_fwr);
-   /* The 2nd axis has strong gain fluctuations, and therefore require
-      limits.  The factor corresponds to the 1 / amplification of (Lt
-      - Rt) when (Lt, Rt) is strongly correlated. (e.g. during
-      dialogues).  It should be bigger than -12 dB to prevent
-      distortion. */
-   float lmr_lim_fwr = lmr_fwr > M9_03DB * lpr_fwr ?
-      lmr_fwr : M9_03DB * lpr_fwr;
-   float lpr_gain = (lpr_fwr + lmr_lim_fwr) /
-      (1 + lpr_fwr + lpr_fwr);
-   float lmr_gain = (lpr_fwr + lmr_lim_fwr) /
-      (1 + lmr_lim_fwr + lmr_lim_fwr);
-   float lmr_unlim_gain = (lpr_fwr + lmr_fwr) /
-      (1 + lmr_fwr + lmr_fwr);
-   float lpr, lmr;
-   float l_agc, r_agc, lpr_agc, lmr_agc;
-   float f, d_gain, c_gain, c_agc_cfk;
-
-#if 0
-   static int counter = 0;
-   static FILE *fp_out;
-
-   if(counter == 0)
-      fp_out = fopen("af_hrtf.log", "w");
-   if(counter % 240 == 0)
-      fprintf(fp_out, "%g %g %g %g %g ", counter * (1.0 / 48000),
-              l_gain, r_gain, lpr_gain, lmr_gain);
-#endif
-
-   /*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/
-   /* AGC adaption */
-   d_gain = (fabs(l_gain - *adapt_l_gain) +
-             fabs(r_gain - *adapt_r_gain)) * 0.5;
-   f = d_gain * (1.0 / MATAGCTRIG);
-   f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
-   *adapt_l_gain = (1 - f) * *adapt_l_gain + f * l_gain;
-   *adapt_r_gain = (1 - f) * *adapt_r_gain + f * r_gain;
-   /* Matrix */
-   l_agc = in[il] * passive_lock(*adapt_l_gain);
-   r_agc = in[ir] * passive_lock(*adapt_r_gain);
-   cf[k] = (l_agc + r_agc) * M_SQRT1_2;
-   if(decode_rear) {
-      lr[kr] = rr[kr] = (l_agc - r_agc) * M_SQRT1_2;
-      /* Stereo rear channel is steered with the same AGC steering as
-         the decoding matrix. Note this requires a fast updating AGC
-         at the order of 20 ms (which is the case here). */
-      lr[kr] *= (l_fwr + l_fwr) /
-         (1 + l_fwr + r_fwr);
-      rr[kr] *= (r_fwr + r_fwr) /
-         (1 + l_fwr + r_fwr);
-   }
-
-   /*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
-   lpr = (in[il] + in[ir]) * M_SQRT1_2;
-   lmr = (in[il] - in[ir]) * M_SQRT1_2;
-   /* AGC adaption */
-   d_gain = fabs(lmr_unlim_gain - *adapt_lmr_gain);
-   f = d_gain * (1.0 / MATAGCTRIG);
-   f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
-   *adapt_lpr_gain = (1 - f) * *adapt_lpr_gain + f * lpr_gain;
-   *adapt_lmr_gain = (1 - f) * *adapt_lmr_gain + f * lmr_gain;
-   /* Matrix */
-   lpr_agc = lpr * passive_lock(*adapt_lpr_gain);
-   lmr_agc = lmr * passive_lock(*adapt_lmr_gain);
-   lf[k] = (lpr_agc + lmr_agc) * M_SQRT1_2;
-   rf[k] = (lpr_agc - lmr_agc) * M_SQRT1_2;
-
-   /*** CENTER FRONT CANCELLATION ***/
-   /* A heuristic approach exploits that Lt + Rt gain contains the
-      information about Lt, Rt correlation.  This effectively reshapes
-      the front and rear "cones" to concentrate Lt + Rt to C and
-      introduce Lt - Rt in L, R. */
-   /* 0.67677 is the emprical lower bound for lpr_gain. */
-   c_gain = 8 * (*adapt_lpr_gain - 0.67677);
-   c_gain = c_gain > 0 ? c_gain : 0;
-   /* c_gain should not be too high, not even reaching full
-      cancellation (~ 0.50 - 0.55 at current AGC implementation), or
-      the center will s0und too narrow. */
-   c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
-   c_agc_cfk = c_gain * cf[k];
-   lf[k] -= c_agc_cfk;
-   rf[k] -= c_agc_cfk;
-   cf[k] += c_agc_cfk + c_agc_cfk;
-#if 0
-   if(counter % 240 == 0)
-      fprintf(fp_out, "%g %g %g %g %g\n",
-              *adapt_l_gain, *adapt_r_gain,
-              *adapt_lpr_gain, *adapt_lmr_gain,
-              c_gain);
-   counter++;
-#endif
-}
-
-static inline void update_ch(af_hrtf_t *s, short *in, const int k)
-{
-    const int fwr_pos = (k + FWRDURATION) % s->dlbuflen;
-    /* Update the full wave rectified total amplitude */
-    /* Input matrix decoder */
-    if(s->decode_mode == HRTF_MIX_MATRIX2CH) {
-       s->l_fwr += abs(in[0]) - fabs(s->fwrbuf_l[fwr_pos]);
-       s->r_fwr += abs(in[1]) - fabs(s->fwrbuf_r[fwr_pos]);
-       s->lpr_fwr += abs(in[0] + in[1]) -
-          fabs(s->fwrbuf_l[fwr_pos] + s->fwrbuf_r[fwr_pos]);
-       s->lmr_fwr += abs(in[0] - in[1]) -
-          fabs(s->fwrbuf_l[fwr_pos] - s->fwrbuf_r[fwr_pos]);
-    }
-    /* Rear matrix decoder */
-    if(s->matrix_mode) {
-       s->lr_fwr += abs(in[2]) - fabs(s->fwrbuf_lr[fwr_pos]);
-       s->rr_fwr += abs(in[3]) - fabs(s->fwrbuf_rr[fwr_pos]);
-       s->lrprr_fwr += abs(in[2] + in[3]) -
-          fabs(s->fwrbuf_lr[fwr_pos] + s->fwrbuf_rr[fwr_pos]);
-       s->lrmrr_fwr += abs(in[2] - in[3]) -
-          fabs(s->fwrbuf_lr[fwr_pos] - s->fwrbuf_rr[fwr_pos]);
-    }
-
-    switch (s->decode_mode) {
-    case HRTF_MIX_51:
-       /* 5/5+1 channel sources */
-       s->lf[k] = in[0];
-       s->cf[k] = in[4];
-       s->rf[k] = in[1];
-       s->fwrbuf_lr[k] = s->lr[k] = in[2];
-       s->fwrbuf_rr[k] = s->rr[k] = in[3];
-       break;
-    case HRTF_MIX_MATRIX2CH:
-       /* Matrix encoded 2 channel sources */
-       s->fwrbuf_l[k] = in[0];
-       s->fwrbuf_r[k] = in[1];
-       matrix_decode(in, k, 0, 1, 1, s->dlbuflen,
-                     s->l_fwr, s->r_fwr,
-                     s->lpr_fwr, s->lmr_fwr,
-                     &(s->adapt_l_gain), &(s->adapt_r_gain),
-                     &(s->adapt_lpr_gain), &(s->adapt_lmr_gain),
-                     s->lf, s->rf, s->lr, s->rr, s->cf);
-       break;
-    case HRTF_MIX_STEREO:
-       /* Stereo sources */
-       s->lf[k] = in[0];
-       s->rf[k] = in[1];
-       s->cf[k] = s->lr[k] = s->rr[k] = 0;
-       break;
-    }
-
-    /* We need to update the bass compensation delay line, too. */
-    // TODO: should this use lf/cf/rf etc. instead?
-    s->ba_l[k] = in[0];
-    s->ba_r[k] = in[1];
-    if (s->decode_mode == HRTF_MIX_51) {
-        s->ba_l[k] += in[4] + in[2];
-        s->ba_r[k] += in[4] + in[3];
-    }
-}
-
-static void clear_coeff(af_hrtf_t *s, float *c)
-{
-    memset(c, 0, s->dlbuflen * sizeof(float));
-}
-
-static void reset(af_hrtf_t *s)
-{
-    clear_coeff(s, s->lf);
-    clear_coeff(s, s->rf);
-    clear_coeff(s, s->lr);
-    clear_coeff(s, s->rr);
-    clear_coeff(s, s->cf);
-    clear_coeff(s, s->cr);
-    clear_coeff(s, s->ba_l);
-    clear_coeff(s, s->ba_r);
-    clear_coeff(s, s->fwrbuf_l);
-    clear_coeff(s, s->fwrbuf_r);
-    clear_coeff(s, s->fwrbuf_lr);
-    clear_coeff(s, s->fwrbuf_rr);
-}
-
-/* Initialization and runtime control */
-static int control(struct af_instance *af, int cmd, void* arg)
-{
-    af_hrtf_t *s = af->priv;
-    int test_output_res;
-
-    switch(cmd) {
-    case AF_CONTROL_REINIT:
-        reset(s);
-        af->data->rate = 48000;
-        mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch);
-        if(af->data->nch == 2) {
-            /* 2 channel input */
-            if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
-                /* Default behavior is stereo mixing. */
-                s->decode_mode = HRTF_MIX_STEREO;
-            }
-        } else if (af->data->nch < 5) {
-            mp_audio_set_channels_old(af->data, 5);
-        }
-        mp_audio_set_format(af->data, AF_FORMAT_S16);
-        test_output_res = af_test_output(af, (struct mp_audio*)arg);
-        // after testing input set the real output format
-        mp_audio_set_num_channels(af->data, 2);
-        s->print_flag = 1;
-        return test_output_res;
-    case AF_CONTROL_RESET:
-        reset(s);
-        return AF_OK;
-    }
-
-    return AF_UNKNOWN;
-}
-
-/* Deallocate memory */
-static void uninit(struct af_instance *af)
-{
-        af_hrtf_t *s = af->priv;
-
-        free(s->lf);
-        free(s->rf);
-        free(s->lr);
-        free(s->rr);
-        free(s->cf);
-        free(s->cr);
-        free(s->ba_l);
-        free(s->ba_r);
-        free(s->ba_ir);
-        free(s->fwrbuf_l);
-        free(s->fwrbuf_r);
-        free(s->fwrbuf_lr);
-        free(s->fwrbuf_rr);
-}
-
-/* Filter data through filter
-
-Two "tricks" are used to compensate the "color" of the KEMAR data:
-
-1. The KEMAR data is refiltered to ensure that the front L, R channels
-on the same side of the ear are equalized (especially in the high
-frequencies).
-
-2. A bass compensation is introduced to ensure that 0-200 Hz are not
-damped (without any real 3D acoustical image, however).
-*/
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
-    af_hrtf_t *s = af->priv;
-
-    if (!data)
-        return 0;
-    struct mp_audio *outframe =
-        mp_audio_pool_get(af->out_pool, &af->fmt_out, data->samples);
-    if (!outframe) {
-        talloc_free(data);
-        return -1;
-    }
-    mp_audio_copy_attributes(outframe, data);
-
-    short *in = data->planes[0]; // Input audio data
-    short *out = outframe->planes[0]; // Output audio data
-    short *end = in + data->samples * data->nch; // Loop end
-    float common, left, right, diff, left_b, right_b;
-    const int dblen = s->dlbuflen, hlen = s->hrflen, blen = s->basslen;
-
-    if(s->print_flag) {
-        s->print_flag = 0;
-        switch (s->decode_mode) {
-        case HRTF_MIX_51:
-          MP_INFO(af, "Using HRTF to mix %s discrete surround into "
-                 "L, R channels\n", s->matrix_mode ? "5+1" : "5");
-          break;
-        case HRTF_MIX_STEREO:
-          MP_INFO(af, "Using HRTF to mix stereo into "
-                 "L, R channels\n");
-          break;
-        case HRTF_MIX_MATRIX2CH:
-          MP_INFO(af, "Using active matrix to decode 2 channel "
-                 "input, HRTF to mix %s matrix surround into "
-                 "L, R channels\n", "3/2");
-          break;
-        default:
-          MP_WARN(af, "bogus decode_mode: %d\n", s->decode_mode);
-          break;
-        }
-
-       if(s->matrix_mode)
-          MP_INFO(af, "Using active matrix to decode rear center "
-                 "channel\n");
-    }
-
-    /* MPlayer's 5 channel layout (notation for the variable):
-     *
-     * 0: L (LF), 1: R (RF), 2: Ls (LR), 3: Rs (RR), 4: C (CF), matrix
-     * encoded: Cs (CR)
-     *
-     * or: L = left, C = center, R = right, F = front, R = rear
-     *
-     * Filter notation:
-     *
-     *      CF
-     * OF        AF
-     *      Ear->
-     * OR        AR
-     *      CR
-     *
-     * or: C = center, A = same side, O = opposite, F = front, R = rear
-     */
-
-    while(in < end) {
-        const int k = s->cyc_pos;
-
-        update_ch(s, in, k);
-
-        /* Simulate a 7.5 ms -20 dB echo of the center channel in the
-           front channels (like reflection from a room wall) - a kind of
-           psycho-acoustically "cheating" to focus the center front
-           channel, which is normally hard to be perceived as front */
-        s->lf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
-        s->rf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
-
-        switch (s->decode_mode) {
-        case HRTF_MIX_51:
-        case HRTF_MIX_MATRIX2CH:
-           /* Mixer filter matrix */
-           common = conv(dblen, hlen, s->cf, s->cf_ir, k + s->cf_o);
-           if(s->matrix_mode) {
-              /* In matrix decoding mode, the rear channel gain must be
-                 renormalized, as there is an additional channel. */
-              matrix_decode(in, k, 2, 3, 0, s->dlbuflen,
-                            s->lr_fwr, s->rr_fwr,
-                            s->lrprr_fwr, s->lrmrr_fwr,
-                            &(s->adapt_lr_gain), &(s->adapt_rr_gain),
-                            &(s->adapt_lrprr_gain), &(s->adapt_lrmrr_gain),
-                            s->lr, s->rr, NULL, NULL, s->cr);
-              common +=
-                 conv(dblen, hlen, s->cr, s->cr_ir, k + s->cr_o) *
-                 M1_76DB;
-              left    =
-                 ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
-                   conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
-                   (conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
-                    conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o)) *
-                   M1_76DB + common);
-              right   =
-                 ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
-                   conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
-                   (conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
-                    conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o)) *
-                   M1_76DB + common);
-           } else {
-              left    =
-                 ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
-                   conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
-                   conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
-                   conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o) +
-                   common);
-              right   =
-                 ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
-                   conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
-                   conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
-                   conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o) +
-                   common);
-           }
-           break;
-        case HRTF_MIX_STEREO:
-           left    =
-              ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
-                conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o));
-           right   =
-              ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
-                conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o));
-           break;
-        default:
-            /* make gcc happy */
-            left = 0.0;
-            right = 0.0;
-            break;
-        }
-
-        /* Bass compensation for the lower frequency cut of the HRTF.  A
-           cross talk of the left and right channel is introduced to
-           match the directional characteristics of higher frequencies.
-           The bass will not have any real 3D perception, but that is
-           OK (note at 180 Hz, the wavelength is about 2 m, and any
-           spatial perception is impossible). */
-        left_b  = conv(dblen, blen, s->ba_l, s->ba_ir, k);
-        right_b = conv(dblen, blen, s->ba_r, s->ba_ir, k);
-        left  += (1 - BASSCROSS) * left_b  + BASSCROSS * right_b;
-        right += (1 - BASSCROSS) * right_b + BASSCROSS * left_b;
-        /* Also mix the LFE channel (if available) */
-        if(data->nch >= 6) {
-            left  += in[5] * M3_01DB;
-            right += in[5] * M3_01DB;
-        }
-
-        /* Amplitude renormalization. */
-        left  *= AMPLNORM;
-        right *= AMPLNORM;
-
-        switch (s->decode_mode) {
-        case HRTF_MIX_51:
-        case HRTF_MIX_STEREO:
-           /* "Cheating": linear stereo expansion to amplify the 3D
-              perception.  Note: Too much will destroy the acoustic space
-              and may even result in headaches. */
-           diff = STEXPAND2 * (left - right);
-           out[0] = av_clip_int16(left  + diff);
-           out[1] = av_clip_int16(right - diff);
-           break;
-        case HRTF_MIX_MATRIX2CH:
-           /* Do attempt any stereo expansion with matrix encoded
-              sources.  The L, R channels are already stereo expanded
-              by the steering, any further stereo expansion will sound
-              very unnatural. */
-           out[0] = av_clip_int16(left);
-           out[1] = av_clip_int16(right);
-           break;
-        }
-
-        /* Next sample... */
-        in = &in[data->nch];
-        out = &out[af->data->nch];
-        (s->cyc_pos)--;
-        if(s->cyc_pos < 0)
-            s->cyc_pos += dblen;
-    }
-
-    talloc_free(data);
-    af_add_output_frame(af, outframe);
-    return 0;
-}
-
-static int allocate(af_hrtf_t *s)
-{
-    if ((s->lf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->rf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->lr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->rr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->cf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->cr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->ba_l = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->ba_r = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->fwrbuf_l =
-         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->fwrbuf_r =
-         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->fwrbuf_lr =
-         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->fwrbuf_rr =
-         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    return 0;
-}
-
-/* Allocate memory and set function pointers */
-static int af_open(struct af_instance* af)
-{
-    int i;
-    af_hrtf_t *s;
-    float fc;
-
-    af->control = control;
-    af->uninit = uninit;
-    af->filter_frame = filter;
-
-    s = af->priv;
-
-    s->dlbuflen = DELAYBUFLEN;
-    s->hrflen = HRTFFILTLEN;
-    s->basslen = BASSFILTLEN;
-
-    s->cyc_pos = s->dlbuflen - 1;
-    /* With a full (two axis) steering matrix decoder, s->matrix_mode
-       should not be enabled lightly (it will also steer the Ls, Rs
-       channels). */
-    s->matrix_mode = 0;
-    s->decode_mode = HRTF_MIX_51;
-
-    switch (s->mode) {
-    case 0: /* Use matrix rear decoding. */
-        s->matrix_mode = 1;
-        break;
-    case 1: /* Input needs matrix decoding. */
-        s->decode_mode = HRTF_MIX_MATRIX2CH;
-        break;
-    case 2:
-        s->matrix_mode = 0;
-        break;
-    }
-
-    s->print_flag = 1;
-
-    if (allocate(s) != 0) {
-        MP_ERR(af, "Memory allocation error.\n");
-        return AF_ERROR;
-    }
-
-    for(i = 0; i < s->dlbuflen; i++)
-        s->lf[i] = s->rf[i] = s->lr[i] = s->rr[i] = s->cf[i] =
-            s->cr[i] = 0;
-
-    s->lr_fwr =
-        s->rr_fwr = 0;
-
-    s->cf_ir = cf_filt + (s->cf_o = pulse_detect(cf_filt));
-    s->af_ir = af_filt + (s->af_o = pulse_detect(af_filt));
-    s->of_ir = of_filt + (s->of_o = pulse_detect(of_filt));
-    s->ar_ir = ar_filt + (s->ar_o = pulse_detect(ar_filt));
-    s->or_ir = or_filt + (s->or_o = pulse_detect(or_filt));
-    s->cr_ir = cr_filt + (s->cr_o = pulse_detect(cr_filt));
-
-    if((s->ba_ir = malloc(s->basslen * sizeof(float))) == NULL) {
-        MP_ERR(af, "Memory allocation error.\n");
-        return AF_ERROR;
-    }
-    fc = 2.0 * BASSFILTFREQ / (float)af->data->rate;
-    if(af_filter_design_fir(s->basslen, s->ba_ir, &fc, LP | KAISER, 4 * M_PI) ==
-       -1) {
-        MP_ERR(af, "Unable to design low-pass "
-               "filter.\n");
-        return AF_ERROR;
-    }
-    for(i = 0; i < s->basslen; i++)
-        s->ba_ir[i] *= BASSGAIN;
-
-    return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_hrtf_t
-const struct af_info af_info_hrtf = {
-    .info = "HRTF Headphone",
-    .name = "hrtf",
-    .open = af_open,
-    .priv_size = sizeof(af_hrtf_t),
-    .options = (const struct m_option[]) {
-        OPT_CHOICE("mode", mode, 0, ({"m", 0}, {"s", 1}, {"0", 2})),
-        {0}
-    },
-};
@@ -1,510 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#ifndef MPLAYER_AF_HRTF_H
-#define MPLAYER_AF_HRTF_H
-
-#define HRTF_MIX_51 0
-#define HRTF_MIX_STEREO 1
-#define HRTF_MIX_MATRIX2CH 2
-
-/* Amplitude scaling factors */
-#define M17_0DB         0.1414213562
-#define M9_03DB         0.3535533906
-#define M6_99DB         0.4472135955
-#define M4_77DB         0.5773502692
-#define M3_01DB         0.7071067812
-#define M1_76DB         0.8164965809
-
-#define DELAYBUFLEN     1024    /* Length of the delay buffer */
-#define HRTFFILTLEN     64      /* HRTF filter length */
-#define IRTHRESH        0.001   /* Impulse response pruning thresh. */
-
-#define AMPLNORM        M6_99DB /* Overall amplitude renormalization */
-
-#define BASSFILTFREQ    180     /* Bass compensation filter cut (Hz) */
-#define BASSFILTLEN     193     /* Bass compensation filter length */
-#define BASSGAIN        M_SQRT2 /* Bass compensation gain */
-#define BASSCROSS       0.35    /* Bass cross talk */
-
-#define FWRDURATION     240     /* FWR average duration (samples) */
-#define MATREARDELAY    720     /* Matrix mode rear delay (samples) */
-
-#define MATAGCTRIG      8.0     /* (Fuzzy) AGC trigger */
-#define MATAGCDECAY     1.0     /* AGC baseline decay rate (1/samp.) */
-#define MATAGCLOCK      0.2     /* AGC range (around 1) where the
-                                   matrix behaves passively */
-#define MATCOMPGAIN     0.37    /* Cross talk compensation gain,
-                                   0.50 - 0.55 is full cancellation. */
-
-#define CFECHODELAY     360     /* Center front echo delay (samples) */
-#define CFECHOAMPL      M17_0DB /* Center front echo amplitude */
-
-#define STEXPAND2       0.07    /* Stereo expansion / 2 */
-
-/* Head related impulse response (HRIR) derived from KEMAR measurement
-   data by Bill Gardner <billg@media.mit.edu> and Keith Martin
-   <kdm@media.mit.edu>
-
-   URL: http://sound.media.mit.edu/KEMAR.html
-
-   Distributed under GPL with authors' permission
-*/
-
-/* EQUALIZED KEMAR HRIR
-
-How to generate these data:
-
-1. You need the MIT Media Lab's KEMAR data, read it into an software
-   capable of signal/time series analysis (like Mathematica, Matlab,
-   ...)
-
-2. Construct an equalizing FIR inverse filter by calculating the
-   transfer function of the front, same side impulse response, then
-   take 1 over the absolute magnitude.
-
-3. Cut the poles in the inverse filter's transfer function
-   specification by limiting the amplification to 2.5 (note, this
-   number assumes that you have correct signal processing
-   normalization of the Fourier transform).
-
-4. Design the FIR inverse filter by calculating the inverse Fourier
-   transform, then chopping the coefficients down to a reasonable
-   number (N = 15 is used here), apply the Kaiser-Bessel window (alpha
-   = 2 is used here).  Note the objective is remove the color bias
-   only (as if you are using an equalizer), _not_ to do a full inverse
-   filtering. (Note: beta = pi*alpha in other notation.)
-
-   For N = 15, alpha = 2, you should get the following impulse
-   response:
-
-    0.001001558668605168,  0.00698457265741865,  0.040453643039829436,
-    0.012230541722147855, -0.11939760844854072,  0.16468099899755967,
-   -0.30297563073747436,   1.3140211791355982,  -0.30297563073747436,
-    0.16468099899755967,  -0.11939760844854072,  0.012230541722147855,
-    0.040453643039829436,  0.00698457265741865,  0.001001558668605168
-
-5. Linearly convolve all KEMAR HRIR with this inverse filter.
-
-6. Resample from 44.1 kHz sampling frequency to 48 kHz.
-*/
-
-/* Center front (-5 degree) - not 0 degree in order to create a clear
-   front image from a finite distance */
-static const float cf_filt[128] = {
-   -0.00008638082319075036, 0.0003198059946385229,
-   -0.0005010631339162132, 0.0011424741331126876,
-   -0.001584220794688753, 0.001742715363246275,
-   -0.0011080796626780694, 0.0001651829990860167,
-   0.005235028337314985, 0.0035223828473357776,
-   0.010057681388431303, -0.033469432129545514,
-   0.013391253316233523, 0.004858462839827063,
-   0.08172161220103627, 0.26158596134500023,
-   -0.12420314583323326, -0.14298458356097565,
-   0.14421897280453896, -0.1635792507629016,
-   -0.02187136722480014, 0.2426863044711817,
-   0.07229814207917194, 0.0942742651913879,
-   0.29856830878076834, 0.2944146162057754,
-   -0.12122157003421209, -0.19640092165631157,
-   0.11623836502034968, -0.05794027397995521,
-   -0.34313138341973776, -0.19487516249168105,
-   0.010118993953802401, -0.09460218797710966,
-   -0.16761521117359582, 0.004154461610153861,
-   0.052768641758969316, -0.00041823982226147407,
-   0.021634960445143514, 0.07562793486871108,
-   0.08895407129506479, 0.039857755093416214,
-   0.044257936180422945, 0.061557584906101664,
-   0.015547268541895703, -0.023908191934932484,
-   0.009498030443468223, 0.03816269003221337,
-   -0.009820500607303615, -0.042003975527908084,
-   -0.03335447117311547, -0.029294510859746596,
-   -0.05212623136198511, -0.073427547153205,
-   -0.061190797824958836, -0.04620925059966413,
-   -0.04204619420702159, -0.02331915902615157,
-   0.00211481411477094, 0.00852563995740107,
-   0.008766809731743399, 0.008666632180812078,
-   0.018279202191625352, 0.02924332751289675,
-   0.022293148257836494, 0.012362146008584188,
-   0.008572582458807008, 0.006491370763597344,
-   -0.0019366944997535774, -0.006318669309634434,
-   -0.006457921690218669, -0.015050265524669974,
-   -0.02110660282616213, -0.017027809096377904,
-   -0.01651052305334348, -0.022770064150046673,
-   -0.01999875754219472, -0.012294792027337775,
-   -0.011506057031057188, -0.011448970577312903,
-   -0.004823572302580925, 0.0022451134042777883,
-   0.004145473526859826, 0.005629030064546135,
-   0.008588029213398785, 0.010092048834844231,
-   0.007182013245552008, 0.0014600979508720656,
-   -0.0038314646272511756, -0.003443901997881347,
-   -0.0029483418254804047, -0.007609357112679647,
-   -0.006518368948030822, -0.004495803701497202,
-   -0.007109113004849672, -0.008346237278084265,
-   -0.005560847336252453, -0.002993453167553188,
-   -0.005122897816824269, -0.004389782626604215,
-   -0.0010912633695218108, -0.0019712029474458835,
-   -0.005870162265802235, -0.005626159534954128,
-   -0.00027254977910844407, 0.0013794425431633785,
-   -0.0005919083190430062, -0.0007861203545416682,
-   -0.0007049560240893946, -0.0032720188494468868,
-   -0.004460645567968504, -0.0032018528193571696,
-   -0.0030579229375062105, -0.003593998902656612,
-   -0.0038032977997776445, -0.004613776010454773,
-   -0.0059796549143736845, -0.00420126194319768,
-   -0.0012374419948287222, 0.0008572699213050608,
-   0.0021490971020081094, 0.00047295283198381995,
-   -0.0009670277915884887, -0.001354440866080231,
-   -0.002962902746547851, -0.00533935813338141,
-   -0.005469203016468759, -0.004355784273189485
-};
-/* Front,   same side (30 degree) */
-static const float af_filt[128] = {
-   -0.004140580614755493, 0.005790934614385445,
-   0.003318916682081112, 0.014257145544366063,
-   0.007328442487127339, -0.06550381777876194,
-   0.03502225818161845, -0.013090579770708259,
-   0.2517776798694195, 0.420770489950659,
-   -0.3418854608834852, -0.24371032493696737,
-   0.04901356150030018, -0.1778083521632833,
-   0.26448004245714163, 0.23245199964546834,
-   0.033053145803936305, 0.46811222821062415,
-   0.5359265986255611, -0.011912195468533787,
-   -0.39763432601411647, -0.034482864386898314,
-   0.029445398240649626, -0.3850940407863262,
-   -0.3272328478175581, -0.14701421403616477,
-   -0.08522137400169517, -0.14936851633336035,
-   -0.09432605283433723, 0.0991200405937827,
-   0.011075012089917331, -0.0051036489980274685,
-   0.0933903289749412, 0.1344189369609565,
-   0.10507466913017807, 0.04240159324684365,
-   0.06559270110638656, 0.026984119875617524,
-   -0.03359846103238096, -0.018000197099174275,
-   0.042031818548178244, 0.03849039666888434,
-   -0.02450829674011345, -0.03407882403088576,
-   -0.029230189282961977, -0.046964865291761734,
-   -0.09458258700116245, -0.1527349330901158,
-   -0.15411577687826097, -0.08761679790956928,
-   -0.033623549089171874, -0.007204768531481949,
-   0.008273425020444852, 0.021368717994908505,
-   0.04366608267875025, 0.05660907333076205,
-   0.06775726495503939, 0.05621881735271431,
-   0.03576231950669927, 0.02500825834889175,
-   0.015423811076054601, 0.007903258334503761,
-   -0.0053873014137761945, -0.006987955469434698,
-   -0.012027972007598602, -0.025228281243816594,
-   -0.026225091797257318, -0.023809293997344882,
-   -0.03250172017712295, -0.03195696301067249,
-   -0.01784813952189948, -0.01663267233760342,
-   -0.016184530450468065, -0.011659883749357463,
-   -0.0035378511240219163, -0.0005485800790443406,
-   0.0018432660108168625, 0.011634844139907534,
-   0.018333603402051105, 0.020447379185133056,
-   0.00850783664147828, 0.0004694148911037838,
-   -0.0017047130409786676, -0.0022409152834483997,
-   -0.000860472174892845, -0.004111075059198666,
-   -0.003527843382056666, -0.009640160874903018,
-   -0.01750044574231376, -0.015613389403672443,
-   -0.010790028120953001, -0.0095313499955768,
-   -0.007469721416726809, -0.0019186578145117315,
-   -0.00014977322572890802, -0.0029803838028179728,
-   -0.006520567233727221, 0.000035015132033882596,
-   0.009245098100543752, 0.009896930052308656,
-   0.008316744929565786, 0.004575207140193997,
-   -0.0000647420103997081, -0.004502916832871627,
-   -0.004225962213251224, -0.002886014126381486,
-   -0.006416834142585976, -0.007156609995423569,
-   -0.008840274447579472, -0.01441763751386817,
-   -0.015435817484659574, -0.00924487254924743,
-   -0.0021571721940235205, 0.0028540722992305453,
-   0.00273577475088536, -0.000788412365513805,
-   -0.0032650029728365907, -0.003880217646231338,
-   -0.0035302087299613778, -0.0038531436176586246,
-   -0.0011921632190514074, -0.0020722967099011938,
-   -0.004985351145629344, -0.0042375588844648735,
-   -0.003030360463006021, -0.0014161075428041471,
-   -0.0005083025643192044, 0.00035096963769606926
-};
-/* Front,   opposite (-30 degree) */
-static const float of_filt[128] = {
-   -0.000013472538374193126, -0.00008048061877079751,
-   0.000043927265781258155, -0.000017931700794858892,
-   -0.000034774602476112886, -0.00009576223008735474,
-   0.0001557797638630691, -0.00018742885883751094,
-   0.00026512448626705716, -0.0001451040203319678,
-   -0.00008263233117758043, 0.0006486245853639179,
-   -0.0010631408451846698, 0.0026571994100746143,
-   0.0014179177997092787, 0.0062326502956616256,
-   -0.008194149324545333, -0.006568029415878379,
-   0.009538759710818582, 0.012309193558632693,
-   0.12336638055838955, 0.046164307101829005,
-   -0.10228706407884815, 0.04047687260345798,
-   -0.00296595313977046, -0.07057949208414134,
-   0.08172114840714612, 0.08736490764127891,
-   0.05105250431333021, 0.11627179512747428,
-   0.20717888490340705, 0.09375052213570291,
-   -0.09784374168330194, -0.010493571845901443,
-   0.053131894303891716, -0.10157443971694806,
-   -0.16264032634244974, -0.05402369511361273,
-   -0.0274403608654217, -0.09860277022495063,
-   -0.06841875821090282, -0.004434574400066223,
-   -0.0005222661652743502, -0.006231881259827263,
-   0.014410397820340159, 0.04885649512730243,
-   0.04361962569042684, 0.03399214029009391,
-   0.04961073933475931, 0.04067325604132289,
-   0.007850647519227257, 0.004564440466905299,
-   0.02257107958021618, 0.008183791928884486,
-   -0.014913479343180557, -0.018685938460856224,
-   -0.01745737397226911, -0.02327177054233603,
-   -0.03723048632685227, -0.044739390162299685,
-   -0.042651220125613766, -0.03730017561004743,
-   -0.029039465434276192, -0.01885087458914294,
-   -0.01207127752277769, -0.006779800724164512,
-   -0.001930416967444157, 0.000029454577995528385,
-   0.0013822760965755472, 0.0014799128583230202,
-   0.0002068200609199832, 0.0022254295286201083,
-   0.005143858159434566, 0.0018580542060917013,
-   -0.0019426046325146259, -0.0014464042108543495,
-   -0.0034430083560735582, -0.009692758426099499,
-   -0.011840035292593485, -0.010716508855893968,
-   -0.012939889036853034, -0.0121846427935653,
-   -0.006198503315630782, -0.0023186723099380305,
-   -0.002679872498314837, -0.003086020446226295,
-   -0.0015709623347698936, -0.0008147490468332398,
-   -0.0012384575726770983, -0.0005212877089109362,
-   0.0017707578744906142, 0.001324932723905786,
-   -0.0017023653780617696, -0.0045108927752919425,
-   -0.005422155613096912, -0.0039489323837623835,
-   -0.005295995750506547, -0.00629706566356189,
-   -0.004685732198036754, -0.0048076735568143,
-   -0.005978864279217503, -0.005928999306332966,
-   -0.004187703549017582, -0.003213999896976475,
-   -0.0028068699816073414, -0.0010889703907593833,
-   0.0003276714243495386, -0.0013015007040186994,
-   -0.003208050402434782, -0.0025115319088208545,
-   -0.0013787553006401076, -0.0018279087370218635,
-   -0.0025904836507747754, -0.002071221947222004,
-   -0.0026424212922485594, -0.0039837031817577,
-   -0.0041635566057380965, -0.004355223489150822,
-   -0.004350395332709937, -0.0036693292471930263,
-   -0.003386384394185026, -0.003972568655001128,
-   -0.004332336840023821, -0.002648767912111827,
-   -0.001384410080218114, -0.0011353792711849466,
-   -0.0013726264946164232, -0.0020075119315034313
-};
-/* Rear,   same side (135 degree) */
-static const float ar_filt[128] = {
-   0.004573315040810066, 0.0013592578059426913,
-   0.01553271930902704, -0.0002356117224941317,
-   -0.05746098219774702, 0.03430688963370445,
-   0.00808371687447385, 0.21893535841158596,
-   0.2984357591724814, -0.3302799746504719,
-   -0.3194029149806245, 0.21633225051331056,
-   0.24371260938097083, -0.08843705549751085,
-   0.03939684701343366, 0.45386926431114494,
-   0.07599118140753386, -0.18114706160474578,
-   0.285640624686038, 0.4049515236666218,
-   -0.05347890222071792, -0.31464359045319074,
-   -0.1033502246468194, -0.04553593949283157,
-   -0.1880747731157464, -0.13629090230626037,
-   -0.10435789106123239, -0.19818232801888755,
-   -0.16701805476330397, -0.022793111199284,
-   0.058049696762683685, 0.007048321372693906,
-   -0.002966419183225961, 0.10140569697797763,
-   0.11648999956673124, 0.05218347182779882,
-   0.028427001212735392, 0.04151900310166159,
-   -0.0006960604221423734, -0.05898623212226975,
-   -0.03801934531173312, -0.029306970535287986,
-   -0.04549125782835908, -0.0599222718506552,
-   -0.058299618975430116, -0.03765579129720727,
-   -0.03559302657499581, -0.020647901025903054,
-   -0.005720957338744348, -0.0041915732688915545,
-   -0.0011470880098346143, 0.008737404798553,
-   0.023444168098121512, 0.024204226042172663,
-   0.01894897166475026, 0.020807655257479588,
-   0.021570431128040954, 0.006800556178576289,
-   -0.009000089216921362, -0.010969824547067934,
-   -0.0033653428332822374, -0.012676936164668659,
-   -0.026739938673413587, -0.023427869194287573,
-   -0.023302007105117244, -0.023647155590533712,
-   -0.021289317515613106, -0.009120487305069884,
-   0.0009251551667728967, 0.00004285344125653763,
-   -0.00009042365479456271, 0.00022573242339446494,
-   0.00720168491586098, 0.007111875505402431,
-   0.003186514817683482, 0.00810087718334745,
-   0.012619557025922575, 0.007854726400013397,
-   -0.0024013592881066267, -0.001452457473161119,
-   -0.0025535188366093945, -0.012428911627809337,
-   -0.013729251536694145, -0.0070099675146427344,
-   -0.007165284278706593, -0.01639289295622301,
-   -0.015831795079778305, -0.007305768485523729,
-   -0.003608863157004021, -0.0032640528878698084,
-   0.0030901263998481944, 0.00749497566124848,
-   0.002515185532327241, 0.00004840875738621367,
-   0.0017596043486043966, 0.0046229941553338144,
-   0.0034259167322926096, 0.003707347634186093,
-   0.0035584806528586328, -0.0019078936035275198,
-   -0.006057343815214898, -0.0069262470468817,
-   -0.004345020728618624, -0.004177623355574794,
-   -0.005373506556122508, -0.006624933928893836,
-   -0.008679541408588839, -0.010718719681595322,
-   -0.011392246979594496, -0.007893917064389902,
-   -0.0027572935365832536, -0.00006064707149834412,
-   -0.0012512537319656323, -0.0024501501002409786,
-   -0.0022106788572895998, -0.00234124933370301,
-   -0.0008953445167066823, 0.0005393670625637734,
-   -0.00033175600142209297, -0.004023994309351289,
-   -0.008655472335784443, -0.009899957354849682,
-   -0.008664952919996412, -0.00553483124503576,
-   -0.003735336089277662, -0.002754824348643885,
-   -0.0026884314856593368, -0.004084181815125924
-};
-/* Rear,   opposite (-135 degree) */
-static const float or_filt[128] = {
-   0.0001220944028243897, -0.000021785381808441314,
-   5.823057988603169e-6, -0.00001217768176447613,
-   -0.00006123604397345513, 5.574117262531134e-6,
-   -0.00004935331914366778, 1.1771577934768211e-6,
-   -0.000059236211621095756, 9.503536190497286e-6,
-   -0.0001494445696103564, 0.00012248858284145305,
-   -0.0000963975321456313, 6.017905197665205e-6,
-   0.00003353395360402643, -0.0001931511015359506,
-   0.0005113536523931485, -0.0005676652619386114,
-   0.0012057159755477467, 0.0009370492250339692,
-   0.004596472288877981, -0.0018831773384237068,
-   -0.008208535225621212, 0.0038178646400751056,
-   0.008726517739105965, 0.06664363898418262,
-   0.06788684221502142, -0.04492315162807267,
-   -0.04019906311255255, 0.026203059677375153,
-   0.013678129114847544, -0.014334962223993527,
-   0.010141709596167392, 0.11559131576945537,
-   0.1586402064538425, 0.059975334707967023,
-   0.004671725963777715, 0.031498678282775874,
-   0.014338626006524587, -0.014749719448472231,
-   -0.02055508237941379, -0.05407690143992048,
-   -0.07767559836886143, -0.05029091786216801,
-   -0.030808335706574427, -0.03401958135442541,
-   -0.030520368430288967, -0.014168302104259355,
-   0.011907621259989802, 0.014286081013069,
-   0.006377467879613449, 0.018546823568277478,
-   0.028297012771618273, 0.025222339408338186,
-   0.021931611353415138, 0.019708894333646355,
-   0.01729258494072014, 0.017468204169564034,
-   0.009729094845051928, -0.002976992018531901,
-   -0.00956986166277019, -0.016125733548332074,
-   -0.02934094241442545, -0.04133767871051455,
-   -0.043536981145416466, -0.0385966307108608,
-   -0.02784453599342459, -0.018995135307247116,
-   -0.012849534096536747, -0.004437491064613308,
-   0.00028385411598204655, 0.003776874988516643,
-   0.008069432041547833, 0.008764754183751848,
-   0.008020908861878062, 0.006830351461360802,
-   0.002218330884267235, -0.0020478725582339444,
-   -0.003997428121462543, -0.007066287373515421,
-   -0.00940847412544698, -0.010938998446237963,
-   -0.011775483016151306, -0.011391103919484287,
-   -0.010586061195163017, -0.009842793078929053,
-   -0.007753202010139829, -0.00569213732353025,
-   -0.006506783349722073, -0.005346134281903736,
-   -0.003913089814898934, -0.0036091443854759727,
-   -0.0020328564301266287, 0.00017932870773467683,
-   0.0032779786679056357, 0.003969695813293966,
-   0.0020339334412434987, -0.00011345940675415259,
-   -0.0018344103399567666, -0.003556764701666365,
-   -0.004263523639408391, -0.002940568582022133,
-   -0.0034341188272627556, -0.006023399920020824,
-   -0.0077456903203677865, -0.007912219312377842,
-   -0.00625202770436523, -0.00530785086116117,
-   -0.005569722659634311, -0.004664448462594344,
-   -0.0037747773914077747, -0.004175649656985592,
-   -0.004659601521384289, -0.005008602967819641,
-   -0.004730625079902729, -0.0034039554356604146,
-   -0.0017110333873406587, -0.0006091938771510242,
-   -0.0016051679050678297, -0.003312864664007262,
-   -0.004505512715977288, -0.004152222189861692,
-   -0.003218596419678823, -0.0027277806209877343,
-   -0.001715005444317267, -0.0012589960071233749,
-   -0.001852908777923165, -0.002540339553144362
-};
-/* Center rear (180 degree) */
-static const float cr_filt[128] = {
-   -0.00005989110716536726, -0.00022790291829128702,
-   0.0002659166098971966, -0.0003774772716776257,
-   0.0004540309551867803, -0.000420238187386368,
-   0.00025518536450885686, 0.00028285526288953955,
-   -0.001016391007574093, 0.0028634984299063795,
-   0.0021574799687976045, 0.01035121276682072,
-   -0.010481720917298163, -0.013197198495899292,
-   0.0031928225328717195, 0.02903137843618603,
-   0.1632429772511569, 0.1047487989875262,
-   -0.10464685060623742, -0.09260196288035998,
-   -0.007514241993554443, 0.013596249226741712,
-   -0.019876166508450258, 0.1255626123599804,
-   0.3648170359521724, 0.19458249753223478,
-   -0.04434070930031298, 0.046582528121935265,
-   0.09484427347230277, -0.03137795311969644,
-   -0.10297437925363695, -0.09351091015917065,
-   -0.1129521092162064, -0.14925322995658827,
-   -0.1231466295584665, -0.06356719756705227,
-   -0.05442277895126282, -0.07568433015661316,
-   -0.023314932828602003, 0.04260950721318558,
-   0.02249026315598923, 0.02048195669571197,
-   0.05651342117268278, 0.05885038917623213,
-   0.03797102097397795, 0.011767394419953451,
-   0.00560502503429481, 0.005051125343961189,
-   -0.012925933188033823, -0.023918884651306566,
-   -0.013251659441678816, -0.010694772488866284,
-   -0.03080486448617846, -0.03661278237783158,
-   -0.0379227303416262, -0.042189005718490775,
-   -0.026595666344596286, -0.009759025956801257,
-   -0.002064986663513004, -0.002420117028098389,
-   -0.006629991977552491, 0.004619970897631026,
-   0.019450642967537877, 0.0173521119057514,
-   0.017641425439988062, 0.02270029598048491,
-   0.018976431925275348, 0.009299852902290885,
-   -0.001695039371619912, -0.00675162574265618,
-   -0.009380968871003034, -0.011208396125485165,
-   -0.01308640049201482, -0.0165636375633249,
-   -0.022004099870933345, -0.025173458684139286,
-   -0.016918759559175375, -0.00865150653575917,
-   -0.006999929082792643, -0.005454830010518988,
-   -0.0021129521131095317, 0.00018717090054046307,
-   -0.0002864344788569993, 0.0017615225381095569,
-   0.006985907557802283, 0.010577308310476465,
-   0.006466104789306027, -0.0014988738575948326,
-   -0.0039669755229277195, -0.0065156971200080235,
-   -0.009343206924192169, -0.0076430644693577495,
-   -0.004395214976600924, -0.003052735340422483,
-   -0.007019103043066595, -0.00974109267696527,
-   -0.007968015032797376, -0.007801513845528344,
-   -0.007535748903681969, -0.003543341967287925,
-   0.0015083125553729722, 0.0023345972556147025,
-   -0.0010043623069557037, -0.0025295765105203746,
-   -0.0023701840891643634, -0.0005908186035024362,
-   0.0029826252289082847, 0.004829048542117764,
-   0.004488360022902081, 0.00002643748103005408,
-   -0.0042100779212597295, -0.006170600558114495,
-   -0.007267149164680168, -0.006825522903494639,
-   -0.006899834372739123, -0.0073493916110062675,
-   -0.009554351265163382, -0.011790297433830197,
-   -0.010645796603734424, -0.0064661575394022106,
-   -0.002026743466524137, -0.0004337034584909932,
-   -0.0011172647031654614, -0.0017947816283674731,
-   -0.00255615052036616, -0.0017721562881944813,
-   -0.0002379619297227554, 0.0007130120121089036
-};
-
-#endif /* MPLAYER_AF_HRTF_H */
@@ -1,86 +0,0 @@
-/*
- * simple voice removal filter
- *
- * copyright (c) 2006 Reynaldo H. Verdejo Pinochet
- * Based on code by Alex Beregszaszi for his 'center' filter.
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "af.h"
-
-// Data for specific instances of this filter
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-        switch(cmd){
-                case AF_CONTROL_REINIT:
-                mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-                mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-                return af_test_output(af,(struct mp_audio*)arg);
-        }
-        return AF_UNKNOWN;
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *c)
-{
-        if (!c)
-                return 0;
-        if (af_make_writeable(af, c) < 0) {
-                talloc_free(c);
-                return 0;
-        }
-
-        float*          a       = c->planes[0];  // Audio data
-        int                     nch     = c->nch;        // Number of channels
-        int                     len     = c->samples*nch;        // Number of samples in current audio block
-        register int  i;
-
-        /*
-                FIXME1 add a low band pass filter to avoid suppressing
-                centered bass/drums
-                FIXME2 better calculated* attenuation factor
-        */
-
-        for(i=0;i<len;i+=nch)
-        {
-                a[i] = (a[i] - a[i+1]) * 0.7;
-                a[i+1]=a[i];
-        }
-
-        af_add_output_frame(af, c);
-        return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
-        af->control     = control;
-        af->filter_frame = filter_frame;
-        return AF_OK;
-}
-
-// Description of this filter
-const struct af_info af_info_karaoke = {
-        .info = "Simple karaoke/voice-removal audio filter",
-        .name = "karaoke",
-        .flags = AF_FLAGS_NOT_REENTRANT,
-        .open = af_open,
-};
@@ -1,851 +0,0 @@
-/*
- * LADSPA plugin loader
- *
- * Written by Ivo van Poorten <ivop@euronet.nl>
- * Copyright (C) 2004, 2005
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-/* ------------------------------------------------------------------------- */
-
-/* Global Includes */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <inttypes.h>
-#include <math.h>
-#include <limits.h>
-
-#include <dlfcn.h>
-#include <ladspa.h>
-
-/* ------------------------------------------------------------------------- */
-
-/* Local Includes */
-
-#include "af.h"
-
-#define _(x) (x)
-
-/* ------------------------------------------------------------------------- */
-
-/* Filter specific data */
-
-typedef struct af_ladspa_s
-{
-    int status;     /**< Status of the filter.
-                     *   Either AF_OK or AF_ERROR
-                     *   Because MPlayer re-inits audio filters that
-                     *   _clearly_ returned AF_ERROR anyway, I use this
-                     *   in play() to skip the processing and return
-                     *   the data unchanged.
-                     */
-
-    int activated;  /**< 0 or 1. Activate LADSPA filters only once, even
-                     *   if the buffers get resized, to avoid a stuttering
-                     *   filter.
-                     */
-
-    char *file;
-    char *label;
-    char *controls;
-
-    char *myname;   /**< It's easy to have a concatenation of file and label */
-
-    void *libhandle;
-    const LADSPA_Descriptor *plugin_descriptor;
-
-    int nports;
-
-    int ninputs;
-    int *inputs;
-
-    int noutputs;
-    int *outputs;
-
-    int ninputcontrols;
-    int *inputcontrolsmap;  /**< Map input port number [0-] to actual port */
-    float *inputcontrols;
-
-    int noutputcontrols;
-    int *outputcontrolsmap;
-    float *outputcontrols;
-
-    int nch;                /**< number of channels */
-    int bufsize;
-    float **inbufs;
-    float **outbufs;
-    LADSPA_Handle *chhandles;
-
-} af_ladspa_t;
-
-/* ------------------------------------------------------------------------- */
-
-static int af_open(struct af_instance *af);
-static int af_ladspa_malloc_failed(char*);
-
-/* ------------------------------------------------------------------------- */
-
-#define OPT_BASE_STRUCT af_ladspa_t
-const struct af_info af_info_ladspa = {
-    .info = "LADSPA plugin loader",
-    .name = "ladspa",
-    .open = af_open,
-    .priv_size = sizeof(af_ladspa_t),
-    .options = (const struct m_option[]) {
-        OPT_STRING("file", file, 0),
-        OPT_STRING("label", label, 0),
-        OPT_STRING("controls", controls, 0),
-        {0}
-    },
-};
-
-/* ------------------------------------------------------------------------- */
-
-/* By lack of a better word (in my vocabulary) this is called 'parse'.
- * Feel free to suggest an alternative.
- */
-
-/** \brief Check for inputs, outputs and controls of a given filter.
- *
- * This function counts and checks all input, output and control ports
- * of the filter that was loaded. If it turns out to be a valid
- * filter for MPlayer use, it prints out a list of all controls and
- * the corresponding range of its value at message level  MSGL_V.
- *
- * \param setup     Current setup of the filter. Must have its
- *                  plugin_descriptor set!
- *
- * \return  Returns AF_OK if it has a valid input/output/controls
- *          configuration. Else, it returns AF_ERROR.
- */
-
-static int af_ladspa_parse_plugin(struct af_instance *af) {
-    af_ladspa_t *setup = af->priv;
-    int p, i;
-    const LADSPA_Descriptor *pdes = setup->plugin_descriptor;
-    LADSPA_PortDescriptor d;
-    LADSPA_PortRangeHint hint;
-
-    if (!setup->libhandle)
-        return AF_ERROR; /* only call parse after a succesful load */
-    if (!setup->plugin_descriptor)
-        return AF_ERROR; /* same as above */
-
-    /* let's do it */
-
-    setup->nports = pdes->PortCount;
-
-    /* allocate memory for all inputs/outputs/controls */
-
-    setup->inputs = calloc(setup->nports, sizeof(int));
-    if (!setup->inputs) return af_ladspa_malloc_failed(setup->myname);
-
-    setup->outputs = calloc(setup->nports, sizeof(int));
-    if (!setup->outputs) return af_ladspa_malloc_failed(setup->myname);
-
-    setup->inputcontrolsmap = calloc(setup->nports, sizeof(int));
-    if (!setup->inputcontrolsmap) return af_ladspa_malloc_failed(setup->myname);
-
-    setup->inputcontrols = calloc(setup->nports, sizeof(float));
-    if (!setup->inputcontrols) return af_ladspa_malloc_failed(setup->myname);
-
-    setup->outputcontrolsmap = calloc(setup->nports, sizeof(int));
-    if (!setup->outputcontrolsmap) return af_ladspa_malloc_failed(setup->myname);
-
-    setup->outputcontrols = calloc(setup->nports, sizeof(float));
-    if (!setup->outputcontrols) return af_ladspa_malloc_failed(setup->myname);
-
-    /* set counts to zero */
-
-    setup->ninputs = 0;
-    setup->noutputs = 0;
-    setup->ninputcontrols = 0;
-    setup->noutputcontrols = 0;
-
-    /* check all ports, see what type it is and set variables according to
-     * what we have found
-     */
-
-    for (p=0; p<setup->nports; p++) {
-        d = pdes->PortDescriptors[p];
-
-        if (LADSPA_IS_PORT_AUDIO(d)) {
-            if (LADSPA_IS_PORT_INPUT(d)) {
-                setup->inputs[setup->ninputs] = p;
-                setup->ninputs++;
-            } else if (LADSPA_IS_PORT_OUTPUT(d)) {
-                setup->outputs[setup->noutputs] = p;
-                setup->noutputs++;
-            }
-        }
-
-        if (LADSPA_IS_PORT_CONTROL(d)) {
-            if (LADSPA_IS_PORT_INPUT(d)) {
-                setup->inputcontrolsmap[setup->ninputcontrols] = p;
-                setup->ninputcontrols++;
-                /* set control to zero. set values after reading the rest
-                 * of the suboptions and check LADSPA_?_HINT's later.
-                 */
-                setup->inputcontrols[p] = 0.0f;
-            } else if (LADSPA_IS_PORT_OUTPUT(d)) {
-                /* read and handle these too, otherwise filters that have them
-                 * will sig11
-                 */
-                setup->outputcontrolsmap[setup->noutputcontrols]=p;
-                setup->noutputcontrols++;
-                setup->outputcontrols[p] = 0.0f;
-            }
-        }
-
-    }
-
-    if (setup->ninputs == 0) {
-        MP_WARN(af, "%s: %s\n", setup->myname,
-                                                _("WARNING! This LADSPA plugin has no audio inputs.\n  The incoming audio signal will be lost."));
-    } else if (setup->ninputs == 1) {
-        MP_VERBOSE(af, "%s: this is a mono effect\n", setup->myname);
-    } else if (setup->ninputs == 2) {
-        MP_VERBOSE(af, "%s: this is a stereo effect\n", setup->myname);
-    } else {
-        MP_VERBOSE(af, "%s: this is a %i-channel effect, "
-               "support is experimental\n", setup->myname, setup->ninputs);
-    }
-
-    if (setup->noutputs == 0) {
-        MP_ERR(af, "%s: %s\n", setup->myname,
-                                                _("This LADSPA plugin has no audio outputs."));
-        return AF_ERROR;
-    }
-
-    if (setup->noutputs != setup->ninputs ) {
-        MP_ERR(af, "%s: %s\n", setup->myname,
-                                                _("The number of audio inputs and audio outputs of the LADSPA plugin differ."));
-        return AF_ERROR;
-    }
-
-    MP_VERBOSE(af, "%s: this plugin has %d input control(s)\n",
-                                        setup->myname, setup->ninputcontrols);
-
-    /* Print list of controls and its range of values it accepts */
-
-    for (i=0; i<setup->ninputcontrols; i++) {
-        p = setup->inputcontrolsmap[i];
-        hint = pdes->PortRangeHints[p];
-        MP_VERBOSE(af, "  --- %d %s [", i, pdes->PortNames[p]);
-
-        if (LADSPA_IS_HINT_BOUNDED_BELOW(hint.HintDescriptor)) {
-            MP_VERBOSE(af, "%0.2f , ", hint.LowerBound);
-        } else {
-            MP_VERBOSE(af, "... , ");
-        }
-
-        if (LADSPA_IS_HINT_BOUNDED_ABOVE(hint.HintDescriptor)) {
-            MP_VERBOSE(af, "%0.2f]\n", hint.UpperBound);
-        } else {
-            MP_VERBOSE(af, "...]\n");
-        }
-
-    }
-
-    return AF_OK;
-}
-
-/* ------------------------------------------------------------------------- */
-
-/* This function might "slightly" look like dlopenLADSPA in the LADSPA SDK :-)
- * But, I changed a few things, because imho it was broken. It did not support
- * relative paths, only absolute paths that start with a /
- * I think ../../some/dir/foobar.so is just as valid. And if one wants to call
- * his library '...somename...so' he's crazy, but it should be allowed.
- * So, search the path first, try plain *filename later.
- * Also, try adding .so first! I like the recursion the SDK did, but it's
- * better the other way around. -af ladspa=cmt:amp_stereo:0.5 is easier to type
- * than -af ladspa=cmt.so:amp_stereo:0.5 :-))
- */
-
-/** \brief dlopen() wrapper
- *
- * This is a wrapper around dlopen(). It tries various variations of the
- * filename (with or without the addition of the .so extension) in various
- * directories specified by the LADSPA_PATH environment variable. If all fails
- * it tries the filename directly as an absolute path to the library.
- *
- * \param filename  filename of the library to load.
- * \param flag      see dlopen(3) for a description of the flags.
- *
- * \return          returns a pointer to the loaded library on success, or
- *                  NULL if it fails to load.
- */
-
-static void* mydlopen(const char *filename, int flag) {
-    char *buf;
-    const char *end, *start, *ladspapath;
-    int endsinso;
-    size_t filenamelen;
-    void *result = NULL;
-
-#if defined(__MINGW32__) || defined(__CYGWIN__)
-    /* For Windows there's only absolute path support.
-     * If you have a Windows machine, feel free to fix this.
-     * (path separator, shared objects extension, et cetera). */
-        MP_VERBOSE(af, "\ton windows, only absolute pathnames "
-                "are supported\n");
-        MP_VERBOSE(af, "\ttrying %s\n", filename);
-        return dlopen(filename, flag);
-#endif
-
-    filenamelen = strlen(filename);
-
-    endsinso = 0;
-    if (filenamelen > 3)
-        endsinso = (strcmp(filename+filenamelen-3, ".so") == 0);
-    if (!endsinso) {
-        buf=malloc(filenamelen+4);
-        strcpy(buf, filename);
-        strcat(buf, ".so");
-        result=mydlopen(buf, flag);
-        free(buf);
-    }
-
-    if (result)
-        return result;
-
-    ladspapath=getenv("LADSPA_PATH");
-
-    if (ladspapath) {
-        start=ladspapath;
-        while (*start != '\0') {
-            int needslash;
-            end=start;
-            while ( (*end != ':') && (*end != '\0') )
-                end++;
-
-            buf=malloc(filenamelen + 2 + (end-start) );
-            if (end > start)
-                strncpy(buf, start, end-start);
-            needslash=0;
-            if (end > start)
-                if (*(end-1) != '/') {
-                    needslash = 1;
-                    buf[end-start] = '/';
-                }
-            strcpy(buf+needslash+(end-start), filename);
-
-            result=dlopen(buf, flag);
-
-            free(buf);
-            if (result)
-                return result;
-
-            start = end;
-            if (*start == ':')
-                start++;
-        } /* end while there's still more in the path */
-    } /* end if there's a ladspapath */
-
-    /* last resort, just open it again, so the dlerror() message is correct */
-    return dlopen(filename,flag);
-}
-
-/* ------------------------------------------------------------------------- */
-
-/** \brief Load a LADSPA Plugin
- *
- * This function loads the LADSPA plugin specified by the file and label
- * that are present in the setup variable. First, it loads the library.
- * If it fails, it returns AF_ERROR. If not, it continues to look for the
- * specified label. If it finds it, it sets the plugin_descriptor inside
- * setup and returns AF_OK. If it doesn't, it returns AF_ERROR. Special case
- * is a label called 'help'. In that case, it prints a list of all available
- * labels (filters) in the library specified by file.
- *
- * \param setup     Current setup of the filter. Contains filename and label.
- *
- * \return  Either AF_ERROR or AF_OK, depending on the success of the operation.
- */
-
-static int af_ladspa_load_plugin(struct af_instance *af) {
-    af_ladspa_t *setup = af->priv;
-    const LADSPA_Descriptor *ladspa_descriptor;
-    LADSPA_Descriptor_Function descriptor_function;
-    int i;
-
-    /* load library */
-    MP_VERBOSE(af, "%s: loading ladspa plugin library %s\n",
-                                                setup->myname, setup->file);
-
-    setup->libhandle = mydlopen(setup->file, RTLD_NOW);
-
-    if (!setup->libhandle) {
-        MP_ERR(af, "%s: %s %s\n\t%s\n", setup->myname,
-                    _("failed to load"), setup->file, dlerror() );
-        return AF_ERROR;
-    }
-
-    MP_VERBOSE(af, "%s: library found.\n", setup->myname);
-
-    /* find descriptor function */
-    dlerror();
-    descriptor_function = (LADSPA_Descriptor_Function) dlsym (setup->libhandle,
-                                                        "ladspa_descriptor");
-
-    if (!descriptor_function) {
-        MP_ERR(af, "%s: %s\n\t%s\n", setup->myname,
-                                _("Couldn't find ladspa_descriptor() function in the specified library file."), dlerror());
-        return AF_ERROR;
-    }
-
-    /* if label == help, list all labels in library and exit */
-
-    if (strcmp(setup->label, "help") == 0) {
-        MP_INFO(af, "%s: %s %s:\n", setup->myname,
-                _("available labels in"), setup->file);
-        for (i=0; ; i++) {
-            ladspa_descriptor = descriptor_function(i);
-            if (ladspa_descriptor == NULL) {
-                return AF_ERROR;
-            }
-            MP_INFO(af, "  %-16s - %s (%lu)\n",
-                    ladspa_descriptor->Label,
-                    ladspa_descriptor->Name,
-                    ladspa_descriptor->UniqueID);
-        }
-    }
-
-    MP_VERBOSE(af, "%s: looking for label\n", setup->myname);
-
-    /* find label in library */
-    for (i=0; ; i++) {
-        ladspa_descriptor = descriptor_function(i);
-        if (ladspa_descriptor == NULL) {
-            MP_ERR(af, "%s: %s\n", setup->myname,
-                                            _("Couldn't find label in plugin library."));
-            return AF_ERROR;
-        }
-        if (strcmp(ladspa_descriptor->Label, setup->label) == 0) {
-            setup->plugin_descriptor = ladspa_descriptor;
-            MP_VERBOSE(af, "%s: %s found\n", setup->myname,
-                                                                setup->label);
-            return AF_OK;
-        }
-    }
-
-    return AF_OK;
-}
-
-/* ------------------------------------------------------------------------- */
-
-/** \brief Print a malloc() failed error message.
- *
- * Generic function which can be called if a call to malloc(), calloc(),
- * strdup(), et cetera, failed. It prints a message to the console and
- * returns AF_ERROR.
- *
- * \return  AF_ERROR
- */
-
-static int af_ladspa_malloc_failed(char *myname) {
-    return AF_ERROR;
-}
-
-/* ------------------------------------------------------------------------- */
-
-/** \brief Controls the filter.
- *
- * Control the behaviour of the filter.
- *
- * Commands:
- * CONTROL_REINIT   Sets the af structure with proper values for number
- *                  of channels, rate, format, et cetera.
- * \param af    Audio filter instance
- * \param cmd   The command to execute
- * \param arg   Arguments to the command
- *
- * \return      Either AF_ERROR or AF_OK, depending on the succes of the
- *              operation.
- */
-
-static int control(struct af_instance *af, int cmd, void *arg) {
-    af_ladspa_t *setup = (af_ladspa_t*) af->priv;
-
-    switch(cmd) {
-    case AF_CONTROL_REINIT:
-        MP_VERBOSE(af, "%s: (re)init\n", setup->myname);
-
-        if (!arg) return AF_ERROR;
-
-        /* accept FLOAT, let af_format do conversion */
-
-        mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-        mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-
-        return af_test_output(af, (struct mp_audio*)arg);
-    }
-
-    return AF_UNKNOWN;
-}
-
-/* ------------------------------------------------------------------------- */
-
-/** \brief Uninitialise the LADSPA Plugin Loader filter.
- *
- * This function deactivates the plugin(s), cleans up, frees all allocated
- * memory and exits.
- *
- * \return  No return value.
- */
-
-static void uninit(struct af_instance *af) {
-    if (af->priv) {
-        af_ladspa_t *setup = (af_ladspa_t*) af->priv;
-        const LADSPA_Descriptor *pdes = setup->plugin_descriptor;
-
-        if (setup->myname) {
-            MP_VERBOSE(af, "%s: cleaning up\n", setup->myname);
-            free(setup->myname);
-        }
-
-        if (setup->chhandles) {
-            for (int i = 0; i < setup->nch; i+=setup->ninputs) {
-                if (pdes->deactivate) pdes->deactivate(setup->chhandles[i]);
-                if (pdes->cleanup) pdes->cleanup(setup->chhandles[i]);
-            }
-            free(setup->chhandles);
-        }
-
-        free(setup->inputcontrolsmap);
-        free(setup->inputcontrols);
-        free(setup->outputcontrolsmap);
-        free(setup->outputcontrols);
-        free(setup->inputs);
-        free(setup->outputs);
-
-        if (setup->inbufs) {
-            for(int i = 0; i < setup->nch; i++)
-                free(setup->inbufs[i]);
-            free(setup->inbufs);
-        }
-
-        if (setup->outbufs) {
-            for (int i = 0; i < setup->nch; i++)
-                free(setup->outbufs[i]);
-            free(setup->outbufs);
-        }
-
-        if (setup->libhandle)
-            dlclose(setup->libhandle);
-    }
-}
-
-/* ------------------------------------------------------------------------- */
-
-/** \brief Process chunk of audio data through the selected LADSPA Plugin.
- *
- * \param af    Pointer to audio filter instance
- * \param data  Pointer to chunk of audio data
- *
- * \return      Either AF_ERROR or AF_OK
- */
-
-static int filter_frame(struct af_instance *af, struct mp_audio *data)
-{
-    if (!data)
-        return 0;
-    af_ladspa_t *setup = af->priv;
-    const LADSPA_Descriptor *pdes = setup->plugin_descriptor;
-    float *audio = (float*)data->planes[0];
-    int nsamples = data->samples*data->nch;
-    int nch = data->nch;
-    int rate = data->rate;
-    int i, p;
-
-    if (setup->status !=AF_OK) {
-        talloc_free(data);
-        return -1;
-    }
-    if (af_make_writeable(af, data) < 0) {
-        talloc_free(data);
-        return -1;
-    }
-
-    /* See if it's the first call. If so, setup inbufs/outbufs, instantiate
-     * plugin, connect ports and activate plugin
-     */
-
-    if ( (setup->bufsize != nsamples/nch) || (setup->nch != nch) ) {
-
-        /* if setup->nch==0, it's the first call, if not, something has
-         * changed and all previous mallocs have to be freed
-         */
-
-        if (setup->nch != 0) {
-            MP_TRACE(af, "%s: bufsize change; free old buffer\n",
-                                                                setup->myname);
-
-            if(setup->inbufs) {
-                for(i=0; i<setup->nch; i++)
-                    free(setup->inbufs[i]);
-                free(setup->inbufs);
-            }
-            if(setup->outbufs) {
-                for(i=0; i<setup->nch; i++)
-                    free(setup->outbufs[i]);
-                free(setup->outbufs);
-            }
-        } /* everything is freed */
-
-        setup->bufsize = nsamples/nch;
-        setup->nch = nch;
-
-        setup->inbufs = calloc(nch, sizeof(float*));
-        setup->outbufs = calloc(nch, sizeof(float*));
-
-        MP_TRACE(af, "%s: bufsize = %d\n",
-                                        setup->myname, setup->bufsize);
-
-        for(i=0; i<nch; i++) {
-            setup->inbufs[i] = calloc(setup->bufsize, sizeof(float));
-            setup->outbufs[i] = calloc(setup->bufsize, sizeof(float));
-        }
-
-        /* only on the first call, there are no handles. */
-
-        if (!setup->chhandles) {
-            setup->chhandles = calloc(nch, sizeof(LADSPA_Handle));
-
-            /* create handles
-             * for stereo effects, create one handle for two channels
-             */
-
-            for(i=0; i<nch; i++) {
-
-                if (i % setup->ninputs) { /* stereo effect */
-                    /* copy the handle from previous channel */
-                    setup->chhandles[i] = setup->chhandles[i-1];
-                    continue;
-                }
-
-                setup->chhandles[i] = pdes->instantiate(pdes, rate);
-            }
-        }
-
-        /* connect input/output ports for each channel/filter instance
-         *
-         * always (re)connect ports
-         */
-
-        for(i=0; i<nch; i++) {
-            pdes->connect_port(setup->chhandles[i],
-                               setup->inputs[i % setup->ninputs],
-                               setup->inbufs[i]);
-            pdes->connect_port(setup->chhandles[i],
-                               setup->outputs[i % setup->ninputs],
-                               setup->outbufs[i]);
-
-            /* connect (input) controls */
-
-            for (p=0; p<setup->nports; p++) {
-                LADSPA_PortDescriptor d = pdes->PortDescriptors[p];
-                if (LADSPA_IS_PORT_CONTROL(d)) {
-                    if (LADSPA_IS_PORT_INPUT(d)) {
-                        pdes->connect_port(setup->chhandles[i], p,
-                                                &(setup->inputcontrols[p]) );
-                    } else {
-                        pdes->connect_port(setup->chhandles[i], p,
-                                                &(setup->outputcontrols[p]) );
-                    }
-                }
-            }
-
-            /* Activate filter (if it isn't already :) ) */
-
-            if (pdes->activate && !setup->activated && i % setup->ninputs == 0)
-                pdes->activate(setup->chhandles[i]);
-
-        } /* All channels/filters done! except for... */
-        setup->activated = 1;
-
-        /* Stereo effect with one channel left. Use same buffer for left
-         * and right. connect it to the second port.
-         */
-
-        for (p = i; p % setup->ninputs; p++) {
-            pdes->connect_port(setup->chhandles[i-1],
-                               setup->inputs[p % setup->ninputs],
-                               setup->inbufs[i-1]);
-            pdes->connect_port(setup->chhandles[i-1],
-                               setup->outputs[p % setup->ninputs],
-                               setup->outbufs[i-1]);
-        } /* done! */
-
-    } /* setup for first call/change of bufsize is done.
-       * normal playing routine follows...
-       */
-
-    /* Right now, I use a separate input and output buffer.
-     * I could change this to in-place processing (inbuf==outbuf), but some
-     * ladspa filters are broken and are not able to handle that. This seems
-     * fast enough, so unless somebody complains, it stays this way :)
-     */
-
-    /* Fill inbufs */
-
-    for (p=0; p<setup->bufsize; p++) {
-        for (i=0; i<nch; i++) {
-            setup->inbufs[i][p] = audio[p*nch + i];
-        }
-    }
-
-    /* Run filter(s) */
-
-    for (i=0; i<nch; i+=setup->ninputs) {
-        pdes->run(setup->chhandles[i], setup->bufsize);
-    }
-
-    /* Extract outbufs */
-
-    for (p=0; p<setup->bufsize; p++) {
-        for (i=0; i<nch; i++) {
-            audio[p*nch + i] = setup->outbufs[i][p];
-        }
-    }
-
-    /* done */
-
-    af_add_output_frame(af, data);
-    return 0;
-}
-
-/* ------------------------------------------------------------------------- */
-
-/** \brief Open LADSPA Plugin Loader Filter
- *
- * \param af    Audio Filter instance
- *
- * \return      Either AF_ERROR or AF_OK
- */
-
-static int af_open(struct af_instance *af) {
-
-    af->control=control;
-    af->uninit=uninit;
-    af->filter_frame = filter_frame;
-
-    af_ladspa_t *setup = af->priv;
-
-    setup->status = AF_ERROR; /* will be set to AF_OK if
-                                                   * all went OK and filter()
-                                                   * should proceed.
-                                                   */
-
-    setup->myname = strdup(af_info_ladspa.name);
-    if (!setup->myname)
-        return af_ladspa_malloc_failed((char*)af_info_ladspa.name);
-
-    if (!setup->file || !setup->file[0]) {
-        MP_ERR(af, "%s: %s\n", setup->myname,
-                                            _("No library file specified."));
-        uninit(af);
-        return AF_ERROR;
-    }
-    if (!setup->label || !setup->label[0]) {
-        MP_ERR(af, "%s: %s\n", setup->myname,
-                                            _("No filter label specified."));
-        uninit(af);
-        return AF_ERROR;
-    }
-
-    free(setup->myname);
-    setup->myname = calloc(strlen(af_info_ladspa.name)+strlen(setup->file)+
-                                                strlen(setup->label)+6, 1);
-    snprintf(setup->myname, strlen(af_info_ladspa.name)+
-            strlen(setup->file)+strlen(setup->label)+6, "%s: (%s:%s)",
-                        af_info_ladspa.name, setup->file, setup->label);
-
-    /* load plugin :) */
-
-    if ( af_ladspa_load_plugin(af) != AF_OK )
-        return AF_ERROR;
-
-    /* see what inputs, outputs and controls this plugin has */
-    if ( af_ladspa_parse_plugin(af) != AF_OK )
-        return AF_ERROR;
-
-    /* ninputcontrols is set by now, read control values from arg */
-
-    float val;
-    char *line = setup->controls;
-    for (int i = 0; i < setup->ninputcontrols; i++) {
-        if (!line || (i != 0 && *line != ',')) {
-            MP_ERR(af, "%s: %s\n", setup->myname,
-                                    _("Not enough controls specified on the command line."));
-            return AF_ERROR;
-        }
-        if (i != 0)
-            line++;
-        if (sscanf(line, "%f", &val) != 1) {
-            MP_ERR(af, "%s: %s\n", setup->myname,
-                                    _("Not enough controls specified on the command line."));
-            return AF_ERROR;
-        }
-        setup->inputcontrols[setup->inputcontrolsmap[i]] = val;
-        line = strchr(line, ',');
-    }
-
-    MP_VERBOSE(af, "%s: input controls: ", setup->myname);
-    for (int i = 0; i < setup->ninputcontrols; i++) {
-        MP_VERBOSE(af, "%0.4f ",
-                        setup->inputcontrols[setup->inputcontrolsmap[i]]);
-    }
-    MP_VERBOSE(af, "\n");
-
-    /* check boundaries of inputcontrols */
-
-    MP_VERBOSE(af, "%s: checking boundaries of input controls\n",
-                                                            setup->myname);
-    for (int i = 0; i < setup->ninputcontrols; i++) {
-        int p = setup->inputcontrolsmap[i];
-        LADSPA_PortRangeHint hint =
-                            setup->plugin_descriptor->PortRangeHints[p];
-        val = setup->inputcontrols[p];
-
-        if (LADSPA_IS_HINT_BOUNDED_BELOW(hint.HintDescriptor) &&
-                val < hint.LowerBound) {
-            MP_ERR(af, "%s: Input control #%d is below lower boundary of %0.4f.\n",
-                                        setup->myname, i, hint.LowerBound);
-            return AF_ERROR;
-        }
-        if (LADSPA_IS_HINT_BOUNDED_ABOVE(hint.HintDescriptor) &&
-                val > hint.UpperBound) {
-            MP_ERR(af, "%s: Input control #%d is above upper boundary of %0.4f.\n",
-                                        setup->myname, i, hint.UpperBound);
-            return AF_ERROR;
-        }
-    }
-    MP_VERBOSE(af, "%s: all controls have sane values\n",
-                                                            setup->myname);
-
-    /* All is well! */
-    setup->status = AF_OK;
-
-    return AF_OK;
-}
-
-/* ------------------------------------------------------------------------- */
@@ -26,7 +26,6 @@
 #include <assert.h>

 #include <libavcodec/avcodec.h>
-#include <libavutil/audioconvert.h>
 #include <libavutil/intreadwrite.h>
 #include <libavutil/common.h>
 #include <libavutil/bswap.h>
@@ -75,7 +74,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
        struct mp_audio *in = arg;
        struct mp_audio orig_in = *in;

-        if (AF_FORMAT_IS_SPECIAL(in->format) || in->nch < s->cfg_min_channel_num)
+        if (!af_fmt_is_pcm(in->format) || in->nch < s->cfg_min_channel_num)
            return AF_DETACH;

        mp_audio_set_format(in, s->in_sampleformat);
@@ -156,6 +155,13 @@ static void uninit(struct af_instance* af)
    }
 }

+static void update_delay(struct af_instance *af)
+{
+    af_ac3enc_t *s = af->priv;
+    af->delay = ((s->pending ? s->pending->samples : 0) + s->input->samples) /
+                (double)s->input->rate;
+}
+
 static int filter_frame(struct af_instance *af, struct mp_audio *audio)
 {
    af_ac3enc_t *s = af->priv;
@@ -166,6 +172,7 @@ static int filter_frame(struct af_instance *af, struct mp_audio *audio)

    talloc_free(s->pending);
    s->pending = audio;
+    update_delay(af);
    return 0;
 }

@@ -177,22 +184,27 @@ static void swap_16(uint16_t *ptr, size_t size)

 // Copy data from input frame to encode frame (because libavcodec wants a full
 // AC3 frame for encoding, while filter input frames can be smaller or larger).
-// Return true if the
-static bool fill_buffer(af_ac3enc_t *s)
+// Return true if the frame is complete.
+static bool fill_buffer(struct af_instance *af)
 {
+    af_ac3enc_t *s = af->priv;
+
+    af->delay = 0;
+
    if (s->pending) {
        int copy = MPMIN(s->in_samples - s->input->samples, s->pending->samples);
        s->input->samples += copy;
        mp_audio_copy(s->input, s->input->samples - copy, s->pending, 0, copy);
        mp_audio_skip_samples(s->pending, copy);
    }
+    update_delay(af);
    return s->input->samples >= s->in_samples;
 }

 static int filter_out(struct af_instance *af)
 {
    af_ac3enc_t *s = af->priv;
-    if (!fill_buffer(s))
+    if (!fill_buffer(af))
        return 0; // need more input

    AVFrame *frame = av_frame_alloc();
@@ -256,6 +268,7 @@ static int filter_out(struct af_instance *af)
    swap_16((uint16_t *)(buf + header_len), s->pkt.size / 2);
    out->samples = frame_size / out->sstride;
    af_add_output_frame(af, out);
+    update_delay(af);
    return 0;
 }

@@ -34,11 +34,14 @@
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>

+#include "config.h"
+
 #include "audio/format.h"
 #include "audio/fmt-conversion.h"
 #include "af.h"

 #include "common/av_common.h"
+#include "common/tags.h"

 #include "options/m_option.h"

@@ -63,6 +66,8 @@ struct priv {

    bool eof;

+    struct mp_tags *metadata;
+
    // options
    char *cfg_graph;
    char **cfg_avopts;
@@ -213,6 +218,12 @@ static int control(struct af_instance *af, int cmd, void *arg)

        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
    }
+    case AF_CONTROL_GET_METADATA:
+        if (p->metadata) {
+            *(struct mp_tags *)arg = *p->metadata;
+            return CONTROL_OK;
+        }
+        return CONTROL_NA;
    case AF_CONTROL_RESET:
        reset(af);
        return AF_OK;
@@ -220,6 +231,17 @@ static int control(struct af_instance *af, int cmd, void *arg)
    return AF_UNKNOWN;
 }

+static void get_metadata_from_av_frame(struct af_instance *af, AVFrame *frame)
+{
+#if HAVE_AVFRAME_METADATA
+    struct priv *p = af->priv;
+    if (!p->metadata)
+        p->metadata = talloc_zero(p, struct mp_tags);
+
+    mp_tags_copy_from_av_dictionary(p->metadata, av_frame_get_metadata(frame));
+#endif
+}
+
 static int filter_frame(struct af_instance *af, struct mp_audio *data)
 {
    struct priv *p = af->priv;
@@ -307,6 +329,7 @@ static int filter_out(struct af_instance *af)
        af->delay = in_time - out_time;
    }

+    get_metadata_from_av_frame(af, frame);
    af_add_output_frame(af, out);
    av_frame_free(&frame);
    return 0;
@@ -28,9 +28,9 @@
 #include <assert.h>

 #include <libavutil/opt.h>
-#include <libavutil/audioconvert.h>
 #include <libavutil/common.h>
 #include <libavutil/samplefmt.h>
+#include <libavutil/channel_layout.h>
 #include <libavutil/mathematics.h>

 #include "common/common.h"
@@ -58,12 +58,30 @@
 #include "options/m_option.h"
 #include "audio/filter/af.h"
 #include "audio/fmt-conversion.h"
+#include "osdep/endian.h"

 struct af_resample_opts {
    int filter_size;
    int phase_shift;
    int linear;
    double cutoff;
+    int normalize;
+};
+
+struct af_resample {
+    int allow_detach;
+    char **avopts;
+    double playback_speed;
+    struct AVAudioResampleContext *avrctx;
+    struct mp_audio avrctx_fmt; // output format of avrctx
+    struct mp_audio pool_fmt; // format used to allocate frames for avrctx output
+    struct mp_audio pre_out_fmt; // format before final conversion (S24)
+    struct AVAudioResampleContext *avrctx_out; // for output channel reordering
+    struct af_resample_opts opts;  // opts requested by the user
+    // At least libswresample keeps a pointer around for this:
+    int reorder_in[MP_NUM_CHANNELS];
+    int reorder_out[MP_NUM_CHANNELS];
+    struct mp_audio_pool *reorder_buffer;

    int in_rate_af; // filter input sample rate
    int in_rate;    // actual rate (used by lavr), adjusted for playback speed
@@ -74,50 +92,53 @@ struct af_resample_opts {
    struct mp_chmap out_channels;
 };

-struct af_resample {
-    int allow_detach;
-    char **avopts;
-    double playback_speed;
-    struct mp_audio *pending;
-    bool avrctx_ok;
-    struct AVAudioResampleContext *avrctx;
-    struct AVAudioResampleContext *avrctx_out; // for output channel reordering
-    struct af_resample_opts ctx;   // opts in the context
-    struct af_resample_opts opts;  // opts requested by the user
-    // At least libswresample keeps a pointer around for this:
-    int reorder_in[MP_NUM_CHANNELS];
-    int reorder_out[MP_NUM_CHANNELS];
-    struct mp_audio_pool *reorder_buffer;
-};
-
 #if HAVE_LIBAVRESAMPLE
-static int get_delay(struct af_resample *s)
+static double get_delay(struct af_resample *s)
 {
-    return avresample_get_delay(s->avrctx);
+    return avresample_get_delay(s->avrctx) / (double)s->in_rate +
+           avresample_available(s->avrctx) / (double)s->out_rate;
 }
 static void drop_all_output(struct af_resample *s)
 {
    while (avresample_read(s->avrctx, NULL, 1000) > 0) {}
 }
-static int get_drain_samples(struct af_resample *s)
+static int get_out_samples(struct af_resample *s, int in_samples)
 {
-    return avresample_get_out_samples(s->avrctx, 0);
+    return avresample_get_out_samples(s->avrctx, in_samples);
 }
 #else
-static int get_delay(struct af_resample *s)
+static double get_delay(struct af_resample *s)
 {
-    return swr_get_delay(s->avrctx, s->ctx.in_rate);
+    int64_t base = s->in_rate * (int64_t)s->out_rate;
+    return swr_get_delay(s->avrctx, base) / (double)base;
 }
 static void drop_all_output(struct af_resample *s)
 {
    while (swr_drop_output(s->avrctx, 1000) > 0) {}
 }
-static int get_drain_samples(struct af_resample *s)
+static int get_out_samples(struct af_resample *s, int in_samples)
 {
-    return 4096; // libswscale does not have this
+#if LIBSWRESAMPLE_VERSION_MAJOR > 1 || LIBSWRESAMPLE_VERSION_MINOR >= 2
+    return swr_get_out_samples(s->avrctx, in_samples);
+#else
+    return av_rescale_rnd(in_samples, s->out_rate, s->in_rate, AV_ROUND_UP)
+           + swr_get_delay(s->avrctx, s->out_rate);
+#endif
 }
 #endif

+static void close_lavrr(struct af_instance *af)
+{
+    struct af_resample *s = af->priv;
+
+    if (s->avrctx)
+        avresample_close(s->avrctx);
+    avresample_free(&s->avrctx);
+    if (s->avrctx_out)
+        avresample_close(s->avrctx_out);
+    avresample_free(&s->avrctx_out);
+}
+
 static int resample_frame(struct AVAudioResampleContext *r,
                          struct mp_audio *out, struct mp_audio *in)
 {
@@ -140,72 +161,82 @@ static int rate_from_speed(int rate, double speed)
    return lrint(rate * speed);
 }

-static bool needs_lavrctx_reconfigure(struct af_resample *s,
-                                      struct mp_audio *in,
-                                      struct mp_audio *out)
+// Return the format libavresample should convert to, given the final output
+// format mp_format. In some cases (S24) we perform an extra conversion step,
+// and signal here what exactly libavresample should output. It will be the
+// input to the final conversion to mp_format.
+static int check_output_conversion(int mp_format)
 {
-    return s->ctx.in_rate_af  != in->rate ||
-           s->ctx.in_format   != in->format ||
-           !mp_chmap_equals(&s->ctx.in_channels, &in->channels) ||
-           s->ctx.out_rate    != out->rate ||
-           s->ctx.out_format  != out->format ||
-           !mp_chmap_equals(&s->ctx.out_channels, &out->channels) ||
-           s->ctx.filter_size != s->opts.filter_size ||
-           s->ctx.phase_shift != s->opts.phase_shift ||
-           s->ctx.linear      != s->opts.linear ||
-           s->ctx.cutoff      != s->opts.cutoff;
-
+    if (mp_format == AF_FORMAT_S24)
+        return AV_SAMPLE_FMT_S32;
+    return af_to_avformat(mp_format);
 }

-static bool test_conversion(int src_format, int dst_format)
+bool af_lavrresample_test_conversion(int src_format, int dst_format)
 {
    return af_to_avformat(src_format) != AV_SAMPLE_FMT_NONE &&
-           af_to_avformat(dst_format) != AV_SAMPLE_FMT_NONE;
+           check_output_conversion(dst_format) != AV_SAMPLE_FMT_NONE;
+}
+
+// mp_chmap_get_reorder() performs:
+//  to->speaker[n] = from->speaker[src[n]]
+// but libavresample does:
+//  to->speaker[dst[n]] = from->speaker[n]
+static void transpose_order(int *map, int num)
+{
+    int nmap[MP_NUM_CHANNELS] = {0};
+    for (int n = 0; n < num; n++) {
+        for (int i = 0; i < num; i++) {
+            if (map[n] == i)
+                nmap[i] = n;
+        }
+    }
+    memcpy(map, nmap, sizeof(nmap));
 }

 static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
-                           struct mp_audio *out)
+                           struct mp_audio *out, bool verbose)
 {
    struct af_resample *s = af->priv;

-    s->avrctx_ok = false;
+    close_lavrr(af);
+
+    s->avrctx = avresample_alloc_context();
+    s->avrctx_out = avresample_alloc_context();
+    if (!s->avrctx || !s->avrctx_out)
+        goto error;

    enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
-    enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
+    enum AVSampleFormat out_samplefmt = check_output_conversion(out->format);
+    enum AVSampleFormat out_samplefmtp = av_get_planar_sample_fmt(out_samplefmt);

-    if (in_samplefmt == AV_SAMPLE_FMT_NONE || out_samplefmt == AV_SAMPLE_FMT_NONE)
-        return AF_ERROR;
+    if (in_samplefmt == AV_SAMPLE_FMT_NONE ||
+        out_samplefmt == AV_SAMPLE_FMT_NONE ||
+        out_samplefmtp == AV_SAMPLE_FMT_NONE)
+        goto error;

-    avresample_close(s->avrctx);
-    avresample_close(s->avrctx_out);
+    s->out_rate    = out->rate;
+    s->in_rate_af  = in->rate;
+    s->in_rate     = rate_from_speed(in->rate, s->playback_speed);
+    s->out_format  = out->format;
+    s->in_format   = in->format;
+    s->out_channels= out->channels;
+    s->in_channels = in->channels;

-    talloc_free(s->pending);
-    s->pending = NULL;
+    av_opt_set_int(s->avrctx, "filter_size",        s->opts.filter_size, 0);
+    av_opt_set_int(s->avrctx, "phase_shift",        s->opts.phase_shift, 0);
+    av_opt_set_int(s->avrctx, "linear_interp",      s->opts.linear, 0);

-    s->ctx.out_rate    = out->rate;
-    s->ctx.in_rate_af  = in->rate;
-    s->ctx.in_rate     = rate_from_speed(in->rate, s->playback_speed);
-    s->ctx.out_format  = out->format;
-    s->ctx.in_format   = in->format;
-    s->ctx.out_channels= out->channels;
-    s->ctx.in_channels = in->channels;
-    s->ctx.filter_size = s->opts.filter_size;
-    s->ctx.phase_shift = s->opts.phase_shift;
-    s->ctx.linear      = s->opts.linear;
-    s->ctx.cutoff      = s->opts.cutoff;
-
-    av_opt_set_int(s->avrctx, "filter_size",        s->ctx.filter_size, 0);
-    av_opt_set_int(s->avrctx, "phase_shift",        s->ctx.phase_shift, 0);
-    av_opt_set_int(s->avrctx, "linear_interp",      s->ctx.linear, 0);
-
-    av_opt_set_double(s->avrctx, "cutoff",          s->ctx.cutoff, 0);
+    av_opt_set_double(s->avrctx, "cutoff",          s->opts.cutoff, 0);

 #if HAVE_LIBSWRESAMPLE
-    av_opt_set_double(s->avrctx, "rematrix_maxval", 1.0, 0);
+    av_opt_set_double(s->avrctx, "rematrix_maxval", s->opts.normalize ? 1 : 1000, 0);
+#else
+    av_opt_set_int(s->avrctx, "normalize_mix_level", s->opts.normalize, 0);
 #endif

    if (mp_set_avopts(af->log, s->avrctx, s->avopts) < 0)
-        return AF_ERROR;
+        goto error;

    struct mp_chmap map_in = in->channels;
    struct mp_chmap map_out = out->channels;
@@ -220,45 +251,86 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
    uint64_t in_ch_layout = mp_chmap_to_lavc_unchecked(&map_in);
    uint64_t out_ch_layout = mp_chmap_to_lavc_unchecked(&map_out);

-    av_opt_set_int(s->avrctx, "in_channel_layout",  in_ch_layout, 0);
-    av_opt_set_int(s->avrctx, "out_channel_layout", out_ch_layout, 0);
-
-    av_opt_set_int(s->avrctx, "in_sample_rate",     s->ctx.in_rate, 0);
-    av_opt_set_int(s->avrctx, "out_sample_rate",    s->ctx.out_rate, 0);
-
-    av_opt_set_int(s->avrctx, "in_sample_fmt",      in_samplefmt, 0);
-    av_opt_set_int(s->avrctx, "out_sample_fmt",     out_samplefmt, 0);
-
-    struct mp_chmap in_lavc;
+    struct mp_chmap in_lavc, out_lavc;
    mp_chmap_from_lavc(&in_lavc, in_ch_layout);
-    mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
-
-    struct mp_chmap out_lavc;
    mp_chmap_from_lavc(&out_lavc, out_ch_layout);
+
+    if (verbose && !mp_chmap_equals(&in_lavc, &out_lavc)) {
+        MP_VERBOSE(af, "Remix: %s -> %s\n", mp_chmap_to_str(&in_lavc),
+                                            mp_chmap_to_str(&out_lavc));
+    }
+
+    if (in_lavc.num != map_in.num) {
+        // For handling NA channels, we would have to add a planarization step.
+        MP_FATAL(af, "Unsupported channel remapping.\n");
+        goto error;
+    }
+
+    mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
+    transpose_order(s->reorder_in, map_in.num);
+
+    if (mp_chmap_equals(&out_lavc, &map_out)) {
+        // No intermediate step required - output new format directly.
+        out_samplefmtp = out_samplefmt;
+    } else {
+        // Verify that we really just reorder and/or insert NA channels.
+        struct mp_chmap withna = out_lavc;
+        mp_chmap_fill_na(&withna, map_out.num);
+        if (withna.num != map_out.num)
+            goto error;
+    }
    mp_chmap_get_reorder(s->reorder_out, &out_lavc, &map_out);

-    // Same configuration; we just reorder.
-    av_opt_set_int(s->avrctx_out, "in_channel_layout",  out_ch_layout, 0);
-    av_opt_set_int(s->avrctx_out, "out_channel_layout", out_ch_layout, 0);
-    av_opt_set_int(s->avrctx_out, "in_sample_fmt",      out_samplefmt, 0);
+    s->avrctx_fmt = *out;
+    mp_audio_set_channels(&s->avrctx_fmt, &out_lavc);
+    mp_audio_set_format(&s->avrctx_fmt, af_from_avformat(out_samplefmtp));
+
+    s->pre_out_fmt = *out;
+    mp_audio_set_format(&s->pre_out_fmt, af_from_avformat(out_samplefmt));
+
+    // If there are NA channels, the final output will have more channels than
+    // the avrctx output. Also, avrctx will output planar (out_samplefmtp was
+    // not overwritten). Allocate the output frame with more channels, so the
+    // NA channels can be trivially added.
+    s->pool_fmt = s->avrctx_fmt;
+    if (map_out.num > out_lavc.num)
+        mp_audio_set_channels(&s->pool_fmt, &map_out);
+
+    // Real conversion; output is input to avrctx_out.
+    av_opt_set_int(s->avrctx, "in_channel_layout",  in_ch_layout, 0);
+    av_opt_set_int(s->avrctx, "out_channel_layout", out_ch_layout, 0);
+    av_opt_set_int(s->avrctx, "in_sample_rate",     s->in_rate, 0);
+    av_opt_set_int(s->avrctx, "out_sample_rate",    s->out_rate, 0);
+    av_opt_set_int(s->avrctx, "in_sample_fmt",      in_samplefmt, 0);
+    av_opt_set_int(s->avrctx, "out_sample_fmt",     out_samplefmtp, 0);
+
+    // Just needs the correct number of channels.
+    int fake_out_ch_layout = av_get_default_channel_layout(map_out.num);
+    if (!fake_out_ch_layout)
+        goto error;
+
+    // Deplanarize if needed.
+    av_opt_set_int(s->avrctx_out, "in_channel_layout",  fake_out_ch_layout, 0);
+    av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
+    av_opt_set_int(s->avrctx_out, "in_sample_fmt",      out_samplefmtp, 0);
    av_opt_set_int(s->avrctx_out, "out_sample_fmt",     out_samplefmt, 0);
-    av_opt_set_int(s->avrctx_out, "in_sample_rate",     s->ctx.out_rate, 0);
-    av_opt_set_int(s->avrctx_out, "out_sample_rate",    s->ctx.out_rate, 0);
+    av_opt_set_int(s->avrctx_out, "in_sample_rate",     s->out_rate, 0);
+    av_opt_set_int(s->avrctx_out, "out_sample_rate",    s->out_rate, 0);

    // API has weird requirements, quoting avresample.h:
    //  * This function can only be called when the allocated context is not open.
    //  * Also, the input channel layout must have already been set.
    avresample_set_channel_mapping(s->avrctx, s->reorder_in);
-    avresample_set_channel_mapping(s->avrctx_out, s->reorder_out);

-    if (avresample_open(s->avrctx) < 0 ||
-        avresample_open(s->avrctx_out) < 0)
-    {
+    if (avresample_open(s->avrctx) < 0 || avresample_open(s->avrctx_out) < 0) {
        MP_ERR(af, "Cannot open Libavresample Context. \n");
-        return AF_ERROR;
+        goto error;
    }
-    s->avrctx_ok = true;
    return AF_OK;
+
+error:
+    close_lavrr(af);
+    return AF_ERROR;
 }


@@ -286,20 +358,20 @@ static int control(struct af_instance *af, int cmd, void *arg)

        if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
            mp_audio_set_format(in, AF_FORMAT_FLOAT);
-        if (af_to_avformat(out->format) == AV_SAMPLE_FMT_NONE)
+        if (check_output_conversion(out->format) == AV_SAMPLE_FMT_NONE)
            mp_audio_set_format(out, in->format);

        int r = ((in->format == orig_in.format) &&
                mp_chmap_equals(&in->channels, &orig_in.channels))
                ? AF_OK : AF_FALSE;

-        if (r == AF_OK && needs_lavrctx_reconfigure(s, in, out))
-            r = configure_lavrr(af, in, out);
+        if (r == AF_OK)
+            r = configure_lavrr(af, in, out, true);
        return r;
    }
    case AF_CONTROL_SET_FORMAT: {
        int format = *(int *)arg;
-        if (format && af_to_avformat(format) == AV_SAMPLE_FMT_NONE)
+        if (format && check_output_conversion(format) == AV_SAMPLE_FMT_NONE)
            return AF_FALSE;

        mp_audio_set_format(af->data, format);
@@ -314,61 +386,72 @@ static int control(struct af_instance *af, int cmd, void *arg)
        return AF_OK;
    case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
        s->playback_speed = *(double *)arg;
-        int new_rate = rate_from_speed(s->ctx.in_rate_af, s->playback_speed);
-        if (new_rate != s->ctx.in_rate && s->avrctx_ok && af->fmt_out.format) {
+        int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
+        if (new_rate != s->in_rate && s->avrctx && af->fmt_out.format) {
            // Before reconfiguring, drain the audio that is still buffered
            // in the resampler.
-            struct mp_audio *pending = talloc_zero(NULL, struct mp_audio);
-            mp_audio_copy_config(pending, &af->fmt_out);
-            pending->samples = get_drain_samples(s);
-            if (pending->samples > 0) {
-                mp_audio_realloc_min(pending, pending->samples);
-                int r = resample_frame(s->avrctx, pending, NULL);
-                pending->samples = MPMAX(r, 0);
-            }
+            af->filter_frame(af, NULL);
            // Reinitialize resampler.
-            configure_lavrr(af, &af->fmt_in, &af->fmt_out);
-            s->pending = pending;
+            configure_lavrr(af, &af->fmt_in, &af->fmt_out, false);
        }
        return AF_OK;
    }
    case AF_CONTROL_RESET:
-        drop_all_output(s);
+        if (s->avrctx)
+            drop_all_output(s);
        return AF_OK;
    }
    return AF_UNKNOWN;
 }

-#undef ctx_opt_set_int
-#undef ctx_opt_set_dbl
-
 static void uninit(struct af_instance *af)
 {
-    struct af_resample *s = af->priv;
-    if (s->avrctx)
-        avresample_close(s->avrctx);
-    avresample_free(&s->avrctx);
-    if (s->avrctx_out)
-        avresample_close(s->avrctx_out);
-    avresample_free(&s->avrctx_out);
-    talloc_free(s->pending);
+    close_lavrr(af);
 }

-static bool needs_reorder(int *reorder, int num_ch)
+// The LSB is always ignored.
+#if BYTE_ORDER == BIG_ENDIAN
+#define SHIFT24(x) ((3-(x))*8)
+#else
+#define SHIFT24(x) (((x)+1)*8)
+#endif
+
+static void extra_output_conversion(struct af_instance *af, struct mp_audio *mpa)
 {
-    for (int n = 0; n < num_ch; n++) {
-        if (reorder[n] != n)
-            return true;
+    if (mpa->format == AF_FORMAT_S32 && af->data->format == AF_FORMAT_S24) {
+        size_t len = mp_audio_psize(mpa) / mpa->bps;
+        for (int s = 0; s < len; s++) {
+            uint32_t val = *((uint32_t *)mpa->planes[0] + s);
+            uint8_t *ptr = (uint8_t *)mpa->planes[0] + s * 3;
+            ptr[0] = val >> SHIFT24(0);
+            ptr[1] = val >> SHIFT24(1);
+            ptr[2] = val >> SHIFT24(2);
+        }
+        mp_audio_set_format(mpa, AF_FORMAT_S24);
    }
-    return false;
 }

-static void reorder_planes(struct mp_audio *mpa, int *reorder)
+// This relies on the tricky way mpa was allocated.
+static void reorder_planes(struct mp_audio *mpa, int *reorder,
+                           struct mp_chmap *newmap)
 {
    struct mp_audio prev = *mpa;
+    mp_audio_set_channels(mpa, newmap);
+
+    // The trailing planes were never written by avrctx, they're the NA channels.
+    int next_na = prev.num_planes;
+
    for (int n = 0; n < mpa->num_planes; n++) {
-        assert(reorder[n] >= 0 && reorder[n] < mpa->num_planes);
-        mpa->planes[n] = prev.planes[reorder[n]];
+        int src = reorder[n];
+        assert(src >= -1 && src < prev.num_planes);
+        if (src >= 0) {
+            mpa->planes[n] = prev.planes[src];
+        } else {
+            assert(next_na < mpa->num_planes);
+            mpa->planes[n] = prev.planes[next_na++];
+            af_fill_silence(mpa->planes[n], mpa->sstride * mpa->samples,
+                            mpa->format);
+        }
    }
 }

@@ -376,27 +459,17 @@ static int filter(struct af_instance *af, struct mp_audio *in)
 {
    struct af_resample *s = af->priv;

-    if (s->pending) {
-        if (s->pending->samples) {
-            af_add_output_frame(af, s->pending);
-        } else {
-            talloc_free(s->pending);
-        }
-        s->pending = NULL;
-    }
+    int samples = get_out_samples(s, in ? in->samples : 0);

-    int samples = avresample_available(s->avrctx) +
-        av_rescale_rnd(get_delay(s) + (in ? in->samples : 0),
-                       s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP);
-    struct mp_audio *out = mp_audio_pool_get(af->out_pool, af->data, samples);
+    struct mp_audio out_format = s->pool_fmt;
+    struct mp_audio *out = mp_audio_pool_get(af->out_pool, &out_format, samples);
    if (!out)
        goto error;
    if (in)
        mp_audio_copy_attributes(out, in);

-    mp_audio_realloc_min(out, out->samples);
-
-    af->delay = get_delay(s) / (double)s->ctx.in_rate;
+    if (!s->avrctx)
+        goto error;

    if (out->samples) {
        out->samples = resample_frame(s->avrctx, out, in);
@@ -404,11 +477,15 @@ static int filter(struct af_instance *af, struct mp_audio *in)
            goto error;
    }

-    if (needs_reorder(s->reorder_out, out->nch)) {
-        if (AF_FORMAT_IS_PLANAR(out->format)) {
-            reorder_planes(out, s->reorder_out);
-        } else if (out->samples) {
-            struct mp_audio *new = mp_audio_pool_get(s->reorder_buffer, out,
+    struct mp_audio real_out = *out;
+    mp_audio_copy_config(out, &s->avrctx_fmt);
+
+    if (out->samples && !mp_audio_config_equals(out, &s->pre_out_fmt)) {
+        assert(af_fmt_is_planar(out->format) && out->format == real_out.format);
+        reorder_planes(out, s->reorder_out, &s->pool_fmt.channels);
+        if (!mp_audio_config_equals(out, &s->pre_out_fmt)) {
+            struct mp_audio *new = mp_audio_pool_get(s->reorder_buffer,
+                                                     &s->pre_out_fmt,
                                                     out->samples);
            if (!new)
                goto error;
@@ -421,12 +498,17 @@ static int filter(struct af_instance *af, struct mp_audio *in)
        }
    }

+    extra_output_conversion(af, out);
+
    talloc_free(in);
    if (out->samples) {
        af_add_output_frame(af, out);
    } else {
        talloc_free(out);
    }
+
+    af->delay = get_delay(s);
+
    return 0;
 error:
    talloc_free(in);
@@ -445,17 +527,9 @@ static int af_open(struct af_instance *af)
    if (s->opts.cutoff <= 0.0)
        s->opts.cutoff = af_resample_default_cutoff(s->opts.filter_size);

-    s->avrctx = avresample_alloc_context();
-    s->avrctx_out = avresample_alloc_context();
    s->reorder_buffer = mp_audio_pool_create(s);

-    if (s->avrctx && s->avrctx_out) {
-        return AF_OK;
-    } else {
-        MP_ERR(af, "Cannot initialize Libavresample Context. \n");
-        uninit(af);
-        return AF_ERROR;
-    }
+    return AF_OK;
 }

 #define OPT_BASE_STRUCT struct af_resample
@@ -464,13 +538,13 @@ const struct af_info af_info_lavrresample = {
    .info = "Sample frequency conversion using libavresample",
    .name = "lavrresample",
    .open = af_open,
-    .test_conversion = test_conversion,
    .priv_size = sizeof(struct af_resample),
    .priv_defaults = &(const struct af_resample) {
        .opts = {
            .filter_size = 16,
            .cutoff      = 0.0,
            .phase_shift = 10,
+            .normalize   = 1,
        },
        .playback_speed = 1.0,
        .allow_detach = 1,
@@ -481,6 +555,7 @@ const struct af_info af_info_lavrresample = {
        OPT_FLAG("linear", opts.linear, 0),
        OPT_DOUBLE("cutoff", opts.cutoff, M_OPT_RANGE, .min = 0, .max = 1),
        OPT_FLAG("detach", allow_detach, 0),
+        OPT_FLAG("normalize", opts.normalize, 0),
        OPT_KEYVALUELIST("o", avopts, 0),
        {0}
    },
@@ -1,117 +0,0 @@
-/*
- * Copyright (C) 2006 Michael Niedermayer
- * Copyright (C) 2004 Alex Beregszaszi
- * based upon af_extrastereo.c by Pierre Lombard
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <inttypes.h>
-#include <math.h>
-#include <limits.h>
-
-#include "af.h"
-
-// Data for specific instances of this filter
-typedef struct af_sinesuppress_s
-{
-    double freq;
-    double decay;
-    double real;
-    double imag;
-    double ref;
-    double pos;
-}af_sinesuppress_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  switch(cmd){
-  case AF_CONTROL_REINIT:{
-    // Sanity check
-    if(!arg) return AF_ERROR;
-
-    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-    mp_audio_set_num_channels(af->data, 1);
-    mp_audio_set_format(af->data, AF_FORMAT_S16);
-
-    return af_test_output(af,(struct mp_audio*)arg);
-  }
-  }
-  return AF_UNKNOWN;
-}
-
-static int play_s16(struct af_instance *af, struct mp_audio *data)
-{
-  if (!data)
-    return 0;
-  if (af_make_writeable(af, data) < 0) {
-    talloc_free(data);
-    return -1;
-  }
-
-  af_sinesuppress_t *s = af->priv;
-  register int i = 0;
-  int16_t *a = (int16_t*)data->planes[0];       // Audio data
-  int len = data->samples*data->nch;            // Number of samples
-
-  for (i = 0; i < len; i++)
-  {
-    double co= cos(s->pos);
-    double si= sin(s->pos);
-
-    s->real += co * a[i];
-    s->imag += si * a[i];
-    s->ref  += co * co;
-
-    a[i] -= (s->real * co + s->imag * si) / s->ref;
-
-    s->real -= s->real * s->decay;
-    s->imag -= s->imag * s->decay;
-    s->ref  -= s->ref  * s->decay;
-
-    s->pos += 2 * M_PI * s->freq / data->rate;
-  }
-
-   MP_VERBOSE(af, "f:%8.2f: amp:%8.2f\n", s->freq, sqrt(s->real*s->real + s->imag*s->imag) / s->ref);
-
-  af_add_output_frame(af, data);
-  return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
-  af->control=control;
-  af->filter_frame = play_s16;
-  return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_sinesuppress_t
-const struct af_info af_info_sinesuppress = {
-    .info = "Sine Suppress",
-    .name = "sinesuppress",
-    .open = af_open,
-    .priv_size = sizeof(af_sinesuppress_t),
-    .options = (const struct m_option[]) {
-        OPT_DOUBLE("freq", freq, 0, OPTDEF_DOUBLE(50.0)),
-        OPT_DOUBLE("decay", decay, 0, OPTDEF_DOUBLE(0.0001)),
-        {0}
-    },
-};
@@ -1,148 +0,0 @@
-/*
- * Copyright (C) 2002 Anders Johansson ajh@watri.uwa.edu.au
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-/* This filter adds a sub-woofer channels to the audio stream by
-   averaging the left and right channel and low-pass filter them. The
-   low-pass filter is implemented as a 4th order IIR Butterworth
-   filter, with a variable cutoff frequency between 10 and 300 Hz. The
-   filter gives 24dB/octave attenuation. There are two runtime
-   controls one for setting which channel to insert the sub-audio into
-   called AF_CONTROL_SUB_CH and one for setting the cutoff frequency
-   called AF_CONTROL_SUB_FC.
-*/
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "common/common.h"
-#include "af.h"
-#include "dsp.h"
-
-// Q value for low-pass filter
-#define Q 1.0
-
-// Analog domain biquad section
-typedef struct{
-  float a[3];           // Numerator coefficients
-  float b[3];           // Denominator coefficients
-} biquad_t;
-
-// S-parameters for designing 4th order Butterworth filter
-static const biquad_t sp[2] = {{{1.0,0.0,0.0},{1.0,0.765367,1.0}},
-                               {{1.0,0.0,0.0},{1.0,1.847759,1.0}}};
-
-// Data for specific instances of this filter
-typedef struct af_sub_s
-{
-  float w[2][4];        // Filter taps for low-pass filter
-  float q[2][2];        // Circular queues
-  float fc;             // Cutoff frequency [Hz] for low-pass filter
-  float k;              // Filter gain;
-  int ch;               // Channel number which to insert the filtered data
-
-}af_sub_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  af_sub_t* s   = af->priv;
-
-  switch(cmd){
-  case AF_CONTROL_REINIT:{
-    // Sanity check
-    if(!arg) return AF_ERROR;
-
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
-    mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-
-    // Design low-pass filter
-    s->k = 1.0;
-    if((-1 == af_filter_szxform(sp[0].a, sp[0].b, Q, s->fc,
-       (float)af->data->rate, &s->k, s->w[0])) ||
-       (-1 == af_filter_szxform(sp[1].a, sp[1].b, Q, s->fc,
-       (float)af->data->rate, &s->k, s->w[1])))
-      return AF_ERROR;
-    return af_test_output(af,(struct mp_audio*)arg);
-  }
-  }
-  return AF_UNKNOWN;
-}
-
-#ifndef IIR
-#define IIR(in,w,q,out) { \
-  float h0 = (q)[0]; \
-  float h1 = (q)[1]; \
-  float hn = (in) - h0 * (w)[0] - h1 * (w)[1];  \
-  out = hn + h0 * (w)[2] + h1 * (w)[3];  \
-  (q)[1] = h0; \
-  (q)[0] = hn; \
-}
-#endif
-
-static int filter_frame(struct af_instance *af, struct mp_audio *data)
-{
-  if (!data)
-    return 0;
-  if (af_make_writeable(af, data) < 0) {
-    talloc_free(data);
-    return -1;
-  }
-
-  struct mp_audio*    c   = data;        // Current working data
-  af_sub_t*     s   = af->priv; // Setup for this instance
-  float*        a   = c->planes[0];      // Audio data
-  int           len = c->samples*c->nch;         // Number of samples in current audio block
-  int           nch = c->nch;    // Number of channels
-  int           ch  = s->ch;     // Channel in which to insert the sub audio
-  register int  i;
-
-  // Run filter
-  for(i=0;i<len;i+=nch){
-    // Average left and right
-    register float x = 0.5 * (a[i] + a[i+1]);
-    IIR(x * s->k, s->w[0], s->q[0], x);
-    IIR(x , s->w[1], s->q[1], a[i+ch]);
-  }
-
-  af_add_output_frame(af, data);
-  return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
-  af->control=control;
-  af->filter_frame = filter_frame;
-  return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_sub_t
-const struct af_info af_info_sub = {
-    .info = "Audio filter for adding a sub-base channel",
-    .name = "sub",
-    .flags = AF_FLAGS_NOT_REENTRANT,
-    .open = af_open,
-    .priv_size = sizeof(af_sub_t),
-    .options = (const struct m_option[]) {
-        OPT_FLOATRANGE("fc", fc, 0, 20, 300, OPTDEF_FLOAT(60.0)),
-        OPT_INTRANGE("ch", ch, 0, 0, AF_NCH - 1, OPTDEF_INT(5)),
-        {0}
-    },
-};
@@ -1,246 +0,0 @@
-/*
- * Filter to do simple decoding of matrixed surround sound.
- * This will provide a (basic) surround-sound effect from
- * audio encoded for Dolby Surround, Pro Logic etc.
- *
- * Original author: Steve Davies <steve@daviesfam.org>
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-/* The principle:  Make rear channels by extracting anti-phase data
-   from the front channels, delay by 20ms and feed to rear in anti-phase
-*/
-
-
-/* SPLITREAR: Define to decode two distinct rear channels - this
-     doesn't work so well in practice because separation in a passive
-     matrix is not high. C (dialogue) to Ls and Rs 14dB or so - so
-     dialogue leaks to the rear.  Still - give it a try and send
-     feedback. Comment this define for old behavior of a single
-     surround sent to rear in anti-phase */
-#define SPLITREAR 1
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "af.h"
-#include "dsp.h"
-
-#define L  32    // Length of fir filter
-#define LD 65536 // Length of delay buffer
-
-// 32 Tap fir filter loop unrolled
-#define FIR(x,w,y) \
-  y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3]  \
-      + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7]  \
-      + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
-      + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \
-      + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \
-      + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \
-      + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \
-      + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31])
-
-// Add to circular queue macro + update index
-#ifdef SPLITREAR
-#define ADDQUE(qi,rq,lq,r,l)\
-  lq[qi]=lq[qi+L]=(l);\
-  rq[qi]=rq[qi+L]=(r);\
-  qi=(qi-1)&(L-1);
-#else
-#define ADDQUE(qi,lq,l)\
-  lq[qi]=lq[qi+L]=(l);\
-  qi=(qi-1)&(L-1);
-#endif
-
-// Macro for updating queue index in delay queues
-#define UPDATEQI(qi) qi=(qi+1)&(LD-1)
-
-// instance data
-typedef struct af_surround_s
-{
-  float lq[2*L]; // Circular queue for filtering left rear channel
-  float rq[2*L]; // Circular queue for filtering right rear channel
-  float w[L];    // FIR filter coefficients for surround sound 7kHz low-pass
-  float* dr;     // Delay queue right rear channel
-  float* dl;     // Delay queue left rear channel
-  float  d;      // Delay time
-  int i;         // Position in circular buffer
-  int wi;        // Write index for delay queue
-  int ri;        // Read index for delay queue
-}af_surround_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  af_surround_t *s = af->priv;
-  switch(cmd){
-  case AF_CONTROL_REINIT:{
-    struct mp_audio *in = arg;
-    float fc;
-    if (!mp_chmap_is_stereo(&in->channels)) {
-        MP_ERR(af, "Only stereo input is supported.\n");
-        return AF_DETACH;
-    }
-
-    mp_audio_set_format(in, AF_FORMAT_FLOAT);
-    mp_audio_copy_config(af->data, in);
-    mp_audio_set_channels_old(af->data, in->nch * 2);
-
-    // Surround filer coefficients
-    fc = 2.0 * 7000.0/(float)af->data->rate;
-    if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
-      MP_ERR(af, "Unable to design low-pass filter.\n");
-      return AF_ERROR;
-    }
-
-    // Free previous delay queues
-    free(s->dl);
-    free(s->dr);
-    // Allocate new delay queues
-    s->dl = calloc(LD,af->data->bps);
-    s->dr = calloc(LD,af->data->bps);
-    if((NULL == s->dl) || (NULL == s->dr))
-      MP_FATAL(af, "Out of memory\n");
-
-    // Initialize delay queue index
-    if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
-      return AF_ERROR;
-//    printf("%i\n",s->wi);
-    s->ri = 0;
-
-    return AF_OK;
-  }
-  }
-  return AF_UNKNOWN;
-}
-
-// The beginnings of an active matrix...
-static const float steering_matrix[][12] = {
-//      LL      RL      LR      RR      LS      RS
-//      LLs     RLs     LRs     RRs     LC      RC
-       {.707,   .0,     .0,     .707,   .5,     -.5,
-        .5878,  -.3928, .3928,  -.5878, .5,     .5},
-};
-
-// Experimental moving average dominance
-//static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
-
-static int filter_frame(struct af_instance *af, struct mp_audio *data)
-{
-  if (!data)
-    return 0;
-  struct mp_audio *outframe =
-    mp_audio_pool_get(af->out_pool, &af->fmt_out, data->samples);
-  if (!outframe) {
-    talloc_free(data);
-    return -1;
-  }
-  mp_audio_copy_attributes(outframe, data);
-
-  af_surround_t* s   = (af_surround_t*)af->priv;
-  const float*   m   = steering_matrix[0];
-  float*         in  = data->planes[0];         // Input audio data
-  float*         out = outframe->planes[0];     // Output audio data
-  float*         end = in + data->samples * data->nch;
-  int            i   = s->i;    // Filter queue index
-  int            ri  = s->ri;   // Read index for delay queue
-  int            wi  = s->wi;   // Write index for delay queue
-
-  while(in < end){
-    /* Dominance:
-       abs(in[0])  abs(in[1]);
-       abs(in[0]+in[1])  abs(in[0]-in[1]);
-       10 * log( abs(in[0]) / (abs(in[1])|1) );
-       10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */
-
-    /* About volume balancing...
-       Surround encoding does the following:
-           Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
-       So S should be extracted as:
-           (Lt-Rt)
-       But we are splitting the S to two output channels, so we
-       must take 3dB off as we split it:
-           Ls=Rs=.707*(Lt-Rt)
-       Trouble is, Lt could be +1, Rt -1, so possibility that S will
-       overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by
-       6dB (/2). This keeps the overall balance, but guarantees no
-       overflow. */
-
-    // Output front left and right
-    out[0] = m[0]*in[0] + m[1]*in[1];
-    out[1] = m[2]*in[0] + m[3]*in[1];
-
-    // Low-pass output @ 7kHz
-    FIR((&s->lq[i]), s->w, s->dl[wi]);
-
-    // Delay output by d ms
-    out[2] = s->dl[ri];
-
-#ifdef SPLITREAR
-    // Low-pass output @ 7kHz
-    FIR((&s->rq[i]), s->w, s->dr[wi]);
-
-    // Delay output by d ms
-    out[3] = s->dr[ri];
-#else
-    out[3] = -out[2];
-#endif
-
-    // Update delay queues indexes
-    UPDATEQI(ri);
-    UPDATEQI(wi);
-
-    // Calculate and save surround in circular queue
-#ifdef SPLITREAR
-    ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
-#else
-    ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
-#endif
-
-    // Next sample...
-    in = &in[data->nch];
-    out = &out[af->data->nch];
-  }
-
-  // Save indexes
-  s->i  = i; s->ri = ri; s->wi = wi;
-
-  talloc_free(data);
-  af_add_output_frame(af, outframe);
-  return 0;
-}
-
-static int af_open(struct af_instance* af){
-  af->control=control;
-  af->filter_frame = filter_frame;
-  return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_surround_t
-const struct af_info af_info_surround =
-{
-    .info = "Surround decoder filter",
-    .name = "surround",
-    .flags = AF_FLAGS_NOT_REENTRANT,
-    .open = af_open,
-    .priv_size = sizeof(af_surround_t),
-    .options = (const struct m_option[]) {
-        OPT_FLOATRANGE("d", d, 0, 0, 1000, OPTDEF_FLOAT(20.0)),
-        {0}
-    },
-};
@@ -1,92 +0,0 @@
-/*
- * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-#include <math.h>
-
-#include "config.h"
-#include "af.h"
-
-typedef struct af_sweep_s{
-    double x;
-    double delta;
-}af_sweept;
-
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  struct mp_audio *data= (struct mp_audio*)arg;
-
-  switch(cmd){
-  case AF_CONTROL_REINIT:
-    mp_audio_copy_config(af->data, data);
-    mp_audio_set_format(af->data, AF_FORMAT_S16);
-
-    return af_test_output(af, data);
-  }
-  return AF_UNKNOWN;
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *data)
-{
-  if (!data)
-    return 0;
-  if (af_make_writeable(af, data) < 0) {
-      talloc_free(data);
-    return 0;
-  }
-
-  af_sweept *s = af->priv;
-  int i, j;
-  int16_t *in = (int16_t*)data->planes[0];
-  int chans   = data->nch;
-  int in_len  = data->samples;
-
-  for(i=0; i<in_len; i++){
-      for(j=0; j<chans; j++)
-          in[i*chans+j]= 32000*sin(s->x*s->x);
-      s->x += s->delta;
-      if(2*s->x*s->delta >= 3.141592) s->x=0;
-  }
-
-  af_add_output_frame(af, data);
-  return 0;
-}
-
-static int af_open(struct af_instance* af){
-  af->control=control;
-  af->filter_frame = filter_frame;
-  return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_sweept
-const struct af_info af_info_sweep = {
-    .info = "sine sweep",
-    .name = "sweep",
-    .open = af_open,
-    .priv_size = sizeof(af_sweept),
-    .options = (const struct m_option[]) {
-        OPT_DOUBLE("delta", delta, 0),
-        {0}
-    },
-};
@@ -30,18 +30,28 @@
 #include "demux/demux.h"

 struct priv {
+    float vol;                  // User-specified non-linear volume
    float level;                // User-specified gain level for each channel
    float rgain;                // Replaygain level
    int rgain_track;            // Enable/disable track based replaygain
    int rgain_album;            // Enable/disable album based replaygain
    float rgain_preamp;         // Set replaygain pre-amplification
    int rgain_clip;             // Enable/disable clipping prevention
+    float replaygain_fallback;
    int soft;                   // Enable/disable soft clipping
    int fast;                   // Use fix-point volume control
    int detach;                 // Detach if gain volume is neutral
    float cfg_volume;
 };

+// Convert to gain value from dB. input <= -200dB will become 0 gain.
+static float from_dB(float in, float k, float mi, float ma)
+{
+    if (in <= -200)
+        return 0.0;
+    return pow(10.0, MPCLAMP(in, mi, ma) / k);
+}
+
 static int control(struct af_instance *af, int cmd, void *arg)
 {
    struct priv *s = af->priv;
@@ -58,7 +68,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
        } else {
            mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
        }
-        if (AF_FORMAT_IS_PLANAR(in->format))
+        if (af_fmt_is_planar(in->format))
            mp_audio_set_format(af->data, af_fmt_to_planar(af->data->format));
        s->rgain = 1.0;
        if ((s->rgain_track || s->rgain_album) && af->replaygain_data) {
@@ -73,7 +83,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
            }

            gain += s->rgain_preamp;
-            af_from_dB(1, &gain, &s->rgain, 20.0, -200.0, 60.0);
+            s->rgain = from_dB(gain, 20.0, -200.0, 60.0);

            MP_VERBOSE(af, "Applying replay-gain: %f\n", s->rgain);

@@ -81,16 +91,21 @@ static int control(struct af_instance *af, int cmd, void *arg)
                s->rgain = MPMIN(s->rgain, 1.0 / peak);
                MP_VERBOSE(af, "...with clipping prevention: %f\n", s->rgain);
            }
+        } else if (s->replaygain_fallback) {
+            s->rgain = from_dB(s->replaygain_fallback, 20.0, -200.0, 60.0);
+            MP_VERBOSE(af, "Applying fallback gain: %f\n", s->rgain);
        }
        if (s->detach && fabs(s->level * s->rgain - 1.0) < 0.00001)
            return AF_DETACH;
        return af_test_output(af, in);
    }
    case AF_CONTROL_SET_VOLUME:
-        s->level = *(float *)arg;
+        s->vol = *(float *)arg;
+        s->level = pow(s->vol, 3);
+        MP_VERBOSE(af, "volume gain: %f\n", s->level);
        return AF_OK;
    case AF_CONTROL_GET_VOLUME:
-        *(float *)arg = s->level;
+        *(float *)arg = s->vol;
        return AF_OK;
    }
    return AF_UNKNOWN;
@@ -143,7 +158,7 @@ static int af_open(struct af_instance *af)
    struct priv *s = af->priv;
    af->control = control;
    af->filter_frame = filter;
-    af_from_dB(1, &s->cfg_volume, &s->level, 20.0, -200.0, 60.0);
+    s->level = from_dB(s->cfg_volume, 20.0, -200.0, 60.0);
    return AF_OK;
 }

@@ -153,7 +168,6 @@ static int af_open(struct af_instance *af)
 const struct af_info af_info_volume = {
    .info = "Volume control audio filter",
    .name = "volume",
-    .flags = AF_FLAGS_NOT_REENTRANT,
    .open = af_open,
    .priv_size = sizeof(struct priv),
    .options = (const struct m_option[]) {
@@ -162,6 +176,7 @@ const struct af_info af_info_volume = {
        OPT_FLAG("replaygain-album", rgain_album, 0),
        OPT_FLOATRANGE("replaygain-preamp", rgain_preamp, 0, -15, 15),
        OPT_FLAG("replaygain-clip", rgain_clip, 0),
+        OPT_FLOATRANGE("replaygain-fallback", replaygain_fallback, 0, -200, 60),
        OPT_FLAG("softclip", soft, 0),
        OPT_FLAG("s16", fast, 0),
        OPT_FLAG("detach", detach, 0),
@@ -1,359 +0,0 @@
-/*
- * design and implementation of different types of digital filters
- *
- * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <string.h>
-#include <math.h>
-#include "dsp.h"
-
-/******************************************************************************
-*  FIR filter implementations
-******************************************************************************/
-
-/* C implementation of FIR filter y=w*x
-
-   n number of filter taps, where mod(n,4)==0
-   w filter taps
-   x input signal must be a circular buffer which is indexed backwards
-*/
-inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w,
-                                const FLOAT_TYPE* x)
-{
-  register FLOAT_TYPE y; // Output
-  y = 0.0;
-  do{
-    n--;
-    y+=w[n]*x[n];
-  }while(n != 0);
-  return y;
-}
-
-/******************************************************************************
-*  FIR filter design
-******************************************************************************/
-
-/* Design FIR filter using the Window method
-
-   n     filter length must be odd for HP and BS filters
-   w     buffer for the filter taps (must be n long)
-   fc    cutoff frequencies (1 for LP and HP, 2 for BP and BS)
-         0 < fc < 1 where 1 <=> Fs/2
-   flags window and filter type as defined in filter.h
-         variables are ored together: i.e. LP|HAMMING will give a
-         low pass filter designed using a hamming window
-   opt   beta constant used only when designing using kaiser windows
-
-   returns 0 if OK, -1 if fail
-*/
-int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
-                         unsigned int flags, FLOAT_TYPE opt)
-{
-  unsigned int  o   = n & 1;            // Indicator for odd filter length
-  unsigned int  end = ((n + 1) >> 1) - o;       // Loop end
-  unsigned int  i;                      // Loop index
-
-  FLOAT_TYPE k1 = 2 * M_PI;             // 2*pi*fc1
-  FLOAT_TYPE k2 = 0.5 * (FLOAT_TYPE)(1 - o);// Constant used if the filter has even length
-  FLOAT_TYPE k3;                        // 2*pi*fc2 Constant used in BP and BS design
-  FLOAT_TYPE g  = 0.0;                  // Gain
-  FLOAT_TYPE t1,t2,t3;                  // Temporary variables
-  FLOAT_TYPE fc1,fc2;                   // Cutoff frequencies
-
-  // Sanity check
-  if(!w || (n == 0)) return -1;
-
-  // Get window coefficients
-  switch(flags & WINDOW_MASK){
-  case(BOXCAR):
-    af_window_boxcar(n,w); break;
-  case(TRIANG):
-    af_window_triang(n,w); break;
-  case(HAMMING):
-    af_window_hamming(n,w); break;
-  case(HANNING):
-    af_window_hanning(n,w); break;
-  case(BLACKMAN):
-    af_window_blackman(n,w); break;
-  case(FLATTOP):
-    af_window_flattop(n,w); break;
-  case(KAISER):
-    af_window_kaiser(n,w,opt); break;
-  default:
-    return -1;
-  }
-
-  if(flags & (LP | HP)){
-    fc1=*fc;
-    // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2
-    fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
-    k1 *= fc1;
-
-    if(flags & LP){ // Low pass filter
-
-      // If the filter length is odd, there is one point which is exactly
-      // in the middle. The value at this point is 2*fCutoff*sin(x)/x,
-      // where x is zero. To make sure nothing strange happens, we set this
-      // value separately.
-      if (o){
-        w[end] = fc1 * w[end] * 2.0;
-        g=w[end];
-      }
-
-      // Create filter
-      for (i=0 ; i<end ; i++){
-        t1 = (FLOAT_TYPE)(i+1) - k2;
-        w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
-        g += 2*w[end-i-1]; // Total gain in filter
-      }
-    }
-    else{ // High pass filter
-      if (!o) // High pass filters must have odd length
-        return -1;
-      w[end] = 1.0 - (fc1 * w[end] * 2.0);
-      g= w[end];
-
-      // Create filter
-      for (i=0 ; i<end ; i++){
-        t1 = (FLOAT_TYPE)(i+1);
-        w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
-        g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter
-      }
-    }
-  }
-
-  if(flags & (BP | BS)){
-    fc1=fc[0];
-    fc2=fc[1];
-    // Cutoff frequencies must be < 1.0 where 1.0 <=> Fs/2
-    fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
-    fc2 = ((fc2 <= 1.0) && (fc2 > 0.0)) ? fc2/2 : 0.25;
-    k3  = k1 * fc2; // 2*pi*fc2
-    k1 *= fc1;      // 2*pi*fc1
-
-    if(flags & BP){ // Band pass
-      // Calculate center tap
-      if (o){
-        g=w[end]*(fc1+fc2);
-        w[end] = (fc2 - fc1) * w[end] * 2.0;
-      }
-
-      // Create filter
-      for (i=0 ; i<end ; i++){
-        t1 = (FLOAT_TYPE)(i+1) - k2;
-        t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
-        t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
-        g += w[end-i-1] * (t3 + t2);   // Total gain in filter
-        w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
-      }
-    }
-    else{ // Band stop
-      if (!o) // Band stop filters must have odd length
-        return -1;
-      w[end] = 1.0 - (fc2 - fc1) * w[end] * 2.0;
-      g= w[end];
-
-      // Create filter
-      for (i=0 ; i<end ; i++){
-        t1 = (FLOAT_TYPE)(i+1);
-        t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
-        t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
-        w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
-        g += 2*w[end-i-1]; // Total gain in filter
-      }
-    }
-  }
-
-  // Normalize gain
-  g=1/g;
-  for (i=0; i<n; i++)
-    w[i] *= g;
-
-  return 0;
-}
-
-/******************************************************************************
-*  IIR filter design
-******************************************************************************/
-
-/* Helper functions for the bilinear transform */
-
-/* Pre-warp the coefficients of a numerator or denominator.
-   Note that a0 is assumed to be 1, so there is no wrapping
-   of it.
-*/
-static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs)
-{
-  FLOAT_TYPE wp;
-  wp = 2.0 * fs * tan(M_PI * fc / fs);
-  a[2] = a[2]/(wp * wp);
-  a[1] = a[1]/wp;
-}
-
-/* Transform the numerator and denominator coefficients of s-domain
-   biquad section into corresponding z-domain coefficients.
-
-   The transfer function for z-domain is:
-
-          1 + alpha1 * z^(-1) + alpha2 * z^(-2)
-   H(z) = -------------------------------------
-          1 + beta1 * z^(-1) + beta2 * z^(-2)
-
-   Store the 4 IIR coefficients in array pointed by coef in following
-   order:
-   beta1, beta2    (denominator)
-   alpha1, alpha2  (numerator)
-
-   Arguments:
-   a       - s-domain numerator coefficients
-   b       - s-domain denominator coefficients
-   k       - filter gain factor. Initially set to 1 and modified by each
-             biquad section in such a way, as to make it the
-             coefficient by which to multiply the overall filter gain
-             in order to achieve a desired overall filter gain,
-             specified in initial value of k.
-   fs      - sampling rate (Hz)
-   coef    - array of z-domain coefficients to be filled in.
-
-   Return: On return, set coef z-domain coefficients and k to the gain
-   required to maintain overall gain = 1.0;
-*/
-static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE* k,
-                               FLOAT_TYPE fs, FLOAT_TYPE *coef)
-{
-  FLOAT_TYPE ad, bd;
-
-  /* alpha (Numerator in s-domain) */
-  ad = 4. * a[2] * fs * fs + 2. * a[1] * fs + a[0];
-  /* beta (Denominator in s-domain) */
-  bd = 4. * b[2] * fs * fs + 2. * b[1] * fs + b[0];
-
-  /* Update gain constant for this section */
-  *k *= ad/bd;
-
-  /* Denominator */
-  *coef++ = (2. * b[0] - 8. * b[2] * fs * fs)/bd; /* beta1 */
-  *coef++ = (4. * b[2] * fs * fs - 2. * b[1] * fs + b[0])/bd; /* beta2 */
-
-  /* Numerator */
-  *coef++ = (2. * a[0] - 8. * a[2] * fs * fs)/ad; /* alpha1 */
-  *coef   = (4. * a[2] * fs * fs - 2. * a[1] * fs + a[0])/ad;   /* alpha2 */
-}
-
-
-
-/* IIR filter design using bilinear transform and prewarp. Transforms
-   2nd order s domain analog filter into a digital IIR biquad link. To
-   create a filter fill in a, b, Q and fs and make space for coef and k.
-
-
-   Example Butterworth design:
-
-   Below are Butterworth polynomials, arranged as a series of 2nd
-   order sections:
-
-   Note: n is filter order.
-
-   n  Polynomials
-   -------------------------------------------------------------------
-   2  s^2 + 1.4142s + 1
-   4  (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1)
-   6  (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1)
-
-   For n=4 we have following equation for the filter transfer function:
-                       1                              1
-   T(s) = --------------------------- * ----------------------------
-          s^2 + (1/Q) * 0.765367s + 1   s^2 + (1/Q) * 1.847759s + 1
-
-   The filter consists of two 2nd order sections since highest s power
-   is 2.  Now we can take the coefficients, or the numbers by which s
-   is multiplied and plug them into a standard formula to be used by
-   bilinear transform.
-
-   Our standard form for each 2nd order section is:
-
-          a2 * s^2 + a1 * s + a0
-   H(s) = ----------------------
-          b2 * s^2 + b1 * s + b0
-
-   Note that Butterworth numerator is 1 for all filter sections, which
-   means s^2 = 0 and s^1 = 0
-
-   Let's convert standard Butterworth polynomials into this form:
-
-             0 + 0 + 1                  0 + 0 + 1
-   --------------------------- * --------------------------
-   1 + ((1/Q) * 0.765367) + 1   1 + ((1/Q) * 1.847759) + 1
-
-   Section 1:
-   a2 = 0; a1 = 0; a0 = 1;
-   b2 = 1; b1 = 0.765367; b0 = 1;
-
-   Section 2:
-   a2 = 0; a1 = 0; a0 = 1;
-   b2 = 1; b1 = 1.847759; b0 = 1;
-
-   Q is filter quality factor or resonance, in the range of 1 to
-   1000. The overall filter Q is a product of all 2nd order stages.
-   For example, the 6th order filter (3 stages, or biquads) with
-   individual Q of 2 will have filter Q = 2 * 2 * 2 = 8.
-
-
-   Arguments:
-   a       - s-domain numerator coefficients, a[1] is always assumed to be 1.0
-   b       - s-domain denominator coefficients
-   Q       - Q value for the filter
-   k       - filter gain factor. Initially set to 1 and modified by each
-             biquad section in such a way, as to make it the
-             coefficient by which to multiply the overall filter gain
-             in order to achieve a desired overall filter gain,
-             specified in initial value of k.
-   fs      - sampling rate (Hz)
-   coef    - array of z-domain coefficients to be filled in.
-
-   Note: Upon return from each call, the k argument will be set to a
-   value, by which to multiply our actual signal in order for the gain
-   to be one. On second call to szxform() we provide k that was
-   changed by the previous section. During actual audio filtering
-   k can be used for gain compensation.
-
-   return -1 if fail 0 if success.
-*/
-int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q, FLOAT_TYPE fc,
-                      FLOAT_TYPE fs, FLOAT_TYPE *k, FLOAT_TYPE *coef)
-{
-  FLOAT_TYPE at[3];
-  FLOAT_TYPE bt[3];
-
-  if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0))
-    return -1;
-
-  memcpy(at,a,3*sizeof(FLOAT_TYPE));
-  memcpy(bt,b,3*sizeof(FLOAT_TYPE));
-
-  bt[1]/=Q;
-
-  /* Calculate a and b and overwrite the original values */
-  af_filter_prewarp(at, fc, fs);
-  af_filter_prewarp(bt, fc, fs);
-  /* Execute bilinear transform */
-  af_filter_bilinear(at, bt, k, fs, coef);
-
-  return 0;
-}
@@ -1,74 +0,0 @@
-/*
- * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#if !defined MPLAYER_DSP_H
-# error Never use filter.h directly; include dsp.h instead.
-#endif
-
-#ifndef MPLAYER_FILTER_H
-#define MPLAYER_FILTER_H
-
-
-// Design and implementation of different types of digital filters
-
-
-// Flags used for filter design
-
-// Filter characteristics
-#define LP          0x00010000 // Low pass
-#define HP          0x00020000 // High pass
-#define BP          0x00040000 // Band pass
-#define BS          0x00080000 // Band stop
-#define TYPE_MASK   0x000F0000
-
-// Window types
-#define BOXCAR      0x00000001
-#define TRIANG      0x00000002
-#define HAMMING     0x00000004
-#define HANNING     0x00000008
-#define BLACKMAN    0x00000010
-#define FLATTOP     0x00000011
-#define KAISER      0x00000012
-#define WINDOW_MASK 0x0000001F
-
-// Parallel filter design
-#define FWD         0x00000001 // Forward indexing of polyphase filter
-#define REW         0x00000002 // Reverse indexing of polyphase filter
-#define ODD         0x00000010 // Make filter HP
-
-// Exported functions
-FLOAT_TYPE af_filter_fir(unsigned int n, const FLOAT_TYPE* w, const FLOAT_TYPE* x);
-
-int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
-                         unsigned int flags, FLOAT_TYPE opt);
-
-int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q,
-                      FLOAT_TYPE fc, FLOAT_TYPE fs, FLOAT_TYPE *k,
-                      FLOAT_TYPE *coef);
-
-/* Add new data to circular queue designed to be used with a FIR
-   filter. xq is the circular queue, in pointing at the new sample, xi
-   current index for xq and n the length of the filter. xq must be n*2
-   long.
-*/
-#define af_filter_updateq(n,xi,xq,in)\
-  xq[xi]=(xq)[(xi)+(n)]=*(in);\
-  xi=(++(xi))&((n)-1);
-
-#endif /* MPLAYER_FILTER_H */
@@ -21,24 +21,6 @@
 #include "common/common.h"
 #include "af.h"

-/* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
- * fail. input <= -200dB will become 0 gain. */
-int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
-{
-  int i = 0;
-  // Sanity check
-  if(!in || !out)
-    return AF_ERROR;
-
-  for(i=0;i<n;i++){
-    if(in[i]<=-200)
-      out[i]=0.0;
-    else
-      out[i]=pow(10.0,MPCLAMP(in[i],mi,ma)/k);
-  }
-  return AF_OK;
-}
-
 /* Convert from ms to sample time */
 int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
 {
@@ -1,212 +0,0 @@
-/*
- * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-/* Calculates a number of window functions. The following window
-   functions are currently implemented: Boxcar, Triang, Hanning,
-   Hamming, Blackman, Flattop and Kaiser. In the function call n is
-   the number of filter taps and w the buffer in which the filter
-   coefficients will be stored.
-*/
-
-#include <math.h>
-#include "dsp.h"
-
-/*
-// Boxcar
-//
-// n window length
-// w buffer for the window parameters
-*/
-void af_window_boxcar(int n, FLOAT_TYPE* w)
-{
-  int i;
-  // Calculate window coefficients
-  for (i=0 ; i<n ; i++)
-    w[i] = 1.0;
-}
-
-
-/*
-// Triang a.k.a Bartlett
-//
-//               |    (N-1)|
-//           2 * |k - -----|
-//               |      2  |
-// w = 1.0 - ---------------
-//                    N+1
-// n window length
-// w buffer for the window parameters
-*/
-void af_window_triang(int n, FLOAT_TYPE* w)
-{
-  FLOAT_TYPE k1  = (FLOAT_TYPE)(n & 1);
-  FLOAT_TYPE k2  = 1/((FLOAT_TYPE)n + k1);
-  int      end = (n + 1) >> 1;
-  int      i;
-
-  // Calculate window coefficients
-  for (i=0 ; i<end ; i++)
-    w[i] = w[n-i-1] = (2.0*((FLOAT_TYPE)(i+1))-(1.0-k1))*k2;
-}
-
-
-/*
-// Hanning
-//                   2*pi*k
-// w = 0.5 - 0.5*cos(------), where 0 < k <= N
-//                    N+1
-// n window length
-// w buffer for the window parameters
-*/
-void af_window_hanning(int n, FLOAT_TYPE* w)
-{
-  int      i;
-  FLOAT_TYPE k = 2*M_PI/((FLOAT_TYPE)(n+1)); // 2*pi/(N+1)
-
-  // Calculate window coefficients
-  for (i=0; i<n; i++)
-    *w++ = 0.5*(1.0 - cos(k*(FLOAT_TYPE)(i+1)));
-}
-
-/*
-// Hamming
-//                        2*pi*k
-// w(k) = 0.54 - 0.46*cos(------), where 0 <= k < N
-//                         N-1
-//
-// n window length
-// w buffer for the window parameters
-*/
-void af_window_hamming(int n,FLOAT_TYPE* w)
-{
-  int      i;
-  FLOAT_TYPE k = 2*M_PI/((FLOAT_TYPE)(n-1)); // 2*pi/(N-1)
-
-  // Calculate window coefficients
-  for (i=0; i<n; i++)
-    *w++ = 0.54 - 0.46*cos(k*(FLOAT_TYPE)i);
-}
-
-/*
-// Blackman
-//                       2*pi*k             4*pi*k
-// w(k) = 0.42 - 0.5*cos(------) + 0.08*cos(------), where 0 <= k < N
-//                        N-1                 N-1
-//
-// n window length
-// w buffer for the window parameters
-*/
-void af_window_blackman(int n,FLOAT_TYPE* w)
-{
-  int      i;
-  FLOAT_TYPE k1 = 2*M_PI/((FLOAT_TYPE)(n-1)); // 2*pi/(N-1)
-  FLOAT_TYPE k2 = 2*k1; // 4*pi/(N-1)
-
-  // Calculate window coefficients
-  for (i=0; i<n; i++)
-    *w++ = 0.42 - 0.50*cos(k1*(FLOAT_TYPE)i) + 0.08*cos(k2*(FLOAT_TYPE)i);
-}
-
-/*
-// Flattop
-//                                        2*pi*k                     4*pi*k
-// w(k) = 0.2810638602 - 0.5208971735*cos(------) + 0.1980389663*cos(------), where 0 <= k < N
-//                                          N-1                        N-1
-//
-// n window length
-// w buffer for the window parameters
-*/
-void af_window_flattop(int n,FLOAT_TYPE* w)
-{
-  int      i;
-  FLOAT_TYPE k1 = 2*M_PI/((FLOAT_TYPE)(n-1)); // 2*pi/(N-1)
-  FLOAT_TYPE k2 = 2*k1;                   // 4*pi/(N-1)
-
-  // Calculate window coefficients
-  for (i=0; i<n; i++)
-    *w++ = 0.2810638602 - 0.5208971735*cos(k1*(FLOAT_TYPE)i)
-                        + 0.1980389663*cos(k2*(FLOAT_TYPE)i);
-}
-
-/* Computes the 0th order modified Bessel function of the first kind.
-// (Needed to compute Kaiser window)
-//
-// y = sum( (x/(2*n))^2 )
-//      n
-*/
-#define BIZ_EPSILON 1E-21 // Max error acceptable
-
-static FLOAT_TYPE besselizero(FLOAT_TYPE x)
-{
-  FLOAT_TYPE temp;
-  FLOAT_TYPE sum   = 1.0;
-  FLOAT_TYPE u     = 1.0;
-  FLOAT_TYPE halfx = x/2.0;
-  int      n     = 1;
-
-  do {
-    temp = halfx/(FLOAT_TYPE)n;
-    u *=temp * temp;
-    sum += u;
-    n++;
-  } while (u >= BIZ_EPSILON * sum);
-  return sum;
-}
-
-/*
-// Kaiser
-//
-// n window length
-// w buffer for the window parameters
-// b beta parameter of Kaiser window, Beta >= 1
-//
-// Beta trades the rejection of the low pass filter against the
-// transition width from passband to stop band.  Larger Beta means a
-// slower transition and greater stop band rejection.  See Rabiner and
-// Gold (Theory and Application of DSP) under Kaiser windows for more
-// about Beta.  The following table from Rabiner and Gold gives some
-// feel for the effect of Beta:
-//
-// All ripples in dB, width of transition band = D*N where N = window
-// length
-//
-// BETA    D       PB RIP   SB RIP
-// 2.120   1.50  +-0.27      -30
-// 3.384   2.23    0.0864    -40
-// 4.538   2.93    0.0274    -50
-// 5.658   3.62    0.00868   -60
-// 6.764   4.32    0.00275   -70
-// 7.865   5.0     0.000868  -80
-// 8.960   5.7     0.000275  -90
-// 10.056  6.4     0.000087  -100
-*/
-void af_window_kaiser(int n, FLOAT_TYPE* w, FLOAT_TYPE b)
-{
-  FLOAT_TYPE tmp;
-  FLOAT_TYPE k1  = 1.0/besselizero(b);
-  int      k2  = 1 - (n & 1);
-  int      end = (n + 1) >> 1;
-  int      i;
-
-  // Calculate window coefficients
-  for (i=0 ; i<end ; i++){
-    tmp = (FLOAT_TYPE)(2*i + k2) / ((FLOAT_TYPE)n - 1.0);
-    w[end-(1&(!k2))+i] = w[end-1-i] = k1 * besselizero(b*sqrt(1.0 - tmp*tmp));
-  }
-}
@@ -1,42 +0,0 @@
-/*
- * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-/* Calculates a number of window functions. The following window
-   functions are currently implemented: Boxcar, Triang, Hanning,
-   Hamming, Blackman, Flattop and Kaiser. In the function call n is
-   the number of filter taps and w the buffer in which the filter
-   coefficients will be stored.
-*/
-
-#if !defined MPLAYER_DSP_H
-# error Never use window.h directly; include dsp.h instead.
-#endif
-
-#ifndef MPLAYER_WINDOW_H
-#define MPLAYER_WINDOW_H
-
-void af_window_boxcar(int n, FLOAT_TYPE* w);
-void af_window_triang(int n, FLOAT_TYPE* w);
-void af_window_hanning(int n, FLOAT_TYPE* w);
-void af_window_hamming(int n, FLOAT_TYPE* w);
-void af_window_blackman(int n, FLOAT_TYPE* w);
-void af_window_flattop(int n, FLOAT_TYPE* w);
-void af_window_kaiser(int n, FLOAT_TYPE* w, FLOAT_TYPE b);
-
-#endif /* MPLAYER_WINDOW_H */
@@ -27,42 +27,68 @@
 #include "common/common.h"
 #include "audio/filter/af.h"

-int af_fmt2bps(int format)
+// number of bytes per sample, 0 if invalid/unknown
+int af_fmt_to_bytes(int format)
 {
-    switch (format & AF_FORMAT_BITS_MASK) {
-    case AF_FORMAT_8BIT:  return 1;
-    case AF_FORMAT_16BIT: return 2;
-    case AF_FORMAT_24BIT: return 3;
-    case AF_FORMAT_32BIT: return 4;
-    case AF_FORMAT_64BIT: return 8;
+    switch (af_fmt_from_planar(format)) {
+    case AF_FORMAT_U8:      return 1;
+    case AF_FORMAT_S16:     return 2;
+    case AF_FORMAT_S24:     return 3;
+    case AF_FORMAT_S32:     return 4;
+    case AF_FORMAT_FLOAT:   return 4;
+    case AF_FORMAT_DOUBLE:  return 8;
    }
+    if (af_fmt_is_spdif(format))
+        return 2;
    return 0;
 }

-int af_fmt2bits(int format)
+int af_fmt_change_bytes(int format, int bytes)
 {
-    return af_fmt2bps(format) * 8;
-}
-
-static int bits_to_mask(int bits)
-{
-    switch (bits) {
-    case 8:  return AF_FORMAT_8BIT;
-    case 16: return AF_FORMAT_16BIT;
-    case 24: return AF_FORMAT_24BIT;
-    case 32: return AF_FORMAT_32BIT;
-    case 64: return AF_FORMAT_64BIT;
-    }
-    return 0;
-}
-
-int af_fmt_change_bits(int format, int bits)
-{
-    if (!af_fmt_is_valid(format))
+    if (!af_fmt_is_valid(format) || !bytes)
        return 0;
-    int mask = bits_to_mask(bits);
-    format = (format & ~AF_FORMAT_BITS_MASK) | mask;
-    return af_fmt_is_valid(format) ? format : 0;
+    for (int fmt = 1; fmt < AF_FORMAT_COUNT; fmt++) {
+        if (af_fmt_to_bytes(fmt) == bytes &&
+            af_fmt_is_float(fmt) == af_fmt_is_float(format) &&
+            af_fmt_is_planar(fmt) == af_fmt_is_planar(format) &&
+            af_fmt_is_spdif(fmt) == af_fmt_is_spdif(format))
+            return fmt;
+    }
+    return 0;
+}
+
+// All formats are considered signed, except explicitly unsigned int formats.
+bool af_fmt_is_unsigned(int format)
+{
+    return format == AF_FORMAT_U8 || format == AF_FORMAT_U8P;
+}
+
+bool af_fmt_is_float(int format)
+{
+    format = af_fmt_from_planar(format);
+    return format == AF_FORMAT_FLOAT || format == AF_FORMAT_DOUBLE;
+}
+
+// true for both unsigned and signed ints
+bool af_fmt_is_int(int format)
+{
+    return format && !af_fmt_is_spdif(format) && !af_fmt_is_float(format);
+}
+
+// false for interleaved and AF_FORMAT_UNKNOWN
+bool af_fmt_is_planar(int format)
+{
+    return format && af_fmt_to_planar(format) == format;
+}
+
+bool af_fmt_is_spdif(int format)
+{
+    return af_format_sample_alignment(format) > 1;
+}
+
+bool af_fmt_is_pcm(int format)
+{
+    return af_fmt_is_valid(format) && !af_fmt_is_spdif(format);
 }

 static const int planar_formats[][2] = {
@@ -99,58 +125,40 @@ int af_fmt_from_planar(int format)
    return format;
 }

-const struct af_fmt_entry af_fmtstr_table[] = {
-    {"u8",          AF_FORMAT_U8},
-    {"s8",          AF_FORMAT_S8},
-    {"u16",         AF_FORMAT_U16},
-    {"s16",         AF_FORMAT_S16},
-    {"u24",         AF_FORMAT_U24},
-    {"s24",         AF_FORMAT_S24},
-    {"u32",         AF_FORMAT_U32},
-    {"s32",         AF_FORMAT_S32},
-    {"float",       AF_FORMAT_FLOAT},
-    {"double",      AF_FORMAT_DOUBLE},
-
-    {"u8p",         AF_FORMAT_U8P},
-    {"s16p",        AF_FORMAT_S16P},
-    {"s32p",        AF_FORMAT_S32P},
-    {"floatp",      AF_FORMAT_FLOATP},
-    {"doublep",     AF_FORMAT_DOUBLEP},
-
-    {"spdif-aac",   AF_FORMAT_S_AAC},
-    {"spdif-ac3",   AF_FORMAT_S_AC3},
-    {"spdif-dts",   AF_FORMAT_S_DTS},
-    {"spdif-dtshd", AF_FORMAT_S_DTSHD},
-    {"spdif-eac3",  AF_FORMAT_S_EAC3},
-    {"spdif-mp3",   AF_FORMAT_S_MP3},
-    {"spdif-truehd",AF_FORMAT_S_TRUEHD},
-
-    {0}
-};
-
 bool af_fmt_is_valid(int format)
 {
-    for (int i = 0; af_fmtstr_table[i].name; i++) {
-        if (af_fmtstr_table[i].format == format)
-            return true;
-    }
-    return false;
+    return format > 0 && format < AF_FORMAT_COUNT;
 }

 const char *af_fmt_to_str(int format)
 {
-    for (int i = 0; af_fmtstr_table[i].name; i++) {
-        if (af_fmtstr_table[i].format == format)
-            return af_fmtstr_table[i].name;
+    switch (format) {
+    case AF_FORMAT_U8:          return "u8";
+    case AF_FORMAT_S16:         return "s16";
+    case AF_FORMAT_S24:         return "s24";
+    case AF_FORMAT_S32:         return "s32";
+    case AF_FORMAT_FLOAT:       return "float";
+    case AF_FORMAT_DOUBLE:      return "double";
+    case AF_FORMAT_U8P:         return "u8p";
+    case AF_FORMAT_S16P:        return "s16p";
+    case AF_FORMAT_S32P:        return "s32p";
+    case AF_FORMAT_FLOATP:      return "floatp";
+    case AF_FORMAT_DOUBLEP:     return "doublep";
+    case AF_FORMAT_S_AAC:       return "spdif-aac";
+    case AF_FORMAT_S_AC3:       return "spdif-ac3";
+    case AF_FORMAT_S_DTS:       return "spdif-dts";
+    case AF_FORMAT_S_DTSHD:     return "spdif-dtshd";
+    case AF_FORMAT_S_EAC3:      return "spdif-eac3";
+    case AF_FORMAT_S_MP3:       return "spdif-mp3";
+    case AF_FORMAT_S_TRUEHD:    return "spdif-truehd";
    }
-
    return "??";
 }

 int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate)
 {
-    assert(!AF_FORMAT_IS_PLANAR(format));
-    int bps      = af_fmt2bps(format);
+    assert(!af_fmt_is_planar(format));
+    int bps      = af_fmt_to_bytes(format);
    int framelen = channels * bps;
    int bytes    = seconds  * bps * samplerate;
    if (bytes % framelen)
@@ -158,64 +166,85 @@ int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int sampler
    return bytes;
 }

-int af_str2fmt_short(bstr str)
-{
-    for (int i = 0; af_fmtstr_table[i].name; i++) {
-        if (!bstrcasecmp0(str, af_fmtstr_table[i].name))
-            return af_fmtstr_table[i].format;
-    }
-    return 0;
-}
-
 void af_fill_silence(void *dst, size_t bytes, int format)
 {
-    bool us = (format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_US;
-    memset(dst, us ? 0x80 : 0, bytes);
+    memset(dst, af_fmt_is_unsigned(format) ? 0x80 : 0, bytes);
 }

-#define FMT_DIFF(type, a, b) (((a) & type) - ((b) & type))
-
 // Returns a "score" that serves as heuristic how lossy or hard a conversion is.
 // If the formats are equal, 1024 is returned. If they are gravely incompatible
 // (like s16<->ac3), INT_MIN is returned. If there is implied loss of precision
 // (like s16->s8), a value <0 is returned.
-int af_format_conversion_score(int dst_format, int src_format)
+static int af_format_conversion_score(int dst_format, int src_format)
 {
    if (dst_format == AF_FORMAT_UNKNOWN || src_format == AF_FORMAT_UNKNOWN)
        return INT_MIN;
    if (dst_format == src_format)
        return 1024;
    // Can't be normally converted
-    if (AF_FORMAT_IS_SPECIAL(dst_format) || AF_FORMAT_IS_SPECIAL(src_format))
+    if (!af_fmt_is_pcm(dst_format) || !af_fmt_is_pcm(src_format))
        return INT_MIN;
    int score = 1024;
-    if (FMT_DIFF(AF_FORMAT_INTERLEAVING_MASK, dst_format, src_format))
+    if (af_fmt_is_planar(dst_format) != af_fmt_is_planar(src_format))
        score -= 1;     // has to (de-)planarize
-    if (FMT_DIFF(AF_FORMAT_SIGN_MASK, dst_format, src_format))
-        score -= 4;     // has to swap sign
-    if (FMT_DIFF(AF_FORMAT_TYPE_MASK, dst_format, src_format)) {
-        int dst_bits = dst_format & AF_FORMAT_BITS_MASK;
-        if ((dst_format & AF_FORMAT_TYPE_MASK) == AF_FORMAT_F) {
-            // For int->float, always prefer 32 bit float.
-            score -= dst_bits == AF_FORMAT_32BIT ? 8 : 0;
+    if (af_fmt_is_float(dst_format) != af_fmt_is_float(src_format)) {
+        int dst_bytes = af_fmt_to_bytes(dst_format);
+        if (af_fmt_is_float(dst_format)) {
+            // For int->float, consider a lower bound on the precision difference.
+            int bytes = (dst_bytes == 4 ? 3 : 6) - af_fmt_to_bytes(src_format);
+            if (bytes >= 0) {
+                score -= 8 * bytes;          // excess precision
+            } else {
+                score += 1024 * (bytes - 1); // precision is lost (i.e. s32 -> float)
+            }
        } else {
-            // For float->int, always prefer highest bit depth int
-            score -= 8 * (AF_FORMAT_64BIT - dst_bits);
+            // float->int is the worst case. Penalize heavily and
+            // prefer highest bit depth int.
+            score -= 1048576 * (8 - dst_bytes);
        }
+        score -= 512; // penalty for any float <-> int conversion
    } else {
-        int bits = FMT_DIFF(AF_FORMAT_BITS_MASK, dst_format, src_format);
-        if (bits > 0) {
-            score -= 8 * bits;          // has to add padding
-        } else if (bits < 0) {
-            score -= 1024 - 8 * bits;   // has to reduce bit depth
+        int bytes = af_fmt_to_bytes(dst_format) - af_fmt_to_bytes(src_format);
+        if (bytes > 0) {
+            score -= 8 * bytes;          // has to add padding
+        } else if (bytes < 0) {
+            score += 1024 * (bytes - 1); // has to reduce bit depth
        }
    }
-    // Consider this the worst case.
-    if (FMT_DIFF(AF_FORMAT_TYPE_MASK, dst_format, src_format))
-        score -= 2048;  // has to convert float<->int
    return score;
 }

+struct entry {
+    int fmt;
+    int score;
+};
+
+static int cmp_entry(const void *a, const void *b)
+{
+#define CMP_INT(a, b) (a > b ? 1 : (a < b ? -1 : 0))
+    return -CMP_INT(((struct entry *)a)->score, ((struct entry *)b)->score);
+}
+
+// Return a list of sample format compatible to src_format, sorted by order
+// of preference. out_formats[0] will be src_format (as long as it's valid),
+// and the list is terminated with 0 (AF_FORMAT_UNKNOWN).
+// Keep in mind that this also returns formats with flipped interleaving
+// (e.g. for s16, it returns [s16, s16p, ...]).
+void af_get_best_sample_formats(int src_format, int out_formats[AF_FORMAT_COUNT])
+{
+    int num = 0;
+    struct entry e[AF_FORMAT_COUNT];
+    for (int fmt = 1; fmt < AF_FORMAT_COUNT; fmt++) {
+        int score = af_format_conversion_score(fmt, src_format);
+        if (score > INT_MIN)
+            e[num++] = (struct entry){fmt, score};
+    }
+    qsort(e, num, sizeof(e[0]), cmp_entry);
+    for (int n = 0; n < num; n++)
+        out_formats[n] = e[n].fmt;
+    out_formats[num] = 0;
+}
+
 // Return the number of samples that make up one frame in this format.
 // You get the byte size by multiplying them with sample size and channel count.
 int af_format_sample_alignment(int format)
@@ -22,96 +22,50 @@
 #ifndef MPLAYER_AF_FORMAT_H
 #define MPLAYER_AF_FORMAT_H

+#include <stddef.h>
 #include <stdbool.h>

-#include "misc/bstr.h"
-
-// Signed/unsigned
-#define AF_FORMAT_SI            (0<<0) // Signed
-#define AF_FORMAT_US            (1<<0) // Unsigned
-#define AF_FORMAT_SIGN_MASK     (1<<0)
-
-// Bits used
-// Some code assumes they're sorted by size.
-#define AF_FORMAT_8BIT          (0<<1)
-#define AF_FORMAT_16BIT         (1<<1)
-#define AF_FORMAT_24BIT         (2<<1)
-#define AF_FORMAT_32BIT         (3<<1)
-#define AF_FORMAT_64BIT         (4<<1)
-#define AF_FORMAT_BITS_MASK     (7<<1)
-
-// Fixed/floating point/special
-#define AF_FORMAT_I             (1<<4) // Int
-#define AF_FORMAT_F             (2<<4) // Foating point
-#define AF_FORMAT_S             (4<<4) // special (IEC61937)
-#define AF_FORMAT_TYPE_MASK     (7<<4)
-
-// Interleaving (planar formats have data for each channel in separate planes)
-#define AF_FORMAT_INTERLEAVED        (0<<7) // must be 0
-#define AF_FORMAT_PLANAR             (1<<7)
-#define AF_FORMAT_INTERLEAVING_MASK  (1<<7)
-
-#define AF_FORMAT_S_CODEC(n)    ((n)<<8)
-#define AF_FORMAT_S_CODEC_MASK  (15 <<8) // 16 codecs max.
-
-#define AF_FORMAT_MASK          ((1<<12)-1)
-
-#define AF_INTP (AF_FORMAT_I|AF_FORMAT_PLANAR)
-#define AF_FLTP (AF_FORMAT_F|AF_FORMAT_PLANAR)
-#define AF_FORMAT_S_(n) (AF_FORMAT_S_CODEC(n)|AF_FORMAT_S|AF_FORMAT_16BIT)
-
-// actual sample formats
 enum af_format {
-    AF_FORMAT_UNKNOWN   = 0,
+    AF_FORMAT_UNKNOWN = 0,

-    AF_FORMAT_U8        = (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_8BIT),
-    AF_FORMAT_S8        = (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_8BIT),
-    AF_FORMAT_U16       = (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_16BIT),
-    AF_FORMAT_S16       = (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_16BIT),
-    AF_FORMAT_U24       = (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_24BIT),
-    AF_FORMAT_S24       = (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_24BIT),
-    AF_FORMAT_U32       = (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_32BIT),
-    AF_FORMAT_S32       = (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_32BIT),
-
-    AF_FORMAT_FLOAT     = (AF_FORMAT_F|AF_FORMAT_32BIT),
-    AF_FORMAT_DOUBLE    = (AF_FORMAT_F|AF_FORMAT_64BIT),
+    AF_FORMAT_U8,
+    AF_FORMAT_S16,
+    AF_FORMAT_S24,
+    AF_FORMAT_S32,
+    AF_FORMAT_FLOAT,
+    AF_FORMAT_DOUBLE,

    // Planar variants
-    AF_FORMAT_U8P       = (AF_INTP|AF_FORMAT_US|AF_FORMAT_8BIT),
-    AF_FORMAT_S16P      = (AF_INTP|AF_FORMAT_SI|AF_FORMAT_16BIT),
-    AF_FORMAT_S32P      = (AF_INTP|AF_FORMAT_SI|AF_FORMAT_32BIT),
-    AF_FORMAT_FLOATP    = (AF_FLTP|AF_FORMAT_32BIT),
-    AF_FORMAT_DOUBLEP   = (AF_FLTP|AF_FORMAT_64BIT),
+    AF_FORMAT_U8P,
+    AF_FORMAT_S16P,
+    AF_FORMAT_S32P,
+    AF_FORMAT_FLOATP,
+    AF_FORMAT_DOUBLEP,

    // All of these use IEC61937 framing, and otherwise pretend to be like PCM.
-    AF_FORMAT_S_AAC     = AF_FORMAT_S_(0),
-    AF_FORMAT_S_AC3     = AF_FORMAT_S_(1),
-    AF_FORMAT_S_DTS     = AF_FORMAT_S_(2),
-    AF_FORMAT_S_DTSHD   = AF_FORMAT_S_(3),
-    AF_FORMAT_S_EAC3    = AF_FORMAT_S_(4),
-    AF_FORMAT_S_MP3     = AF_FORMAT_S_(5),
-    AF_FORMAT_S_TRUEHD  = AF_FORMAT_S_(6),
+    AF_FORMAT_S_AAC,
+    AF_FORMAT_S_AC3,
+    AF_FORMAT_S_DTS,
+    AF_FORMAT_S_DTSHD,
+    AF_FORMAT_S_EAC3,
+    AF_FORMAT_S_MP3,
+    AF_FORMAT_S_TRUEHD,
+
+    AF_FORMAT_COUNT
 };

-#define AF_FORMAT_IS_IEC61937(f) (((f) & AF_FORMAT_TYPE_MASK) == AF_FORMAT_S)
-#define AF_FORMAT_IS_SPECIAL(f) AF_FORMAT_IS_IEC61937(f)
-#define AF_FORMAT_IS_FLOAT(f) (!!((f) & AF_FORMAT_F))
-// false for interleaved and AF_FORMAT_UNKNOWN
-#define AF_FORMAT_IS_PLANAR(f) (!!((f) & AF_FORMAT_PLANAR))
-
-struct af_fmt_entry {
-    const char *name;
-    int format;
-};
-
-extern const struct af_fmt_entry af_fmtstr_table[];
-
-int af_str2fmt_short(bstr str);
 const char *af_fmt_to_str(int format);

-int af_fmt2bps(int format);
-int af_fmt2bits(int format);
-int af_fmt_change_bits(int format, int bits);
+int af_fmt_to_bytes(int format);
+int af_fmt_change_bytes(int format, int bytes);
+
+bool af_fmt_is_valid(int format);
+bool af_fmt_is_unsigned(int format);
+bool af_fmt_is_float(int format);
+bool af_fmt_is_int(int format);
+bool af_fmt_is_planar(int format);
+bool af_fmt_is_spdif(int format);
+bool af_fmt_is_pcm(int format);

 int af_fmt_to_planar(int format);
 int af_fmt_from_planar(int format);
@@ -119,11 +73,9 @@ int af_fmt_from_planar(int format);
 // Amount of bytes that contain audio of the given duration, aligned to frames.
 int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate);

-bool af_fmt_is_valid(int format);
-
 void af_fill_silence(void *dst, size_t bytes, int format);

-int af_format_conversion_score(int dst_format, int src_format);
+void af_get_best_sample_formats(int src_format, int out_formats[AF_FORMAT_COUNT]);

 int af_format_sample_alignment(int format);

@@ -69,10 +69,10 @@ bool mixer_audio_initialized(struct mixer *mixer)
    return !!mixer->ao;
 }

-float mixer_getneutralvolume(struct mixer *mixer)
+float mixer_getmaxvolume(struct mixer *mixer)
 {
    // gain == 1
-    return mixer->softvol ? 1.0 / mixer->opts->softvol_max * 100.0 * 100.0 : 100;
+    return mixer->softvol ? mixer->opts->softvol_max : 100;
 }

 static void checkvolume(struct mixer *mixer)
@@ -85,8 +85,8 @@ static void checkvolume(struct mixer *mixer)
        float gain;
        if (!af_control_any_rev(mixer->af, AF_CONTROL_GET_VOLUME, &gain))
            gain = 1.0;
-        vol.left = (gain / (mixer->opts->softvol_max / 100.0)) * 100.0;
-        vol.right = (gain / (mixer->opts->softvol_max / 100.0)) * 100.0;
+        vol.left = gain * 100.0;
+        vol.right = gain * 100.0;
    } else {
        MP_DBG(mixer, "Reading volume from AO.\n");
        // Rely on the values not changing if the query is not supported
@@ -122,14 +122,14 @@ void mixer_getvolume(struct mixer *mixer, float *l, float *r)

 static void setvolume_internal(struct mixer *mixer, float l, float r)
 {
-    struct ao_control_vol vol = {.left = l, .right = r};
    if (!mixer->softvol) {
        MP_DBG(mixer, "Setting volume on AO.\n");
+        struct ao_control_vol vol = {.left = l, .right = r};
        if (ao_control(mixer->ao, AOCONTROL_SET_VOLUME, &vol) != CONTROL_OK)
            MP_ERR(mixer, "Failed to change audio output volume.\n");
        return;
    }
-    float gain = (l + r) / 2.0 / 100.0 * mixer->opts->softvol_max / 100.0;
+    float gain = (l + r) / 2.0 / 100.0;
    if (!af_control_any_rev(mixer->af, AF_CONTROL_SET_VOLUME, &gain)) {
        if (gain == 1.0)
            return;
@@ -144,8 +144,9 @@ void mixer_setvolume(struct mixer *mixer, float l, float r)
 {
    checkvolume(mixer);  // to check mute status

-    mixer->vol_l = av_clipf(l, 0, 100);
-    mixer->vol_r = av_clipf(r, 0, 100);
+    float max = mixer_getmaxvolume(mixer);
+    mixer->vol_l = MPCLAMP(l, 0, max);
+    mixer->vol_r = MPCLAMP(r, 0, max);
    if (mixer->ao && !(mixer->emulate_mute && mixer->muted))
        setvolume_internal(mixer, mixer->vol_l, mixer->vol_r);
 }
@@ -226,7 +227,7 @@ void mixer_setbalance(struct mixer *mixer, float val)
        return;
    }

-    /* make all other channels pass thru since by default pan blocks all */
+    /* make all other channels pass through since by default pan blocks all */
    for (int i = 2; i < AF_NCH; i++) {
        float level[AF_NCH] = {0};
        level[i] = 1.f;
@@ -242,9 +243,8 @@ char *mixer_get_volume_restore_data(struct mixer *mixer)
 {
    if (!mixer->driver[0])
        return NULL;
-    return talloc_asprintf(NULL, "%s:%f:%f:%d:%f", mixer->driver, mixer->vol_l,
-                           mixer->vol_r, mixer->muted_by_us,
-                           mixer->opts->softvol_max);
+    return talloc_asprintf(NULL, "%s:%f:%f:%d", mixer->driver, mixer->vol_l,
+                           mixer->vol_r, mixer->muted_by_us);
 }

 static void probe_softvol(struct mixer *mixer)
@@ -315,11 +315,10 @@ static void restore_volume(struct mixer *mixer)
    char *data = mixer->opts->mixer_restore_volume_data;
    if (!mixer->persistent_volume && data && data[0]) {
        char drv[40];
-        float v_l, v_r, s;
+        float v_l, v_r;
        int m;
-        if (sscanf(data, "%39[^:]:%f:%f:%d:%f", drv, &v_l, &v_r, &m, &s) == 5) {
-            float diff = fabs(mixer->opts->softvol_max - s);
-            if (strcmp(mixer->driver, drv) == 0 && diff < 0.01) {
+        if (sscanf(data, "%39[^:]:%f:%f:%d", drv, &v_l, &v_r, &m) == 4) {
+            if (strcmp(mixer->driver, drv) == 0) {
                force_vol_l = v_l;
                force_vol_r = v_r;
                force_mute = !!m;
@@ -44,7 +44,7 @@ void mixer_setmute(struct mixer *mixer, bool mute);
 bool mixer_getmute(struct mixer *mixer);
 void mixer_getbalance(struct mixer *mixer, float *bal);
 void mixer_setbalance(struct mixer *mixer, float bal);
-float mixer_getneutralvolume(struct mixer *mixer);
+float mixer_getmaxvolume(struct mixer *mixer);
 char *mixer_get_volume_restore_data(struct mixer *mixer);

 #endif /* MPLAYER_MIXER_H */
@@ -191,14 +191,14 @@ static struct ao *ao_init(bool probing, struct mpv_global *global,
            snprintf(rdevice, sizeof(rdevice), "%s", ao->device ? ao->device : "");
            talloc_free(ao);
            return ao_init(probing, global, input_ctx, encode_lavc_ctx,
-                           samplerate, format, channels, rdevice, redirect, args);
+                           samplerate, format, channels, rdevice, redirect, NULL);
        }
        goto fail;
    }

-    ao->sstride = af_fmt2bps(ao->format);
+    ao->sstride = af_fmt_to_bytes(ao->format);
    ao->num_planes = 1;
-    if (AF_FORMAT_IS_PLANAR(ao->format)) {
+    if (af_fmt_is_planar(ao->format)) {
        ao->num_planes = ao->channels.num;
    } else {
        ao->sstride *= ao->channels.num;
@@ -307,7 +307,8 @@ done:
 // Uninitialize and destroy the AO. Remaining audio must be dropped.
 void ao_uninit(struct ao *ao)
 {
-    ao->api->uninit(ao);
+    if (ao)
+        ao->api->uninit(ao);
    talloc_free(ao);
 }

@@ -385,28 +386,26 @@ bool ao_eof_reached(struct ao *ao)
 // Query the AO_EVENT_*s as requested by the events parameter, and return them.
 int ao_query_and_reset_events(struct ao *ao, int events)
 {
-    int actual_events = 0;
-    if (atomic_load(&ao->request_reload)) // don't need to reset it
-        actual_events |= AO_EVENT_RELOAD;
-    if (atomic_load(&ao->request_hotplug))
-        actual_events |= AO_EVENT_HOTPLUG;
-    return actual_events & events;
+    return atomic_fetch_and(&ao->events_, ~(unsigned)events) & events;
+}
+
+static void ao_add_events(struct ao *ao, int events)
+{
+    atomic_fetch_or(&ao->events_, events);
+    if (ao->input_ctx)
+        mp_input_wakeup(ao->input_ctx);
 }

 // Request that the player core destroys and recreates the AO. Fully thread-safe.
 void ao_request_reload(struct ao *ao)
 {
-    atomic_store(&ao->request_reload, true);
-    if (ao->input_ctx)
-        mp_input_wakeup(ao->input_ctx);
+    ao_add_events(ao, AO_EVENT_RELOAD);
 }

 // Notify the player that the device list changed. Fully thread-safe.
 void ao_hotplug_event(struct ao *ao)
 {
-    atomic_store(&ao->request_hotplug, true);
-    if (ao->input_ctx)
-        mp_input_wakeup(ao->input_ctx);
+    ao_add_events(ao, AO_EVENT_HOTPLUG);
 }

 bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
@@ -415,12 +414,15 @@ bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
    if (mp_msg_test(ao->log, MSGL_DEBUG)) {
        for (int i = 0; i < s->num_chmaps; i++) {
            struct mp_chmap c = s->chmaps[i];
-            MP_DBG(ao, "chmap_sel #%d: %s\n", i, mp_chmap_to_str(&c));
+            MP_DBG(ao, "chmap_sel #%d: %s (%s)\n", i, mp_chmap_to_str(&c),
+                   mp_chmap_to_str_hr(&c));
        }
    }
    bool r = mp_chmap_sel_adjust(s, map);
-    if (r)
-        MP_DBG(ao, "result: %s\n", mp_chmap_to_str(map));
+    if (r) {
+        MP_DBG(ao, "result: %s (%s)\n", mp_chmap_to_str(map),
+               mp_chmap_to_str_hr(map));
+    }
    return r;
 }

@@ -496,7 +498,6 @@ bool ao_hotplug_check_update(struct ao_hotplug *hp)
 {
    if (hp->ao && ao_query_and_reset_events(hp->ao, AO_EVENT_HOTPLUG)) {
        hp->needs_update = true;
-        atomic_store(&hp->ao->request_hotplug, false);
        return true;
    }
    return false;
@@ -101,7 +101,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
        long get_vol, set_vol;
        float f_multi;

-        if (AF_FORMAT_IS_SPECIAL(ao->format))
+        if (!af_fmt_is_pcm(ao->format))
            return CONTROL_FALSE;

        snd_mixer_selem_id_alloca(&sid);
@@ -204,22 +204,26 @@ alsa_error:
 }

 static const int mp_to_alsa_format[][2] = {
-    {AF_FORMAT_S8,          SND_PCM_FORMAT_S8},
    {AF_FORMAT_U8,          SND_PCM_FORMAT_U8},
-    {AF_FORMAT_U16,         SND_PCM_FORMAT_U16},
    {AF_FORMAT_S16,         SND_PCM_FORMAT_S16},
-    {AF_FORMAT_U32,         SND_PCM_FORMAT_U32},
    {AF_FORMAT_S32,         SND_PCM_FORMAT_S32},
-    {AF_FORMAT_U24,
-            MP_SELECT_LE_BE(SND_PCM_FORMAT_U24_3LE, SND_PCM_FORMAT_U24_3BE)},
    {AF_FORMAT_S24,
            MP_SELECT_LE_BE(SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S24_3BE)},
    {AF_FORMAT_FLOAT,       SND_PCM_FORMAT_FLOAT},
+    {AF_FORMAT_DOUBLE,      SND_PCM_FORMAT_FLOAT64},
    {AF_FORMAT_UNKNOWN,     SND_PCM_FORMAT_UNKNOWN},
 };

 static int find_alsa_format(int af_format)
 {
+    if (af_fmt_is_spdif(af_format)) {
+        if (af_format == AF_FORMAT_S_MP3) {
+            return SND_PCM_FORMAT_MPEG;
+        } else {
+            return SND_PCM_FORMAT_S16;
+        }
+    }
+
    af_format = af_fmt_from_planar(af_format);
    for (int n = 0; mp_to_alsa_format[n][0] != AF_FORMAT_UNKNOWN; n++) {
        if (mp_to_alsa_format[n][0] == af_format)
@@ -251,7 +255,10 @@ static const int alsa_to_mp_channels[][2] = {
    {SND_CHMAP_TRL,     MP_SP(TBL)},
    {SND_CHMAP_TRR,     MP_SP(TBR)},
    {SND_CHMAP_TRC,     MP_SP(TBC)},
+    {SND_CHMAP_RRC,     MP_SP(SDR)},
+    {SND_CHMAP_RLC,     MP_SP(SDL)},
    {SND_CHMAP_MONO,    MP_SP(FC)},
+    {SND_CHMAP_NA,      MP_SPEAKER_ID_NA},
    {SND_CHMAP_LAST,    MP_SPEAKER_ID_COUNT}
 };

@@ -275,6 +282,20 @@ static int find_alsa_channel(int mp_channel)
    return SND_CHMAP_UNKNOWN;
 }

+static int mp_chmap_from_alsa(struct mp_chmap *dst, snd_pcm_chmap_t *src)
+{
+    *dst = (struct mp_chmap) {0};
+
+    if (src->channels > MP_NUM_CHANNELS)
+        return -1;
+
+    dst->num = src->channels;
+    for (int c = 0; c < dst->num; c++)
+        dst->speaker[c] = find_mp_channel(src->pos[c]);
+
+    return 0;
+}
+
 static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
 {
    struct priv *p = ao->priv;
@@ -285,16 +306,12 @@ static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
        return false;

    for (int i = 0; maps[i] != NULL; i++) {
-        if (maps[i]->map.channels > MP_NUM_CHANNELS) {
-            MP_VERBOSE(ao, "skipping ALSA channel map with too many channels.\n");
-            continue;
-        }
-
-        struct mp_chmap entry = {.num = maps[i]->map.channels};
-        for (int c = 0; c < entry.num; c++)
-            entry.speaker[c] = find_mp_channel(maps[i]->map.pos[c]);
+        struct mp_chmap entry;
+        mp_chmap_from_alsa(&entry, &maps[i]->map);

        if (mp_chmap_is_valid(&entry)) {
+            if (maps[i]->type == SND_CHMAP_TYPE_VAR)
+                mp_chmap_reorder_norm(&entry);
            MP_VERBOSE(ao, "Got supported channel map: %s (type %s)\n",
                       mp_chmap_to_str(&entry),
                       snd_pcm_chmap_type_name(maps[i]->type));
@@ -366,8 +383,9 @@ static char *append_params(void *ta_parent, const char *device, const char *p)
 static int try_open_device(struct ao *ao, const char *device)
 {
    struct priv *p = ao->priv;
+    int err;

-    if (AF_FORMAT_IS_IEC61937(ao->format)) {
+    if (af_fmt_is_spdif(ao->format)) {
        void *tmp = talloc_new(NULL);
        char *params = talloc_asprintf(tmp,
                        "AES0=%d,AES1=%d,AES2=0,AES3=%d",
@@ -376,15 +394,29 @@ static int try_open_device(struct ao *ao, const char *device)
                        map_iec958_srate(ao->samplerate));
        const char *ac3_device = append_params(tmp, device, params);
        MP_VERBOSE(ao, "opening device '%s' => '%s'\n", device, ac3_device);
-        int err = snd_pcm_open
-                    (&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, 0);
+        err = snd_pcm_open(&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, 0);
+        if (err < 0) {
+            // Some spdif-capable devices do not accept the AES0 parameter,
+            // and instead require the iec958 pseudo-device (they will play
+            // noise otherwise). Unfortunately, ALSA gives us no way to map
+            // these devices, so try it for the default device only.
+            bstr dev;
+            bstr_split_tok(bstr0(device), ":", &dev, &(bstr){0});
+            if (bstr_equals0(dev, "default")) {
+                ac3_device = append_params(tmp, "iec958", params);
+                MP_VERBOSE(ao, "got error %d; opening iec fallback device '%s'\n",
+                           err, ac3_device);
+                err = snd_pcm_open
+                            (&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, 0);
+            }
+        }
        talloc_free(tmp);
-        if (!err)
-            return 0;
+    } else {
+        MP_VERBOSE(ao, "opening device '%s'\n", device);
+        err = snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, 0);
    }

-    MP_VERBOSE(ao, "opening device '%s'\n", device);
-    return snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, 0);
+    return err;
 }

 static void uninit(struct ao *ao)
@@ -404,14 +436,12 @@ alsa_error: ;
 #define INIT_OK 0
 #define INIT_ERROR -1
 #define INIT_BRAINDEATH -2
-static int init_device(struct ao *ao)
+static int init_device(struct ao *ao, bool second_try)
 {
    struct priv *p = ao->priv;
    int err;

    const char *device = "default";
-    if (AF_FORMAT_IS_IEC61937(ao->format))
-        device = "iec958";
    if (ao->device)
        device = ao->device;
    if (p->cfg_device && p->cfg_device[0])
@@ -432,38 +462,28 @@ static int init_device(struct ao *ao)
    err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
    CHECK_ALSA_ERROR("Unable to get initial parameters");

-    if (AF_FORMAT_IS_IEC61937(ao->format)) {
-        if (ao->format == AF_FORMAT_S_MP3) {
-            p->alsa_fmt = SND_PCM_FORMAT_MPEG;
-        } else {
-            p->alsa_fmt = SND_PCM_FORMAT_S16;
-        }
-    } else {
+    int try_formats[AF_FORMAT_COUNT];
+    af_get_best_sample_formats(ao->format, try_formats);
+    for (int n = 0; try_formats[n]; n++) {
+        ao->format = try_formats[n];
        p->alsa_fmt = find_alsa_format(ao->format);
-    }
-    if (p->alsa_fmt == SND_PCM_FORMAT_UNKNOWN) {
-        p->alsa_fmt = SND_PCM_FORMAT_S16;
-        ao->format = AF_FORMAT_S16;
+        if (snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt) >= 0)
+            break;
    }

-    err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
-    if (err < 0) {
-        if (AF_FORMAT_IS_IEC61937(ao->format))
-            CHECK_ALSA_ERROR("Unable to set IEC61937 format");
-        MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n",
-                af_fmt_to_str(ao->format));
-        p->alsa_fmt = SND_PCM_FORMAT_S16;
-        ao->format = AF_FORMAT_S16;
+    if (!ao->format) {
+        MP_ERR(ao, "Can't find appropriate sample format.\n");
+        goto alsa_error;
    }

    err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
    CHECK_ALSA_ERROR("Unable to set format");

-    snd_pcm_access_t access = AF_FORMAT_IS_PLANAR(ao->format)
+    snd_pcm_access_t access = af_fmt_is_planar(ao->format)
                                    ? SND_PCM_ACCESS_RW_NONINTERLEAVED
                                    : SND_PCM_ACCESS_RW_INTERLEAVED;
    err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access);
-    if (err < 0 && AF_FORMAT_IS_PLANAR(ao->format)) {
+    if (err < 0 && af_fmt_is_planar(ao->format)) {
        ao->format = af_fmt_from_planar(ao->format);
        access = SND_PCM_ACCESS_RW_INTERLEAVED;
        err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access);
@@ -471,8 +491,11 @@ static int init_device(struct ao *ao)
    CHECK_ALSA_ERROR("Unable to set access type");

    struct mp_chmap dev_chmap = ao->channels;
-    if (AF_FORMAT_IS_IEC61937(ao->format) || p->cfg_ignore_chmap) {
+    if (af_fmt_is_spdif(ao->format) || p->cfg_ignore_chmap) {
        dev_chmap.num = 0; // disable chmap API
+    } else if (dev_chmap.num == 1 && dev_chmap.speaker[0] == MP_SPEAKER_ID_FC) {
+        // As yet another ALSA API inconsistency, mono is not reported correctly.
+        dev_chmap.num = 0;
    } else if (query_chmaps(ao, &dev_chmap)) {
        ao->channels = dev_chmap;
    } else {
@@ -492,11 +515,12 @@ static int init_device(struct ao *ao)
    }

    if (num_channels != ao->channels.num) {
+        int req = ao->channels.num;
        mp_chmap_from_channels_alsa(&ao->channels, num_channels);
        if (!mp_chmap_is_valid(&ao->channels))
            mp_chmap_from_channels(&ao->channels, 2);
-        MP_ERR(ao, "Couldn't get requested number of channels, fallback to %s.\n",
-               mp_chmap_to_str(&ao->channels));
+        MP_ERR(ao, "Asked for %d channels, got %d - fallback to %s.\n", req,
+               num_channels, mp_chmap_to_str(&ao->channels));
    }

    // Some ALSA drivers have broken delay reporting, so disable the ALSA
@@ -546,10 +570,11 @@ static int init_device(struct ao *ao)

        err = snd_pcm_set_chmap(p->alsa, alsa_chmap);
        if (err == -ENXIO) {
-            // I consider this an ALSA bug: the channel map was reported as
-            // supported, but we still can't set it. It happens virtually
-            // always with dmix, though.
-            MP_VERBOSE(ao, "Device does not support requested channel map (%s)\n",
+            // A device my not be able to set any channel map, even channel maps
+            // that were reported as supported. This is either because the ALSA
+            // device is broken (dmix), or because the driver has only 1
+            // channel map per channel count, and setting the map is not needed.
+            MP_VERBOSE(ao, "device returned ENXIO when setting channel map %s\n",
                       mp_chmap_to_str(&dev_chmap));
        } else {
            CHECK_ALSA_WARN("Channel map setup failed");
@@ -564,17 +589,45 @@ static int init_device(struct ao *ao)
        if (snd_pcm_chmap_print(alsa_chmap, sizeof(tmp), tmp) > 0)
            MP_VERBOSE(ao, "channel map reported by ALSA: %s\n", tmp);

-        struct mp_chmap chmap = {.num = alsa_chmap->channels};
-        for (int c = 0; c < chmap.num; c++)
-            chmap.speaker[c] = find_mp_channel(alsa_chmap->pos[c]);
+        struct mp_chmap chmap;
+        mp_chmap_from_alsa(&chmap, alsa_chmap);

        MP_VERBOSE(ao, "which we understand as: %s\n", mp_chmap_to_str(&chmap));

        if (p->cfg_ignore_chmap) {
            MP_VERBOSE(ao, "user set ignore-chmap; ignoring the channel map.\n");
-        } else if (AF_FORMAT_IS_IEC61937(ao->format)) {
+        } else if (af_fmt_is_spdif(ao->format)) {
            MP_VERBOSE(ao, "using spdif passthrough; ignoring the channel map.\n");
        } else if (mp_chmap_is_valid(&chmap)) {
+            // Is it one that contains NA channels?
+            struct mp_chmap without_na = chmap;
+            mp_chmap_remove_na(&without_na);
+
+            if (mp_chmap_is_valid(&without_na) &&
+                !mp_chmap_equals(&without_na, &chmap) &&
+                !mp_chmap_equals(&chmap, &ao->channels) &&
+                !second_try)
+            {
+                // Sometimes, ALSA will advertise certain chmaps, but it's not
+                // possible to set them. This can happen with dmix: as of
+                // alsa 1.0.28, dmix can do stereo only, but advertises the
+                // surround chmaps of the underlying device. In this case,
+                // e.g. setting 6 channels will succeed, but requesting  5.1
+                // afterwards will fail. Then it will return something like
+                // "FL FR NA NA NA NA" as channel map. This means we would
+                // have to pad stereo output to 6 channels with silence, which
+                // would require lots of extra processing. You can't change
+                // the number of channels to 2 either, because the hw params
+                // are already set! So just fuck it and reopen the device with
+                // the chmap "cleaned out" of NA entries.
+                MP_VERBOSE(ao, "Working around braindead ALSA behavior.\n");
+                err = snd_pcm_close(p->alsa);
+                p->alsa = NULL;
+                CHECK_ALSA_ERROR("pcm close error");
+                ao->channels = without_na;
+                return INIT_BRAINDEATH;
+            }
+
            if (mp_chmap_equals(&chmap, &ao->channels)) {
                MP_VERBOSE(ao, "which is what we requested.\n");
            } else if (chmap.num == ao->channels.num) {
@@ -584,41 +637,14 @@ static int init_device(struct ao *ao)
                MP_WARN(ao, "ALSA channel map conflicts with channel count!\n");
            }
        } else {
-            // Is it one that contains NA channels?
-            struct mp_chmap chmap2 = {0};
-            for (int c = 0; c < alsa_chmap->channels; c++) {
-                int alsa_ch = alsa_chmap->pos[c];
-                if (alsa_ch != SND_CHMAP_NA)
-                    chmap2.speaker[chmap2.num++] = find_mp_channel(alsa_ch);
-            }
-
-            if (mp_chmap_is_valid(&chmap2)) {
-                // Sometimes, ALSA will advertise certain chmaps, but it's not
-                // possible to set them. This can happen with dmix: as of
-                // alsa 1.0.28, dmix can do stereo only, but advertises the
-                // surround chmaps of the underlying device. In this case,
-                // requesting e.g. 5.1 will fail, but it will still allow
-                // setting 6 channels. Then it will return something like
-                // "FL FR NA NA NA NA" as channel map. This means we would
-                // have to pad stereo output to 6 channels with silence, which
-                // is way too complicated in the general case. You can't change
-                // the number of channels to 2 either, because the hw params
-                // are already set! So just fuck it and reopen the device with
-                // the chmap "cleaned out" of NA entries.
-                MP_VERBOSE(ao, "Working around braindead ALSA behavior.\n");
-                err = snd_pcm_close(p->alsa);
-                p->alsa = NULL;
-                CHECK_ALSA_ERROR("pcm close error");
-                ao->channels = chmap2;
-                return INIT_BRAINDEATH;
-            }
-
            MP_WARN(ao, "Got unknown channel map from ALSA.\n");
        }

        // mpv and ALSA use different conventions for mono
-        if (ao->channels.num == 1)
+        if (ao->channels.num == 1) {
+            MP_VERBOSE(ao, "assuming we actually got MONO from ALSA.\n");
            ao->channels.speaker[0] = MP_SP(FC);
+        }

        free(alsa_chmap);
    }
@@ -683,9 +709,9 @@ static int init(struct ao *ao)

    MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());

-    int r = init_device(ao);
+    int r = init_device(ao, false);
    if (r == INIT_BRAINDEATH)
-        r = init_device(ao); // retry with normalized channel layout
+        r = init_device(ao, true); // retry with normalized channel layout
    return r == INIT_OK ? 0 : -1;
 }

@@ -822,7 +848,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
        return 0;

    do {
-        if (AF_FORMAT_IS_PLANAR(ao->format)) {
+        if (af_fmt_is_planar(ao->format)) {
            res = snd_pcm_writen(p->alsa, data, samples);
        } else {
            res = snd_pcm_writei(p->alsa, data[0], samples);
@@ -896,15 +922,18 @@ static void list_devs(struct ao *ao, struct ao_device_list *list)
        char *name = snd_device_name_get_hint(hints[n], "NAME");
        char *desc = snd_device_name_get_hint(hints[n], "DESC");
        char *io = snd_device_name_get_hint(hints[n], "IOID");
-        if (io && strcmp(io, "Output") != 0)
-            continue;
-        char desc2[1024];
-        snprintf(desc2, sizeof(desc2), "%s", desc ? desc : "");
-        for (int i = 0; desc2[i]; i++) {
-            if (desc2[i] == '\n')
-                desc2[i] = '/';
+        if (io && strcmp(io, "Output") == 0) {
+            char desc2[1024];
+            snprintf(desc2, sizeof(desc2), "%s", desc ? desc : "");
+            for (int i = 0; desc2[i]; i++) {
+                if (desc2[i] == '\n')
+                    desc2[i] = '/';
+            }
+            ao_device_list_add(list, ao, &(struct ao_device_desc){name, desc2});
        }
-        ao_device_list_add(list, ao, &(struct ao_device_desc){name, desc2});
+        free(name);
+        free(desc);
+        free(io);
    }

    snd_device_name_free_hint(hints);
@@ -25,18 +25,22 @@
 #include "options/m_option.h"
 #include "misc/ring.h"
 #include "common/msg.h"
-#include "audio/out/ao_coreaudio_properties.h"
-#include "audio/out/ao_coreaudio_utils.h"
+#include "ao_coreaudio_chmap.h"
+#include "ao_coreaudio_properties.h"
+#include "ao_coreaudio_utils.h"

 struct priv {
    AudioDeviceID device;
    AudioUnit audio_unit;

    uint64_t hw_latency_us;
-};

-bool ca_layout_to_mp_chmap(struct ao *ao, AudioChannelLayout *layout,
-                           struct mp_chmap *chmap);
+    AudioStreamBasicDescription original_asbd;
+    AudioStreamID original_asbd_stream;
+
+    int change_physical_format;
+    int exclusive;
+};

 static int64_t ca_get_hardware_latency(struct ao *ao) {
    struct priv *p = ao->priv;
@@ -52,12 +56,8 @@ static int64_t ca_get_hardware_latency(struct ao *ao) {
            &size);
    CHECK_CA_ERROR("cannot get audio unit latency");

-    uint32_t frames = 0;
-    err = CA_GET_O(p->device, kAudioDevicePropertyLatency, &frames);
-    CHECK_CA_ERROR("cannot get device latency");
-
    uint64_t audiounit_latency_us = audiounit_latency_sec * 1e6;
-    uint64_t device_latency_us    = ca_frames_to_us(ao, frames);
+    uint64_t device_latency_us    = ca_get_device_latency_us(ao, p->device);

    MP_VERBOSE(ao, "audiounit latency [us]: %lld\n", audiounit_latency_us);
    MP_VERBOSE(ao, "device latency [us]: %lld\n", device_latency_us);
@@ -74,12 +74,14 @@ static OSStatus render_cb_lpcm(void *ctx, AudioUnitRenderActionFlags *aflags,
 {
    struct ao *ao   = ctx;
    struct priv *p  = ao->priv;
-    AudioBuffer buf = buffer_list->mBuffers[0];
+    void *planes[MP_NUM_CHANNELS] = {0};
+
+    for (int n = 0; n < ao->num_planes; n++)
+        planes[n] = buffer_list->mBuffers[n].mData;

    int64_t end = mp_time_us();
    end += p->hw_latency_us + ca_get_latency(ts) + ca_frames_to_us(ao, frames);
-
-    ao_read_data(ao, &buf.mData, frames, end);
+    ao_read_data(ao, planes, frames, end);
    return noErr;
 }

@@ -122,8 +124,8 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
    return CONTROL_UNKNOWN;
 }

-static bool init_chmap(struct ao *ao);
 static bool init_audiounit(struct ao *ao, AudioStreamBasicDescription asbd);
+static void init_physical_format(struct ao *ao);

 static bool reinit_device(struct ao *ao) {
    struct priv *p = ao->priv;
@@ -144,8 +146,10 @@ coreaudio_error:

 static int init(struct ao *ao)
 {
-    if (AF_FORMAT_IS_IEC61937(ao->format)) {
-        MP_WARN(ao, "detected IEC61937, redirecting to coreaudio_exclusive\n");
+    struct priv *p = ao->priv;
+
+    if (!af_fmt_is_pcm(ao->format) || p->exclusive) {
+        MP_VERBOSE(ao, "redirecting to coreaudio_exclusive\n");
        ao->redirect = "coreaudio_exclusive";
        return CONTROL_ERROR;
    }
@@ -153,10 +157,11 @@ static int init(struct ao *ao)
    if (!reinit_device(ao))
        goto coreaudio_error;

-    if (!init_chmap(ao))
-        goto coreaudio_error;
+    if (p->change_physical_format)
+        init_physical_format(ao);

-    ao->format = af_fmt_from_planar(ao->format);
+    if (!ca_init_chmap(ao, p->device))
+        goto coreaudio_error;

    AudioStreamBasicDescription asbd;
    ca_fill_asbd(ao, &asbd);
@@ -170,95 +175,71 @@ coreaudio_error:
    return CONTROL_ERROR;
 }

-static AudioChannelLayout* ca_query_layout(struct ao *ao, void *talloc_ctx)
+static void init_physical_format(struct ao *ao)
 {
    struct priv *p = ao->priv;
-    OSStatus err;
-    uint32_t psize;
-    AudioChannelLayout *r = NULL;
+    OSErr err;

-    AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
-        .mSelector = kAudioDevicePropertyPreferredChannelLayout,
-        .mScope    = kAudioDevicePropertyScopeOutput,
-        .mElement  = kAudioObjectPropertyElementWildcard,
-    };
+    AudioStreamBasicDescription asbd;
+    ca_fill_asbd(ao, &asbd);

-    err = AudioObjectGetPropertyDataSize(p->device, &p_addr, 0, NULL, &psize);
-    CHECK_CA_ERROR("could not get device preferred layout (size)");
+    AudioStreamID *streams;
+    size_t n_streams;

-    r = talloc_size(talloc_ctx, psize);
+    err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams,
+                       &streams, &n_streams);
+    CHECK_CA_ERROR("could not get number of streams");

-    err = AudioObjectGetPropertyData(p->device, &p_addr, 0, NULL, &psize, r);
-    CHECK_CA_ERROR("could not get device preferred layout (get)");
+    MP_VERBOSE(ao, "Found %zd substream(s).\n", n_streams);

-coreaudio_error:
-    return r;
-}
+    for (int i = 0; i < n_streams; i++) {
+        AudioStreamRangedDescription *formats;
+        size_t n_formats;

-static AudioChannelLayout* ca_query_stereo_layout(struct ao *ao, void *talloc_ctx)
-{
-    struct priv *p = ao->priv;
-    OSStatus err;
-    const int nch = 2;
-    uint32_t channels[nch];
-    AudioChannelLayout *r = NULL;
+        MP_VERBOSE(ao, "Looking at formats in substream %d...\n", i);

-    AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
-        .mSelector = kAudioDevicePropertyPreferredChannelsForStereo,
-        .mScope    = kAudioDevicePropertyScopeOutput,
-        .mElement  = kAudioObjectPropertyElementWildcard,
-    };
+        err = CA_GET_ARY(streams[i], kAudioStreamPropertyAvailablePhysicalFormats,
+                         &formats, &n_formats);

-    uint32_t psize = sizeof(channels);
-    err = AudioObjectGetPropertyData(p->device, &p_addr, 0, NULL, &psize, channels);
-    CHECK_CA_ERROR("could not get device preferred stereo layout");
+        if (!CHECK_CA_WARN("could not get number of stream formats"))
+            continue; // try next one

-    psize = sizeof(AudioChannelLayout) + nch * sizeof(AudioChannelDescription);
-    r = talloc_zero_size(talloc_ctx, psize);
-    r->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
-    r->mNumberChannelDescriptions = nch;

-    AudioChannelDescription desc = {0};
-    desc.mChannelFlags = kAudioChannelFlags_AllOff;
+        uint32_t direction;
+        err = CA_GET(streams[i], kAudioStreamPropertyDirection, &direction);
+        CHECK_CA_ERROR("could not get stream direction");
+        if (direction != 0) {
+            MP_VERBOSE(ao, "Not an output stream.\n");
+            continue;
+        }

-    for(int i = 0; i < nch; i++) {
-        desc.mChannelLabel = channels[i];
-        r->mChannelDescriptions[i] = desc;
+        AudioStreamBasicDescription best_asbd = {0};
+
+        for (int j = 0; j < n_formats; j++) {
+            AudioStreamBasicDescription *stream_asbd = &formats[j].mFormat;
+
+            ca_print_asbd(ao, "- ", stream_asbd);
+
+            if (!best_asbd.mFormatID || ca_asbd_is_better(&asbd, &best_asbd,
+                                                          stream_asbd))
+                best_asbd = *stream_asbd;
+        }
+
+        if (best_asbd.mFormatID) {
+            p->original_asbd_stream = streams[i];
+            err = CA_GET(p->original_asbd_stream,
+                         kAudioStreamPropertyPhysicalFormat,
+                         &p->original_asbd);
+            CHECK_CA_WARN("could not get current physical stream format");
+
+            if (!ca_change_physical_format_sync(ao, streams[i], best_asbd))
+                p->original_asbd = (AudioStreamBasicDescription){0};
+            break;
+        }
    }

 coreaudio_error:
-    return r;
-}
-
-static bool init_chmap(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-    void *ta_ctx = talloc_new(NULL);
-
-    struct mp_chmap_sel chmap_sel = {.tmp = p};
-    struct mp_chmap chmap = {0};
-
-    AudioChannelLayout *ml = ca_query_layout(ao, ta_ctx);
-    if (ml && ca_layout_to_mp_chmap(ao, ml, &chmap))
-        mp_chmap_sel_add_map(&chmap_sel, &chmap);
-
-    AudioChannelLayout *sl = ca_query_stereo_layout(ao, ta_ctx);
-    if (sl && ca_layout_to_mp_chmap(ao, sl, &chmap))
-        mp_chmap_sel_add_map(&chmap_sel, &chmap);
-
-    talloc_free(ta_ctx);
-
-    if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels)) {
-        MP_ERR(ao, "could not select a suitable channel map among the "
-                   "hardware supported ones. Make sure to configure your "
-                   "output device correctly in 'Audio MIDI Setup.app'\n");
-        goto coreaudio_error;
-    }
-
-    return true;
-
-coreaudio_error:
-    return false;
+    return;
 }

 static bool init_audiounit(struct ao *ao, AudioStreamBasicDescription asbd)
@@ -351,13 +332,23 @@ static void uninit(struct ao *ao)
    AudioOutputUnitStop(p->audio_unit);
    AudioUnitUninitialize(p->audio_unit);
    AudioComponentInstanceDispose(p->audio_unit);
+
+    if (p->original_asbd.mFormatID) {
+        OSStatus err = CA_SET(p->original_asbd_stream,
+                              kAudioStreamPropertyPhysicalFormat,
+                              &p->original_asbd);
+        CHECK_CA_WARN("could not restore physical stream format");
+    }
 }

 static OSStatus hotplug_cb(AudioObjectID id, UInt32 naddr,
                           const AudioObjectPropertyAddress addr[],
-                           void *ctx) {
-    reinit_device(ctx);
-    ao_hotplug_event(ctx);
+                           void *ctx)
+{
+    struct ao *ao = ctx;
+    MP_VERBOSE(ao, "Handling potential hotplug event...\n");
+    reinit_device(ao);
+    ao_hotplug_event(ao);
    return noErr;
 }

@@ -381,8 +372,8 @@ static int hotplug_init(struct ao *ao)
        err = AudioObjectAddPropertyListener(
            kAudioObjectSystemObject, &addr, hotplug_cb, (void *)ao);
        if (err != noErr) {
-            char *c1 = fourcc_repr(ao, hotplug_properties[i]);
-            char *c2 = fourcc_repr(ao, err);
+            char *c1 = fourcc_repr(hotplug_properties[i]);
+            char *c2 = fourcc_repr(err);
            MP_ERR(ao, "failed to set device listener %s (%s)", c1, c2);
            goto coreaudio_error;
        }
@@ -406,163 +397,13 @@ static void hotplug_uninit(struct ao *ao)
        err = AudioObjectRemovePropertyListener(
            kAudioObjectSystemObject, &addr, hotplug_cb, (void *)ao);
        if (err != noErr) {
-            char *c1 = fourcc_repr(ao, hotplug_properties[i]);
-            char *c2 = fourcc_repr(ao, err);
+            char *c1 = fourcc_repr(hotplug_properties[i]);
+            char *c2 = fourcc_repr(err);
            MP_ERR(ao, "failed to set device listener %s (%s)", c1, c2);
        }
    }
 }

-
-// Channel Mapping functions
-static const int speaker_map[][2] = {
-    { kAudioChannelLabel_Left,                 MP_SPEAKER_ID_FL   },
-    { kAudioChannelLabel_Right,                MP_SPEAKER_ID_FR   },
-    { kAudioChannelLabel_Center,               MP_SPEAKER_ID_FC   },
-    { kAudioChannelLabel_LFEScreen,            MP_SPEAKER_ID_LFE  },
-    { kAudioChannelLabel_LeftSurround,         MP_SPEAKER_ID_BL   },
-    { kAudioChannelLabel_RightSurround,        MP_SPEAKER_ID_BR   },
-    { kAudioChannelLabel_LeftCenter,           MP_SPEAKER_ID_FLC  },
-    { kAudioChannelLabel_RightCenter,          MP_SPEAKER_ID_FRC  },
-    { kAudioChannelLabel_CenterSurround,       MP_SPEAKER_ID_BC   },
-    { kAudioChannelLabel_LeftSurroundDirect,   MP_SPEAKER_ID_SL   },
-    { kAudioChannelLabel_RightSurroundDirect,  MP_SPEAKER_ID_SR   },
-    { kAudioChannelLabel_TopCenterSurround,    MP_SPEAKER_ID_TC   },
-    { kAudioChannelLabel_VerticalHeightLeft,   MP_SPEAKER_ID_TFL  },
-    { kAudioChannelLabel_VerticalHeightCenter, MP_SPEAKER_ID_TFC  },
-    { kAudioChannelLabel_VerticalHeightRight,  MP_SPEAKER_ID_TFR  },
-    { kAudioChannelLabel_TopBackLeft,          MP_SPEAKER_ID_TBL  },
-    { kAudioChannelLabel_TopBackCenter,        MP_SPEAKER_ID_TBC  },
-    { kAudioChannelLabel_TopBackRight,         MP_SPEAKER_ID_TBR  },
-
-    // unofficial extensions
-    { kAudioChannelLabel_RearSurroundLeft,     MP_SPEAKER_ID_SDL  },
-    { kAudioChannelLabel_RearSurroundRight,    MP_SPEAKER_ID_SDR  },
-    { kAudioChannelLabel_LeftWide,             MP_SPEAKER_ID_WL   },
-    { kAudioChannelLabel_RightWide,            MP_SPEAKER_ID_WR   },
-    { kAudioChannelLabel_LFE2,                 MP_SPEAKER_ID_LFE2 },
-
-    { kAudioChannelLabel_HeadphonesLeft,       MP_SPEAKER_ID_DL   },
-    { kAudioChannelLabel_HeadphonesRight,      MP_SPEAKER_ID_DR   },
-
-    { kAudioChannelLabel_Unknown,              MP_SPEAKER_ID_UNKNOWN0 },
-};
-
-static int ca_label_to_mp_speaker_id(AudioChannelLabel label)
-{
-    for (int i = 0; speaker_map[i][1] >= 0; i++)
-        if (speaker_map[i][0] == label)
-            return speaker_map[i][1];
-    return -1;
-}
-
-static void ca_log_layout(struct ao *ao, int l, AudioChannelLayout *layout)
-{
-    if (!mp_msg_test(ao->log, l))
-        return;
-
-    AudioChannelDescription *descs = layout->mChannelDescriptions;
-
-    mp_msg(ao->log, l, "layout: tag: <%u>, bitmap: <%u>, "
-                       "descriptions <%u>\n",
-                       (unsigned) layout->mChannelLayoutTag,
-                       (unsigned) layout->mChannelBitmap,
-                       (unsigned) layout->mNumberChannelDescriptions);
-
-    for (int i = 0; i < layout->mNumberChannelDescriptions; i++) {
-        AudioChannelDescription d = descs[i];
-        mp_msg(ao->log, l, " - description %d: label <%u, %u>, "
-            " flags: <%u>, coords: <%f, %f, %f>\n", i,
-            (unsigned) d.mChannelLabel,
-            (unsigned) ca_label_to_mp_speaker_id(d.mChannelLabel),
-            (unsigned) d.mChannelFlags,
-            d.mCoordinates[0],
-            d.mCoordinates[1],
-            d.mCoordinates[2]);
-    }
-}
-
-static AudioChannelLayout *ca_layout_to_custom_layout(
-    struct ao *ao, void *talloc_ctx, AudioChannelLayout *l)
-{
-    AudioChannelLayoutTag tag = l->mChannelLayoutTag;
-    AudioChannelLayout *r;
-    OSStatus err;
-
-    if (tag == kAudioChannelLayoutTag_UseChannelBitmap) {
-        uint32_t psize;
-        err = AudioFormatGetPropertyInfo(
-            kAudioFormatProperty_ChannelLayoutForBitmap,
-            sizeof(uint32_t), &l->mChannelBitmap, &psize);
-        CHECK_CA_ERROR("failed to convert channel bitmap to descriptions (info)");
-        r = talloc_size(NULL, psize);
-        err = AudioFormatGetProperty(
-            kAudioFormatProperty_ChannelLayoutForBitmap,
-            sizeof(uint32_t), &l->mChannelBitmap, &psize, r);
-        CHECK_CA_ERROR("failed to convert channel bitmap to descriptions (get)");
-    } else if (tag != kAudioChannelLayoutTag_UseChannelDescriptions) {
-        uint32_t psize;
-        err = AudioFormatGetPropertyInfo(
-            kAudioFormatProperty_ChannelLayoutForTag,
-            sizeof(AudioChannelLayoutTag), &l->mChannelLayoutTag, &psize);
-        r = talloc_size(NULL, psize);
-        CHECK_CA_ERROR("failed to convert channel tag to descriptions (info)");
-        err = AudioFormatGetProperty(
-            kAudioFormatProperty_ChannelLayoutForTag,
-            sizeof(AudioChannelLayoutTag), &l->mChannelLayoutTag, &psize, r);
-        CHECK_CA_ERROR("failed to convert channel tag to descriptions (get)");
-    } else {
-        r = l;
-    }
-
-    return r;
-coreaudio_error:
-    return NULL;
-}
-
-bool ca_layout_to_mp_chmap(struct ao *ao, AudioChannelLayout *layout,
-                           struct mp_chmap *chmap)
-{
-    void *talloc_ctx = talloc_new(NULL);
-
-    MP_DBG(ao, "input channel layout:\n");
-    ca_log_layout(ao, MSGL_DEBUG, layout);
-
-    AudioChannelLayout *l = ca_layout_to_custom_layout(ao, talloc_ctx, layout);
-    if (!l)
-        goto coreaudio_error;
-
-    MP_VERBOSE(ao, "converted input channel layout:\n");
-    ca_log_layout(ao, MSGL_V, l);
-
-    if (l->mNumberChannelDescriptions > MP_NUM_CHANNELS) {
-        MP_VERBOSE(ao, "layout has too many descriptions (%u, max: %d)\n",
-                   (unsigned) l->mNumberChannelDescriptions, MP_NUM_CHANNELS);
-        return false;
-    }
-
-    for (int n = 0; n < l->mNumberChannelDescriptions; n++) {
-        AudioChannelLabel label = l->mChannelDescriptions[n].mChannelLabel;
-        uint8_t speaker = ca_label_to_mp_speaker_id(label);
-        if (speaker < 0) {
-            MP_VERBOSE(ao, "channel label=%u unusable to build channel "
-                           "bitmap, skipping layout\n", (unsigned) label);
-            goto coreaudio_error;
-        } else {
-            chmap->speaker[n] = speaker;
-            chmap->num = n + 1;
-        }
-    }
-
-    talloc_free(talloc_ctx);
-    return chmap->num > 0;
-coreaudio_error:
-    MP_VERBOSE(ao, "converted input channel layout (failed):\n");
-    ca_log_layout(ao, MSGL_V, layout);
-    talloc_free(talloc_ctx);
-    return false;
-}
-
 #define OPT_BASE_STRUCT struct priv

 const struct ao_driver audio_out_coreaudio = {
@@ -577,4 +418,9 @@ const struct ao_driver audio_out_coreaudio = {
    .hotplug_uninit = hotplug_uninit,
    .list_devs      = ca_get_device_list,
    .priv_size      = sizeof(struct priv),
+    .options = (const struct m_option[]){
+        OPT_FLAG("change-physical-format", change_physical_format, 0),
+        OPT_FLAG("exclusive", exclusive, 0),
+        {0}
+    },
 };
@@ -0,0 +1,267 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "common/common.h"
+
+#include "ao_coreaudio_utils.h"
+
+#include "ao_coreaudio_chmap.h"
+
+static const int speaker_map[][2] = {
+    { kAudioChannelLabel_Left,                 MP_SPEAKER_ID_FL   },
+    { kAudioChannelLabel_Right,                MP_SPEAKER_ID_FR   },
+    { kAudioChannelLabel_Center,               MP_SPEAKER_ID_FC   },
+    { kAudioChannelLabel_LFEScreen,            MP_SPEAKER_ID_LFE  },
+    { kAudioChannelLabel_LeftSurround,         MP_SPEAKER_ID_BL   },
+    { kAudioChannelLabel_RightSurround,        MP_SPEAKER_ID_BR   },
+    { kAudioChannelLabel_LeftCenter,           MP_SPEAKER_ID_FLC  },
+    { kAudioChannelLabel_RightCenter,          MP_SPEAKER_ID_FRC  },
+    { kAudioChannelLabel_CenterSurround,       MP_SPEAKER_ID_BC   },
+    { kAudioChannelLabel_LeftSurroundDirect,   MP_SPEAKER_ID_SL   },
+    { kAudioChannelLabel_RightSurroundDirect,  MP_SPEAKER_ID_SR   },
+    { kAudioChannelLabel_TopCenterSurround,    MP_SPEAKER_ID_TC   },
+    { kAudioChannelLabel_VerticalHeightLeft,   MP_SPEAKER_ID_TFL  },
+    { kAudioChannelLabel_VerticalHeightCenter, MP_SPEAKER_ID_TFC  },
+    { kAudioChannelLabel_VerticalHeightRight,  MP_SPEAKER_ID_TFR  },
+    { kAudioChannelLabel_TopBackLeft,          MP_SPEAKER_ID_TBL  },
+    { kAudioChannelLabel_TopBackCenter,        MP_SPEAKER_ID_TBC  },
+    { kAudioChannelLabel_TopBackRight,         MP_SPEAKER_ID_TBR  },
+
+    // unofficial extensions
+    { kAudioChannelLabel_RearSurroundLeft,     MP_SPEAKER_ID_SDL  },
+    { kAudioChannelLabel_RearSurroundRight,    MP_SPEAKER_ID_SDR  },
+    { kAudioChannelLabel_LeftWide,             MP_SPEAKER_ID_WL   },
+    { kAudioChannelLabel_RightWide,            MP_SPEAKER_ID_WR   },
+    { kAudioChannelLabel_LFE2,                 MP_SPEAKER_ID_LFE2 },
+
+    { kAudioChannelLabel_HeadphonesLeft,       MP_SPEAKER_ID_DL   },
+    { kAudioChannelLabel_HeadphonesRight,      MP_SPEAKER_ID_DR   },
+
+    { kAudioChannelLabel_Unknown,              MP_SPEAKER_ID_NA   },
+
+    { 0,                                       -1                 },
+};
+
+static int ca_label_to_mp_speaker_id(AudioChannelLabel label)
+{
+    for (int i = 0; speaker_map[i][1] >= 0; i++)
+        if (speaker_map[i][0] == label)
+            return speaker_map[i][1];
+    return -1;
+}
+
+static void ca_log_layout(struct ao *ao, int l, AudioChannelLayout *layout)
+{
+    if (!mp_msg_test(ao->log, l))
+        return;
+
+    AudioChannelDescription *descs = layout->mChannelDescriptions;
+
+    mp_msg(ao->log, l, "layout: tag: <%u>, bitmap: <%u>, "
+                       "descriptions <%u>\n",
+                       (unsigned) layout->mChannelLayoutTag,
+                       (unsigned) layout->mChannelBitmap,
+                       (unsigned) layout->mNumberChannelDescriptions);
+
+    for (int i = 0; i < layout->mNumberChannelDescriptions; i++) {
+        AudioChannelDescription d = descs[i];
+        mp_msg(ao->log, l, " - description %d: label <%u, %u>, "
+            " flags: <%u>, coords: <%f, %f, %f>\n", i,
+            (unsigned) d.mChannelLabel,
+            (unsigned) ca_label_to_mp_speaker_id(d.mChannelLabel),
+            (unsigned) d.mChannelFlags,
+            d.mCoordinates[0],
+            d.mCoordinates[1],
+            d.mCoordinates[2]);
+    }
+}
+
+static AudioChannelLayout *ca_layout_to_custom_layout(struct ao *ao,
+                                                      void *talloc_ctx,
+                                                      AudioChannelLayout *l)
+{
+    AudioChannelLayoutTag tag = l->mChannelLayoutTag;
+    AudioChannelLayout *r;
+    OSStatus err;
+
+    if (tag == kAudioChannelLayoutTag_UseChannelBitmap) {
+        uint32_t psize;
+        err = AudioFormatGetPropertyInfo(
+            kAudioFormatProperty_ChannelLayoutForBitmap,
+            sizeof(uint32_t), &l->mChannelBitmap, &psize);
+        CHECK_CA_ERROR("failed to convert channel bitmap to descriptions (info)");
+        r = talloc_size(NULL, psize);
+        err = AudioFormatGetProperty(
+            kAudioFormatProperty_ChannelLayoutForBitmap,
+            sizeof(uint32_t), &l->mChannelBitmap, &psize, r);
+        CHECK_CA_ERROR("failed to convert channel bitmap to descriptions (get)");
+    } else if (tag != kAudioChannelLayoutTag_UseChannelDescriptions) {
+        uint32_t psize;
+        err = AudioFormatGetPropertyInfo(
+            kAudioFormatProperty_ChannelLayoutForTag,
+            sizeof(AudioChannelLayoutTag), &l->mChannelLayoutTag, &psize);
+        r = talloc_size(NULL, psize);
+        CHECK_CA_ERROR("failed to convert channel tag to descriptions (info)");
+        err = AudioFormatGetProperty(
+            kAudioFormatProperty_ChannelLayoutForTag,
+            sizeof(AudioChannelLayoutTag), &l->mChannelLayoutTag, &psize, r);
+        CHECK_CA_ERROR("failed to convert channel tag to descriptions (get)");
+    } else {
+        r = l;
+    }
+
+    return r;
+coreaudio_error:
+    return NULL;
+}
+
+static bool ca_layout_to_mp_chmap(struct ao *ao, AudioChannelLayout *layout,
+                                  struct mp_chmap *chmap)
+{
+    void *talloc_ctx = talloc_new(NULL);
+
+    MP_DBG(ao, "input channel layout:\n");
+    ca_log_layout(ao, MSGL_DEBUG, layout);
+
+    AudioChannelLayout *l = ca_layout_to_custom_layout(ao, talloc_ctx, layout);
+    if (!l)
+        goto coreaudio_error;
+
+    MP_VERBOSE(ao, "converted input channel layout:\n");
+    ca_log_layout(ao, MSGL_V, l);
+
+    if (l->mNumberChannelDescriptions > MP_NUM_CHANNELS) {
+        MP_VERBOSE(ao, "layout has too many descriptions (%u, max: %d)\n",
+                   (unsigned) l->mNumberChannelDescriptions, MP_NUM_CHANNELS);
+        return false;
+    }
+
+    for (int n = 0; n < l->mNumberChannelDescriptions; n++) {
+        AudioChannelLabel label = l->mChannelDescriptions[n].mChannelLabel;
+        int speaker = ca_label_to_mp_speaker_id(label);
+        if (speaker < 0) {
+            MP_VERBOSE(ao, "channel label=%u unusable to build channel "
+                           "bitmap, skipping layout\n", (unsigned) label);
+            goto coreaudio_error;
+        } else {
+            chmap->speaker[n] = speaker;
+            chmap->num = n + 1;
+        }
+    }
+
+    talloc_free(talloc_ctx);
+    return chmap->num > 0;
+coreaudio_error:
+    MP_VERBOSE(ao, "converted input channel layout (failed):\n");
+    ca_log_layout(ao, MSGL_V, layout);
+    talloc_free(talloc_ctx);
+    return false;
+}
+
+static AudioChannelLayout* ca_query_layout(struct ao *ao,
+                                           AudioDeviceID device,
+                                           void *talloc_ctx)
+{
+    OSStatus err;
+    uint32_t psize;
+    AudioChannelLayout *r = NULL;
+
+    AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
+        .mSelector = kAudioDevicePropertyPreferredChannelLayout,
+        .mScope    = kAudioDevicePropertyScopeOutput,
+        .mElement  = kAudioObjectPropertyElementWildcard,
+    };
+
+    err = AudioObjectGetPropertyDataSize(device, &p_addr, 0, NULL, &psize);
+    CHECK_CA_ERROR("could not get device preferred layout (size)");
+
+    r = talloc_size(talloc_ctx, psize);
+
+    err = AudioObjectGetPropertyData(device, &p_addr, 0, NULL, &psize, r);
+    CHECK_CA_ERROR("could not get device preferred layout (get)");
+
+coreaudio_error:
+    return r;
+}
+
+static AudioChannelLayout* ca_query_stereo_layout(struct ao *ao,
+                                                  AudioDeviceID device,
+                                                  void *talloc_ctx)
+{
+    OSStatus err;
+    const int nch = 2;
+    uint32_t channels[nch];
+    AudioChannelLayout *r = NULL;
+
+    AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
+        .mSelector = kAudioDevicePropertyPreferredChannelsForStereo,
+        .mScope    = kAudioDevicePropertyScopeOutput,
+        .mElement  = kAudioObjectPropertyElementWildcard,
+    };
+
+    uint32_t psize = sizeof(channels);
+    err = AudioObjectGetPropertyData(device, &p_addr, 0, NULL, &psize, channels);
+    CHECK_CA_ERROR("could not get device preferred stereo layout");
+
+    psize = sizeof(AudioChannelLayout) + nch * sizeof(AudioChannelDescription);
+    r = talloc_zero_size(talloc_ctx, psize);
+    r->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
+    r->mNumberChannelDescriptions = nch;
+
+    AudioChannelDescription desc = {0};
+    desc.mChannelFlags = kAudioChannelFlags_AllOff;
+
+    for(int i = 0; i < nch; i++) {
+        desc.mChannelLabel = channels[i];
+        r->mChannelDescriptions[i] = desc;
+    }
+
+coreaudio_error:
+    return r;
+}
+
+bool ca_init_chmap(struct ao *ao, AudioDeviceID device)
+{
+    void *ta_ctx = talloc_new(NULL);
+
+    struct mp_chmap_sel chmap_sel = {.tmp = ta_ctx};
+    struct mp_chmap chmap = {0};
+
+    mp_chmap_sel_add_map(&chmap_sel, &(struct mp_chmap)MP_CHMAP_INIT_MONO);
+
+    AudioChannelLayout *ml = ca_query_layout(ao, device, ta_ctx);
+    if (ml && ca_layout_to_mp_chmap(ao, ml, &chmap))
+        mp_chmap_sel_add_map(&chmap_sel, &chmap);
+
+    AudioChannelLayout *sl = ca_query_stereo_layout(ao, device, ta_ctx);
+    if (sl && ca_layout_to_mp_chmap(ao, sl, &chmap))
+        mp_chmap_sel_add_map(&chmap_sel, &chmap);
+
+    if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels)) {
+        MP_ERR(ao, "could not select a suitable channel map among the "
+                   "hardware supported ones. Make sure to configure your "
+                   "output device correctly in 'Audio MIDI Setup.app'\n");
+        goto coreaudio_error;
+    }
+
+    talloc_free(ta_ctx);
+    return true;
+
+coreaudio_error:
+    talloc_free(ta_ctx);
+    return false;
+}
@@ -1,6 +1,4 @@
 /*
- * Copyright (C) 2002 Anders Johansson ajh@atri.curtin.edu.au
- *
 * This file is part of mpv.
 *
 * mpv is free software; you can redistribute it and/or modify
@@ -17,15 +15,11 @@
 * with mpv.  If not, see <http://www.gnu.org/licenses/>.
 */

-#ifndef MPLAYER_DSP_H
-#define MPLAYER_DSP_H
+#ifndef MPV_COREAUDIO_CHMAP_H
+#define MPV_COREAUDIO_CHMAP_H

-/* Implementation of routines used for DSP */
+#include <AudioToolbox/AudioToolbox.h>

-/* Size of floating point type used in routines */
-#define FLOAT_TYPE float
+bool ca_init_chmap(struct ao *ao, AudioDeviceID device);

-#include "window.h"
-#include "filter.h"
-
-#endif /* MPLAYER_DSP_H */
+#endif
@@ -41,297 +41,100 @@
 #include "internal.h"
 #include "audio/format.h"
 #include "osdep/timer.h"
+#include "osdep/atomics.h"
 #include "options/m_option.h"
 #include "common/msg.h"
+#include "audio/out/ao_coreaudio_chmap.h"
 #include "audio/out/ao_coreaudio_properties.h"
 #include "audio/out/ao_coreaudio_utils.h"

-static bool ca_format_is_digital(AudioStreamBasicDescription asbd)
-{
-    switch (asbd.mFormatID)
-    case 'IAC3':
-    case 'iac3':
-    case  kAudioFormat60958AC3:
-    case  kAudioFormatAC3:
-        return true;
-    return false;
-}
-
-static bool ca_stream_supports_digital(struct ao *ao, AudioStreamID stream)
-{
-    AudioStreamRangedDescription *formats = NULL;
-    size_t n_formats;
-
-    OSStatus err =
-        CA_GET_ARY(stream, kAudioStreamPropertyAvailablePhysicalFormats,
-                   &formats, &n_formats);
-
-    CHECK_CA_ERROR("Could not get number of stream formats.");
-
-    for (int i = 0; i < n_formats; i++) {
-        AudioStreamBasicDescription asbd = formats[i].mFormat;
-        ca_print_asbd(ao, "supported format:", &(asbd));
-        if (ca_format_is_digital(asbd)) {
-            talloc_free(formats);
-            return true;
-        }
-    }
-
-    talloc_free(formats);
-coreaudio_error:
-    return false;
-}
-
-static bool ca_device_supports_digital(struct ao *ao, AudioDeviceID device)
-{
-    AudioStreamID *streams = NULL;
-    size_t n_streams;
-
-    /* Retrieve all the output streams. */
-    OSStatus err =
-        CA_GET_ARY_O(device, kAudioDevicePropertyStreams, &streams, &n_streams);
-
-    CHECK_CA_ERROR("could not get number of streams.");
-
-    for (int i = 0; i < n_streams; i++) {
-        if (ca_stream_supports_digital(ao, streams[i])) {
-            talloc_free(streams);
-            return true;
-        }
-    }
-
-    talloc_free(streams);
-
-coreaudio_error:
-    return false;
-}
-
-static OSStatus ca_property_listener(
-    AudioObjectPropertySelector selector,
-    AudioObjectID object, uint32_t n_addresses,
-    const AudioObjectPropertyAddress addresses[],
-    void *data)
-{
-    void *talloc_ctx = talloc_new(NULL);
-
-    for (int i = 0; i < n_addresses; i++) {
-        if (addresses[i].mSelector == selector) {
-            if (data) *(volatile int *)data = 1;
-            break;
-        }
-    }
-    talloc_free(talloc_ctx);
-    return noErr;
-}
-
-static OSStatus ca_stream_listener(
-    AudioObjectID object, uint32_t n_addresses,
-    const AudioObjectPropertyAddress addresses[],
-    void *data)
-{
-    return ca_property_listener(kAudioStreamPropertyPhysicalFormat,
-                                object, n_addresses, addresses, data);
-}
-
-static OSStatus ca_device_listener(
-    AudioObjectID object, uint32_t n_addresses,
-    const AudioObjectPropertyAddress addresses[],
-    void *data)
-{
-    return ca_property_listener(kAudioDevicePropertyDeviceHasChanged,
-                                object, n_addresses, addresses, data);
-}
-
-static OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid) {
-    *pid = getpid();
-    OSStatus err = CA_SET(device, kAudioDevicePropertyHogMode, pid);
-    if (err != noErr)
-        *pid = -1;
-
-    return err;
-}
-
-static OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid) {
-    if (*pid == getpid()) {
-        *pid = -1;
-        return CA_SET(device, kAudioDevicePropertyHogMode, &pid);
-    }
-    return noErr;
-}
-
-static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device,
-                                 uint32_t val, bool *changed) {
-    *changed = false;
-
-    AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
-        .mSelector = kAudioDevicePropertySupportsMixing,
-        .mScope    = kAudioObjectPropertyScopeGlobal,
-        .mElement  = kAudioObjectPropertyElementMaster,
-    };
-
-    if (AudioObjectHasProperty(device, &p_addr)) {
-        OSStatus err;
-        Boolean writeable = 0;
-        err = CA_SETTABLE(device, kAudioDevicePropertySupportsMixing,
-                          &writeable);
-
-        if (!CHECK_CA_WARN("can't tell if mixing property is settable")) {
-            return err;
-        }
-
-        if (!writeable)
-            return noErr;
-
-        err = CA_SET(device, kAudioDevicePropertySupportsMixing, &val);
-        if (err != noErr)
-            return err;
-
-        if (!CHECK_CA_WARN("can't set mix mode")) {
-            return err;
-        }
-
-        *changed = true;
-    }
-
-    return noErr;
-}
-
-static OSStatus ca_disable_mixing(struct ao *ao,
-                                  AudioDeviceID device, bool *changed) {
-    return ca_change_mixing(ao, device, 0, changed);
-}
-
-static OSStatus ca_enable_mixing(struct ao *ao,
-                                 AudioDeviceID device, bool changed) {
-    if (changed) {
-        bool dont_care = false;
-        return ca_change_mixing(ao, device, 1, &dont_care);
-    }
-
-    return noErr;
-}
-
-static OSStatus ca_change_device_listening(AudioDeviceID device,
-                                           void *flag, bool enabled)
-{
-    AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
-        .mSelector = kAudioDevicePropertyDeviceHasChanged,
-        .mScope    = kAudioObjectPropertyScopeGlobal,
-        .mElement  = kAudioObjectPropertyElementMaster,
-    };
-
-    if (enabled) {
-        return AudioObjectAddPropertyListener(
-            device, &p_addr, ca_device_listener, flag);
-    } else {
-        return AudioObjectRemovePropertyListener(
-            device, &p_addr, ca_device_listener, flag);
-    }
-}
-
-static OSStatus ca_enable_device_listener(AudioDeviceID device, void *flag) {
-    return ca_change_device_listening(device, flag, true);
-}
-
-static OSStatus ca_disable_device_listener(AudioDeviceID device, void *flag) {
-    return ca_change_device_listening(device, flag, false);
-}
-
-static bool ca_change_format(struct ao *ao, AudioStreamID stream,
-                             AudioStreamBasicDescription change_format)
-{
-    OSStatus err = noErr;
-    AudioObjectPropertyAddress p_addr;
-    volatile int stream_format_changed = 0;
-
-    ca_print_asbd(ao, "setting stream format:", &change_format);
-
-    /* Install the callback. */
-    p_addr = (AudioObjectPropertyAddress) {
-        .mSelector = kAudioStreamPropertyPhysicalFormat,
-        .mScope    = kAudioObjectPropertyScopeGlobal,
-        .mElement  = kAudioObjectPropertyElementMaster,
-    };
-
-    err = AudioObjectAddPropertyListener(stream, &p_addr, ca_stream_listener,
-                                         (void *)&stream_format_changed);
-    if (!CHECK_CA_WARN("can't add property listener during format change")) {
-        return false;
-    }
-
-    /* Change the format. */
-    err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &change_format);
-    if (!CHECK_CA_WARN("error changing physical format")) {
-        return false;
-    }
-
-    /* The AudioStreamSetProperty is not only asynchronious,
-     * it is also not Atomic, in its behaviour.
-     * Therefore we check 5 times before we really give up. */
-    bool format_set = false;
-    for (int i = 0; !format_set && i < 5; i++) {
-        for (int j = 0; !stream_format_changed && j < 50; j++)
-            mp_sleep_us(10000);
-
-        if (stream_format_changed) {
-            stream_format_changed = 0;
-        } else {
-            MP_VERBOSE(ao, "reached timeout\n");
-        }
-
-        AudioStreamBasicDescription actual_format;
-        err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format);
-
-        ca_print_asbd(ao, "actual format in use:", &actual_format);
-        if (actual_format.mSampleRate == change_format.mSampleRate &&
-            actual_format.mFormatID == change_format.mFormatID &&
-            actual_format.mFramesPerPacket == change_format.mFramesPerPacket) {
-            format_set = true;
-        }
-    }
-
-    err = AudioObjectRemovePropertyListener(stream, &p_addr, ca_stream_listener,
-                                            (void *)&stream_format_changed);
-
-    if (!CHECK_CA_WARN("can't remove property listener")) {
-        return false;
-    }
-
-    return format_set;
-}
-
-
 struct priv {
    AudioDeviceID device;   // selected device

    bool paused;

-    // digital render callback
+    // audio render callback
    AudioDeviceIOProcID render_cb;

    // pid set for hog mode, (-1) means that hog mode on the device was
    // released. hog mode is exclusive access to a device
    pid_t hog_pid;

-    // stream selected for digital playback by the detection in init
    AudioStreamID stream;

    // stream index in an AudioBufferList
    int stream_idx;

-    // format we changed the stream to: for the digital case each application
-    // sets the stream format for a device to what it needs
+    // format we changed the stream to, and the original format to restore
    AudioStreamBasicDescription stream_asbd;
    AudioStreamBasicDescription original_asbd;

    bool changed_mixing;
-    int stream_asbd_changed;
-    bool reload_requested;
+
+    atomic_bool reload_requested;

    uint32_t hw_latency_us;
 };

-static OSStatus render_cb_digital(
+static OSStatus property_listener_cb(
+    AudioObjectID object, uint32_t n_addresses,
+    const AudioObjectPropertyAddress addresses[],
+    void *data)
+{
+    struct ao *ao = data;
+    struct priv *p = ao->priv;
+
+    // Check whether we need to reset the compressed output stream.
+    AudioStreamBasicDescription f;
+    OSErr err = CA_GET(p->stream, kAudioStreamPropertyVirtualFormat, &f);
+    CHECK_CA_WARN("could not get stream format");
+    if (err != noErr || !ca_asbd_equals(&p->stream_asbd, &f)) {
+        if (atomic_compare_exchange_strong(&p->reload_requested,
+                                           &(bool){false}, true))
+        {
+            ao_request_reload(ao);
+            MP_INFO(ao, "Stream format changed! Reloading.\n");
+        }
+    }
+
+    return noErr;
+}
+
+static OSStatus enable_property_listener(struct ao *ao, bool enabled)
+{
+    struct priv *p = ao->priv;
+
+    uint32_t selectors[] = {kAudioDevicePropertyDeviceHasChanged,
+                            kAudioHardwarePropertyDevices};
+    AudioDeviceID devs[] = {p->device,
+                            kAudioObjectSystemObject};
+    assert(MP_ARRAY_SIZE(selectors) == MP_ARRAY_SIZE(devs));
+
+    OSStatus status = noErr;
+    for (int n = 0; n < MP_ARRAY_SIZE(devs); n++) {
+        AudioObjectPropertyAddress addr = {
+            .mScope    = kAudioObjectPropertyScopeGlobal,
+            .mElement  = kAudioObjectPropertyElementMaster,
+            .mSelector = selectors[n],
+        };
+        AudioDeviceID device = devs[n];
+
+        OSStatus status2;
+        if (enabled) {
+            status2 = AudioObjectAddPropertyListener(
+                device, &addr, property_listener_cb, ao);
+        } else {
+            status2 = AudioObjectRemovePropertyListener(
+                device, &addr, property_listener_cb, ao);
+        }
+        if (status == noErr)
+            status = status2;
+    }
+
+    return status;
+}
+
+static OSStatus render_cb_compressed(
        AudioDeviceID device, const AudioTimeStamp *ts,
        const void *in_data, const AudioTimeStamp *in_ts,
        AudioBufferList *out_data, const AudioTimeStamp *out_ts, void *ctx)
@@ -355,58 +158,112 @@ static OSStatus render_cb_digital(

    ao_read_data(ao, &buf.mData, pseudo_frames, end);

-    // Check whether we need to reset the digital output stream.
-    if (p->stream_asbd_changed) {
-        p->stream_asbd_changed = 0;
-        if (!p->reload_requested && ca_stream_supports_digital(ao, p->stream)) {
-            p->reload_requested = true;
-            ao_request_reload(ao);
-            MP_INFO(ao, "Stream format changed! Reloading.\n");
-        }
-    }
-
    return noErr;
 }

-static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd);
+// Apparently, audio devices can have multiple sub-streams. It's not clear to
+// me what devices with multiple streams actually do. So only select the first
+// one that fulfills some minimum requirements.
+// If this is not sufficient, we could duplicate the device list entries for
+// each sub-stream, and make it explicit.
+static int select_stream(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    AudioStreamID *streams;
+    size_t n_streams;
+    OSStatus err;
+
+    /* Get a list of all the streams on this device. */
+    err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams,
+                       &streams, &n_streams);
+    CHECK_CA_ERROR("could not get number of streams");
+    for (int i = 0; i < n_streams; i++) {
+        uint32_t direction;
+        err = CA_GET(streams[i], kAudioStreamPropertyDirection, &direction);
+        CHECK_CA_WARN("could not get stream direction");
+        if (err == noErr && direction != 0) {
+            MP_VERBOSE(ao, "Substream %d is not an output stream.\n", i);
+            continue;
+        }
+
+        if (af_fmt_is_pcm(ao->format) || ca_stream_supports_compressed(ao,
+                                                                   streams[i]))
+        {
+            MP_VERBOSE(ao, "Using substream %d/%zd.\n", i, n_streams);
+            p->stream = streams[i];
+            p->stream_idx = i;
+            break;
+        }
+    }
+
+    talloc_free(streams);
+
+    if (p->stream_idx < 0) {
+        MP_ERR(ao, "No useable substream found.\n");
+        goto coreaudio_error;
+    }
+
+    return 0;
+
+coreaudio_error:
+    return -1;
+}
+
+static int find_best_format(struct ao *ao, AudioStreamBasicDescription *out_fmt)
+{
+    struct priv *p = ao->priv;
+
+    // Build ASBD for the input format
+    AudioStreamBasicDescription asbd;
+    ca_fill_asbd(ao, &asbd);
+    ca_print_asbd(ao, "our format:", &asbd);
+
+    *out_fmt = (AudioStreamBasicDescription){0};
+
+    AudioStreamRangedDescription *formats;
+    size_t n_formats;
+    OSStatus err;
+
+    err = CA_GET_ARY(p->stream, kAudioStreamPropertyAvailablePhysicalFormats,
+                     &formats, &n_formats);
+    CHECK_CA_ERROR("could not get number of stream formats");
+
+    for (int j = 0; j < n_formats; j++) {
+        AudioStreamBasicDescription *stream_asbd = &formats[j].mFormat;
+
+        ca_print_asbd(ao, "- ", stream_asbd);
+
+        if (!out_fmt->mFormatID || ca_asbd_is_better(&asbd, out_fmt, stream_asbd))
+            *out_fmt = *stream_asbd;
+    }
+
+    talloc_free(formats);
+
+    if (!out_fmt->mFormatID) {
+        MP_ERR(ao, "no format found\n");
+        return -1;
+    }
+
+    return 0;
+coreaudio_error:
+    return -1;
+}

 static int init(struct ao *ao)
 {
    struct priv *p = ao->priv;

    OSStatus err = ca_select_device(ao, ao->device, &p->device);
-    CHECK_CA_ERROR("failed to select device");
+    CHECK_CA_ERROR_L(coreaudio_error_nounlock, "failed to select device");

    ao->format = af_fmt_from_planar(ao->format);

-    bool supports_digital = false;
-    /* Probe whether device support S/PDIF stream output if input is AC3 stream,
-     * or anything else IEC61937-framed. */
-    if (AF_FORMAT_IS_IEC61937(ao->format)) {
-        if (ca_device_supports_digital(ao, p->device))
-            supports_digital = true;
+    if (!af_fmt_is_pcm(ao->format) && !af_fmt_is_spdif(ao->format)) {
+        MP_ERR(ao, "Unsupported format.\n");
+        goto coreaudio_error_nounlock;
    }

-    if (!supports_digital) {
-        MP_ERR(ao, "selected device doesn't support digital formats\n");
-        goto coreaudio_error;
-    } // closes if (!supports_digital)
-
-    // Build ASBD for the input format
-    AudioStreamBasicDescription asbd;
-    ca_fill_asbd(ao, &asbd);
-
-    return init_digital(ao, asbd);
-
-coreaudio_error:
-    return CONTROL_ERROR;
-}
-
-static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
-{
-    struct priv *p = ao->priv;
-    OSStatus err = noErr;
-
    uint32_t is_alive = 1;
    err = CA_GET(p->device, kAudioDevicePropertyDeviceIsAlive, &is_alive);
    CHECK_CA_WARN("could not check whether device is alive");
@@ -420,107 +277,69 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
    err = ca_disable_mixing(ao, p->device, &p->changed_mixing);
    CHECK_CA_WARN("failed to disable mixing");

-    AudioStreamID *streams;
-    size_t n_streams;
-
-    /* Get a list of all the streams on this device. */
-    err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams,
-                       &streams, &n_streams);
-
-    CHECK_CA_ERROR("could not get number of streams");
-
-    for (int i = 0; i < n_streams && p->stream_idx < 0; i++) {
-        bool digital = ca_stream_supports_digital(ao, streams[i]);
-
-        if (digital) {
-            err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat,
-                         &p->original_asbd);
-            if (!CHECK_CA_WARN("could not get stream's physical format to "
-                               "revert to, getting the next one"))
-                continue;
-
-            AudioStreamRangedDescription *formats;
-            size_t n_formats;
-
-            err = CA_GET_ARY(streams[i],
-                             kAudioStreamPropertyAvailablePhysicalFormats,
-                             &formats, &n_formats);
-
-            if (!CHECK_CA_WARN("could not get number of stream formats"))
-                continue; // try next one
-
-            int req_rate_format = -1;
-            int max_rate_format = -1;
-
-            p->stream = streams[i];
-            p->stream_idx = i;
-
-            for (int j = 0; j < n_formats; j++)
-                if (ca_format_is_digital(formats[j].mFormat)) {
-                    // select the digital format that has exactly the same
-                    // samplerate. If an exact match cannot be found, select
-                    // the format with highest samplerate as backup.
-                    if (formats[j].mFormat.mSampleRate == asbd.mSampleRate) {
-                        req_rate_format = j;
-                        break;
-                    } else if (max_rate_format < 0 ||
-                        formats[j].mFormat.mSampleRate >
-                        formats[max_rate_format].mFormat.mSampleRate)
-                        max_rate_format = j;
-                }
-
-            if (req_rate_format >= 0)
-                p->stream_asbd = formats[req_rate_format].mFormat;
-            else
-                p->stream_asbd = formats[max_rate_format].mFormat;
-
-            talloc_free(formats);
-        }
-    }
-
-    talloc_free(streams);
-
-    if (p->stream_idx < 0) {
-        MP_WARN(ao , "can't find any digital output stream format\n");
-        goto coreaudio_error;
-    }
-
-    if (!ca_change_format(ao, p->stream, p->stream_asbd))
+    if (select_stream(ao) < 0)
        goto coreaudio_error;

-    void *changed = (void *) &(p->stream_asbd_changed);
-    err = ca_enable_device_listener(p->device, changed);
-    CHECK_CA_ERROR("cannot install format change listener during init");
+    AudioStreamBasicDescription hwfmt;
+    if (find_best_format(ao, &hwfmt) < 0)
+        goto coreaudio_error;

-    if (p->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
-        MP_WARN(ao, "stream has non-native byte order, output may fail\n");
+    err = CA_GET(p->stream, kAudioStreamPropertyPhysicalFormat,
+                 &p->original_asbd);
+    CHECK_CA_ERROR("could not get stream's original physical format");
+
+    // Even if changing the physical format fails, we can try using the current
+    // virtual format.
+    ca_change_physical_format_sync(ao, p->stream, hwfmt);
+
+    if (!ca_init_chmap(ao, p->device))
+        goto coreaudio_error;
+
+    err = CA_GET(p->stream, kAudioStreamPropertyVirtualFormat, &p->stream_asbd);
+    CHECK_CA_ERROR("could not get stream's virtual format");
+
+    ca_print_asbd(ao, "virtual format", &p->stream_asbd);
+
+    int new_format = ca_asbd_to_mp_format(&p->stream_asbd);
+
+    // If both old and new formats are spdif, avoid changing it due to the
+    // imperfect mapping between mp and CA formats.
+    if (!(af_fmt_is_spdif(ao->format) && af_fmt_is_spdif(new_format)))
+        ao->format = new_format;
+
+    if (!ao->format || af_fmt_is_planar(ao->format)) {
+        MP_ERR(ao, "hardware format not supported\n");
+        goto coreaudio_error;
+    }

    ao->samplerate = p->stream_asbd.mSampleRate;
-    ao->bps = ao->samplerate *
-                  (p->stream_asbd.mBytesPerPacket /
-                   p->stream_asbd.mFramesPerPacket);

-    uint32_t latency_frames = 0;
-    err = CA_GET_O(p->device, kAudioDevicePropertyLatency, &latency_frames);
-    if (err != noErr) {
-        CHECK_CA_WARN("cannot get device latency");
-        latency_frames = 0;
+    if (ao->channels.num != p->stream_asbd.mChannelsPerFrame) {
+        // We really expect that ca_init_chmap() fixes the layout to the HW's.
+        MP_ERR(ao, "number of channels changed, and unknown channel layout!\n");
+        goto coreaudio_error;
    }

-    p->hw_latency_us = ca_frames_to_us(ao, latency_frames);
+    p->hw_latency_us = ca_get_device_latency_us(ao, p->device);
    MP_VERBOSE(ao, "base latency: %d microseconds\n", (int)p->hw_latency_us);

+    err = enable_property_listener(ao, true);
+    CHECK_CA_ERROR("cannot install format change listener during init");
+
    err = AudioDeviceCreateIOProcID(p->device,
-                                    (AudioDeviceIOProc)render_cb_digital,
+                                    (AudioDeviceIOProc)render_cb_compressed,
                                    (void *)ao,
                                    &p->render_cb);
-    CHECK_CA_ERROR("failed to register digital render callback");
+    CHECK_CA_ERROR("failed to register audio render callback");

    return CONTROL_TRUE;

 coreaudio_error:
+    err = enable_property_listener(ao, false);
+    CHECK_CA_WARN("can't remove format change listener");
    err = ca_unlock_device(p->device, &p->hog_pid);
    CHECK_CA_WARN("can't release hog mode");
+coreaudio_error_nounlock:
    return CONTROL_ERROR;
 }

@@ -529,8 +348,7 @@ static void uninit(struct ao *ao)
    struct priv *p = ao->priv;
    OSStatus err = noErr;

-    void *changed = (void *) &(p->stream_asbd_changed);
-    err = ca_disable_device_listener(p->device, changed);
+    err = enable_property_listener(ao, false);
    CHECK_CA_WARN("can't remove device listener, this may cause a crash");

    err = AudioDeviceStop(p->device, p->render_cb);
@@ -539,7 +357,7 @@ static void uninit(struct ao *ao)
    err = AudioDeviceDestroyIOProcID(p->device, p->render_cb);
    CHECK_CA_WARN("failed to remove device render callback");

-    if (!ca_change_format(ao, p->stream, p->original_asbd))
+    if (!ca_change_physical_format_sync(ao, p->stream, p->original_asbd))
        MP_WARN(ao, "can't revert to original device format");

    err = ca_enable_mixing(ao, p->device, p->changed_mixing);
@@ -554,7 +372,7 @@ static void audio_pause(struct ao *ao)
    struct priv *p = ao->priv;

    OSStatus err = AudioDeviceStop(p->device, p->render_cb);
-    CHECK_CA_WARN("can't stop digital device");
+    CHECK_CA_WARN("can't stop audio device");
 }

 static void audio_resume(struct ao *ao)
@@ -562,7 +380,7 @@ static void audio_resume(struct ao *ao)
    struct priv *p = ao->priv;

    OSStatus err = AudioDeviceStart(p->device, p->render_cb);
-    CHECK_CA_WARN("can't start digital device");
+    CHECK_CA_WARN("can't start audio device");
 }

 #define OPT_BASE_STRUCT struct priv
@@ -577,7 +395,6 @@ const struct ao_driver audio_out_coreaudio_exclusive = {
    .list_devs = ca_get_device_list,
    .priv_size = sizeof(struct priv),
    .priv_defaults = &(const struct priv){
-        .stream_asbd_changed = 0,
        .hog_pid = -1,
        .stream = 0,
        .stream_idx = -1,
@@ -28,6 +28,7 @@
 #include "audio/out/ao_coreaudio_properties.h"
 #include "osdep/timer.h"
 #include "osdep/endian.h"
+#include "osdep/semaphore.h"
 #include "audio/format.h"

 CFStringRef cfstr_from_cstr(char *str)
@@ -88,7 +89,7 @@ OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device)
    OSStatus err = noErr;
    *device = kAudioObjectUnknown;

-    if (name) {
+    if (name && name[0]) {
        CFStringRef uid = cfstr_from_cstr(name);
        AudioValueTranslation v = (AudioValueTranslation) {
            .mInputData = &uid,
@@ -116,9 +117,8 @@ OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device)

    if (mp_msg_test(ao->log, MSGL_V)) {
        char *desc;
-        err = CA_GET_STR(*device, kAudioObjectPropertyName, &desc);
-        CHECK_CA_WARN("could not get selected audio device name");
-        if (err == noErr) {
+        OSStatus err2 = CA_GET_STR(*device, kAudioObjectPropertyName, &desc);
+        if (err2 == noErr) {
            MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
                           desc, *device);
            talloc_free(desc);
@@ -129,7 +129,7 @@ coreaudio_error:
    return err;
 }

-char *fourcc_repr(void *talloc_ctx, uint32_t code)
+char *fourcc_repr_buf(char *buf, size_t buf_size, uint32_t code)
 {
    // Extract FourCC letters from the uint32_t and finde out if it's a valid
    // code that is made of letters.
@@ -146,64 +146,119 @@ char *fourcc_repr(void *talloc_ctx, uint32_t code)
            valid_fourcc = false;
    }

-    char *repr;
    if (valid_fourcc)
-        repr = talloc_asprintf(talloc_ctx, "'%c%c%c%c'",
-                               fcc[0], fcc[1], fcc[2], fcc[3]);
+        snprintf(buf, buf_size, "'%c%c%c%c'", fcc[0], fcc[1], fcc[2], fcc[3]);
    else
-        repr = talloc_asprintf(NULL, "%u", (unsigned int)code);
+        snprintf(buf, buf_size, "%u", (unsigned int)code);

-    return repr;
+    return buf;
 }

 bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message)
 {
    if (code == noErr) return true;

-    char *error_string = fourcc_repr(NULL, code);
-    mp_msg(ao->log, level, "%s (%s)\n", message, error_string);
-    talloc_free(error_string);
+    mp_msg(ao->log, level, "%s (%s)\n", message, fourcc_repr(code));

    return false;
 }

-void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd)
+static void ca_fill_asbd_raw(AudioStreamBasicDescription *asbd, int mp_format,
+                             int samplerate, int num_channels)
 {
-    asbd->mSampleRate       = ao->samplerate;
+    asbd->mSampleRate       = samplerate;
    // Set "AC3" for other spdif formats too - unknown if that works.
-    asbd->mFormatID         = AF_FORMAT_IS_IEC61937(ao->format) ?
+    asbd->mFormatID         = af_fmt_is_spdif(mp_format) ?
                              kAudioFormat60958AC3 :
                              kAudioFormatLinearPCM;
-    asbd->mChannelsPerFrame = ao->channels.num;
-    asbd->mBitsPerChannel   = af_fmt2bits(ao->format);
+    asbd->mChannelsPerFrame = num_channels;
+    asbd->mBitsPerChannel   = af_fmt_to_bytes(mp_format) * 8;
    asbd->mFormatFlags      = kAudioFormatFlagIsPacked;

-    if ((ao->format & AF_FORMAT_TYPE_MASK) == AF_FORMAT_F)
-        asbd->mFormatFlags |= kAudioFormatFlagIsFloat;
+    int channels_per_buffer = num_channels;
+    if (af_fmt_is_planar(mp_format)) {
+        asbd->mFormatFlags |= kAudioFormatFlagIsNonInterleaved;
+        channels_per_buffer = 1;
+    }

-    if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI)
+    if (af_fmt_is_float(mp_format)) {
+        asbd->mFormatFlags |= kAudioFormatFlagIsFloat;
+    } else if (!af_fmt_is_unsigned(mp_format)) {
        asbd->mFormatFlags |= kAudioFormatFlagIsSignedInteger;
+    }

    if (BYTE_ORDER == BIG_ENDIAN)
        asbd->mFormatFlags |= kAudioFormatFlagIsBigEndian;

    asbd->mFramesPerPacket = 1;
    asbd->mBytesPerPacket = asbd->mBytesPerFrame =
-        asbd->mFramesPerPacket * asbd->mChannelsPerFrame *
+        asbd->mFramesPerPacket * channels_per_buffer *
        (asbd->mBitsPerChannel / 8);
 }

+void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd)
+{
+    ca_fill_asbd_raw(asbd, ao->format, ao->samplerate, ao->channels.num);
+}
+
+bool ca_formatid_is_compressed(uint32_t formatid)
+{
+    switch (formatid)
+    case 'IAC3':
+    case 'iac3':
+    case  kAudioFormat60958AC3:
+    case  kAudioFormatAC3:
+        return true;
+    return false;
+}
+
+// This might be wrong, but for now it's sufficient for us.
+static uint32_t ca_normalize_formatid(uint32_t formatID)
+{
+    return ca_formatid_is_compressed(formatID) ? kAudioFormat60958AC3 : formatID;
+}
+
+bool ca_asbd_equals(const AudioStreamBasicDescription *a,
+                    const AudioStreamBasicDescription *b)
+{
+    int flags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsFloat |
+                kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsBigEndian;
+    bool spdif = ca_formatid_is_compressed(a->mFormatID) &&
+                 ca_formatid_is_compressed(b->mFormatID);
+
+    return (a->mFormatFlags & flags) == (b->mFormatFlags & flags) &&
+           a->mBitsPerChannel == b->mBitsPerChannel &&
+           ca_normalize_formatid(a->mFormatID) ==
+                ca_normalize_formatid(b->mFormatID) &&
+           (spdif || a->mBytesPerPacket == b->mBytesPerPacket) &&
+           (spdif || a->mChannelsPerFrame == b->mChannelsPerFrame) &&
+           a->mSampleRate == b->mSampleRate;
+}
+
+// Return the AF_FORMAT_* (AF_FORMAT_S16 etc.) corresponding to the asbd.
+int ca_asbd_to_mp_format(const AudioStreamBasicDescription *asbd)
+{
+    for (int fmt = 1; fmt < AF_FORMAT_COUNT; fmt++) {
+        AudioStreamBasicDescription mp_asbd = {0};
+        ca_fill_asbd_raw(&mp_asbd, fmt, asbd->mSampleRate, asbd->mChannelsPerFrame);
+        if (ca_asbd_equals(&mp_asbd, asbd))
+            return af_fmt_is_spdif(fmt) ? AF_FORMAT_S_AC3 : fmt;
+    }
+    return 0;
+}
+
 void ca_print_asbd(struct ao *ao, const char *description,
                   const AudioStreamBasicDescription *asbd)
 {
    uint32_t flags  = asbd->mFormatFlags;
-    char *format    = fourcc_repr(NULL, asbd->mFormatID);
+    char *format    = fourcc_repr(asbd->mFormatID);
+    int mpfmt       = ca_asbd_to_mp_format(asbd);

    MP_VERBOSE(ao,
-       "%s %7.1fHz %" PRIu32 "bit [%s]"
-       "[%" PRIu32 "][%" PRIu32 "][%" PRIu32 "]"
-       "[%" PRIu32 "][%" PRIu32 "] "
-       "%s %s %s%s%s%s\n",
+       "%s %7.1fHz %" PRIu32 "bit %s "
+       "[%" PRIu32 "][%" PRIu32 "bpp][%" PRIu32 "fbp]"
+       "[%" PRIu32 "bpf][%" PRIu32 "ch] "
+       "%s %s %s%s%s%s (%s)\n",
       description, asbd->mSampleRate, asbd->mBitsPerChannel, format,
       asbd->mFormatFlags, asbd->mBytesPerPacket, asbd->mFramesPerPacket,
       asbd->mBytesPerFrame, asbd->mChannelsPerFrame,
@@ -212,9 +267,49 @@ void ca_print_asbd(struct ao *ao, const char *description,
       (flags & kAudioFormatFlagIsSignedInteger) ? "S" : "U",
       (flags & kAudioFormatFlagIsPacked) ? " packed" : "",
       (flags & kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
-       (flags & kAudioFormatFlagIsNonInterleaved) ? " P" : "");
+       (flags & kAudioFormatFlagIsNonInterleaved) ? " P" : "",
+       mpfmt ? af_fmt_to_str(mpfmt) : "-");
+}

-    talloc_free(format);
+// Return whether new is an improvement over old. Assume a higher value means
+// better quality, and we always prefer the value closest to the requested one,
+// which is still larger than the requested one.
+// Equal values prefer the new one (so ca_asbd_is_better() checks other params).
+static bool value_is_better(double req, double old, double new)
+{
+    if (new >= req) {
+        return old < req || new <= old;
+    } else {
+        return old < req && new >= old;
+    }
+}
+
+// Return whether new is an improvement over old (req is the requested format).
+bool ca_asbd_is_better(AudioStreamBasicDescription *req,
+                       AudioStreamBasicDescription *old,
+                       AudioStreamBasicDescription *new)
+{
+    if (new->mChannelsPerFrame > MP_NUM_CHANNELS)
+        return false;
+    if (old->mChannelsPerFrame > MP_NUM_CHANNELS)
+        return true;
+    if (req->mFormatID != new->mFormatID)
+        return false;
+    if (req->mFormatID != old->mFormatID)
+        return true;
+
+    if (!value_is_better(req->mBitsPerChannel, old->mBitsPerChannel,
+                         new->mBitsPerChannel))
+        return false;
+
+    if (!value_is_better(req->mSampleRate, old->mSampleRate, new->mSampleRate))
+        return false;
+
+    if (!value_is_better(req->mChannelsPerFrame, old->mChannelsPerFrame,
+                         new->mChannelsPerFrame))
+        return false;
+
+    return true;
 }

 int64_t ca_frames_to_us(struct ao *ao, uint32_t frames)
@@ -232,3 +327,229 @@ int64_t ca_get_latency(const AudioTimeStamp *ts)

    return (out - now) * 1e-3;
 }
+
+bool ca_stream_supports_compressed(struct ao *ao, AudioStreamID stream)
+{
+    AudioStreamRangedDescription *formats = NULL;
+    size_t n_formats;
+
+    OSStatus err =
+        CA_GET_ARY(stream, kAudioStreamPropertyAvailablePhysicalFormats,
+                   &formats, &n_formats);
+
+    CHECK_CA_ERROR("Could not get number of stream formats.");
+
+    for (int i = 0; i < n_formats; i++) {
+        AudioStreamBasicDescription asbd = formats[i].mFormat;
+        if (ca_formatid_is_compressed(asbd.mFormatID)) {
+            talloc_free(formats);
+            return true;
+        }
+    }
+
+    talloc_free(formats);
+coreaudio_error:
+    return false;
+}
+
+bool ca_device_supports_compressed(struct ao *ao, AudioDeviceID device)
+{
+    AudioStreamID *streams = NULL;
+    size_t n_streams;
+
+    /* Retrieve all the output streams. */
+    OSStatus err =
+        CA_GET_ARY_O(device, kAudioDevicePropertyStreams, &streams, &n_streams);
+
+    CHECK_CA_ERROR("could not get number of streams.");
+
+    for (int i = 0; i < n_streams; i++) {
+        if (ca_stream_supports_compressed(ao, streams[i])) {
+            talloc_free(streams);
+            return true;
+        }
+    }
+
+    talloc_free(streams);
+
+coreaudio_error:
+    return false;
+}
+
+OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid)
+{
+    *pid = getpid();
+    OSStatus err = CA_SET(device, kAudioDevicePropertyHogMode, pid);
+    if (err != noErr)
+        *pid = -1;
+
+    return err;
+}
+
+OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid)
+{
+    if (*pid == getpid()) {
+        *pid = -1;
+        return CA_SET(device, kAudioDevicePropertyHogMode, &pid);
+    }
+    return noErr;
+}
+
+static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device,
+                                 uint32_t val, bool *changed)
+{
+    *changed = false;
+
+    AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
+        .mSelector = kAudioDevicePropertySupportsMixing,
+        .mScope    = kAudioObjectPropertyScopeGlobal,
+        .mElement  = kAudioObjectPropertyElementMaster,
+    };
+
+    if (AudioObjectHasProperty(device, &p_addr)) {
+        OSStatus err;
+        Boolean writeable = 0;
+        err = CA_SETTABLE(device, kAudioDevicePropertySupportsMixing,
+                          &writeable);
+
+        if (!CHECK_CA_WARN("can't tell if mixing property is settable")) {
+            return err;
+        }
+
+        if (!writeable)
+            return noErr;
+
+        err = CA_SET(device, kAudioDevicePropertySupportsMixing, &val);
+        if (err != noErr)
+            return err;
+
+        if (!CHECK_CA_WARN("can't set mix mode")) {
+            return err;
+        }
+
+        *changed = true;
+    }
+
+    return noErr;
+}
+
+OSStatus ca_disable_mixing(struct ao *ao, AudioDeviceID device, bool *changed)
+{
+    return ca_change_mixing(ao, device, 0, changed);
+}
+
+OSStatus ca_enable_mixing(struct ao *ao, AudioDeviceID device, bool changed)
+{
+    if (changed) {
+        bool dont_care = false;
+        return ca_change_mixing(ao, device, 1, &dont_care);
+    }
+
+    return noErr;
+}
+
+int64_t ca_get_device_latency_us(struct ao *ao, AudioDeviceID device)
+{
+    uint32_t latency_frames = 0;
+    uint32_t latency_properties[] = {
+        kAudioDevicePropertyLatency,
+        kAudioDevicePropertyBufferFrameSize,
+        kAudioDevicePropertySafetyOffset,
+    };
+    for (int n = 0; n < MP_ARRAY_SIZE(latency_properties); n++) {
+        uint32_t temp;
+        OSStatus err = CA_GET_O(device, latency_properties[n], &temp);
+        CHECK_CA_WARN("cannot get device latency");
+        if (err == noErr) {
+            latency_frames += temp;
+            MP_VERBOSE(ao, "Latency property %s: %d frames\n",
+                       fourcc_repr(latency_properties[n]), (int)temp);
+        }
+    }
+
+    return ca_frames_to_us(ao, latency_frames);
+}
+
+static OSStatus ca_change_format_listener(
+    AudioObjectID object, uint32_t n_addresses,
+    const AudioObjectPropertyAddress addresses[],
+    void *data)
+{
+    sem_t *sem = data;
+    sem_post(sem);
+    return noErr;
+}
+
+bool ca_change_physical_format_sync(struct ao *ao, AudioStreamID stream,
+                                    AudioStreamBasicDescription change_format)
+{
+    OSStatus err = noErr;
+    bool format_set = false;
+
+    ca_print_asbd(ao, "setting stream physical format:", &change_format);
+
+    sem_t wakeup;
+    if (sem_init(&wakeup, 0, 0)) {
+        MP_WARN(ao, "OOM\n");
+        return false;
+    }
+
+    AudioStreamBasicDescription prev_format;
+    err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &prev_format);
+    CHECK_CA_ERROR("can't get current physical format");
+
+    /* Install the callback. */
+    AudioObjectPropertyAddress p_addr = {
+        .mSelector = kAudioStreamPropertyPhysicalFormat,
+        .mScope    = kAudioObjectPropertyScopeGlobal,
+        .mElement  = kAudioObjectPropertyElementMaster,
+    };
+
+    err = AudioObjectAddPropertyListener(stream, &p_addr,
+                                         ca_change_format_listener,
+                                         &wakeup);
+    CHECK_CA_ERROR("can't add property listener during format change");
+
+    /* Change the format. */
+    err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &change_format);
+    CHECK_CA_WARN("error changing physical format");
+
+    /* The AudioStreamSetProperty is not only asynchronous,
+     * it is also not Atomic, in its behaviour. */
+    struct timespec timeout = mp_rel_time_to_timespec(0.5);
+    AudioStreamBasicDescription actual_format = {0};
+    while (1) {
+        err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format);
+        if (!CHECK_CA_WARN("could not retrieve physical format"))
+            break;
+
+        format_set = ca_asbd_equals(&change_format, &actual_format);
+        if (format_set)
+            break;
+
+        if (sem_timedwait(&wakeup, &timeout)) {
+            MP_VERBOSE(ao, "reached timeout\n");
+            break;
+        }
+    }
+
+    ca_print_asbd(ao, "actual format in use:", &actual_format);
+
+    if (!format_set) {
+        MP_WARN(ao, "changing physical format failed\n");
+        // Some drivers just fuck up and get into a broken state. Restore the
+        // old format in this case.
+        err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &prev_format);
+        CHECK_CA_WARN("error restoring physical format");
+    }
+
+    err = AudioObjectRemovePropertyListener(stream, &p_addr,
+                                            ca_change_format_listener,
+                                            &wakeup);
+    CHECK_CA_ERROR("can't remove property listener");
+
+coreaudio_error:
+    sem_destroy(&wakeup);
+    return format_set;
+}
+
@@ -31,7 +31,9 @@
 CFStringRef cfstr_from_cstr(char *str);
 char *cfstr_get_cstr(CFStringRef cfstr);

-char *fourcc_repr(void *talloc_ctx, uint32_t code);
+char *fourcc_repr_buf(char *buf, size_t buf_size, uint32_t code);
+#define fourcc_repr(code) fourcc_repr_buf((char[40]){0}, 40, code)
+
 bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message);

 #define CHECK_CA_ERROR_L(label, message) \
@@ -52,11 +54,28 @@ bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message);
 void ca_get_device_list(struct ao *ao, struct ao_device_list *list);
 OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device);

+bool ca_formatid_is_compressed(uint32_t formatid);
 void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd);
 void ca_print_asbd(struct ao *ao, const char *description,
                   const AudioStreamBasicDescription *asbd);
+bool ca_asbd_equals(const AudioStreamBasicDescription *a,
+                    const AudioStreamBasicDescription *b);
+int ca_asbd_to_mp_format(const AudioStreamBasicDescription *asbd);
+bool ca_asbd_is_better(AudioStreamBasicDescription *req,
+                       AudioStreamBasicDescription *old,
+                       AudioStreamBasicDescription *new);

 int64_t ca_frames_to_us(struct ao *ao, uint32_t frames);
 int64_t ca_get_latency(const AudioTimeStamp *ts);

+bool ca_device_supports_compressed(struct ao *ao, AudioDeviceID device);
+bool ca_stream_supports_compressed(struct ao *ao, AudioStreamID stream);
+OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid);
+OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid);
+OSStatus ca_disable_mixing(struct ao *ao, AudioDeviceID device, bool *changed);
+OSStatus ca_enable_mixing(struct ao *ao, AudioDeviceID device, bool changed);
+int64_t ca_get_device_latency_us(struct ao *ao, AudioDeviceID device);
+bool ca_change_physical_format_sync(struct ao *ao, AudioStreamID stream,
+                                    AudioStreamBasicDescription change_format);
+
 #endif /* MPV_COREAUDIO_UTILS_H */
@@ -288,7 +288,7 @@ static int InitDirectSound(struct ao *ao)
    /* Set DirectSound Cooperative level, ie what control we want over Windows
     * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
     * settings of the primary buffer, but also that only the sound of our
-     * application will be hearable when it will have the focus.
+     * application will be audible when it will have the focus.
     * !!! (this is not really working as intended yet because to set the
     * cooperative level you need the window handle of your application, and
     * I don't know of any easy way to get it. Especially since we might play
@@ -444,7 +444,7 @@ static int init(struct ao *ao)
    int format = af_fmt_from_planar(ao->format);
    int rate = ao->samplerate;

-    if (!AF_FORMAT_IS_IEC61937(format)) {
+    if (!af_fmt_is_spdif(format)) {
        struct mp_chmap_sel sel = {0};
        mp_chmap_sel_add_waveext(&sel);
        if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
@@ -456,7 +456,7 @@ static int init(struct ao *ao)
    case AF_FORMAT_U8:
        break;
    default:
-        if (AF_FORMAT_IS_IEC61937(format))
+        if (af_fmt_is_spdif(format))
            break;
        MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
                   af_fmt_to_str(format));
@@ -465,7 +465,7 @@ static int init(struct ao *ao)
    //set our audio parameters
    ao->samplerate = rate;
    ao->format = format;
-    ao->bps = ao->channels.num * rate * af_fmt2bps(format);
+    ao->bps = ao->channels.num * rate * af_fmt_to_bytes(format);
    int buffersize = ao->bps * p->cfg_buffersize / 1000;
    MP_VERBOSE(ao, "Samplerate:%iHz Channels:%i Format:%s\n", rate,
               ao->channels.num, af_fmt_to_str(format));
@@ -478,7 +478,7 @@ static int init(struct ao *ao)
                    ? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
    wformat.Format.nChannels = ao->channels.num;
    wformat.Format.nSamplesPerSec = rate;
-    if (AF_FORMAT_IS_IEC61937(format)) {
+    if (af_fmt_is_spdif(format)) {
        // Whether it also works with e.g. DTS is unknown, but probably does.
        wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
        wformat.Format.wBitsPerSample = 16;
@@ -486,7 +486,7 @@ static int init(struct ao *ao)
    } else {
        wformat.Format.wFormatTag = (ao->channels.num > 2)
                                    ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
-        int bps = af_fmt2bps(format);
+        int bps = af_fmt_to_bytes(format);
        wformat.Format.wBitsPerSample = bps * 8;
        wformat.Format.nBlockAlign = wformat.Format.nChannels * bps;
    }
@@ -616,9 +616,9 @@ static int check_free_buffer_size(struct ao *ao)
    space = p->buffer_size - (p->write_offset - play_offset);
    // |              | <-- const --> |                |                 |
    // buffer start   play_cursor     write_cursor     p->write_offset   buffer end
-    // play_cursor is the actual postion of the play cursor
+    // play_cursor is the actual position of the play cursor
    // write_cursor is the position after which it is assumed to be save to write data
-    // p->write_offset is the postion where we actually write the data to
+    // p->write_offset is the position where we actually write the data to
    if (space > p->buffer_size)
        space -= p->buffer_size;                        // p->write_offset < play_offset
    // Check for buffer underruns. An underrun happens if DirectSound
@@ -25,7 +25,6 @@
 #include <limits.h>

 #include <libavutil/common.h>
-#include <libavutil/audioconvert.h>

 #include "config.h"
 #include "options/options.h"
@@ -59,33 +58,28 @@ struct priv {
    bool shutdown;
 };

-static void select_format(struct ao *ao, AVCodec *codec)
+static bool supports_format(AVCodec *codec, int format)
 {
-    int best_score = INT_MIN;
-    int best_format = 0;
-
-    // Check the encoder's list of supported formats.
    for (const enum AVSampleFormat *sampleformat = codec->sample_fmts;
         sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
         ++sampleformat)
    {
-        int fmt = af_from_avformat(*sampleformat);
-        if (!fmt) {
-            MP_WARN(ao, "unsupported lavc format %s\n",
-                    av_get_sample_fmt_name(*sampleformat));
-            continue;
-        }
-        int score = af_format_conversion_score(fmt, ao->format);
-        if (score > best_score) {
-            best_score = score;
-            best_format = fmt;
-        }
+        if (af_from_avformat(*sampleformat) == format)
+            return true;
    }
+    return false;
+}

-    if (best_format) {
-        ao->format = best_format;
-    } else {
-        MP_ERR(ao, "sample format not found\n"); // shouldn't happen
+static void select_format(struct ao *ao, AVCodec *codec)
+{
+    int formats[AF_FORMAT_COUNT];
+    af_get_best_sample_formats(ao->format, formats);
+
+    for (int n = 0; formats[n]; n++) {
+        if (supports_format(codec, formats[n])) {
+            ao->format = formats[n];
+            break;
+        }
    }
 }

@@ -136,7 +130,7 @@ static int init(struct ao *ao)

    select_format(ao, codec);

-    ac->sample_size = af_fmt2bps(ao->format);
+    ac->sample_size = af_fmt_to_bytes(ao->format);
    ac->stream->codec->sample_fmt = af_to_avformat(ao->format);
    ac->stream->codec->bits_per_raw_sample = ac->sample_size * 8;

@@ -241,7 +235,7 @@ static int encode(struct ao *ao, double apts, void **data)
        frame->format = af_to_avformat(ao->format);
        frame->nb_samples = ac->aframesize;

-        size_t num_planes = AF_FORMAT_IS_PLANAR(ao->format) ? ao->channels.num : 1;
+        size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;
        assert(num_planes <= AV_NUM_DATA_POINTERS);
        for (int n = 0; n < num_planes; n++)
            frame->extended_data[n] = data[n];
@@ -350,7 +344,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
    double pts = ectx->last_audio_in_pts;
    pts += ectx->samples_since_last_pts / (double)ao->samplerate;

-    size_t num_planes = AF_FORMAT_IS_PLANAR(ao->format) ? ao->channels.num : 1;
+    size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;

    void *tempdata = NULL;
    void *padded[MP_NUM_CHANNELS];
@@ -56,6 +56,8 @@ struct priv {
    // called with sizes not aligned to this, a rounded size will be returned.
    // (This is not needed by the AO API, but many AOs behave this way.)
    int outburst;       // samples
+
+    char **channel_layouts;
 };

 static void drain(struct ao *ao)
@@ -88,8 +90,19 @@ static int init(struct ao *ao)

    ao->untimed = priv->untimed;

-    struct mp_chmap_sel sel = {0};
-    mp_chmap_sel_add_any(&sel);
+    struct mp_chmap_sel sel = {.tmp = ao};
+    if (priv->channel_layouts) {
+        for (int n = 0; priv->channel_layouts[n]; n++) {
+            struct mp_chmap map = {0};
+            if (!mp_chmap_from_str(&map, bstr0(priv->channel_layouts[n]))) {
+                MP_FATAL(ao, "Invalid channel map in option.\n");
+                return -1;
+            }
+            mp_chmap_sel_add_map(&sel, &map);
+        }
+    } else {
+        mp_chmap_sel_add_any(&sel);
+    }
    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
        mp_chmap_from_channels(&ao->channels, 2);

@@ -231,6 +244,7 @@ const struct ao_driver audio_out_null = {
        OPT_FLOATRANGE("latency", latency_sec, 0, 0, 100),
        OPT_FLAG("broken-eof", broken_eof, 0),
        OPT_FLAG("broken-delay", broken_delay, 0),
+        OPT_STRINGLIST("channel-layouts", channel_layouts, 0),
        {0}
    },
 };
@@ -94,40 +94,20 @@ static const struct mp_chmap oss_layouts[MP_NUM_CHANNELS + 1] = {
 #define AFMT_S16_NE MP_SELECT_LE_BE(AFMT_S16_LE, AFMT_S16_BE)
 #endif

-#if !defined(AFMT_U16_NE) && defined(AFMT_U16_LE) && defined(AFMT_U16_BE)
-#define AFMT_U16_NE MP_SELECT_LE_BE(AFMT_U16_LE, AFMT_U16_BE)
-#endif
-
-#if !defined(AFMT_U24_NE) && defined(AFMT_U24_LE) && defined(AFMT_U24_BE)
-#define AFMT_U24_NE MP_SELECT_LE_BE(AFMT_U24_LE, AFMT_U24_BE)
-#endif
-
 #if !defined(AFMT_S24_NE) && defined(AFMT_S24_LE) && defined(AFMT_S24_BE)
 #define AFMT_S24_NE MP_SELECT_LE_BE(AFMT_S24_LE, AFMT_S24_BE)
 #endif

-#if !defined(AFMT_U32_NE) && defined(AFMT_U32_LE) && defined(AFMT_U32_BE)
-#define AFMT_U32_NE AFMT_U32MP_SELECT_LE_BE(AFMT_U32_LE, AFMT_U32_BE)
-#endif
-
 #if !defined(AFMT_S32_NE) && defined(AFMT_S32_LE) && defined(AFMT_S32_BE)
 #define AFMT_S32_NE AFMT_S32MP_SELECT_LE_BE(AFMT_S32_LE, AFMT_S32_BE)
 #endif

 static const int format_table[][2] = {
    {AFMT_U8,           AF_FORMAT_U8},
-    {AFMT_S8,           AF_FORMAT_S8},
-    {AFMT_U16_NE,       AF_FORMAT_U16},
    {AFMT_S16_NE,       AF_FORMAT_S16},
-#ifdef AFMT_U24_NE
-    {AFMT_U24_NE,       AF_FORMAT_U24},
-#endif
 #ifdef AFMT_S24_NE
    {AFMT_S24_NE,       AF_FORMAT_S24},
 #endif
-#ifdef AFMT_U32_NE
-    {AFMT_U32_NE,       AF_FORMAT_U32},
-#endif
 #ifdef AFMT_S32_NE
    {AFMT_S32_NE,       AF_FORMAT_S32},
 #endif
@@ -204,7 +184,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
            return CONTROL_OK;
 #endif

-        if (AF_FORMAT_IS_SPECIAL(ao->format))
+        if (!af_fmt_is_pcm(ao->format))
            return CONTROL_TRUE;

        if ((fd = open(p->oss_mixer_device, O_RDONLY)) != -1) {
@@ -267,7 +247,7 @@ static bool try_format(struct ao *ao, int *format)
    struct priv *p = ao->priv;

    int oss_format = format2oss(*format);
-    if (oss_format == -1 && AF_FORMAT_IS_IEC61937(*format))
+    if (oss_format == -1 && af_fmt_is_spdif(*format))
        oss_format = AFMT_AC3;

    if (oss_format == -1) {
@@ -323,7 +303,7 @@ static int reopen_device(struct ao *ao, bool allow_format_changes)
    fcntl(p->audio_fd, F_SETFD, FD_CLOEXEC);
 #endif

-    if (AF_FORMAT_IS_IEC61937(format)) {
+    if (af_fmt_is_spdif(format)) {
        if (ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
            goto fail;
        // Probably could be fixed by setting number of channels; needs testing.
@@ -333,7 +313,8 @@ static int reopen_device(struct ao *ao, bool allow_format_changes)
        }
    }

-    int try_formats[] = {format, AF_FORMAT_S32, AF_FORMAT_S24, AF_FORMAT_S16, 0};
+    int try_formats[AF_FORMAT_COUNT];
+    af_get_best_sample_formats(format, try_formats);
    for (int n = 0; try_formats[n]; n++) {
        format = try_formats[n];
        if (try_format(ao, &format))
@@ -347,7 +328,7 @@ static int reopen_device(struct ao *ao, bool allow_format_changes)

    MP_VERBOSE(ao, "sample format: %s\n", af_fmt_to_str(format));

-    if (!AF_FORMAT_IS_IEC61937(format)) {
+    if (!af_fmt_is_spdif(format)) {
        struct mp_chmap_sel sel = {0};
        for (int n = 0; n < MP_NUM_CHANNELS + 1; n++)
            mp_chmap_sel_add_map(&sel, &oss_layouts[n]);
@@ -412,7 +393,7 @@ static int reopen_device(struct ao *ao, bool allow_format_changes)
        }
    }

-    p->outburst -= p->outburst % (channels.num * af_fmt2bps(format)); // round down
+    p->outburst -= p->outburst % (channels.num * af_fmt_to_bytes(format)); // round down

    return 0;

@@ -73,7 +73,7 @@ static void fput32le(uint32_t val, FILE *fp)
 static void write_wave_header(struct ao *ao, FILE *fp, uint64_t data_length)
 {
    uint16_t fmt = ao->format == AF_FORMAT_FLOAT ? WAV_ID_FLOAT_PCM : WAV_ID_PCM;
-    int bits = af_fmt2bits(ao->format);
+    int bits = af_fmt_to_bytes(ao->format) * 8;

    // Master RIFF chunk
    fput32le(WAV_ID_RIFF, fp);
@@ -135,7 +135,7 @@ static int init(struct ao *ao)
        case AF_FORMAT_FLOAT:
             break;
        default:
-            if (!AF_FORMAT_IS_IEC61937(ao->format))
+            if (!af_fmt_is_spdif(ao->format))
                ao->format = AF_FORMAT_S16;
            break;
        }
@@ -146,7 +146,7 @@ static int init(struct ao *ao)
    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
        return -1;

-    ao->bps = ao->channels.num * ao->samplerate * af_fmt2bps(ao->format);
+    ao->bps = ao->channels.num * ao->samplerate * af_fmt_to_bytes(ao->format);

    MP_INFO(ao, "File: %s (%s)\nPCM: Samplerate: %d Hz Channels: %d Format: %s\n",
            priv->outputfilename,
@@ -210,7 +210,7 @@ static pa_encoding_t map_digital_format(int format)
    case AF_FORMAT_S_AAC:   return PA_ENCODING_MPEG2_AAC_IEC61937;
 #endif
    default:
-        if (AF_FORMAT_IS_IEC61937(format))
+        if (af_fmt_is_spdif(format))
            return PA_ENCODING_ANY;
        return PA_ENCODING_PCM;
    }