Fix typo...

(cherry picked from commit e85bcc1223)

.gitignore

@@ -14,8 +14,8 @@
 /input/input_conf.h
 /tags
 /TAGS
+/video/out/gl_video_shaders.h
 /video/out/x11_icon.inc
-/video/out/opengl/nnedi3_weights.inc
 /demux/ebml_defs.c
 /demux/ebml_types.h
 /sub/osd_font.h

.travis.yml

@@ -2,7 +2,7 @@ language: c
 
 os:
   - linux
-  - osx
+#  - osx
 env:
   matrix:
     - LIBAV=libav-stable
@@ -47,7 +47,7 @@ notifications:
   email: false
   irc:
     channels:
-      - "irc.freenode.org#mpv-devel"
+      - "irc.freenode.org#mpv-player-dev"
     on_success: change
     on_failure: always
 

CONTRIBUTING.md

@@ -1,30 +0,0 @@
-Reporting bugs
-==============
-
-- State your platform and mpv version.
-- Make sure you're actually using the latest mpv release. Linux distributions
-  in particular love redistributing ancient and unmaintained mpv releases.
-- Attach a log file made with ``-v`` or ``--log-file=output.txt``.
-- Avoid attaching text files packed in archive files (zip/rar/...), unless
-  explicitly asked so. There are better means, such as using http://sprunge.us
-- If you think this is file specific, upload a sample file and link it.
-- Don't report multiple unrelated/barely related bugs in one issue.
-- Read: https://mpv.io/bug-reports/
-
-Asking questions
-================
-
-Preferably, asking questions should be done via IRC (#mpv on irc.freenode.net),
-but asking questions by opening a github issue is also barely tolerated.
-
-When asking on IRC (#mpv on irc.freenode.net), don't ask to ask, just state
-your problem. Stay long enough until someone gets a chance to reply. Sometimes
-it can take hours.
-
-Sending patches
-===============
-
-See: https://github.com/mpv-player/mpv/blob/master/DOCS/contribute.md
-
-In particular, be aware that all changes to GPL code must come with the
-implicit agreement that they might be LGPLv2.1+ relicensed at a later point.

Copyright

@@ -2,18 +2,9 @@ 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 LGPLv2.1+, BSD, MIT, ISC, and possibly others. Look at the copyright
+such as LGPLv2+, 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 LGPLv2.1+.
-
-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.
-
-Some GPLv3 or LGPLv3 code in mpv can be enabled by building with --enable-gpl3,
-and in this case, the resulting binaries must be distributed under GPLv3.
+to know details. Files without Copyright notice are licensed as LGPLv2+.
 
 For information about authors and contributors, consult the git log, which
 contains the complete SVN and CVS history as well.
@@ -26,5 +17,3 @@ 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".

DOCS/client-api-changes.rst

@@ -4,7 +4,9 @@ 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, you should also look at ``interface-changes.rst``.
+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.
 
 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),
@@ -18,43 +20,12 @@ 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
 ===========
 
 ::
 
- --- mpv 0.12.0 ---
- 1.20   - deprecate "GL_MP_D3D_interfaces"/"glMPGetD3DInterface", and introduce
-          "GL_MP_MPGetNativeDisplay"/"glMPGetNativeDisplay" (this is a
-          backwards-compatible rename)
- --- mpv 0.11.0 ---
- --- mpv 0.10.0 ---
- 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 ---
+ --- mpv 0.8.0 is released ---
  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)
@@ -67,7 +38,7 @@ 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 ---
+ --- mpv 0.7.0 is released ---
  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
@@ -90,7 +61,7 @@ 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 ---
+ --- mpv 0.6.0 is released ---
  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
@@ -99,7 +70,7 @@ 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 ---
+ --- mpv 0.5.0 is released ---
  1.3    - add MPV_MAKE_VERSION()
  1.2    - remove "stream-time-pos" property (no replacement)
  1.1    - remap dvdnav:// to dvd://
@@ -111,6 +82,6 @@ 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 ---
+ --- mpv 0.4.0 is released ---
  1.0    - the API is declared stable
 

DOCS/client_api_examples/README.md

@@ -1,114 +1,30 @@
-# Client API examples
+Client API examples
+===================
 
 All these examples use the mpv client API through libmpv.
 
-## Where are the docs?
-
-The libmpv C API is documented directly in the header files (on normal Unix
-systems, this is in `/usr/include/mpv/client.h`.
-
-libmpv merely gives you access to mpv's command interface, which is documented
-here:
-* Options (`mpv_set_option()` and friends): http://mpv.io/manual/master/#options
-* Commands (`mpv_command()` and friends): http://mpv.io/manual/master/#list-of-input-commands
-* Properties (`mpv_set_property()` and friends): http://mpv.io/manual/master/#properties
-
-Essentially everything is done with them, including loading a file, retrieving
-playback progress, and so on.
-
-## Methods of embedding the video window
-
-All of these examples concentrate on how to integrate mpv's video renderers
-with your own GUI. This is generally the hardest part. libmpv enforces a
-somewhat efficient video output method, rather than e.g. returning a RGBA
-surface in memory for each frame. The latter would be prohibitively inefficient,
-because it would require conversion on the CPU. The goal is also not requiring
-the API users to reinvent their own video rendering/scaling/presentation
-mechanisms.
-
-There are currently 2 methods of embedding video.
-
-### Native window embedding
-
-This uses the platform's native method of nesting multiple windows. For example,
-Linux/X11 can nest a window from a completely different process. The nested
-window can redraw contents on its own, and receive user input if the user
-interacts with this window.
-
-libmpv lets you specify a parent window for its own video window via the `wid`
-option. Then libmpv will create its window with your window as parent, and
-render its video inside of it.
-
-This method is highly OS-dependent. Some behavior is OS-specific. There are
-problems with focusing on X11 (the ancient X11 focus policy mismatches with
-that of modern UI toolkits - it's normally worked around, but this is not
-easily possible with raw window embedding). It seems to have stability problems
-on OSX when using the Qt toolkit.
-
-### OpenGL embedding
-
-This method lets you use libmpv's OpenGL renderer directly. You create an
-OpenGL context, and then use `mpv_opengl_cb_draw()` to render the video on
-each frame.
-
-This is more complicated, because libmpv will work directly on your own OpenGL
-state. It's also not possible to have mpv automatically receive user input.
-You will have to simulate this with the `mouse`/`keypress`/`keydown`/`keyup`
-commands.
-
-You also get much more flexibility. For example, you can actually render your
-own OSD on top of the video, something that is not possible with raw window
-embedding.
-
-### Which one to use?
-
-Due to the various platform-specific behavior and problems (in particular on
-OSX), OpenGL embedding is recommended. If you're not comfortable with requiring
-OpenGL, or want to support "direct" video output such as vdpau (which might
-win when it comes to performance and energy-saving), you should probably
-support both methods if possible.
-
-## List of examples
-
-### simple
-
-Very primitive terminal-only example. Shows some most basic API usage.
-
-### cocoa
+cocoa
+-----
 
 Shows how to embed the mpv video window in Objective-C/Cocoa.
 
-### cocoa-openglcb
-
-Similar to the cocoa sample, but shows how to integrate mpv's OpenGL renderer
-using libmpv's opengl-cb API. Since it does not require complicated interaction
-with Cocoa elements from different libraries, it's more robust.
-
-### qt
+qt
+--
 
 Shows how to embed the mpv video window in Qt (using normal desktop widgets).
 
-### qt_opengl
+qml
+---
 
-Shows how to use mpv's OpenGL video renderer in Qt. This uses the opengl-cb API
-for video. Since it does not rely on embedding "foreign" native Windows, it's
-usually more robust, potentially faster, and it's easier to control how your
-GUI interacts with the video. You can do your own OpenGL rendering on top of
-the video as well.
+Shows how to use mpv's OpenGL video renderer in QtQuick2 with QML.
 
-### qml
-
-Shows how to use mpv's OpenGL video renderer in QtQuick2 with QML. Uses the
-opengl-cb API for video. Since the video is a normal QML element, it's trivial
-to create OSD overlays with QML-native graphical elements as well.
-
-### qml_direct
+qml_direct
+----------
 
 Alternative example, which typically avoids a FBO indirection. Might be
-slightly faster, but is less flexible and harder to use. In particular, the
-video is not a normal QML element. Uses the opengl-cb API for video.
+slightly faster, but is less flexible and harder to use.
 
-### sdl
+simple
+------
 
-Show how to embed the mpv OpenGL renderer in SDL. Uses the opengl-cb API for
-video.
+Very primitive terminal-only example. Shows some most basic API usage.

DOCS/client_api_examples/cocoa-openglcb/cocoa-openglcb.m

@@ -1,304 +0,0 @@
-// 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;
-}

DOCS/client_api_examples/cocoa/cocoabasic.m

@@ -1,7 +1,9 @@
-// Plays a video from the command line in a view provided by the client
-// application.
+// 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.
 
-// Build with: clang -o cocoabasic cocoabasic.m `pkg-config --libs --cflags mpv` -framework cocoa
+// Build with: clang -o cocoabasic cocoabasic.m `pkg-config --libs --cflags mpv`
 
 #include <mpv/client.h>
 
@@ -31,7 +33,6 @@ static inline void check_error(int status)
     mpv_handle *mpv;
     dispatch_queue_t queue;
     NSWindow *w;
-    NSView *wrapper;
 }
 @end
 
@@ -54,19 +55,11 @@ static void wakeup(void *);
     [self->w makeMainWindow];
     [self->w makeKeyAndOrderFront:nil];
 
-    NSRect frame = [[self->w contentView] bounds];
-    self->wrapper = [[NSView alloc] initWithFrame:frame];
-    [self->wrapper setAutoresizingMask:NSViewWidthSizable|NSViewHeightSizable];
-    [[self->w contentView] addSubview:self->wrapper];
-    [self->wrapper release];
-
     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: @"mpv_command('stop')" action:@selector(mpv_stop) keyEquivalent:@""];
-    [sm addItemWithTitle: @"mpv_command('quit')" action:@selector(mpv_quit) keyEquivalent:@""];
-    [sm addItemWithTitle: @"quit" action:@selector(terminate:) keyEquivalent:@"q"];
+    [sm addItemWithTitle: @"Shutdown mpv" action:@selector(shutdown) keyEquivalent:@"s"];
     [NSApp setMenu:m];
     [NSApp activateIgnoringOtherApps:YES];
 }
@@ -100,7 +93,7 @@ static void wakeup(void *);
             exit(1);
         }
 
-        int64_t wid = (intptr_t) self->wrapper;
+        int64_t wid = (intptr_t) [self->w contentView];
         check_error(mpv_set_option(mpv, "wid", MPV_FORMAT_INT64, &wid));
 
         // Maybe set some options here, like default key bindings.
@@ -129,28 +122,24 @@ static void wakeup(void *);
 - (void) handleEvent:(mpv_event *)event
 {
     switch (event->event_id) {
-    case MPV_EVENT_SHUTDOWN: {
-        mpv_detach_destroy(mpv);
+    case MPV_EVENT_SHUTDOWN:
+        // Clean up and shut down.
+        mpv_terminate_destroy(mpv);
         mpv = NULL;
-        printf("event: shutdown\n");
+        dispatch_async(dispatch_get_main_queue(), ^{
+            [[NSApplication sharedApplication] terminate:nil];
+        });
         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);
     }
 
-    case MPV_EVENT_VIDEO_RECONFIG: {
+    case MPV_EVENT_VIDEO_RECONFIG:
         dispatch_async(dispatch_get_main_queue(), ^{
-            NSArray *subviews = [self->wrapper subviews];
-            if ([subviews count] > 0) {
-                // mpv's events view
-                NSView *eview = [self->wrapper subviews][0];
-                [self->w makeFirstResponder:eview];
-            }
+            [self->w selectNextKeyView:nil];
         });
-    }
 
     default:
         printf("event: %s\n", mpv_event_name(event->event_id));
@@ -177,18 +166,13 @@ static void wakeup(void *context) {
 // Ostensibly, mpv's window would be hooked up to this.
 - (BOOL) windowShouldClose:(id)sender
 {
-    return NO;
+    [self shutdown];
+    if (self->w)
+        [self->w release];
+    return YES;
 }
 
-- (void) mpv_stop
-{
-    if (mpv) {
-        const char *args[] = {"stop", NULL};
-        mpv_command(mpv, args);
-    }
-}
-
-- (void) mpv_quit
+- (void) shutdown
 {
     if (mpv) {
         const char *args[] = {"quit", NULL};
@@ -205,5 +189,5 @@ int main(int argc, const char * argv[]) {
         app.delegate = delegate;
         [app run];
     }
-    return 0;
+    return EXIT_SUCCESS;
 }

DOCS/client_api_examples/qml/main.cpp

@@ -46,8 +46,9 @@ public:
     void render()
     {
         QOpenGLFramebufferObject *fbo = framebufferObject();
+        int vp[4] = {0, 0, fbo->width(), fbo->height()};
         window->resetOpenGLState();
-        mpv_opengl_cb_draw(mpv_gl, fbo->handle(), fbo->width(), fbo->height());
+        mpv_opengl_cb_render(mpv_gl, fbo->handle(), vp);
         window->resetOpenGLState();
     }
 };

DOCS/client_api_examples/qml/main.qml

@@ -50,7 +50,7 @@ Item {
                 text: "Make video look like on a Smart TV"
                 onClicked: {
                     if (checkbox.checked) {
-                        renderer.command(["vo_cmdline", "sharpen=5.0"])
+                        renderer.command(["vo_cmdline", "scale=sharpen3:scale-param1=5.0"])
                     } else {
                         renderer.command(["vo_cmdline", ""])
                     }

DOCS/client_api_examples/qml/mpvtest.pro

@@ -3,7 +3,6 @@ QT += qml quick
 HEADERS += main.h
 SOURCES += main.cpp
 
-QT_CONFIG -= no-pkg-config
 CONFIG += link_pkgconfig debug
 PKGCONFIG += mpv
 

DOCS/client_api_examples/qml_direct/main.cpp

@@ -39,13 +39,15 @@ void MpvRenderer::paint()
 {
     window->resetOpenGLState();
 
+    // Render to the whole window.
+    int vp[4] = {0, 0, size.width(), -size.height()};
+
     // This uses 0 as framebuffer, which indicates that mpv will render directly
     // to the frontbuffer. Note that mpv will always switch framebuffers
     // explicitly. Some QWindow setups (such as using QQuickWidget) actually
     // want you to render into a FBO in the beforeRendering() signal, and this
     // code won't work there.
-    // The negation is used for rendering with OpenGL's flipped coordinates.
-    mpv_opengl_cb_draw(mpv_gl, 0, size.width(), -size.height());
+    mpv_opengl_cb_render(mpv_gl, 0, vp);
 
     window->resetOpenGLState();
 }
@@ -94,8 +96,6 @@ void MpvObject::handleWindowChanged(QQuickWindow *win)
             this, &MpvObject::sync, Qt::DirectConnection);
     connect(win, &QQuickWindow::sceneGraphInvalidated,
             this, &MpvObject::cleanup, Qt::DirectConnection);
-    connect(win, &QQuickWindow::frameSwapped,
-            this, &MpvObject::swapped, Qt::DirectConnection);
     win->setClearBeforeRendering(false);
 }
 
@@ -110,11 +110,6 @@ void MpvObject::sync()
     renderer->size = window()->size() * window()->devicePixelRatio();
 }
 
-void MpvObject::swapped()
-{
-    mpv_opengl_cb_report_flip(mpv_gl, 0);
-}
-
 void MpvObject::cleanup()
 {
     if (renderer) {

DOCS/client_api_examples/qml_direct/main.h

@@ -37,7 +37,6 @@ public:
 public slots:
     void command(const QVariant& params);
     void sync();
-    void swapped();
     void cleanup();
 signals:
     void onUpdate();

DOCS/client_api_examples/qml_direct/mpvtest.pro

@@ -3,7 +3,6 @@ QT += qml quick
 HEADERS += main.h
 SOURCES += main.cpp
 
-QT_CONFIG -= no-pkg-config
 CONFIG += link_pkgconfig debug
 PKGCONFIG += mpv
 

DOCS/client_api_examples/qt/qtexample.pro

@@ -5,7 +5,6 @@ greaterThan(QT_MAJOR_VERSION, 4): QT += widgets
 TARGET = qtexample
 TEMPLATE = app
 
-QT_CONFIG -= no-pkg-config
 CONFIG += link_pkgconfig debug
 PKGCONFIG += mpv
 

DOCS/client_api_examples/qt_opengl/main.cpp

@@ -1,13 +0,0 @@
-#include <QApplication>
-#include "mainwindow.h"
-
-int main(int argc, char *argv[])
-{
-    QApplication a(argc, argv);
-    // Qt sets the locale in the QApplication constructor, but libmpv requires
-    // the LC_NUMERIC category to be set to "C", so change it back.
-    setlocale(LC_NUMERIC, "C");
-    MainWindow w;
-    w.show();
-    return a.exec();
-}

DOCS/client_api_examples/qt_opengl/mainwindow.cpp

@@ -1,52 +0,0 @@
-#include "mainwindow.h"
-#include "mpvwidget.h"
-#include <QPushButton>
-#include <QSlider>
-#include <QLayout>
-#include <QFileDialog>
-
-MainWindow::MainWindow(QWidget *parent) : QWidget(parent)
-{
-    m_mpv = new MpvWidget(this);
-    m_slider = new QSlider();
-    m_slider->setOrientation(Qt::Horizontal);
-    m_openBtn = new QPushButton("Open");
-    m_playBtn = new QPushButton("Pause");
-    QHBoxLayout *hb = new QHBoxLayout();
-    hb->addWidget(m_openBtn);
-    hb->addWidget(m_playBtn);
-    QVBoxLayout *vl = new QVBoxLayout();
-    vl->addWidget(m_mpv);
-    vl->addWidget(m_slider);
-    vl->addLayout(hb);
-    setLayout(vl);
-    connect(m_slider, SIGNAL(sliderMoved(int)), SLOT(seek(int)));
-    connect(m_openBtn, SIGNAL(clicked()), SLOT(openMedia()));
-    connect(m_playBtn, SIGNAL(clicked()), SLOT(pauseResume()));
-    connect(m_mpv, SIGNAL(positionChanged(int)), m_slider, SLOT(setValue(int)));
-    connect(m_mpv, SIGNAL(durationChanged(int)), this, SLOT(setSliderRange(int)));
-}
-
-void MainWindow::openMedia()
-{
-    QString file = QFileDialog::getOpenFileName(0, "Open a video");
-    if (file.isEmpty())
-        return;
-    m_mpv->command(QStringList() << "loadfile" << file);
-}
-
-void MainWindow::seek(int pos)
-{
-    m_mpv->command(QVariantList() << "seek" << pos << "absolute");
-}
-
-void MainWindow::pauseResume()
-{
-    const bool paused = m_mpv->getProperty("pause").toBool();
-    m_mpv->setProperty("pause", !paused);
-}
-
-void MainWindow::setSliderRange(int duration)
-{
-    m_slider->setRange(0, duration);
-}

DOCS/client_api_examples/qt_opengl/mainwindow.h

@@ -1,27 +0,0 @@
-#ifndef MainWindow_H
-#define MainWindow_H
-
-#include <QtWidgets/QWidget>
-
-class MpvWidget;
-class QSlider;
-class QPushButton;
-class MainWindow : public QWidget
-{
-    Q_OBJECT
-public:
-    explicit MainWindow(QWidget *parent = 0);
-public Q_SLOTS:
-    void openMedia();
-    void seek(int pos);
-    void pauseResume();
-private Q_SLOTS:
-    void setSliderRange(int duration);
-private:
-    MpvWidget *m_mpv;
-    QSlider *m_slider;
-    QPushButton *m_openBtn;
-    QPushButton *m_playBtn;
-};
-
-#endif // MainWindow_H

DOCS/client_api_examples/qt_opengl/mpvwidget.cpp

@@ -1,129 +0,0 @@
-#include "mpvwidget.h"
-#include <stdexcept>
-#include <QtGui/QOpenGLContext>
-#include <QtCore/QMetaObject>
-
-static void wakeup(void *ctx)
-{
-    QMetaObject::invokeMethod((MpvWidget*)ctx, "on_mpv_events", Qt::QueuedConnection);
-}
-
-static void *get_proc_address(void *ctx, const char *name) {
-    Q_UNUSED(ctx);
-    QOpenGLContext *glctx = QOpenGLContext::currentContext();
-    if (!glctx)
-        return NULL;
-    return (void *)glctx->getProcAddress(QByteArray(name));
-}
-
-MpvWidget::MpvWidget(QWidget *parent, Qt::WindowFlags f)
-    : QOpenGLWidget(parent, f)
-{
-    mpv = mpv::qt::Handle::FromRawHandle(mpv_create());
-    if (!mpv)
-        throw std::runtime_error("could not create mpv context");
-
-    mpv_set_option_string(mpv, "terminal", "yes");
-    mpv_set_option_string(mpv, "msg-level", "all=v");
-    if (mpv_initialize(mpv) < 0)
-        throw std::runtime_error("could not initialize mpv context");
-
-    // Make use of the MPV_SUB_API_OPENGL_CB API.
-    mpv::qt::set_option_variant(mpv, "vo", "opengl-cb");
-
-    // Request hw decoding, just for testing.
-    mpv::qt::set_option_variant(mpv, "hwdec", "auto");
-
-    mpv_gl = (mpv_opengl_cb_context *)mpv_get_sub_api(mpv, MPV_SUB_API_OPENGL_CB);
-    if (!mpv_gl)
-        throw std::runtime_error("OpenGL not compiled in");
-    mpv_opengl_cb_set_update_callback(mpv_gl, MpvWidget::on_update, (void *)this);
-    connect(this, SIGNAL(frameSwapped()), SLOT(swapped()));
-
-    mpv_observe_property(mpv, 0, "duration", MPV_FORMAT_DOUBLE);
-    mpv_observe_property(mpv, 0, "time-pos", MPV_FORMAT_DOUBLE);
-    mpv_set_wakeup_callback(mpv, wakeup, this);
-}
-
-MpvWidget::~MpvWidget()
-{
-    makeCurrent();
-    if (mpv_gl)
-        mpv_opengl_cb_set_update_callback(mpv_gl, NULL, NULL);
-    // Until this call is done, we need to make sure the player remains
-    // alive. This is done implicitly with the mpv::qt::Handle instance
-    // in this class.
-    mpv_opengl_cb_uninit_gl(mpv_gl);
-}
-
-void MpvWidget::command(const QVariant& params)
-{
-    mpv::qt::command_variant(mpv, params);
-}
-
-void MpvWidget::setProperty(const QString& name, const QVariant& value)
-{
-    mpv::qt::set_property_variant(mpv, name, value);
-}
-
-QVariant MpvWidget::getProperty(const QString &name) const
-{
-    return mpv::qt::get_property_variant(mpv, name);
-}
-
-void MpvWidget::initializeGL()
-{
-    int r = mpv_opengl_cb_init_gl(mpv_gl, NULL, get_proc_address, NULL);
-    if (r < 0)
-        throw std::runtime_error("could not initialize OpenGL");
-}
-
-void MpvWidget::paintGL()
-{
-    mpv_opengl_cb_draw(mpv_gl, defaultFramebufferObject(), width(), -height());
-}
-
-void MpvWidget::swapped()
-{
-    mpv_opengl_cb_report_flip(mpv_gl, 0);
-}
-
-void MpvWidget::on_mpv_events()
-{
-    // Process all events, until the event queue is empty.
-    while (mpv) {
-        mpv_event *event = mpv_wait_event(mpv, 0);
-        if (event->event_id == MPV_EVENT_NONE) {
-            break;
-        }
-        handle_mpv_event(event);
-    }
-}
-
-void MpvWidget::handle_mpv_event(mpv_event *event)
-{
-    switch (event->event_id) {
-    case MPV_EVENT_PROPERTY_CHANGE: {
-        mpv_event_property *prop = (mpv_event_property *)event->data;
-        if (strcmp(prop->name, "time-pos") == 0) {
-            if (prop->format == MPV_FORMAT_DOUBLE) {
-                double time = *(double *)prop->data;
-                Q_EMIT positionChanged(time);
-            }
-        } else if (strcmp(prop->name, "duration") == 0) {
-            if (prop->format == MPV_FORMAT_DOUBLE) {
-                double time = *(double *)prop->data;
-                Q_EMIT durationChanged(time);
-            }
-        }
-        break;
-    }
-    default: ;
-        // Ignore uninteresting or unknown events.
-    }
-}
-
-void MpvWidget::on_update(void *ctx)
-{
-    QMetaObject::invokeMethod((MpvWidget*)ctx, "update");
-}

DOCS/client_api_examples/qt_opengl/mpvwidget.h

@@ -1,38 +0,0 @@
-#ifndef PLAYERWINDOW_H
-#define PLAYERWINDOW_H
-
-#include <QtWidgets/QOpenGLWidget>
-#include <mpv/client.h>
-#include <mpv/opengl_cb.h>
-#include <mpv/qthelper.hpp>
-
-class MpvWidget Q_DECL_FINAL: public QOpenGLWidget
-{
-    Q_OBJECT
-public:
-    MpvWidget(QWidget *parent = 0, Qt::WindowFlags f = 0);
-    ~MpvWidget();
-    void command(const QVariant& params);
-    void setProperty(const QString& name, const QVariant& value);
-    QVariant getProperty(const QString& name) const;
-    QSize sizeHint() const { return QSize(480, 270);}
-Q_SIGNALS:
-    void durationChanged(int value);
-    void positionChanged(int value);
-protected:
-    void initializeGL() Q_DECL_OVERRIDE;
-    void paintGL() Q_DECL_OVERRIDE;
-private Q_SLOTS:
-    void swapped();
-    void on_mpv_events();
-private:
-    void handle_mpv_event(mpv_event *event);
-    static void on_update(void *ctx);
-
-    mpv::qt::Handle mpv;
-    mpv_opengl_cb_context *mpv_gl;
-};
-
-
-
-#endif // PLAYERWINDOW_H

DOCS/client_api_examples/qt_opengl/qt_opengl.pro

@@ -1,13 +0,0 @@
-CONFIG -= app_bundle
-QT += widgets
-
-QT_CONFIG -= no-pkg-config
-CONFIG += link_pkgconfig debug
-PKGCONFIG += mpv
-
-HEADERS = \
-    mpvwidget.h \
-    mainwindow.h
-SOURCES = main.cpp \
-    mpvwidget.cpp \
-    mainwindow.cpp

DOCS/client_api_examples/sdl/main.c

@@ -1,158 +0,0 @@
-// 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;
-}

DOCS/compile-windows.md

@@ -66,9 +66,9 @@ echo "MXE_TARGETS := i686-w64-mingw32.static" >> settings.mk
 # echo "MXE_TARGETS := x86_64-w64-mingw32.static" >> settings.mk
 
 # Build required packages. The following provide a minimum required to build
-# a reasonable mpv binary (though not an absolute minimum).
+# mpv.
 
-make gcc ffmpeg libass jpeg lua
+make gcc ffmpeg libass jpeg pthreads
 
 # Add MXE binaries to $PATH
 export PATH=/opt/mxe/usr/bin/:$PATH
@@ -79,7 +79,6 @@ export PATH=/opt/mxe/usr/bin/:$PATH
 cd ..
 git clone https://github.com/mpv-player/mpv.git
 cd mpv
-python ./bootstrap.py
 DEST_OS=win32 TARGET=i686-w64-mingw32.static ./waf configure
 # Or, if 64 bit version,
 # DEST_OS=win32 TARGET=x86_64-w64-mingw32.static ./waf configure
@@ -134,10 +133,10 @@ 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-lua51
+pacman -S mingw-w64-x86_64-ffmpeg mingw-w64-x86_64-libjpeg-turbo mingw-w64-x86_64-lua
 
 # Install additional (optional) dependencies
-pacman -S mingw-w64-x86_64-libdvdnav mingw-w64-x86_64-libguess
+pacman -S mingw-w64-x86_64-libdvdnav mingw-w64-x86_64-mpg123 mingw-w64-x86_64-libguess
 ```
 
 For a 32-bit build, install ``mingw-w64-i686-*`` packages instead.
@@ -173,3 +172,40 @@ mv -f /mingw64/bin/libmpv.dll.a /mingw64/lib/
 sed -i 's_/mingw64/bin_/mingw64/lib_' /mingw64/lib/pkgconfig/mpv.pc
 rmdir /mingw64/bin/pkgconfig
 ```
+
+Additional dependencies
+=======================
+
+pthreads
+--------
+
+mpv will use a pthreads wrapper by default. Either pthreads-win32 or
+winpthreads should work. The latter is packaged with most MinGW-w64
+environments, including MSYS2, so it shouldn't be a problem. If you don't have
+a pthreads wrapper or you want to build mpv without one, configure with:
+
+```bash
+./waf configure --enable-win32-internal-pthreads
+```
+
+libwaio
+-------
+
+If you want to use ``--input-file``, you need libwaio. It's available from
+git://midipix.org/waio
+
+To compile libwaio in MSYS2, run:
+
+```bash
+git clone git://midipix.org/waio && cd waio
+
+# 32-bit build, run from mingw32_shell.bat
+./build-mingw-nt32 lib-static CC=gcc AR=ar
+cp -r include/waio /mingw32/include
+cp lib32/libwaio.a /mingw32/lib
+
+# 64-bit build, run from mingw64_shell.bat
+./build-mingw-nt64 lib-static CC=gcc AR=ar
+cp -r include/waio /mingw64/include
+cp lib64/libwaio.a /mingw64/lib
+```

DOCS/contribute.md

@@ -4,8 +4,8 @@ How to contribute
 General
 -------
 
-The main contact for mpv development is IRC, specifically #mpv
-and #mpv-devel on Freenode.
+The main contact for mpv development is IRC, specifically #mpv-player
+and #mpv-player-dev on Freenode.
 
 Sending patches
 ---------------
@@ -14,11 +14,6 @@ Sending patches
   ``git format-patch``. diffs posted as pastebins (especially if the http link
   returns HTML) just cause extra work for everyone, because they lack commit
   message and authorship information.
-- 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).
-- When creating pull requests, be sure to test your changes. If you didn't,
-  please say so in the pull request message.
 - Write informative commit messages. Use present tense to describe the
   situation with the patch applied, and past tense for the situation before
   the change.
@@ -29,8 +24,8 @@ Sending patches
 
   For example, you fixed a crash in af_volume.c:
 
-  - Bad: ``fixed the bug (wtf?)``
-  - Good: ``af_volume: fix crash due to null pointer access``
+  Bad: ``fixed the bug (wtf?)``
+  Good: ``af_volume: fix crash due to null pointer access``
 
   Having a prefix gives context, and is especially useful when trying to find
   a specific change by looking at the history, or when running ``git blame``.
@@ -55,10 +50,7 @@ Sending patches
   something like reformatting a whole file, and hiding an actual functional
   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. Changes to the user
-  interface (options, properties, commands) should be documented in
-  interface-changes.rst. Changes to libmpv should be documented in
-  client-api-changes.rst.
+  add a new input property, document it in input.rst.
 
 Code formatting
 ---------------
@@ -126,8 +118,6 @@ mpv uses C99 with K&R formatting, with some exceptions.
         very_long_condition_b)
     {
         code();
-    } else {
-        ...
     }
     ```
 - Remove any trailing whitespace.

DOCS/interface-changes.rst

@@ -1,115 +0,0 @@
-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 0.15.0 ---
-    - change "yadif" video filter defaults
- --- mpv 0.14.0 ---
-    - vo_opengl interpolation now requires --video-sync=display-... to be set
-    - change some vo_opengl defaults (including changing tscale)
-    - add "vsync-ratio", "estimated-display-fps" properties
-    - add --rebase-start-time option
-      This is a breaking change to start time handling. Instead of making start
-      time handling an aspect of different options and properties (like
-      "time-pos" vs. "playback-time"), make it dependent on the new option. For
-      compatibility, the "time-start" property now always returns 0, so code
-      which attempted to handle rebasing manually will not break.
- --- mpv 0.13.0 ---
-    - remove VO opengl-cb frame queue suboptions (no replacement)
- --- mpv 0.12.0 ---
-    - remove --use-text-osd (useless; fontconfig isn't a requirement anymore,
-      and text rendering is also lazily initialized)
-    - some time properties (at least "playback-time", "time-pos",
-      "time-remaining", "playtime-remaining") now are unavailable if the time
-      is unknown, instead of just assuming that the internal playback position
-      is 0
-    - add --audio-fallback-to-null option
-    - replace vf_format outputlevels suboption with "video-output-levels" global
-      property/option; also remove "colormatrix-output-range" property
-    - vo_opengl: remove sharpen3/sharpen5 scale filters, add sharpen sub-option
- --- 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 ---

DOCS/man/af.rst

@@ -11,10 +11,6 @@ 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.
 
@@ -63,18 +59,11 @@ 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.
 
-``lavcac3enc[=tospdif[:bitrate[:minch]]]``
+``lavcac3enc[=tospdif[:bitrate[:minchn]]]``
     Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports
     16-bit native-endian input format, maximum 6 channels. The output is
     big-endian when outputting a raw AC-3 stream, native-endian when
@@ -103,10 +92,63 @@ Available filters are:
         :5ch: 448
         :6ch: 448
 
-    ``minch=<n>``
-        If the input channel number is less than ``<minch>``, the filter will
+    ``minchn=<n>``
+        If the input channel number is less than ``<minchn>``, 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
@@ -164,7 +206,7 @@ Available filters are:
 
     .. admonition:: Examples
 
-        ``mpv --af=channels=4:[0-1,1-0,2-2,3-3] media.avi``
+        ``mpv --af=channels=4:[0-1,1-0,0-2,1-3] media.avi``
             Would change the number of channels to 4 and set up 4 routes that
             swap channel 0 and channel 1 and leave channel 2 and 3 intact.
             Observe that if media containing two channels were played back,
@@ -183,21 +225,14 @@ Available filters are:
         automatically up- and downmix standard channel layouts.
 
 ``format=format:srate:channels:out-format:out-srate:out-channels``
-    Does not do any format conversion itself. Rather, it may cause the
-    filter system to insert necessary conversion filters before or after this
-    filter if needed. It is primarily useful for controlling the audio format
-    going into other filters. To specify the format for audio output, see
-    ``--audio-format``, ``--audio-samplerate``, and ``--audio-channels``. This
-    filter is able to force a particular format, whereas ``--audio-*``
-    may be overridden by the ao based on output compatibility.
+    Force a specific audio format/configuration without actually changing the
+    audio data. Keep in mind that the filter system might auto-insert actual
+    conversion filters before or after this filter if needed.
 
     All parameters are optional. The first 3 parameters restrict what the filter
-    accepts as input. They will therefore cause conversion filters to be
-    inserted before this one.  The ``out-`` parameters tell the filters or audio
-    outputs following this filter how to interpret the data without actually
-    doing a conversion. Setting these will probably just break things unless you
-    really know you want this for some reason, such as testing or dealing with
-    broken media.
+    accepts as input. The ``out-`` parameters change the audio format, without
+    actually doing a conversion. The data will be 'reinterpreted' by the
+    filters or audio outputs following this filter.
 
     ``<format>``
         Force conversion to this format. Use ``--af=format=format=help`` to get
@@ -217,9 +252,21 @@ Available filters are:
 
     ``<out-channels>``
 
-    *NOTE*: this filter used to be named ``force``. The old ``format`` filter
-    used to do conversion itself, unlike this one which lets the filter system
-    handle the conversion.
+    See also ``--audio-format``, ``--audio-samplerate``, and
+    ``--audio-channels`` for related options. Keep in mind that
+    ``--audio-channels`` does not actually force the number of
+    channels in most cases, while this filter can do this.
+
+    *NOTE*: this filter used to be named ``force``. Also, unlike the old
+    ``format`` filter, this does not do any actual conversion anymore.
+    Conversion is done by other, automatically inserted filters.
+
+``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
@@ -245,10 +292,6 @@ 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
@@ -312,6 +355,64 @@ 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
@@ -340,6 +441,36 @@ 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``.)
@@ -363,6 +494,45 @@ 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).
@@ -395,10 +565,9 @@ Available filters are:
              Scale tempo in sync with speed (default).
         pitch
              Reverses effect of filter. Scales pitch without altering tempo.
-             Add this to your ``input.conf`` to step by musical semi-tones::
-
-                [ multiply speed 0.9438743126816935
-                ] multiply speed 1.059463094352953
+             Add ``[ multiply speed 0.9438743126816935`` and
+             ``] multiply speed 1.059463094352953`` to your ``input.conf``
+             to step by musical semi-tones.
 
              .. warning::
 
@@ -431,20 +600,6 @@ Available filters are:
             Changing playback speed would change pitch, leaving audio tempo at
             1.2x.
 
-``rubberband``
-    High quality pitch correction with librubberband. This can be used in place
-    of ``scaletempo``, and will be used to adjust audio pitch when playing
-    at speed different from normal.
-
-    This filter has a number of sub-options. You can list them with
-    ``mpv --af=rubberband=help``. This will also show the default values
-    for each option. The options are not documented here, because they are
-    merely passed to librubberband. Look at the librubberband documentation
-    to learn what each option does:
-    http://breakfastquay.com/rubberband/code-doc/classRubberBand_1_1RubberBandStretcher.html
-    (The mapping of the mpv rubberband filter sub-option names and values to
-    those of librubberband follows a simple pattern: ``"Option" + Name + Value``.)
-
 ``lavfi=graph``
     Filter audio using FFmpeg's libavfilter.
 
@@ -459,3 +614,4 @@ Available filters are:
 
     ``o=<string>``
         AVOptions.
+

DOCS/man/ao.rst

@@ -20,8 +20,10 @@ normal driver parameters.
 
     See ``--ao=help`` for a list of compiled-in audio output drivers. The
     driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
-    where PulseAudio is used. On BSD systems, ``--ao=oss`` or ``--ao=sndio``
-    may work (the latter being experimental).
+    where PulseAudio is used. On Windows, ``--ao=wasapi`` is preferred,
+    though it might cause trouble sometimes, in which case ``--ao=dsound``
+    should be used. On BSD systems, ``--ao=oss`` or `--ao=sndio`` may work
+    (the latter being experimental). On OS X systems, use ``--ao=coreaudio``.
 
 .. admonition:: Examples
 
@@ -53,14 +55,6 @@ 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.
-    ``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,
-        or for static setups with a specially engineered ALSA configuration (in
-        this case you should always force the same layout with ``--audio-channels``,
-        or it will work only for files which use the layout implicit to your
-        ALSA device).
 
     .. note::
 
@@ -105,10 +99,8 @@ Available audio output drivers are:
         ``--audio-device`` to select the device (use ``--audio-device=help``
         to get a list of all devices and their mpv name).
 
-        You can also try `using the upmix plugin <http://git.io/vfuAy>`_.
-        This setup enables multichannel audio on the ``default`` device
-        with automatic upmixing with shared access, so playing stereo
-        and multichannel audio at the same time will work as expected.
+        You can also try
+        `Using the upmix plugin <https://github.com/mpv-player/mpv/wiki/ALSA:-Surround-Sound-and-Upmixing>`_.
 
 ``oss``
     OSS audio output driver
@@ -137,13 +129,16 @@ Available audio output drivers are:
         Automatically create connections to output ports (default: enabled).
         When enabled, the maximum number of output channels will be limited to
         the number of available output ports.
-    ``std-channel-layout=waveext|any``
-        Select the standard channel layout (default: waveext). JACK itself has no
+    ``std-channel-layout=alsa|waveext|any``
+        Select the standard channel layout (default: alsa). JACK itself has no
         notion of channel layouts (i.e. assigning which speaker a given
         channel is supposed to map to) - it just takes whatever the application
         outputs, and reroutes it to whatever the user defines. This means the
         user and the application are in charge of dealing with the channel
-        layout. ``waveext`` uses WAVE_FORMAT_EXTENSIBLE order, which, even
+        layout. ``alsa`` uses the old MPlayer layout, which is inspired by
+        ALSA's standard layouts. In this mode, ao_jack will refuse to play 3
+        or 7 channels (because these do not really have a defined meaning in
+        MPlayer). ``waveext`` uses WAVE_FORMAT_EXTENSIBLE order, which, even
         though it was defined by Microsoft, is the standard on many systems.
         The value ``any`` makes JACK accept whatever comes from the audio
         filter chain, regardless of channel layout and without reordering. This
@@ -157,23 +152,12 @@ 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 affect 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
 
@@ -203,7 +187,19 @@ Available audio output drivers are:
         (it used to be required to get good behavior on old PulseAudio versions).
 
         If you have stuttering video when using pulse, try to enable this
-        option. (Or try to update PulseAudio.)
+        option. (Or alternatively, try to update PulseAudio.)
+
+``dsound`` (Windows only)
+    DirectX DirectSound audio output driver
+
+    .. note:: This driver is for compatibility with old systems.
+
+    ``device=<devicenum>``
+        Sets the device number to use. Playing a file with ``-v`` will show a
+        list of available devices.
+
+    ``buffersize=<ms>``
+        DirectSound buffer size in milliseconds (default: 200).
 
 ``sdl``
     SDL 1.2+ audio output driver. Should work on any platform supported by SDL
@@ -242,7 +238,7 @@ Available audio output drivers are:
     ``speed``
         Simulated audio playback speed as a multiplier. Usually, a real audio
         device will not go exactly as fast as the system clock. It will deviate
-        just a little, and this option helps to simulate this.
+        just a little, and this option helps simulating this.
 
     ``latency``
         Simulated device latency. This is additional to EOF.
@@ -254,10 +250,6 @@ 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
 
@@ -302,19 +294,19 @@ Available audio output drivers are:
 ``wasapi``
     Audio output to the Windows Audio Session API.
 
+    ``device=<id>``
+        Uses the requested endpoint instead of the system's default audio
+        endpoint. Both the number and the ID String are valid; the ID String
+        is guaranteed to not change unless the driver is uninstalled.
+
+        Also supports searching active devices by name. If more than one
+        device matches the name, refuses loading it.
+
+        To get a list of the valid devices, give ``help`` as the id. The
+        list is the same as the ``list`` suboption, but stops the player
+        initialization.
     ``exclusive``
         Requests exclusive, direct hardware access. By definition prevents
         sound playback of any other program until mpv exits.
-    ``device=<id>``
-        Uses the requested endpoint instead of the system's default audio
-        endpoint. Both an ordinal number (0,1,2,...) and the GUID
-        String are valid; the GUID string is guaranteed to not change
-        unless the driver is uninstalled.
-
-        Also supports searching active devices by human-readable name. If more
-        than one device matches the name, refuses loading it.
-
-        This option is mostly deprecated in favour of the more general
-        ``--audio-device`` option. That said, ``--audio-device=help`` will give
-        a list of valid device GUIDs (prefixed with ``wasapi/``), as well as
-        their human readable names, which should work here.
+    ``list``
+        Lists all audio endpoints (output devices) present in the system.

DOCS/man/encode.rst

@@ -93,8 +93,8 @@ You can encode files from one format/codec to another using this facility.
         Completely empties the options list.
 
 ``--oafirst``
-    Force the audio stream to become the first stream in the output.
-    By default, the order is unspecified.
+    Force the audio stream to become the first stream in the output. By default
+    the order is unspecified.
 
 ``--ovc=<codec>``
     Specifies the output video codec. This can be a comma separated list of
@@ -111,7 +111,7 @@ You can encode files from one format/codec to another using this facility.
 
     .. admonition:: Examples
 
-        ``"--ovc=mpeg4 --ovcopts=qscale=5"``
+        ``"--ovc=mpeg4 --oacopts=qscale=5"``
             selects constant quantizer scale 5 for MPEG-4 encoding.
 
         ``"--ovc=libx264 --ovcopts=crf=23"``
@@ -134,8 +134,8 @@ You can encode files from one format/codec to another using this facility.
         Completely empties the options list.
 
 ``--ovfirst``
-    Force the video stream to become the first stream in the output.
-    By default, the order is unspecified.
+    Force the video stream to become the first stream in the output. By default
+    the order is unspecified.
 
 ``--ocopyts``
     Copies input pts to the output video (not supported by some output

DOCS/man/ipc.rst

@@ -79,27 +79,6 @@ 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``).
-
 If the first character (after skipping whitespace) is not ``{``, the command
 will be interpreted as non-JSON text command, as they are used in input.conf
 (or ``mpv_command_string()`` in the client API). Additionally, line starting
@@ -111,7 +90,7 @@ rely on this.
 Commands
 --------
 
-In addition to the commands described in `List of Input Commands`_, a few
+Additionally to  the commands described in `List of Input Commands`_, a few
 extra commands can also be used as part of the protocol:
 
 ``client_name``
@@ -189,7 +168,7 @@ extra commands can also be used as part of the protocol:
 
 ``unobserve_property``
     Undo ``observe_property`` or ``observe_property_string``. This requires the
-    numeric id passed to the observed command as argument.
+    numeric id passed to the observe command as argument.
 
     Example:
 
@@ -228,9 +207,7 @@ extra commands can also be used as part of the protocol:
 
 ``get_version``
     Returns the client API version the C API of the remote mpv instance
-    provides.
-
-    See also: ``DOCS/client-api-changes.rst``.
+    provides. (Also see ``DOCS/client-api-changes.rst``.)
 
 UTF-8
 -----

DOCS/man/lua.rst

@@ -73,10 +73,6 @@ 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, ...)``
@@ -98,9 +94,6 @@ 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
@@ -198,7 +191,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
     be called (unless the user remapped the key with another binding).
 
     The ``name`` argument should be a short symbolic string. It allows the user
-    to remap the key binding via input.conf using the ``script-message``
+    to remap the key binding via input.conf using the ``script_message``
     command, and the name of the key binding (see below for
     an example). The name should be unique across other bindings in the same
     script - if not, the previous binding with the same name will be
@@ -220,8 +213,8 @@ The ``mp`` module is preloaded, although it can be loaded manually with
             has an ``is_mouse`` entry, which tells whether the event was caused
             by a mouse button.
 
-    Internally, key bindings are dispatched via the ``script-message-to`` or
-    ``script-binding`` input commands and ``mp.register_script_message``.
+    Internally, key bindings are dispatched via the ``script_message_to`` or
+    ``script_binding`` input commands and ``mp.register_script_message``.
 
     Trying to map multiple commands to a key will essentially prefer a random
     binding, while the other bindings are not called. It is guaranteed that
@@ -244,7 +237,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 
     ::
 
-        y script-binding something
+        y script_binding something
 
 
     This will print the message when the key ``y`` is pressed. (``x`` will
@@ -255,7 +248,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 
     ::
 
-        y script-binding fooscript.something
+        y script_binding fooscript.something
 
 ``mp.add_forced_key_binding(...)``
     This works almost the same as ``mp.add_key_binding``, but registers the
@@ -333,6 +326,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
     the timer is re-added after the function fn is run.
 
     Returns a timer object. The timer object provides the following methods:
+
         ``stop()``
             Disable the timer. Does nothing if the timer is already disabled.
             This will remember the current elapsed time when stopping, so that
@@ -363,23 +357,6 @@ The ``mp`` module is preloaded, although it can be loaded manually with
             (``true``). This value is used when the timer expires (but before
             the timer callback function fn is run).
 
-    Note that these are method, and you have to call them using ``:`` instead
-    of ``.`` (Refer to http://www.lua.org/manual/5.2/manual.html#3.4.9 .)
-
-    Example:
-
-    ::
-
-        seconds = 0
-        timer = mp.add_periodic_timer(1, function()
-            print("called every second")
-            # stop it after 10 seconds
-            seconds = seconds + 1
-            if seconds >= 10 then
-                timer:kill()
-            end
-        end)
-
 
 ``mp.get_opt(key)``
     Return a setting from the ``--script-opts`` option. It's up to the user and
@@ -389,7 +366,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 ``mp.get_script_name()``
     Return the name of the current script. The name is usually made of the
     filename of the script, with directory and file extension removed. If
-    there are several scripts which would have the same name, it's made unique
+    there are several script which would have the same name, it's made unique
     by appending a number.
 
     .. admonition:: Example
@@ -518,8 +495,8 @@ Example implementation::
         optionB = -0.5,
         optionC = true,
     }
-    read_options(options, "myscript")
-    print(options.optionA)
+    options.read_options(options, "myscript")
+    print(option.optionA)
 
 
 The config file will be stored in ``lua-settings/identifier.conf`` in mpv's user
@@ -630,10 +607,6 @@ 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])``
@@ -650,14 +623,6 @@ strictly part of the guaranteed API.
     trailing text is returned as 3rd return value. (The 3rd return value is
     always there, but with ``trail`` set, no error is raised.)
 
-``utils.format_json(v)``
-    Format the given Lua table (or value) as a JSON string and return it. On
-    error, returns ``nil, error``. (Errors usually only happen on value types
-    incompatible with JSON.)
-
-    The argument value uses similar conventions as ``mp.set_property_native()``
-    to distinguish empty objects and arrays.
-
 ``utils.to_string(v)``
     Turn the given value into a string. Formats tables and their contents. This
     doesn't do anything special; it is only needed because Lua is terrible.
@@ -694,33 +659,6 @@ 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.
 

DOCS/man/mpv.rst

@@ -32,10 +32,8 @@ INTERACTIVE CONTROL
 ===================
 
 mpv has a fully configurable, command-driven control layer which allows you
-to control mpv using keyboard, mouse, or remote control (there is no
-LIRC support - configure remotes as input devices instead).
-
-See the ``--input-`` options for ways to customize it.
+to control mpv using keyboard, mouse, joystick or remote control (with
+LIRC). See the ``--input-`` options for ways to customize it.
 
 Keyboard Control
 ----------------
@@ -50,7 +48,7 @@ UP and DOWN
 
 Ctrl+LEFT and Ctrl+RIGHT
     Seek to the previous/next subtitle. Subject to some restrictions and
-    might not always work; see ``sub_seek`` command.
+    might not work always; see ``sub_seek`` command.
 
 [ and ]
     Decrease/increase current playback speed by 10%.
@@ -116,7 +114,11 @@ o (also P)
     Show progression bar, elapsed time and total duration on the OSD.
 
 O
-    Toggle OSD states between normal and playback time/duration.
+    Toggle OSD states: none / seek / seek + timer / seek + timer + total time.
+
+d
+    Toggle frame dropping states: none / skip display / skip decoding (see
+    ``--framedrop``).
 
 v
     Toggle subtitle visibility.
@@ -128,19 +130,11 @@ x and z
     Adjust subtitle delay by +/- 0.1 seconds.
 
 l
-    Set/clear A-B loop points. See ``ab-loop`` command for details.
-
-L
-    Toggle infinite looping.
+    Set/clear A-B loop points. See ``ab_loop`` command for details.
 
 Ctrl + and Ctrl -
     Adjust audio delay by +/- 0.1 seconds.
 
-u
-    Switch between applying no style overrides to SSA/ASS subtitles, and
-    overriding them almost completely with the normal subtitle style. See
-    ``--ass-style-override`` for more info.
-
 V
     Toggle subtitle VSFilter aspect compatibility mode. See
     ``--ass-vsfilter-aspect-compat`` for more info.
@@ -155,10 +149,6 @@ 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.
 
@@ -171,7 +161,7 @@ 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
@@ -192,19 +182,23 @@ corresponding adjustment, or the software equalizer (``--vf=eq``).)
 7 and 8
     Adjust saturation.
 
-Alt+0 (and command+0 on OSX)
-    Resize video window to half its original size.
+(The following keys are valid only on OS X.)
 
-Alt+1 (and command+1 on OSX)
+command + 0
+    Resize video window to half its original size.
+    (On other platforms, you can bind keys to change the ``window-scale``
+    property.)
+
+command + 1
     Resize video window to its original size.
 
-Alt+2 (and command+2 on OSX)
+command + 2
     Resize video window to double its original size.
 
-command + f (OSX only)
+command + f
     Toggle fullscreen (see also ``--fs``).
 
-command + [ and command + ] (OSX only)
+command + [ and command + ]
     Set video window alpha.
 
 (The following keys are valid if you have a keyboard with multimedia keys.)
@@ -222,10 +216,7 @@ PREVIOUS and NEXT
 support.)
 
 h and k
-    Select previous/next tv-channel.
-
-H and K
-    Select previous/next dvb-channel.
+    Select previous/next channel.
 
 Mouse Control
 -------------
@@ -261,7 +252,7 @@ parser puts several options into a single string, and passes them to a
 component at once, instead of using multiple options on the level of the
 command line.
 
-The suboption parser can quote strings with ``"`` and ``[...]``.
+The suboption parser can quote strings with ``"``, ``'``, and ``[...]``.
 Additionally, there is a special form of quoting with ``%n%`` described below.
 
 For example, the ``opengl`` VO can take multiple options:
@@ -275,19 +266,17 @@ 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 receives the ``"``
+so the example quotes the string twice, ensuring that mpv recieves the ``"``
 quotes.
 
-The ``[...]`` form of quotes wraps everything between ``[`` and ``]``. It's
+The ``[...]`` from of quotes wraps everything between ``[`` and ``]``. It's
 useful with shells that don't interpret these characters in the middle of
-an argument (like bash). These quotes are balanced (since mpv 0.9.0): the ``[``
-and ``]`` nest, and the quote terminates on the last ``]`` that has no matching
-``[`` within the string. (For example, ``[a[b]c]`` results in ``a[b]c``.)
+an argument (like bash).
 
-The fixed-length quoting syntax is intended for use with external
-scripts and programs.
+A special kind of string-escaping intended for use with external scripts and
+programs is started with ``%``.
 
-It is started with ``%`` and has the following format::
+It has the following format::
 
     %n%string_of_length_n
 
@@ -304,7 +293,7 @@ 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 lets you pass suboptions as a nested data structure
+``MPV_FORMAT_NODE``, which let's you pass suboptions as a nested data structure
 of maps and arrays. (``--vo`` supports this in the same way, although this
 fact is undocumented.)
 
@@ -323,23 +312,12 @@ console controls. (Which makes it suitable for playing data piped to stdin.)
 
 For paths passed to suboptions, the situation is further complicated by the
 need to escape special characters. To work this around, the path can be
-additionally wrapped in the fixed-length syntax, e.g. ``%n%string_of_length_n``
-(see above).
+additionally wrapped in the ``%n%string_of_length_n`` syntax (see above).
 
 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.) 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)
-================ ===============================================================
-
+required.)
 
 Per-File Options
 ----------------
@@ -390,12 +368,10 @@ 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``. For details and platform
-specifics (in particular Windows paths) see the `FILES`_ section.
-
+user-specific one is ``~/.config/mpv/mpv.conf``.
 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
+``option=<value>``; everything after a *#* is considered a comment. Options
 that work without values can be enabled by setting them to *yes* and disabled by
 setting them to *no*. Even suboptions can be specified in this way.
 
@@ -413,11 +389,11 @@ Escaping spaces and special characters
 
 This is done like with command line options. The shell is not involved here,
 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 ``"``,
-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 do
-this manually. (This is a mess and should probably be changed at some point.)
+characters like spaces. A config entry can be quoted with ``"`` and ``'``,
+as well as with the ``%n%`` syntax 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 this
+manually. (This is a mess and should probably be changed at some point.)
 
 Putting Command Line Options into the Configuration File
 --------------------------------------------------------
@@ -460,38 +436,7 @@ description (shown by ``--profile=help``) can be defined with the
 ``profile-desc`` option. To end the profile, start another one or use the
 profile name ``default`` to continue with normal options.
 
-.. admonition:: Example mpv config file with profiles
-
-    ::
-
-        # normal top-level option
-        fullscreen=yes
-
-        # a profile that can be enabled with --profile=big-cache
-        [big-cache]
-        cache=123400
-        demuxer-readahead-secs=20
-
-        [slow]
-        profile-desc="some profile name"
-        vo=opengl:scale=ewa_lanczos:scale-radius=16
-
-        [fast]
-        vo=vdpau
-
-        # using a profile again extends it
-        [slow]
-        framedrop=no
-        # you can also include other profiles
-        profile=big-cache
-
-
-Auto profiles
--------------
-
-Some profiles are loaded automatically. The following example demonstrates this:
-
-.. admonition:: Auto profile loading
+.. admonition:: Example mpv profile
 
     ::
 
@@ -510,29 +455,24 @@ Some profiles are loaded automatically. The following example demonstrates this:
         [ao.alsa]
         device=spdif
 
-The profile name follows the schema ``type.name``, where type can be ``vo``
-to match the value the ``--vo`` option is set to, ``ao`` for ``--ao``,
-``protocol`` for the input/output protocol in use (see ``--list-protocols``),
-and ``extension`` for the extension of the path of the currently played file
-(*not* the file format).
-
-This feature is very limited, and there are no other auto profiles.
 
 TAKING SCREENSHOTS
 ==================
 
 Screenshots of the currently played file can be taken using the 'screenshot'
 input mode command, which is by default bound to the ``s`` key. Files named
-``mpv-shotNNNN.jpg`` will be saved in the working directory, using the first
-available number - no files will be overwritten. In pseudo-GUI mode, the
-screenshot will be saved somewhere else. See `PSEUDO GUI MODE`_.
+``shotNNNN.jpg`` will be saved in the working directory, using the first
+available number - no files will be overwritten.
 
 A screenshot will usually contain the unscaled video contents at the end of the
 video filter chain and subtitles. By default, ``S`` takes screenshots without
 subtitles, while ``s`` includes subtitles.
 
-Unlike with MPlayer, the ``screenshot`` video filter is not required. This
-filter was never required in mpv, and has been removed.
+The ``screenshot`` video filter is not required when using a recommended GUI
+video output driver. It should normally not be added to the config file, as
+taking screenshots is handled by the VOs, and adding the screenshot filter will
+break hardware decoding. (The filter may still be useful for taking screenshots
+at a certain point within the video chain when using multiple video filters.)
 
 TERMINAL STATUS LINE
 ====================
@@ -565,26 +505,19 @@ 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: 2.500/13``, where the first number is average
-  number of vsyncs per video frame (e.g. 2.5 when playing 24Hz videos on 60Hz
-  screens), which might jitter if the ratio doesn't round off, or there are
-  mistimed frames (``vsync-ratio``), and the second number of estimated number
-  of vsyncs which took too long (``vo-delayed-frame-count`` property). The
-  latter is a heuristic, as it's generally not possible to determine this with
-  certainty.
 - 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. May also be incremented on "hiccups" and when the 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 frame dropping is enabled with the ``--framedrop`` options.
-  (``drop-frame-count`` property.)
+  to the display as well, e.g.: ``Dropped: 4/34``. This should almost never
+  happen, unless decoder-framedropping is enabled with one of the
+  ``--framedrop`` options, the stream contains errors, or a weird codec is in
+  use. (``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,
   the second value shows *additional* data buffered in the stream cache in
-  kilobytes. (``demuxer-cache-duration`` and ``cache-used`` properties.)
+  kilobytes. (``demuxer-cache-duration`` and ``cached-used`` properties.)
 
 
 PROTOCOLS
@@ -606,13 +539,17 @@ PROTOCOLS
     Play a path from  Samba share.
 
 ``bd://[title][/device]`` ``--bluray-device=PATH``
-    Play a Blu-ray disc. Currently, this does not accept ISO files. Instead,
+    Play a Blu-Ray disc. Currently, this does not accept ISO files. Instead,
     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. DVD menus are not supported. If no title is given, the longest
-    title is auto-selected.
+    Play a DVD. If you want dvdnav menus, use ``dvd://menu``. If no title
+    is given, the longest title is auto-selected.
 
     ``dvdnav://`` is an old alias for ``dvd://`` and does exactly the same
     thing.
@@ -657,10 +594,6 @@ PROTOCOLS
     ``PATH`` itself should start with a third ``/`` to make the path an
     absolute path.
 
-``fd://123``
-    Read data from the given file descriptor (for example 123). This is similar
-    to piping data to stdin via ``-``, but can use an arbitrary file descriptor.
-
 ``edl://[edl specification as in edl-mpv.rst]``
     Stitch together parts of multiple files and play them.
 
@@ -670,40 +603,6 @@ PROTOCOLS
 ``memory://data``
     Use the ``data`` part as source data.
 
-PSEUDO GUI MODE
-===============
-
-mpv has no official GUI, other than the OSC (`ON SCREEN CONTROLLER`_), which
-is not a full GUI and is not meant to be. However, to compensate for the lack
-of expected GUI behavior, mpv will in some cases start with some settings
-changed to behave slightly more like a GUI mode.
-
-Currently this happens only in the following cases:
-
-- if started using the ``mpv.desktop`` file on Linux (e.g. started from menus
-  or file associations provided by desktop environments)
-- if started from explorer.exe on Windows (technically, if it was started on
-  Windows, and all of the stdout/stderr/stdin handles are unset)
-- manually adding ``--profile=pseudo-gui`` to the command line
-
-This mode implicitly adds ``--profile=pseudo-gui`` to the command line, with
-the ``pseudo-gui`` profile being predefined with the following contents:
-
-::
-
-    [pseudo-gui]
-    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
-enhance the default profile, rather than overwrite it.
-
-The profile always overrides other settings in ``mpv.conf``.
-
-
 .. include:: options.rst
 
 .. include:: ao.rst
@@ -726,15 +625,6 @@ The profile always overrides other settings in ``mpv.conf``.
 
 .. include:: changes.rst
 
-
-EMBEDDING INTO OTHER PROGRAMS (LIBMPV)
-======================================
-
-mpv can be embedded into other programs as video/audio playback backend. The
-recommended way to do so is using libmpv. See ``libmpv/client.h`` in the mpv
-source code repository. This provides a C API. Bindings for other languages
-might be available (see wiki).
-
 ENVIRONMENT VARIABLES
 =====================
 
@@ -806,7 +696,7 @@ libdvdcss:
 
         key
            is the default method. libdvdcss will use a set of calculated
-           player keys to try to get the disc key. This can fail if the drive
+           player keys to try and get the disc key. This can fail if the drive
            does not recognize any of the player keys.
 
         disc
@@ -859,8 +749,6 @@ 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``
@@ -869,15 +757,11 @@ 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
-    directory, while most Linux distributions will set it to ``/etc/mpv/``).
+    mpv system-wide settings (depends on ``--prefix`` passed to configure)
 
 ``~/.config/mpv/mpv.conf``
-    mpv user settings (see `CONFIGURATION FILES`_ section)
+    mpv user settings
 
 ``~/.config/mpv/input.conf``
     key bindings (see `INPUT.CONF`_ section)
@@ -922,7 +806,7 @@ locations are different. They are generally located under ``%APPDATA%/mpv/``.
 For example, the path to mpv.conf is ``%APPDATA%/mpv/mpv.conf``, which maps to
 a system and user-specific path, for example
 
-    ``C:\users\USERNAME\AppData\Roaming\mpv\mpv.conf``
+    ``C:\users\USERNAME\Application Data\mpv\mpv.conf``
 
 You can find the exact path by running ``echo %APPDATA%\mpv\mpv.conf`` in cmd.exe.
 
@@ -932,23 +816,61 @@ 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
 one from the directory with higher priority will be loaded.
 
-A third config directory with the lowest priority is the directory named ``mpv``
-in the same directory as ``mpv.exe``. This used to be the directory with the
+A third config directory with lowest priority is the directory named ``mpv``
+in the same directory as ``mpv.exe``. This used to be the directory with
 highest priority, but is now discouraged to use and might be removed in the
 future.
 
 Note that mpv likes to mix ``/`` and ``\`` path separators for simplicity.
 kernel32.dll accepts this, but cmd.exe does not.
+
+EXAMPLES OF MPV USAGE
+=====================
+
+Blu-ray playback:
+    - ``mpv bd:////path/to/disc``
+    - ``mpv bd:// --bluray-device=/path/to/disc``
+
+Play in Japanese with English subtitles:
+    ``mpv dvd://1 --alang=ja --slang=en``
+
+Play only chapters 5, 6, 7:
+    ``mpv dvd://1 --chapter=5-7``
+
+Play only titles 5, 6, 7:
+    ``mpv dvd://5-7``
+
+Play a multi-angle DVD:
+    ``mpv dvd://1 --dvd-angle=2``
+
+Play from a different DVD device:
+    ``mpv dvd://1 --dvd-device=/dev/dvd2``
+
+Play DVD video from a directory with VOB files:
+    ``mpv dvd://1 --dvd-device=/path/to/directory/``
+
+Stream from HTTP:
+    ``mpv http://example.com/example.avi``
+
+Stream using RTSP:
+    ``mpv rtsp://server.example.com/streamName``
+
+Play a libavfilter graph:
+    ``mpv avdevice://lavfi:mandelbrot``
+
+AUTHORS
+=======
+
+mpv is a MPlayer fork based on mplayer2, which in turn is a fork of MPlayer.
+
+MPlayer was initially written by Arpad Gereoffy. See the ``AUTHORS`` file for
+a list of some of the many other contributors.
+
+MPlayer is (C) 2000-2013 The MPlayer Team
+
+This man page was written mainly by Gabucino, Jonas Jermann and Diego Biurrun.

DOCS/man/osc.rst

@@ -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
@@ -169,35 +169,34 @@ Configurable Options
 
 ``showwindowed``
     | Default: yes
-    | Enable the OSC when windowed
+    | Show OSC when windowed?
 
 ``showfullscreen``
     | Default: yes
-    | Enable the OSC when fullscreen
+    | Show OSC when fullscreen?
 
 ``scalewindowed``
-    | Default: 1.0
-    | Scale factor of the OSC when windowed
+    | Default: 1
+    | Scaling of the controller when windowed
 
 ``scalefullscreen``
-    | Default: 1.0
-    | Scale factor of the OSC when fullscreen
+    | Default: 1
+    | Scaling of the controller when fullscreen
 
 ``scaleforcedwindow``
-    | Default: 2.0
-    | Scale factor of the OSC when rendered on a forced (dummy) window
+    | Default: 2
+    | Scaling of the controller when rendered on a forced (dummy) window
 
 ``vidscale``
     | Default: yes
-    | Scale the OSC with the video
-    | ``no`` tries to keep the OSC size constant as much as the window size allows
+    | Scale the controller with the video?
 
 ``valign``
     | Default: 0.8
     | Vertical alignment, -1 (top) to 1 (bottom)
 
 ``halign``
-    | Default: 0.0
+    | Default: 0
     | Horizontal alignment, -1 (left) to 1 (right)
 
 ``boxalpha``
@@ -235,14 +234,6 @@ 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
 ~~~~~~~~~~~~~~~
 

DOCS/man/vf.rst

@@ -17,10 +17,6 @@ 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.
 
@@ -101,7 +97,7 @@ Available filters are:
 
     All parameters are optional.
 
-    ``<w>:<h>``
+    ``<w>,<h>``
         scaled width/height (default: original width/height)
 
         :0:      scaled d_width/d_height
@@ -151,15 +147,15 @@ Available filters are:
         :yes: Enable accurate rounding.
 
 ``dsize[=w:h:aspect-method:r:aspect]``
-    Changes the intended display aspect at an arbitrary point in the
+    Changes the intended display size/aspect at an arbitrary point in the
     filter chain. Aspect can be given as a fraction (4/3) or floating point
-    number (1.33). Note that this filter does *not* do any scaling itself; it
+    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 the correct aspect.
+    auto-scaling to correct aspect.
 
     ``<w>,<h>``
-        New aspect ratio given by a display width and height. Unlike older mpv
-        versions or MPlayer, this does not set the display size.
+        New display width and height.
 
         Can also be these special values:
 
@@ -202,7 +198,7 @@ Available filters are:
     ``<aspect>``
         Force an aspect ratio.
 
-``format=fmt=<value>:colormatrix=<value>:...``
+``format[=fmt[:outfmt]]``
     Restricts the color space for the next filter without doing any conversion.
     Use together with the scale filter for a real conversion.
 
@@ -211,128 +207,19 @@ Available filters are:
         For a list of available formats, see ``format=fmt=help``.
 
     ``<fmt>``
-        Format name, e.g. rgb15, bgr24, 420p, etc. (default: don't change).
+        Format name, e.g. rgb15, bgr24, 420p, etc. (default: yuyv).
     ``<outfmt>``
-        Format name that should be substituted for the output. If they do not
-        have the same bytes per pixel and chroma subsamplimg, it will fail.
-    ``<colormatrix>``
-        Controls the YUV to RGB color space conversion when playing video. There
-        are various standards. Normally, BT.601 should be used for SD video, and
-        BT.709 for HD video. (This is done by default.) Using incorrect color space
-        results in slightly under or over saturated and shifted colors.
+        Format name that should be substituted for the output. If this is not
+        100% compatible with the ``<fmt>`` value, it will crash.
 
-        These options are not always supported. Different video outputs provide
-        varying degrees of support. The ``opengl`` and ``vdpau`` video output
-        drivers usually offer full support. The ``xv`` output can set the color
-        space if the system video driver supports it, but not input and output
-        levels. The ``scale`` video filter can configure color space and input
-        levels, but only if the output format is RGB (if the video output driver
-        supports RGB output, you can force this with ``-vf scale,format=rgba``).
-
-        If this option is set to ``auto`` (which is the default), the video's
-        color space flag will be used. If that flag is unset, the color space
-        will be selected automatically. This is done using a simple heuristic that
-        attempts to distinguish SD and HD video. If the video is larger than
-        1279x576 pixels, BT.709 (HD) will be used; otherwise BT.601 (SD) is
-        selected.
-
-        Available color spaces are:
-
-        :auto:          automatic selection (default)
-        :bt.601:        ITU-R BT.601 (SD)
-        :bt.709:        ITU-R BT.709 (HD)
-        :bt.2020-ncl:   ITU-R BT.2020 non-constant luminance system
-        :bt.2020-cl:    ITU-R BT.2020 constant luminance system
-        :smpte-240m:    SMPTE-240M
-
-    ``<colorlevels>``
-        YUV color levels used with YUV to RGB conversion. This option is only
-        necessary when playing broken files which do not follow standard color
-        levels or which are flagged wrong. If the video does not specify its
-        color range, it is assumed to be limited range.
-
-        The same limitations as with ``<colormatrix>`` apply.
-
-        Available color ranges are:
-
-        :auto:      automatic selection (normally limited range) (default)
-        :limited:   limited range (16-235 for luma, 16-240 for chroma)
-        :full:      full range (0-255 for both luma and chroma)
-
-    ``<primaries>``
-        RGB primaries the source file was encoded with. Normally this should be set
-        in the file header, but when playing broken or mistagged files this can be
-        used to override the setting.
-
-        This option only affects video output drivers that perform color
-        management, for example ``opengl`` with the ``target-prim`` or
-        ``icc-profile`` suboptions set.
-
-        If this option is set to ``auto`` (which is the default), the video's
-        primaries flag will be used. If that flag is unset, the color space will
-        be selected automatically, using the following heuristics: If the
-        ``<colormatrix>`` is set or determined as BT.2020 or BT.709, the
-        corresponding primaries are used. Otherwise, if the video height is
-        exactly 576 (PAL), BT.601-625 is used. If it's exactly 480 or 486 (NTSC),
-        BT.601-525 is used. If the video resolution is anything else, BT.709 is
-        used.
-
-        Available primaries are:
-
-        :auto:         automatic selection (default)
-        :bt.601-525:   ITU-R BT.601 (SD) 525-line systems (NTSC, SMPTE-C)
-        :bt.601-625:   ITU-R BT.601 (SD) 625-line systems (PAL, SECAM)
-        :bt.709:       ITU-R BT.709 (HD) (same primaries as sRGB)
-        :bt.2020:      ITU-R BT.2020 (UHD)
-        :apple:        Apple RGB
-        :adobe:        Adobe RGB (1998)
-        :prophoto:     ProPhoto RGB (ROMM)
-        :cie1931:      CIE 1931 RGB
-
-     ``<gamma>``
-        Gamma function the source file was encoded with. Normally this should be set
-        in the file header, but when playing broken or mistagged files this can be
-        used to override the setting.
-
-        This option only affects video output drivers that perform color management.
-
-        If this option is set to ``auto`` (which is the default), the gamma will
-        be set to BT.1886 for YCbCr content, sRGB for RGB content and Linear for
-        XYZ content.
-
-        Available gamma functions are:
-
-        :auto:         automatic selection (default)
-        :bt.1886:      ITU-R BT.1886 (approximation of BT.601/BT.709/BT.2020 curve)
-        :srgb:         IEC 61966-2-4 (sRGB)
-        :linear:       Linear light
-        :gamma1.8:     Pure power curve (gamma 1.8)
-        :gamma2.2:     Pure power curve (gamma 2.2)
-        :gamma2.8:     Pure power curve (gamma 2.8)
-        :prophoto:     ProPhoto RGB (ROMM) curve
-
-    ``<stereo-in>``
-        Set the stereo mode the video is assumed to be encoded in. Takes the
-        same values as the ``--video-stereo-mode`` option.
-
-    ``<stereo-out>``
-        Set the stereo mode the video should be displayed as. Takes the
-        same values as the ``--video-stereo-mode`` option.
-
-    ``<rotate>``
-        Set the rotation the video is assumed to be encoded with in degrees.
-        The special value ``-1`` uses the input format.
-
-    ``<dw>``, ``<dh>``
-        Set the display size. Note that setting the display size such that
-        the video is scaled in both directions instead of just changing the
-        aspect ratio is an implementation detail, and might change later.
-
-    ``<dar>``
-        Set the display aspect ratio of the video frame. This is a float,
-        but values such as ``[16:9]`` can be passed too (``[...]`` for quoting
-        to prevent the option parser from interpreting the ``:`` character).
+        .. admonition:: Examples
 
+            ====================== =====================
+            Valid                  Invalid (will crash)
+            ====================== =====================
+            ``format=rgb24:bgr24`` ``format=rgb24:420p``
+            ``format=yuyv:uyvy``
+            ====================== =====================
 
 ``noformat[=fmt]``
     Restricts the color space for the next filter without doing any conversion.
@@ -393,6 +280,41 @@ 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
@@ -419,6 +341,23 @@ 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
@@ -438,8 +377,9 @@ Available filters are:
         generating an occasional mismatched frame, but it may also cause an
         excessive number of frames to be dropped during high motion sequences.
         Conversely, setting it to -1 will make ``pullup`` match fields more
-        easily. This may help process video with slight blurring between the
-        fields, but may also cause interlaced frames in the output.
+        easily. This may help processing of video where there is slight
+        blurring between the fields, but may also cause there to be interlaced
+        frames in the output.
 
     ``mp`` (metric plane)
         This option may be set to ``u`` or ``v`` to use a chroma plane instead of the
@@ -449,32 +389,55 @@ 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:interlaced-only]``
+``yadif=[mode[:enabled=yes|no]]``
     Yet another deinterlacing filter
 
     ``<mode>``
         :frame: Output 1 frame for each frame.
-        :field: Output 1 frame for each field (default).
+        :field: Output 1 frame for each field.
         :frame-nospatial: Like ``frame`` but skips spatial interlacing check.
         :field-nospatial: Like ``field`` but skips spatial interlacing check.
 
-    ``<interlaced-only>``
-        :no:  Deinterlace all frames.
-        :yes: Only deinterlace frames marked as interlaced (default).
+    ``<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).
 
-    This filter is automatically inserted when using the ``d`` key (or any
+    This filter, is automatically inserted when using the ``D`` key (or any
     other key that toggles the ``deinterlace`` property or when using the
     ``--deinterlace`` switch), assuming the video output does not have native
     deinterlacing support.
 
     If you just want to set the default mode, put this filter and its options
-    into ``--vf-defaults`` instead, and enable deinterlacing with ``d`` or
+    into ``--vf-defaults`` instead, and enable deinterlacing with ``D`` or
     ``--deinterlace``.
 
-    Also, note that the ``d`` key is stupid enough to insert a deinterlacer twice
+    Also note that the ``D`` key is stupid enough to insert an interlacer twice
     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.
+
+``screenshot``
+    Optional filter for screenshot support. This is only needed if the video
+    output does not provide working direct screenshot support. Note that it is
+    not always safe to insert this filter by default. See `TAKING SCREENSHOTS`_
+    for details.
+
 ``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
@@ -482,9 +445,9 @@ Available filters are:
 
     ``<bottom-margin>``
         Adds a black band at the bottom of the frame. The SSA/ASS renderer can
-        place subtitles there (with ``--sub-use-margins``).
+        place subtitles there (with ``--ass-use-margins``).
     ``<top-margin>``
-        Black band on the top for toptitles  (with ``--sub-use-margins``).
+        Black band on the top for toptitles  (with ``--ass-use-margins``).
 
     .. admonition:: Examples
 
@@ -583,8 +546,6 @@ 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.
@@ -643,11 +604,10 @@ Available filters are:
     ``buffered-frames``
         Maximum number of decoded video frames that should be buffered before
         the filter (default: 4). This specifies the maximum number of frames
-        the script can request in reverse direction.
-        E.g. if ``buffered-frames=5``, and the script just requested frame 15,
-        it can still request frame 10, but frame 9 is not available anymore.
-        If it requests frame 30, mpv will decode 15 more frames, and keep only
-        frames 25-30.
+        the script can requests backwards. E.g. if ``buffered-frames=5``, and
+        the script just requested frame 15, it can still request frame 10, but
+        frame 9 is not available anymore. If it requests frame 30, mpv will
+        decode 15 more frames, and keep only frames 25-30.
 
         The actual number of buffered frames also depends on the value of the
         ``concurrent-frames`` option. Currently, both option values are
@@ -695,7 +655,7 @@ 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 convenience features are
+    is completely different, and absolutely no conveniencve features are
     provided. There's no type checking either, and you can trigger crashes.
 
     .. admonition:: Example:
@@ -720,7 +680,7 @@ Available filters are:
 ``vavpp``
     VA-AP-API video post processing. Works with ``--vo=vaapi`` and ``--vo=opengl``
     only. Currently deinterlaces. This filter is automatically inserted if
-    deinterlacing is requested (either using the ``d`` key, by default mapped to
+    deinterlacing is requested (either using the ``D`` key, by default mapped to
     the command ``cycle deinterlace``, or the ``--deinterlace`` option).
 
     ``deint=<method>``
@@ -737,14 +697,10 @@ 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
-    (either using the ``d`` key, by default mapped to the command
+    (either using the ``D`` key, by default mapped to the command
     ``cycle deinterlace``, or the ``--deinterlace`` option). When enabling
     deinterlacing, it is always preferred over software deinterlacer filters
     if the ``vdpau`` VO is used, and also if ``opengl`` is used and hardware
@@ -785,23 +741,14 @@ 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 to smooth out
+    useless, except for debugging. (Note that this won't help smoothing out
     latencies with decoding, because the filter will never output a frame if
     the buffer isn't full, except on EOF.)

DOCS/man/vo.rst

@@ -20,10 +20,13 @@ normal driver parameters.
 
     See ``--vo=help`` for a list of compiled-in video output drivers.
 
-    The recommended output driver is ``--vo=opengl-hq``. All other drivers are
-    for compatibility or special purposes. By default, ``--vo=opengl`` is used,
-    but if that appears not to work, it fallback to other drivers (in the same
-    order as listed by ``--vo=help``).
+    The recommended output drivers are ``--vo=vdpau`` and ``--vo=opengl-hq``.
+    All other drivers are just for compatibility or special purposes.
+
+.. admonition:: Example
+
+    ``--vo=opengl,xv,``
+        Try the ``opengl`` driver, then the ``xv`` driver, then others.
 
 Available video output drivers are:
 
@@ -66,12 +69,6 @@ Available video output drivers are:
     ``no-colorkey``
         Disables color-keying.
 
-    ``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.
-
 ``x11`` (X11 only)
     Shared memory video output driver without hardware acceleration that works
     whenever X11 is present.
@@ -105,7 +102,7 @@ Available video output drivers are:
         Select deinterlacing mode (default: 0). In older versions (as well as
         MPlayer/mplayer2) you could use this option to enable deinterlacing.
         This doesn't work anymore, and deinterlacing is enabled with either
-        the ``d`` key (by default mapped to the command ``cycle deinterlace``),
+        the ``D`` key (by default mapped to the command ``cycle deinterlace``),
         or the ``--deinterlace`` option. Also, to select the default deint mode,
         you should use something like ``--vf-defaults=vdpaupp:deint-mode=temporal``
         instead of this sub-option.
@@ -293,29 +290,6 @@ 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. It's also enabled automatically if nothing uses
-        features which require FBOs.
-
-        This option might be silently removed in the future.
-
     ``scale=<filter>``
 
         ``bilinear``
@@ -334,71 +308,36 @@ Available video output drivers are:
             This filter corresponds to the old ``lanczos3`` alias if the default
             radius is used, while ``lanczos2`` corresponds to a radius of 2.
 
-            (This filter is an alias for ``sinc``-windowed ``sinc``)
-
         ``ewa_lanczos``
             Elliptic weighted average Lanczos scaling. Also known as Jinc.
-            Relatively slow, but very good quality. The radius can be
-            controlled with ``scale-radius``. Increasing the radius makes the
+            Relatively slow, but very good quality. The number of taps can
+            be controlled with ``scale-radius``. Adding extra taps makes the
             filter sharper but adds more ringing.
 
-            (This filter is an alias for ``jinc``-windowed ``jinc``)
-
-        ``ewa_lanczossharp``
-            A slightly sharpened version of ewa_lanczos, preconfigured to use
-            an ideal radius and parameter. If your hardware can run it, this is
-            probably what you should use by default.
+            This filter supports antiringing (see ``scale-antiring``).
 
         ``mitchell``
-            Mitchell-Netravali. The ``B`` and ``C`` parameters can be set with
-            ``scale-param1`` and ``scale-param2``. This filter is very good at
-            downscaling (see ``dscale``).
-
-        ``oversample``
-            A version of nearest neighbour that (naively) oversamples pixels,
-            so that pixels overlapping edges get linearly interpolated instead
-            of rounded. This essentially removes the small imperfections and
-            judder artifacts caused by nearest-neighbour interpolation, in
-            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``).
+            Mitchell-Netravali. The ``b`` and ``c`` parameters can be set with
+            ``scale-param1`` and ``scale-param2``. Both are set to 1/3 by default.
+            This filter is very good at downscaling (see ``scale-down``).
 
         There are some more filters, but most are not as useful. For a complete
         list, pass ``help`` as value, e.g.::
 
             mpv --vo=opengl:scale=help
 
-    ``scale-param1=<value>``, ``scale-param2=<value>``
-        Set filter parameters. Ignored if the filter is not tunable.
-        Currently, this affects the following filter parameters:
+    ``scale-param1=<value>``
+        Set filter parameters. Ignored if the filter is not tunable. These are
+        unset by default, and use the filter specific default if applicable.
 
-        bcspline
-            Spline parameters (``B`` and ``C``). Defaults to 0.5 for both.
+    ``scale-param2=<value>``
+        See ``scale-param1``.
 
-        gaussian
-            Scale parameter (``t``). Increasing this makes the result blurrier.
-            Defaults to 1.
+    ``scale-radius=<r>``
+        Set radius for filters listed below, must be a float number between 1.0
+        and 16.0. Defaults to be 3.0 if not specified.
 
-        oversample
-            Minimum distance to an edge before interpolation is used. Setting
-            this to 0 will always interpolate edges, whereas setting it to 0.5
-            will never interpolate, thus behaving as if the regular nearest
-            neighbour algorithm was used. Defaults to 0.0.
-
-    ``scale-blur=<value>``
-        Kernel scaling factor (also known as a blur factor). Decreasing this
-        makes the result sharper, increasing it makes it blurrier (default 0).
-        If set to 0, the kernel's preferred blur factor is used. Note that
-        setting this too low (eg. 0.5) leads to bad results. It's generally
-        recommended to stick to values between 0.8 and 1.2.
-
-    ``scale-radius=<value>``
-        Set radius for filters listed below, must be a float number between 0.5
-        and 16.0. Defaults to the filter's preferred radius if not specified.
-
-            ``sinc`` and derivatives, ``jinc`` and derivatives, ``gaussian``, ``box`` and ``triangle``
+            ``sinc``, ``lanczos``, ``ewa_lanczos``, ``blackman``, ``gaussian``
 
         Note that depending on filter implementation details and video scaling
         ratio, the radius that actually being used might be different
@@ -410,35 +349,7 @@ Available video output drivers are:
         between 0.0 and 1.0. The default value of 0.0 disables antiringing
         entirely.
 
-        Note that this doesn't affect the special filters ``bilinear`` and
-        ``bicubic_fast``.
-
-    ``scale-window=<window>``
-        (Advanced users only) Choose a custom windowing function for the kernel.
-        Defaults to the filter's preferred window if unset. Use
-        ``scale-window=help`` to get a list of supported windowing functions.
-
-    ``scale-wparam=<window>``
-        (Advanced users only) Configure the parameter for the window function
-        given by ``scale-window``. Ignored if the window is not tunable.
-        Currently, this affects the following window parameters:
-
-        kaiser
-            Window parameter (alpha). Defaults to 6.33.
-        blackman
-            Window parameter (alpha). Defaults to 0.16.
-        gaussian
-            Scale parameter (t). Increasing this makes the window wider.
-            Defaults to 1.
-
-    ``scaler-lut-size=<4..10>``
-        Set the size of the lookup texture for scaler kernels (default: 6).
-        The actual size of the texture is ``2^N`` for an option value of ``N``.
-        So the lookup texture with the default setting uses 64 samples.
-
-        All weights are bilinearly interpolated from those samples, so
-        increasing the size of lookup table might improve the accuracy of
-        scaler.
+        Note that this currently only affects ``ewa_lanczos``.
 
     ``scaler-resizes-only``
         Disable the scaler if the video image is not resized. In that case,
@@ -446,13 +357,26 @@ Available video output drivers are:
         will reproduce the source image perfectly if no scaling is performed.
         Note that this option never affects ``cscale``.
 
-    ``pbo``
-        Enable use of PBOs. On some drivers this can be faster, especially if
-        the source video size is huge (e.g. so called "4K" video). On other
-        drivers it might be slower or cause latency issues.
+    ``srgb``
+        Convert and color correct the output to sRGB before displaying it on
+        the screen. This option enables ``linear-scaling``.
 
-        In theory, this can sometimes lead to sporadic and temporary image
-        corruption (because reupload is not retried when it fails).
+        This option is equivalent to using ``icc-profile`` with an sRGB ICC
+        profile, but it is implemented without a 3DLUT and does not require
+        LittleCMS 2. If both ``srgb`` and ``icc-profile`` are present, the
+        latter takes precedence, as they are somewhat redundant.
+
+        Note: When playing back BT.2020 content with this option enabled, out
+        of gamut colors will be numerically clipped, which can potentially
+        change the hue and/or luminance. If this is not desired, it is
+        recommended to use ``icc-profile`` with an sRGB ICC profile instead,
+        when playing back wide-gamut BT.2020 content.
+
+    ``pbo``
+        Enable use of PBOs. This is slightly faster, but can sometimes lead to
+        sporadic and temporary image corruption (in theory, because reupload
+        is not retried when it fails), and perhaps actually triggers slower
+        paths with drivers that don't support PBOs properly.
 
     ``dither-depth=<N|no|auto>``
         Set dither target depth to N. Default: no.
@@ -483,245 +407,53 @@ Available video output drivers are:
 
     ``temporal-dither``
         Enable temporal dithering. (Only active if dithering is enabled in
-        general.) This changes between 8 different dithering patterns on each
+        general.) This changes between 8 different dithering pattern on each
         frame by changing the orientation of the tiled dithering matrix.
         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
+        Check for OpenGL errors, i.e. call ``glGetError()``. Also request a
         debug OpenGL context (which does nothing with current graphics drivers
         as of this writing).
 
-    ``interpolation``
-        Reduce stuttering caused by mismatches in the video fps and display
-        refresh rate (also known as judder).
-
-        .. warning:: This requires setting the ``--video-sync`` option to one
-                     of the ``display-`` modes, or it will be silently disabled.
-                     This was not required before mpv 0.14.0.
-
-        This essentially attempts to interpolate the missing frames by
-        convoluting the video along the temporal axis. The filter used can be
-        controlled using the ``tscale`` setting.
-
-        Note that this relies on vsync to work, see ``swapinterval`` for more
-        information.
-
     ``swapinterval=<n>``
         Interval in displayed frames between two buffer swaps.
-        1 is equivalent to enable VSYNC, 0 to disable VSYNC. Defaults to 1 if
-        not specified.
-
-        Note that this depends on proper OpenGL vsync support. On some platforms
-        and drivers, this only works reliably when in fullscreen mode. It may
-        also require driver-specific hacks if using multiple monitors, to
-        ensure mpv syncs to the right one. Compositing window managers can
-        also lead to bad results, as can missing or incorrect display FPS
-        information (see ``--display-fps``).
-
-    ``dscale=<filter>``
-        Like ``scale``, but apply these filters on downscaling instead. If this
-        option is unset, the filter implied by ``scale`` will be applied.
+        1 is equivalent to enable VSYNC, 0 to disable VSYNC.
 
     ``cscale=<filter>``
         As ``scale``, but for interpolating chroma information. If the image
-        is not subsampled, this option is ignored entirely.
+        is not subsampled, this option is ignored entirely. Note that the
+        implementation is currently always done as a single pass, so using
+        it with separable filters will result in slow performance for very
+        little visible benefit.
 
-    ``tscale=<filter>``
-        The filter used for interpolating the temporal axis (frames). This is
-        only used if ``interpolation`` is enabled. The only valid choices
-        for ``tscale`` are separable convolution filters (use ``tscale=help``
-        to get a list). The default is ``mitchell``.
+    ``scale-down=<filter>``
+        Like ``scale``, but apply these filters on downscaling instead. If this
+        option is unset, the filter implied by ``scale`` will be applied.
 
-        Note that the maximum supported filter radius is currently 3, due to
-        limitations in the number of video textures that can be loaded
-        simultaneously.
+    ``cscale-param1``, ``cscale-param2``, ``cscale-radius``, ``cscale-antiring``
+        Set filter parameters and radius for ``cscale``.
 
-    ``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``,
-        respectively.
-
-        See the corresponding options for ``scale``.
+        See ``scale-param1``, ``scale-param2``, ``scale-radius`` and
+        ``scale-antiring``.
 
     ``linear-scaling``
-        Scale in linear light. It should only be used with a ``fbo-format``
-        that has at least 16 bit precision.
+        Scale in linear light. This is automatically enabled if ``srgb``,
+        ``icc-profile`` or ``sigmoid-upscaling`` is set. It should only
+        be used with a ``fbo-format`` that has at least 16 bit precision.
 
-    ``correct-downscaling``
+    ``fancy-downscaling``
         When using convolution based filters, extend the filter size
-        when downscaling. Increases quality, but reduces performance while
-        downscaling.
+        when downscaling. Trades quality for reduced downscaling performance.
 
-        This will perform slightly sub-optimally for anamorphic video (but still
-        better than without it) since it will extend the size to match only the
-        milder of the scale factors between the axes.
-
-    ``prescale=<filter>``
-        This option provides non-convolution-based filters for upscaling. These
-        filters resize the video to multiple of the original size (all currently
-        supported prescalers can only perform image doubling in a single pass).
-        Generally another convolution based filter (the main scaler) will be
-        applied after prescaler to match the target display size.
-
-        ``none``
-            Disable all prescalers. This is the default.
-
-        ``superxbr``
-            A relatively fast prescaler originally developed for pixel art.
-
-            Some parameters can be tuned with ``superxbr-sharpness`` and
-            ``superxbr-edge-strength`` options.
-
-        ``nnedi3``
-            An artificial neural network based deinterlacer, which can be used
-            to upscale images.
-
-            Extremely slow and requires a recent mid or high end graphics card
-            to work smoothly (as of 2015).
-
-        Note that all the filters above are designed (or implemented) to process
-        luma plane only and probably won't work as intended for video in RGB
-        format.
-
-    ``prescale-passes=<1..5>``
-        The number of passes to apply the prescaler (defaults to be 1). Setting
-        it to 2 will perform a 4x upscaling.
-
-    ``prescale-downscaling-threshold=<0..32>``
-        This option prevents "overkill" use of prescalers, which can be caused
-        by misconfiguration, or user trying to play a video with much larger
-        size. With this option, user can specify the maximal allowed downscaling
-        ratio in both dimension. To satisfy it, the number of passes for
-        prescaler will be reduced, and if necessary prescaler could also be
-        disabled.
-
-        The default value is 2.0, and should be able to prevent most seemingly
-        unreasonable use of prescalers. Most user would probably want to set it
-        to a smaller value between 1.0 and 1.5 for better performance.
-
-        A value less than 1.0 will disable the check.
-
-    ``nnedi3-neurons=<16|32|64|128>``
-        Specify the neurons for nnedi3 prescaling (defaults to be 32). The
-        rendering time is expected to be linear to the number of neurons.
-
-    ``nnedi3-window=<8x4|8x6>``
-        Specify the size of local window for sampling in nnedi3 prescaling
-        (defaults to be ``8x4``). The ``8x6`` window produces sharper images,
-        but is also slower.
-
-    ``nnedi3-upload=<ubo|shader>``
-        Specify how to upload the NN weights to GPU. Depending on the graphics
-        card, driver, shader compiler and nnedi3 settings, both method can be
-        faster or slower.
-
-        ``ubo``
-            Upload these weights via uniform buffer objects. This is the
-            default. (requires OpenGL 3.1 / GLES 3.0)
-
-        ``shader``
-            Hard code all the weights into the shader source code. (requires
-            OpenGL 3.3 / GLES 3.0)
-
-
-    ``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 (>4) are
-        practically useless. (Default 1)
-
-    ``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 16)
-
-        If you increase the ``deband-iterations``, you should probably
-        decrease this to compensate.
-
-    ``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)
+        This is automatically disabled for anamorphic video, because this
+        feature doesn't work correctly with different scale factors in
+        different directions.
 
     ``sigmoid-upscaling``
         When upscaling, use a sigmoidal color transform to avoid emphasizing
-        ringing artifacts. This also implies ``linear-scaling``.
+        ringing artifacts. This also enables ``linear-scaling``.
 
     ``sigmoid-center``
         The center of the sigmoid curve used for ``sigmoid-upscaling``, must
@@ -731,19 +463,13 @@ 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.
 
-    ``sharpen=<value>``
-        If set to a value other than 0, enable an unsharp masking filter.
-        Positive values will sharpen the image (but add more ringing and
-        aliasing). Negative values will blur the image. If your GPU is powerful
-        enough, consider alternatives like the ``ewa_lanczossharp`` scale
-        filter, or the ``scale-blur`` sub-option.
-
-        (This feature is the replacement for the old ``sharpen3`` and
-        ``sharpen5`` scalers.)
+    ``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 improve results when doing framedropping.
+        Slower, but might help getting better results when doing framedropping.
         Can completely ruin performance. The details depend entirely on the
         OpenGL driver.
 
@@ -754,31 +480,6 @@ Available video output drivers are:
 
         X11/GLX only.
 
-    ``vsync-fences=<N>``
-        Synchronize the CPU to the Nth past frame using the ``GL_ARB_sync``
-        extension. A value of 0 disables this behavior (default). A value of
-        1 means it will synchronize to the current frame after rendering it.
-        Like ``glfinish`` and ``waitvsync``, this can lower or ruin performance.
-        Its advantage is that it can span multiple frames, and effectively limit
-        the number of frames the GPU queues ahead (which also has an influence
-        on vsync).
-
-    ``dwmflush=<no|windowed|yes|auto>``
-        Calls ``DwmFlush`` after swapping buffers on Windows (default: auto).
-        It also sets ``SwapInterval(0)`` to ignore the OpenGL timing. Values
-        are: no (disabled), windowed (only in windowed mode), yes (also in
-        full screen).
-
-        The value ``auto`` will try to determine whether the compositor is
-        active, and calls ``DwmFlush`` only if it seems to be.
-
-        This may help to get more consistent frame intervals, especially with
-        high-fps clips - which might also reduce dropped frames. Typically, a
-        value of ``windowed`` should be enough, since full screen may bypass the
-        DWM.
-
-        Windows only.
-
     ``sw``
         Continue even if a software renderer is detected.
 
@@ -793,39 +494,20 @@ Available video output drivers are:
             Cocoa/OS X
         win
             Win32/WGL
-        angle
-            Direct3D11 through the OpenGL ES translation layer ANGLE. This
-            supports almost everything the ``win`` backend does, except ICC
-            profiles, high bit depth video input, and the ``nnedi3`` prescaler.
-        dxinterop (experimental)
-            Win32, using WGL for rendering and Direct3D 9Ex for presentation.
-            Works on Nvidia and AMD only.
-        x11
-            X11/GLX
+        x11, x11es
+            X11/GLX (the ``es`` variant forces GLES)
         wayland
             Wayland/EGL
-        drm-egl
-            DRM/EGL
-        x11egl
-            X11/EGL
-
-    ``es=<mode>``
-        Select whether to use GLES:
-
-        yes
-            Try to prefer ES over Desktop GL
-        no
-            Try to prefer desktop GL over ES
-        auto
-            Use the default for each backend (default)
+        x11egl, x11egles
+            X11/EGL (the ``es`` variant forces GLES)
 
     ``fbo-format=<fmt>``
         Selects the internal format of textures used for FBOs. The format can
-        influence performance and quality of the video output.
+        influence performance and quality of the video output. (FBOs are not
+        always used, and typically only when using extended scalers.)
         ``fmt`` can be one of: rgb, rgba, rgb8, rgb10, rgb10_a2, rgb16, rgb16f,
         rgb32f, rgba12, rgba16, rgba16f, rgba32f.
-        Default: ``auto``, which maps to rgba16 on desktop GL, and rgb10_a2 on
-        GLES (e.g. ANGLE).
+        Default: rgba.
 
     ``gamma=<0.1..2.0>``
         Set a gamma value (default: 1.0). If gamma is adjusted in other ways
@@ -841,85 +523,29 @@ Available video output drivers are:
         0.8
             Pitch black room
 
-    ``gamma-auto``
-        Automatically corrects the gamma value depending on ambient lighting
-        conditions (adding a gamma boost for dark rooms).
-
-        With ambient illuminance of 64lux, mpv will pick the 1.0 gamma value
-        (no boost), and slightly increase the boost up until 0.8 for 16lux.
-
-        NOTE: Only implemented on OS X.
-
-    ``target-prim=<value>``
-        Specifies the primaries of the display. Video colors will be adapted
-        to this colorspace if necessary. Valid values are:
-
-        auto
-            Disable any adaptation (default)
-        bt.470m
-            ITU-R BT.470 M
-        bt.601-525
-            ITU-R BT.601 (525-line SD systems, eg. NTSC), SMPTE 170M/240M
-        bt.601-625
-            ITU-R BT.601 (625-line SD systems, eg. PAL/SECAM), ITU-R BT.470 B/G
-        bt.709
-            ITU-R BT.709 (HD), IEC 61966-2-4 (sRGB), SMPTE RP177 Annex B
-        bt.2020
-            ITU-R BT.2020 (UHD)
-        apple
-            Apple RGB
-        adobe
-            Adobe RGB (1998)
-        prophoto
-            ProPhoto RGB (ROMM)
-        cie1931
-            CIE 1931 RGB (not to be confused with CIE XYZ)
-
-    ``target-trc=<value>``
-        Specifies the transfer characteristics (gamma) of the display. Video
-        colors will be adjusted to this curve. Valid values are:
-
-        auto
-            Disable any adaptation (default)
-        bt.1886
-            ITU-R BT.1886 curve, without the brightness drop (approx. 1.961)
-        srgb
-            IEC 61966-2-4 (sRGB)
-        linear
-            Linear light output
-        gamma1.8
-            Pure power curve (gamma 1.8), also used for Apple RGB
-        gamma2.2
-            Pure power curve (gamma 2.2)
-        gamma2.8
-            Pure power curve (gamma 2.8), also used for BT.470-BG
-        prophoto
-            ProPhoto RGB (ROMM)
-
     ``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``, ``target-trc`` and ``icc-profile-auto`` options.
+        Needs LittleCMS 2 support compiled in. This option overrides the ``srgb``
+        property, as using both is somewhat redundant. It also enables
+        ``linear-scaling``.
+
 
     ``icc-profile-auto``
         Automatically select the ICC display profile currently specified by
         the display settings of the operating system.
 
-        NOTE: On Windows, the default profile must be an ICC profile. WCS
-        profiles are not supported.
+        NOTE: Only implemented on OS X and X11
 
-    ``icc-cache-dir=<dirname>``
-        Store and load the 3D LUTs created from the ICC profile in this directory.
+    ``icc-cache=<file>``
+        Store and load the 3D LUT created from the ICC profile in this file.
         This can be used to speed up loading, since LittleCMS 2 can take a while
-        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.
+        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.
 
     ``icc-intent=<value>``
-        Specifies the ICC intent used for the color transformation (when using
-        ``icc-profile``).
+        Specifies the ICC Intent used for transformations between color spaces.
+        This affects the rendering when using ``icc-profile`` or ``srgb`` and
+        also affects the way DCP XYZ content gets converted to RGB.
 
         0
             perceptual
@@ -935,33 +561,9 @@ Available video output drivers are:
         Default is 128x256x64.
         Sizes must be a power of two, and 512 at most.
 
-    ``blend-subtitles=<yes|video|no>``
-        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 ``post-shader``. It
-        also increases subtitle performance when using ``interpolation``.
+    ``alpha=<blend|yes|no>``
+        Decides what to do if the input has an alpha component (default: blend).
 
-        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
-        black margins below a video (for example).
-
-        If ``video`` is selected, the behavior is similar to ``yes``, but subs
-        are drawn at the video's native resolution, and scaled along with the
-        video.
-
-        .. warning:: This changes the way subtitle colors are handled. Normally,
-                     subtitle colors are assumed to be in sRGB and color managed
-                     as such. Enabling this makes them treated as being in the
-                     video's color space instead. This is good if you want
-                     things like softsubbed ASS signs to match the video colors,
-                     but may cause SRT subtitles or similar to look slightly off.
-
-    ``alpha=<blend-tiles|blend|yes|no>``
-        Decides what to do if the input has an alpha component.
-
-        blend-tiles
-            Blend the frame against a 16x16 gray/white tiles background (default).
         blend
             Blend the frame against a black background.
         yes
@@ -974,6 +576,10 @@ Available video output drivers are:
         no
             Ignore alpha component.
 
+    ``chroma-location=<auto|center|left>``
+        Set the YUV chroma sample location. auto means use the bitstream
+        flags (default: auto).
+
     ``rectangle-textures``
         Force use of rectangle textures (default: no). Normally this shouldn't
         have any advantages over normal textures. Note that hardware decoding
@@ -983,16 +589,65 @@ Available video output drivers are:
         Color used to draw parts of the mpv window not covered by video.
         See ``--osd-color`` option how colors are defined.
 
+    ``smoothmotion``
+        Reduce stuttering caused by mismatches in video fps and display
+        refresh rate (also known as judder).
+
+        Instead of drawing source frames for variable durations, smoothmotion
+        will blend frames that overlap the transition between two frames in
+        the source material.
+
+        For example, a 24 Hz clip played back on a 60 Hz display would normally
+        result in a pattern like this::
+
+            A A A B B C C C D D E E E F F
+
+        which has different lengths, alternating between 3 and 2. This
+        difference in frame duration is what causes judder.
+
+        With smoothmotion enabled, the pattern changes to::
+
+            A A A+B B B C C C+D D D E E E+F F F
+
+        where A+B is a blend of A and B. In this pattern, each frame gets a
+        (consistent) duration of roughly 2.5 - resulting in smooth motion.
+
+        GPU drivers or compositing window managers overriding vsync behavior
+        can lead to bad results. If the framerate is close to or over the
+        display refresh rate, results can be bad as well.
+
+        .. note:: On systems other than Linux or OS X, you currently must set
+                  the ``--display-fps`` option, or the results will be bad.
+
+    ``smoothmotion-threshold=<0.0-0.5>``
+        Mix threshold at which interpolation is skipped (default: 0.0 – never
+        skip).
+
+        For example, with a ``smoothmotion-threshold`` of 0.1, if the
+        smoothmotion algorithm would try to blend two frames by a ratio of
+        95% A + 5% B, it would simply display A instead. (Since the
+        distance, 0.05, is lower than the threshold)
+
+        Setting this to 0.5 would be similar to disabling smoothmotion
+        completely, since it would always just display the nearest frame.
+
 ``opengl-hq``
     Same as ``opengl``, but with default settings for high quality rendering.
 
     This is equivalent to::
 
-        --vo=opengl:scale=spline36:cscale=spline36:dscale=mitchell:dither-depth=auto:correct-downscaling:sigmoid-upscaling:deband:es=no
+        --vo=opengl:scale=spline36:scale-down=mitchell:dither-depth=auto:fbo-format=rgba16:fancy-downscaling:sigmoid-upscaling
 
     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.
@@ -1028,7 +683,7 @@ Available video output drivers are:
 
     ``deint-mode=<mode>``
         Select deinterlacing algorithm. Note that by default deinterlacing is
-        initially always off, and needs to be enabled with the ``d`` key
+        initially always off, and needs to be enabled with the ``D`` key
         (default key binding for ``cycle deinterlace``).
 
         This option doesn't apply if libva supports video post processing (vpp).
@@ -1053,12 +708,6 @@ 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.
 
@@ -1126,48 +775,27 @@ Available video output drivers are:
 ``opengl-cb``
     For use with libmpv direct OpenGL embedding; useless in any other contexts.
     (See ``<mpv/opengl_cb.h>``.)
+    Usually, ``opengl-cb`` renders frames asynchronously by client and this
+    can cause some frame drops. In order to provide a way to handle this
+    situation, ``opengl-cb`` has its own frame queue and calls update callback
+    more frequently if the queue is not empty regardless of existence of new frame.
+    Once the queue is filled, ``opengl-cb`` drops frames automatically.
 
-    This also supports many of the suboptions the ``opengl`` VO has. Run
+    With default options, ``opengl-cb`` renders only the latest frame and drops
+    all frames handed over while waiting render function after update callback.
+
+    ``frame-queue-size=<1..100>``
+        The maximum count of frames which the frame queue can hold (default: 1)
+
+    ``frame-drop-mode=<pop|clear>``
+        Select the behavior when the frame queue is full.
+
+        pop
+            Drop the oldest frame in the frame queue. (default)
+        clear
+            Drop all frames in the frame queue.
+
+    This also supports many of the suboptions the ``opengl`` VO has. Runs
     ``mpv --vo=opengl-cb:help`` for a list.
 
     This also supports the ``vo_cmdline`` command.
-
-``rpi`` (Raspberry Pi)
-    Native video output on the Raspberry Pi using the MMAL API.
-
-    ``display=<number>``
-        Select the display number on which the video overlay should be shown
-        (default: 0).
-
-    ``layer=<number>``
-        Select the dispmanx layer on which the video overlay should be shown
-        (default: -10). Note that mpv will also use the 2 layers above the
-        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.
-
-    ``osd=<yes|no>``
-        Enabled by default. If disabled with ``no``, no OSD layer is created.
-        This also means there will be no subtitles rendered.
-
-``drm`` (Direct Rendering Manager)
-    Video output driver using Kernel Mode Setting / Direct Rendering Manager.
-    Should be used when one doesn't want to install full-blown graphical
-    environment (e.g. no X). Does not support hardware acceleration (if you
-    need this, check the ``drm-egl`` backend for ``opengl`` VO).
-
-    ``connector=<number>``
-        Select the connector to use (usually this is a monitor.) If set to -1,
-        mpv renders the output on the first available connector. (default: -1)
-
-    ``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)

DOCS/mplayer-changes.rst

@@ -46,13 +46,6 @@ Input
 * Allow customizing whether a key binding for seeking shows the video time, the
   OSD bar, or nothing (see the `Input Command Prefixes`_ section).
 * Support mapping multiple commands to one key.
-* 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
 ~~~~~
@@ -94,8 +87,6 @@ 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
 ~~~~~~~~~~~~~~~~
@@ -236,6 +227,7 @@ Command Line Switches
     ``-idx``                    ``--index``
     ``-lavdopts ...``           ``--vd-lavc-...``
     ``-lavfdopts``              ``--demuxer-lavf-...``
+    ``-lircconf``               ``--input-lirc-conf``
     ``-loop 0``                 ``--loop=inf``
     ``-mixer-channel``          AO suboptions (``alsa``, ``oss``)
     ``-mixer``                  AO suboptions (``alsa``, ``oss``)
@@ -247,6 +239,7 @@ Command Line Switches
     ``-noar``                   ``--no-input-appleremote``
     ``-noautosub``              ``--no-sub-auto``
     ``-noconsolecontrols``      ``--no-input-terminal``
+    ``-nojoystick``             ``--no-input-joystick``
     ``-nosound``                ``--no-audio``
     ``-osdlevel``               ``--osd-level``
     ``-panscanrange``           ``--video-zoom``, ``--video-pan-x/y``
@@ -279,7 +272,7 @@ Command Line Switches
     ``-xineramascreen``         ``--screen`` (different values)
     ``-xy W``                   ``--autofit=W``
     ``-zoom``                   Inverse available as ``--video-unscaled``
-    ``dvdnav://``               Removed.
+    ``dvdnav://``               ``dvdnav://menu``
     ``dvd://1``                 ``dvd://0`` (0-based offset)
     =========================== ========================================
 
@@ -399,8 +392,7 @@ Slave mode
 
   Assuming the system supports ``/dev/stdin``.
 
-  (The option was added back in 0.5.1 and sets exactly these options. It was
-  removed in 0.10.x again.)
+  (The option was readded in 0.5.1 and sets exactly these options.)
 
 * A JSON RPC protocol giving access to the client API is also supported. See
   `JSON IPC`_ for more information.

DOCS/release-policy.md

@@ -1,106 +1,64 @@
 Release Policy
 ==============
 
-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.
+Every few months, a new release is cut off of the master branch and is assigned
+a 0.X.0 version number.
 
-There is only one release branch that keeps track of the latest version and
-will not be maintained separately.
+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.
 
-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.
+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.
 
 Releases other than the latest release are unsupported and unmaintained.
 
 Release procedure
 -----------------
 
-- Merge master into branch release/current.
+- Create branch release/0.X or cherry-pick commits to the relevant branch.
 
-- 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
 =============
 
-Features
---------
+Changes
+-------
 
-New
-~~~
+- List of changes.
 
-- List of new features
+Bug fixes
+---------
 
-Removed
-~~~~~~~
+- List of bug fixes.
 
-- List of removed features
+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
+- List of new features.
 
 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.
 ```
 
@@ -108,6 +66,6 @@ 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.

DOCS/tech-overview.txt

@@ -75,7 +75,8 @@ 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 setting the
+    as in it doesn't require valgrind.) You can enable it by passing the option
+    --leak-report as first parameter, or better, setting the
     MPV_LEAK_REPORT environment variable to "1":
         export MPV_LEAK_REPORT=1
     This will list all unfree'd allocations on exit.

DOCS/waf-buildsystem.rst

@@ -1,7 +1,7 @@
 waf build system overview
 =========================
 
-mpv's new build system is based on waf and it should completely replace the
+mpv's new build system is based on waf and it should completly replace the
 custom ./configure + Makefile based system inherited from MPlayer.
 
 Goals and the choice of waf
@@ -20,9 +20,8 @@ shortcomings:
    part is this pieces are spread apart in the configure and copy pasted for
    any single case. That brings us to..
 
-2) --enable-feature has to be overridden by the user and helps them understand that
-   they have libraries missing and should install them for the feature to be
-   enabled.
+2) --enable-feature has to override the user and help him understand that he
+   has libraries missing and should install them for the feature to be enabled.
 
 3) Must be customizable, hackable, pleasant to the developer eyes and to work
    with in general.
@@ -65,7 +64,7 @@ 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 succeeds a ``#define HAVE_VDPAU 1`` will be added to
+``>= 0.2``. If the check succeds 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

README.md

@@ -1,54 +1,17 @@
-![http://mpv.io/](https://raw.githubusercontent.com/mpv-player/mpv.io/master/source/images/mpv-logo-128.png)
-
-## mpv
-
---------------
-
-
-* [Overview](#overview)
-* [Downloads](#downloads)
-* [Compilation](#compilation)
-* [FFmpeg vs. Libav](#ffmpeg-vs-libav)
-* [Release cycle](#release-cycle)
-* [Bug reports](#bug-reports)
-* [Contributing](#contributing)
-* [Relation to MPlayer and mplayer2](#relation-to-mplayer-and-mplayer2)
-* [Wiki](https://github.com/mpv-player/mpv/wiki)
-* [Man pages](http://mpv.io/manual/master/)
-* [Contact](#contact)
-* [License](#license)
-
-## Overview
+mpv
+===
 
+Overview
+--------
 
 **mpv** is a media player based on MPlayer and mplayer2. It supports a wide
 variety of video file formats, audio and video codecs, and subtitle types.
 
-Releases can be found on the [release list][releases].
-
-## System requirements
-
-- A not too ancient Linux, or Windows Vista or later, or OSX 10.8 or later.
-- A somewhat capable CPU. Hardware decoding might sometimes help if the CPU
-  is too slow to decode video realtime, but must be explicitly enabled with
-  the `--hwdec` option. On Windows, a CPU with SSE4 instruction set is required
-  to get decent hardware decoding performance.
-- A not too crappy GPU. mpv is not intended to be used with bad GPUs. There are
-  many caveats with drivers or system compositors causing tearing, stutter,
-  etc. On Windows, you might want to make sure the graphics drivers are
-  current, especially OpenGL. In some cases, ancient fallback video output
-  methods can help (such as `--vo=xv` on Linux), but this use is not
-  recommended or supported.
-
-
-## Downloads
-
-
-For semi-official builds and third-party packages please see
-[mpv.io](http://mpv.io/installation/).
-
-## Compilation
+If you are wondering what's different from mplayer2 and MPlayer you can read
+more about the [changes][changes].
 
+Compilation
+-----------
 
 Compiling with full features requires development files for several
 external libraries. Below is a list of some important requirements.
@@ -72,6 +35,9 @@ 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
@@ -79,13 +45,13 @@ Essential dependencies (incomplete list):
 - Audio output development headers (libasound/ALSA, pulseaudio)
 - FFmpeg libraries (libavutil libavcodec libavformat libswscale libavfilter
   and either libswresample or libavresample)
-  At least FFmpeg 2.4.0 or Libav 11 is required.
+  At least FFmpeg 2.1.4 or Libav 10 is required.
 - zlib
 - iconv (normally provided by the system libc)
-- libass (OSD, OSC, text subtitles)
+- libass
 - Lua (optional, required for the OSC pseudo-GUI and youtube-dl integration)
 - libjpeg (optional, used for screenshots only)
-- uchardet (optional, for subtitle charset detection)
+- Enca (optional, for subtitle charset detection)
 - vdpau and vaapi libraries for hardware decoding on Linux (optional)
 
 Libass dependencies:
@@ -111,20 +77,23 @@ the separately available build wrapper ([mpv-build][mpv-build]) that first compi
 libraries and libass, and then compiles the player statically linked against
 those.
 
+If you are running Mac OSX and using homebrew we provide [homebrew-mpv][homebrew-mpv], an up
+to date formula that compiles mpv with sensible dependencies and defaults for
+OSX.
+
 If you want to build a Windows binary, you either have to use MSYS2 and MinGW,
 or cross-compile from Linux with MinGW. See [Windows compilation][windows_compilation].
 
-
-## FFmpeg vs. Libav
-
+FFmpeg vs. Libav
+----------------
 
 Generally, mpv should work with the latest release as well as the git version
 of both FFmpeg and Libav. But FFmpeg is preferred, and some mpv features work
-with FFmpeg only (subtitle formats in particular).
+with FFmpeg only. See the [wiki article][ffmpeg_vs_libav] about the issue.
 
 
-## Release cycle
-
+Release cycle
+-------------
 
 Every few months, a new release is cut off of the master branch and is assigned
 a 0.X.0 version number.
@@ -142,64 +111,40 @@ Releases other than the latest release are unsupported and unmaintained.
 
 See the [release policy document][release-policy] for more information.
 
-## Bug reports
-
+Bug reports
+-----------
 
 Please use the [issue tracker][issue-tracker] provided by GitHub to send us bug
 reports or feature requests.
 
-## Contributing
-
+Contributing
+------------
 
 For small changes you can just send us pull requests through GitHub. For bigger
 changes come and talk to us on IRC before you start working on them. It will
 make code review easier for both parties later on.
 
-## Relation to MPlayer and mplayer2
-
-
-mpv is based on mplayer2, which in turn is based on the original MPlayer
-(also called mplayer, mplayer-svn, mplayer1). Many changes have been made, a
-large part of which is incompatible or completely changes how the player
-behaves. Although there are still many similarities to its ancestors, mpv
-should generally be treated as a completely different program.
-
-mpv was forked because we wanted to modernize MPlayer. This includes
-removing cruft (including features which stopped making sense 10 years ago),
-and of course adding modern features. Such huge and intrusive changes made it
-infeasible to work directly with MPlayer, which is mostly focused on
-preservation, so a fork had to be made. (Actually, mpv is based on mplayer2,
-which already started this process of removing cruft.)
-
-In general, mpv should be considered a completely new program, rather than a
-MPlayer drop-in replacement.
-
-If you are wondering what's different from mplayer2 and MPlayer, an incomplete
-list of changes is located [here][mplayer-changes].
-
-## Contact
-
+Contact
+-------
 
 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` on `irc.freenode.net`
- - **Developer IRC Channel**: `#mpv-devel` on `irc.freenode.net`
+ - **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])
 
 To contact the `mpv` team in private write to `mpv-team@googlegroups.com`. Use
 only if discretion is required.
 
-[releases]: https://github.com/mpv-player/mpv/releases
+[changes]: https://github.com/mpv-player/mpv/blob/master/DOCS/man/changes.rst
 [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
-[mplayer-changes]: https://github.com/mpv-player/mpv/blob/master/DOCS/mplayer-changes.rst
-
-## License
-
-
-Mostly GPLv2 or later. See [details.](https://github.com/mpv-player/mpv/blob/master/Copyright)

TOOLS/file2string.pl

@@ -9,8 +9,8 @@ use warnings;
 # the exact contents of the original file.
 # FIXME: why not a char array?
 
-# treat only alphanumeric and punctuations (excluding " and ?) as safe
-my $unsafe_chars = qr{[^][A-Za-z0-9!#%&'()*+,./:;<=>^_{|}~ -]};
+# treat only alphanumeric and not-" punctuation as safe
+my $unsafe_chars = qr{[^][A-Za-z0-9!#%&'()*+,./:;<=>?^_{|}~ -]};
 
 for my $file (@ARGV) {
     open my $fh, '<:raw', $file or next;

TOOLS/gen-x11-icon.sh

@@ -5,7 +5,7 @@
 
 conv() {
     echo
-    echo `identify -format "icon: %w %h" "$1"`
+    identify -format "icon: %w %h" "$1"
     convert "$1" -depth 8 rgba:-
 }
 

TOOLS/idet.sh

@@ -47,7 +47,7 @@ judge()
         progressive=$((progressive + progressive1))
         undetermined=$((undetermined + undetermined1))
     done <<EOF
-$(testfun "$@" | sed 's/:/: /g')
+$(testfun "$@")
 EOF
 
     interlaced=$((bff + tff))
@@ -95,18 +95,18 @@ case "$verdict" in
         exit 0
         ;;
     interlaced-tff)
-        judge "$@" --vf-clr --vf-pre=lavfi=\[setfield=tff\],pullup
+        judge "$@" --vf-clr --vf-pre=pullup --field-dominance=top
         case "$verdict" in
             progressive)
                 [ -n "$ILDETECT_DRY_RUN" ] || \
-                    $ILDETECT_MPV "$@" --vf-pre=lavfi=\[setfield=tff\],pullup
+                    $ILDETECT_MPV "$@" --vf-pre=pullup --field-dominance=top
                 r=$?
                 [ $r -eq 0 ] || exit $((r | 16))
                 exit 1
                 ;;
             *)
                 [ -n "$ILDETECT_DRY_RUN" ] || \
-                    $ILDETECT_MPV "$@" --vf-pre=lavfi=\[setfield=tff\],yadif
+                    $ILDETECT_MPV "$@" --vf-pre=yadif --field-dominance=top
                 r=$?
                 [ $r -eq 0 ] || exit $((r | 16))
                 exit 2
@@ -114,18 +114,18 @@ case "$verdict" in
         esac
         ;;
     interlaced-bff)
-        judge "$@" --vf-clr --vf-pre=lavfi=\[setfield=bff\],pullup
+        judge "$@" --vf-clr --vf-pre=pullup --field-dominance=bottom
         case "$verdict" in
             progressive)
                 [ -n "$ILDETECT_DRY_RUN" ] || \
-                    $ILDETECT_MPV "$@" --vf-pre=lavfi=\[setfield=bff\],pullup
+                    $ILDETECT_MPV "$@" --vf-pre=pullup --field-dominance=bottom
                 r=$?
                 [ $r -eq 0 ] || exit $((r | 16))
                 exit 1
                 ;;
             *)
                 [ -n "$ILDETECT_DRY_RUN" ] || \
-                    $ILDETECT_MPV "$@" --vf-pre=lavfi=\[setfield=bff\],yadif
+                    $ILDETECT_MPV "$@" --vf-pre=yadif --field-dominance=bottom
                 r=$?
                 [ $r -eq 0 ] || exit $((r | 16))
                 exit 2

TOOLS/lib/Parse/Matroska/Definitions.pm

@@ -84,7 +84,7 @@ sub elem_by_hexid {
 # used by elem when setting the 'valname' key
 use constant TYPE_MAP => {
     uint    => 'uint64_t',
-    str     => 'char *',
+    str     => 'struct bstr',
     binary  => 'struct bstr',
     ebml_id => 'uint32_t',
     float   => 'double',

TOOLS/lua/README.md

@@ -10,5 +10,3 @@ to mpv's command line.
 
 Where appropriate, they may also be placed in ~/.config/mpv/scripts/ from
 where they will be automatically loaded when mpv starts.
-
-Some of these are just for testing mpv internals.

TOOLS/lua/audio-hotplug-test.lua

@@ -1,13 +0,0 @@
-local utils = require("mp.utils")
-
-mp.observe_property("audio-device-list", "native", function(name, val)
-    print("Audio device list changed:")
-    for index, e in ipairs(val) do
-        print("  - '" .. e.name .. "' (" .. e.description .. ")")
-    end
-end)
-
-mp.observe_property("audio-out-detected-device", "native", function(name, val)
-    print("Detected audio device changed:")
-    print("  - '" .. val)
-end)

TOOLS/lua/autodeint.lua

@@ -2,16 +2,21 @@
 -- 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 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.
+-- 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.
 --
--- 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.
+-- 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.
 --
--- OPTIONS:
 -- The default detection time may be overridden by adding
 --
 -- --script-opts=autodeint.detect_seconds=<number of seconds>
@@ -32,7 +37,6 @@ 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)))
@@ -65,29 +69,30 @@ 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",@%s:pullup,@%s:lavfi=graph="idet"',
-                              detect_label, pullup_label, ivtc_detect_label)
+    local cmd = string.format('vf add @%s:lavfi=graph="idet"', detect_label)
     if not mp.command(cmd) then
-        mp.msg.error("failed to insert detection filters")
+        mp.msg.error("failed to insert detection filter")
         return
     end
 
     -- wait to gather data
-    timer = mp.add_timeout(detect_seconds, select_filter)
+    timer = mp.add_timeout(detect_seconds, judge_field_dominance)
 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(label)
+function judge()
     -- get the metadata
-    local result = mp.get_property_native(string.format("vf-metadata/%s", label))
+    local result = mp.get_property_native(string.format("vf-metadata/%s", detect_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"])
@@ -95,10 +100,10 @@ function judge(label)
     num_interlaced   = num_tff + num_bff
     num_determined   = num_interlaced + num_progressive
 
-    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)
+    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)
 
     if num_determined < num_undetermined then
         mp.msg.warn("majority undetermined frames")
@@ -114,12 +119,11 @@ function judge(label)
     end
 end
 
-function select_filter()
-    -- handle the first detection filter results
-    verdict = judge(detect_label)
+function judge_field_dominance()
+    verdict = judge()
     if verdict == progressive then
-        mp.msg.info("progressive: doing nothing")
         stop_detect()
+        mp.msg.info("progressive: doing nothing")
         return
     elseif verdict == interlaced_tff then
         mp.set_property("field-dominance", "top")
@@ -129,8 +133,21 @@ function select_filter()
         mp.set_property("field-dominance", "auto")
     end
 
-    -- handle the ivtc detection filter results
-    verdict = judge(ivtc_detect_label)
+    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()
+
     if verdict == progressive then
         mp.msg.info(string.format("telecinied with %s field dominance: using pullup", mp.get_property("field-dominance")))
         stop_detect()
@@ -140,6 +157,8 @@ function select_filter()
         mp.set_property("deinterlace","yes")
         stop_detect()
     end
+
+    return
 end
 
 mp.add_key_binding("ctrl+d", script_name, start_detect)

TOOLS/lua/autoload.lua

@@ -4,19 +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 5000 * 2 files when starting a file (before + after).
-MAXENTRIES = 5000
-
-function Set (t)
-    local set = {}
-    for _, v in pairs(t) do set[v] = true end
-    return set
-end
-
-EXTENSIONS = Set {
-    'mkv', 'avi', 'mp4', 'ogv', 'webm', 'rmvb', 'flv', 'wmv', 'mpeg', 'mpg', 'm4v', '3gp',
-    'mp3', 'wav', 'ogv', 'flac', 'm4a', 'wma',
-}
+-- Add at most 5 * 2 files when starting a file (before + after).
+MAXENTRIES = 5
 
 mputils = require 'mp.utils'
 
@@ -29,56 +18,20 @@ function add_files_at(index, files)
     end
 end
 
-function get_extension(path)
-    match = string.match(path, "%.([^%.]+)$" )
-    if match == nil then
-        return "nomatch"
-    else
-        return match
-    end
-end
-
-table.filter = function(t, iter)
-    for i = #t, 1, -1 do
-        if not iter(t[i]) then
-            table.remove(t, i)
-        end
-    end
-end
-
 function find_and_add_entries()
     local path = mp.get_property("path", "")
     local dir, filename = mputils.split_path(path)
     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
         return
     end
-    table.filter(files, function (v, k)
-        local ext = get_extension(v)
-        if ext == nil then
-            return false
-        end
-        return EXTENSIONS[string.lower(ext)]
-    end)
-    table.sort(files, function (a, b)
-        return string.lower(a) < string.lower(b)
-    end)
-
+    table.sort(files)
     if dir == "." then
         dir = ""
     end
-
     local pl = mp.get_property_native("playlist", {})
     local pl_current = mp.get_property_number("playlist-pos", 0) + 1
     -- Find the current pl entry (dir+"/"+filename) in the sorted dir list
@@ -92,7 +45,6 @@ function find_and_add_entries()
     if current == nil then
         return
     end
-
     local append = {[-1] = {}, [1] = {}}
     for direction = -1, 1, 2 do -- 2 iterations, with direction = -1 and +1
         for i = 1, MAXENTRIES do
@@ -101,7 +53,6 @@ function find_and_add_entries()
             if file == nil or file[1] == "." then
                 break
             end
-
             local filepath = dir .. file
             if pl_e then
                 -- If there's a playlist entry, and it's the same file, stop.
@@ -109,7 +60,6 @@ function find_and_add_entries()
                     break
                 end
             end
-
             if direction == -1 then
                 if pl_current == 1 then -- never add additional entries in the middle
                     mp.msg.info("Prepending " .. file)
@@ -121,7 +71,6 @@ function find_and_add_entries()
             end
         end
     end
-
     add_files_at(pl_current + 1, append[1])
     add_files_at(pl_current, append[-1])
 end

TOOLS/lua/drc-control.lua

@@ -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)

TOOLS/lua/ontop-playback.lua

@@ -1,19 +0,0 @@
---makes mpv disable ontop when pausing and re-enable it again when resuming playback
---please note that this won't do anything if ontop was not enabled before pausing
-
-local was_ontop = false
-
-mp.observe_property("pause", "bool", function(name, value)
-    local ontop = mp.get_property_native("ontop")
-    if value then
-        if ontop then
-            mp.set_property_native("ontop", false)
-            was_ontop = true
-        end
-    else
-        if was_ontop and not ontop then
-            mp.set_property_native("ontop", true)
-        end
-        was_ontop = false
-    end
-end)

TOOLS/lua/pause-when-minimize.lua

@@ -2,6 +2,8 @@
 -- if it's brought back again. If the player was already paused when minimizing,
 -- then try not to mess with the pause state.
 
+-- Note: currently works with X11 only.
+
 local did_minimize = false
 
 mp.observe_property("window-minimized", "bool", function(name, value)

TOOLS/lua/status-line.lua

@@ -1,7 +1,5 @@
 -- 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)
@@ -30,7 +28,7 @@ function update_status_line()
     atsl(mp.get_property_osd("time-pos"))
 
     atsl(" / ");
-    atsl(mp.get_property_osd("duration"));
+    atsl(mp.get_property_osd("length"));
 
     atsl(" (")
     atsl(mp.get_property_osd("percent-pos", -1))

TOOLS/lua/zones.lua

@@ -1,74 +0,0 @@
---  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)

TOOLS/matroska.pl

@@ -64,7 +64,7 @@ sub generate_c_header {
 
         # Output a counter variable for each element
         # (presence/absence for scalars, item count for arrays)
-        for my $subel (sort values %{$el->{subelements}}) {
+        for my $subel (values %{$el->{subelements}}) {
             print "    int n_$subel->{fieldname};\n"
         }
         print "};\n";

TOOLS/osxbundle.py

@@ -39,17 +39,8 @@ def apply_plist_template(plist_file, version):
     for line in fileinput.input(plist_file, inplace=1):
         print (line.rstrip().replace('${VERSION}', version))
 
-def bundle_version():
-    if os.path.exists('VERSION'):
-        x = open('VERSION')
-        version = x.read()
-        x.close()
-    else:
-        version = sh("./version.sh").strip()
-    return version
-
 def main():
-    version = bundle_version().rstrip()
+    version = sh("./version.sh --print").strip()
 
     usage = "usage: %prog [options] arg"
     parser = OptionParser(usage)

TOOLS/osxbundle/mpv.app/Contents/Resources/mpv.conf

@@ -1 +1,3 @@
-profile=pseudo-gui
+screenshot-template=~/Desktop/shot%n
+quiet
+idle=once

TOOLS/stats-conv.py

@@ -1,15 +1,11 @@
 #!/usr/bin/env python3
-from pyqtgraph.Qt import QtGui, QtCore
-import pyqtgraph as pg
+import matplotlib.pyplot as plot
 import sys
 import re
 
 filename = sys.argv[1]
 
-events = ".*"
-if len(sys.argv) > 2:
-    events = sys.argv[2]
-event_regex = re.compile(events)
+event_regex = re.compile(".*")
 
 """
 This script is meant to display stats written by mpv --dump-stats=filename.
@@ -30,8 +26,8 @@ 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
-    'range-timed' <ts1> <ts2> <name>        like start/end, but explicit times
+    'value-timed' <ts> <float> <name>
+                                a value for an event at the given timestamp
     <name>                      singular event (same as 'signal')
 
 """
@@ -39,8 +35,8 @@ Currently, the following event types are supported:
 class G:
     events = {}
     start = None
-    markers = ["o", "s", "t", "d"]
-    curveno = {}
+    # http://matplotlib.org/api/markers_api.html#module-matplotlib.markers
+    markers = ["o", "8", "s", "p", "*", "h", "+", "x", "D"]
 
 def find_marker():
     if len(G.markers) == 0:
@@ -66,10 +62,6 @@ def get_event(event, evtype):
         G.events[event] = e
     return G.events[event]
 
-colors = [(0.0, 0.5, 0.0), (0.0, 0.0, 1.0), (0.0, 0.0, 0.0), (1.0, 0.0, 0.0), (0.75, 0.75, 0), (0.0, 0.75, 0.75), (0.75, 0, 0.75)]
-def mkColor(t):
-    return pg.mkColor(int(t[0] * 255), int(t[1] * 255), int(t[2] * 255))
-
 SCALE = 1e6 # microseconds to seconds
 
 for line in [line.split("#")[0].strip() for line in open(filename, "r")]:
@@ -99,15 +91,6 @@ 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
@@ -133,33 +116,20 @@ for e, index in zip(G.sevents, range(len(G.sevents))):
     else:
         e.vals = [(x, y * (m - index) / m) for (x, y) in e.vals]
 
-pg.setConfigOption('background', 'w')
-pg.setConfigOption('foreground', 'k')
-app = QtGui.QApplication([])
-win = pg.GraphicsWindow()
-#win.resize(1500, 900)
-
+fig = plot.figure()
+fig.hold(True)
 ax = [None, None]
 plots = 2 if hasval else 1
-ax[0] = win.addPlot()
+ax[0] = fig.add_subplot(plots, 1, 1)
 if hasval:
-    win.nextRow()
-    ax[1] = win.addPlot()
-    ax[1].setXLink(ax[0])
-for cur in ax:
-   if cur is not None:
-       cur.addLegend(offset = (-1, 1))
+    ax[1] = fig.add_subplot(plots, 1, 2, sharex=ax[0])
+legends = [[], []]
 for e in G.sevents:
     cur = ax[1 if e.type == "value" else 0]
-    args = {'name': e.name,'antialias':True}
+    pl, = cur.plot([x for x,y in e.vals], [y for x,y in e.vals], label=e.name)
     if e.type == "event-signal":
-        args['symbol'] = e.marker
-        args['pen'] = None
-    else:
-        if not cur in G.curveno:
-            G.curveno[cur] = 0
-        args['pen'] = pg.mkPen(mkColor(colors[G.curveno[cur] % len(colors)]), width=0)
-        G.curveno[cur] += 1
-    n = cur.plot([x for x,y in e.vals], [y for x,y in e.vals], **args)
-
-QtGui.QApplication.instance().exec_()
+        plot.setp(pl, marker = e.marker, linestyle = "None")
+for cur in ax:
+    if cur is not None:
+        cur.legend()
+plot.show()

TOOLS/umpv

@@ -21,7 +21,7 @@ script after that will start a new mpv instance.
 
 Note that you can control the mpv instance by writing to the command fifo:
 
-    echo "cycle fullscreen" > ~/.umpv_fifo
+    echo "cycle fullscreen" > ~/.umpv-fifo
 
 Note: you can supply custom mpv path and options with the MPV environment
       variable. The environment variable will be split on whitespace, and the

TOOLS/youtube-dl_mpv.sh

@@ -0,0 +1,22 @@
+#!/bin/sh
+# Example of script for using mpv with youtube-dl
+set -e
+
+cookies_dir="$(mktemp -d /tmp/youtube-dl_mpv.XXXX)"
+cookies_file="${cookies_dir}/cookies"
+user_agent="$(youtube-dl --dump-user-agent)" # or set whatever you want
+
+video_url=$(youtube-dl \
+    --user-agent="${user_agent}" \
+    --cookies="${cookies_file}" \
+    --get-url \
+    "$@")
+
+set -f
+${MPV:-mpv} \
+    --cookies \
+    --cookies-file="${cookies_file}" \
+    --user-agent="${user_agent}" \
+    -- ${video_url}
+
+rm -rf "${cookies_dir}"

TOOLS/zsh.pl

@@ -7,15 +7,15 @@ use warnings;
 
 my $mpv = $ARGV[0] || 'mpv';
 
-my @opts = parse_main_opts('--list-options', '^ (\-\-[^\s\*]*)\*?\s*(.*)');
+my @opts = parse_main_opts("$mpv --list-options", '^ (\-\-[^\s\*]*)\*?\s*(.*)');
 
-my @ao = parse_opts('--ao=help', '^  ([^\s\:]*)\s*: (.*)');
-my @vo = parse_opts('--vo=help', '^  ([^\s\:]*)\s*: (.*)');
+my @ao = parse_opts("$mpv --ao=help", '^  ([^\s\:]*)\s*: (.*)');
+my @vo = parse_opts("$mpv --vo=help", '^  ([^\s\:]*)\s*: (.*)');
 
-my @af = parse_opts('--af=help', '^  ([^\s\:]*)\s*: (.*)');
-my @vf = parse_opts('--vf=help', '^  ([^\s\:]*)\s*: (.*)');
+my @af = parse_opts("$mpv --af=help", '^  ([^\s\:]*)\s*: (.*)');
+my @vf = parse_opts("$mpv --vf=help", '^  ([^\s\:]*)\s*: (.*)');
 
-my @protos = parse_opts('--list-protocols', '^ ([^\s]*)');
+my @protos = parse_opts("$mpv --list-protocols", '^ ([^\s]*)');
 
 my ($opts_str, $ao_str, $vo_str, $af_str, $vf_str, $protos_str);
 
@@ -34,35 +34,36 @@ chomp $af_str;
 $vf_str .= qq{      '$_' \\\n} foreach (@vf);
 chomp $vf_str;
 
-$protos_str = join(' ', @protos);
+$protos_str .= qq{$_ } foreach (@protos);
+chomp $protos_str;
 
 my $profile_comp = <<'EOS';
-    local -a profiles
-    local current
-    for current in "${(@f)$($words[1] --profile=help)}"; do
-      current=${current//\*/\\\*}
-      current=${current//\:/\\\:}
-      current=${current//\[/\\\[}
-      current=${current//\]/\\\]}
-      if [[ $current =~ $'\t'([^$'\t']*)$'\t'(.*) ]]; then
-        if [[ -n $match[2] ]]; then
-          current="$match[1][$match[2]]"
-        else
-          current="$match[1]"
+      local -a profiles
+      local current
+      for current in "${(@f)$($words[1] --profile=help)}"; do
+        current=${current//\*/\\\*}
+        current=${current//\:/\\\:}
+        current=${current//\[/\\\[}
+        current=${current//\]/\\\]}
+        if [[ $current =~ $'\t'([^$'\t']*)$'\t'(.*) ]]; then
+          if [[ -n $match[2] ]]; then
+            current="$match[1][$match[2]]"
+          else
+            current="$match[1]"
+          fi
+          profiles=($profiles $current)
         fi
-        profiles=($profiles $current)
+      done
+      if [[ $state == show-profile ]]; then
+        # For --show-profile, only one allowed
+        if (( ${#profiles} > 0 )); then
+          _values 'profile' $profiles && rc=0
+        fi
+      else
+        # For --profile, multiple allowed
+        profiles=($profiles 'help[list profiles]')
+        _values -s , 'profile(s)' $profiles && rc=0
       fi
-    done
-    if [[ $state == show-profile ]]; then
-      # For --show-profile, only one allowed
-      if (( ${#profiles} > 0 )); then
-        _values 'profile' $profiles && rc=0
-      fi
-    else
-      # For --profile, multiple allowed
-      profiles=($profiles 'help[list profiles]')
-      _values -s , 'profile(s)' $profiles && rc=0
-    fi
 EOS
 chomp $profile_comp;
 
@@ -82,27 +83,39 @@ $opts_str
 
 case \$state in
   ao)
-    _values -s , 'audio outputs' \\
+    local -a values
+    values=(
 $ao_str
-    && rc=0
+    )
+
+    _describe -t values 'audio outputs' values && rc=0
   ;;
 
   vo)
-    _values -s , 'video outputs' \\
+    local -a values
+    values=(
 $vo_str
-    && rc=0
+    )
+
+    _describe -t values 'video outputs' values && rc=0
   ;;
 
   af)
-    _values -s , 'audio filters' \\
+    local -a values
+    values=(
 $af_str
-    && rc=0
+    )
+
+    _describe -t values 'audio filters' values && rc=0
   ;;
 
   vf)
-    _values -s , 'video filters' \\
+    local -a values
+    values=(
 $vf_str
-    && rc=0
+    )
+
+    _describe -t values 'video filters' values && rc=0
   ;;
 
   profile|show-profile)
@@ -120,7 +133,7 @@ $profile_comp
     _tags files urls
     while _tags; do
       _requested files expl 'media file' _files -g \\
-        "*.(#i)(asf|asx|avi|flac|flv|m1v|m2p|m2v|m4v|mjpg|mka|mkv|mov|mp3|mp4|mpe|mpeg|mpg|ogg|ogm|ogv|opus|qt|rm|ts|vob|wav|webm|wma|wmv)(-.)" && rc=0
+        "*.(#i)(asf|asx|avi|flac|flv|m1v|m2p|m2v|m4v|mjpg|mka|mkv|mov|mp3|mp4|mpe|mpeg|mpg|ogg|ogm|ogv|qt|rm|ts|vob|wav|webm|wma|wmv)(-.)" && rc=0
       if _requested urls; then
         while _next_label urls expl URL; do
           _urls "\$expl[@]" && rc=0
@@ -141,7 +154,7 @@ sub parse_main_opts {
     my ($cmd, $regex) = @_;
 
     my @list;
-    my @lines = call_mpv($cmd);
+    my @lines = split /\n/, `$cmd`;
 
     foreach my $line (@lines) {
         my ($name, $desc) = ($line =~ /^$regex/) or next;
@@ -206,7 +219,7 @@ sub parse_opts {
     my ($cmd, $regex) = @_;
 
     my @list;
-    my @lines = call_mpv($cmd);
+    my @lines = split /\n/, `$cmd`;
 
     foreach my $line (@lines) {
         if ($line !~ /^$regex/) {
@@ -218,7 +231,7 @@ sub parse_opts {
         if (defined $2) {
             my $desc = $2;
             $desc =~ s/\:/\\:/g;
-            $entry .= "[$desc]";
+            $entry .= ':' . $desc;
         }
 
         push @list, $entry
@@ -226,14 +239,3 @@ 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;
-}

VERSION

@@ -1 +1 @@
-0.15.0
+0.8.1

audio/audio.c

@@ -24,7 +24,7 @@
 #include <libavutil/frame.h>
 #include <libavutil/version.h>
 
-#include "mpv_talloc.h"
+#include "talloc.h"
 #include "common/common.h"
 #include "fmt-conversion.h"
 #include "audio.h"
@@ -34,7 +34,7 @@ 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_fmt_to_bytes(mpa->format);
+    mpa->bps = af_fmt2bps(mpa->format);
     if (af_fmt_is_planar(mpa->format)) {
         mpa->spf = 1;
         mpa->num_planes = mpa->nch;
@@ -58,6 +58,13 @@ 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;
@@ -86,13 +93,9 @@ 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,
-             ch, mpa->channels.num, af_fmt_to_str(mpa->format));
+             mp_chmap_to_str(&mpa->channels), mpa->channels.num,
+             af_fmt_to_str(mpa->format));
     return buf;
 }
 
@@ -156,8 +159,8 @@ 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]);
@@ -188,18 +191,8 @@ int mp_audio_get_allocated_size(struct mp_audio *mpa)
 {
     int size = 0;
     for (int n = 0; n < mpa->num_planes; n++) {
-        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: ;
+        int s = mpa->allocated[n] ? mpa->allocated[n]->size / mpa->sstride : 0;
+        size = n == 0 ? s : MPMIN(size, s);
     }
     return size;
 }
@@ -323,23 +316,21 @@ struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe)
     }
 
     // If we can't handle the format (e.g. too many channels), bail out.
-    if (!mp_audio_config_valid(new))
+    if (!mp_audio_config_valid(new) || avframe->nb_extended_buf)
         goto fail;
 
-    for (int n = 0; n < AV_NUM_DATA_POINTERS + avframe->nb_extended_buf; n++) {
-        AVBufferRef *buf = n < AV_NUM_DATA_POINTERS ? avframe->buf[n]
-                            : avframe->extended_buf[n - AV_NUM_DATA_POINTERS];
-        if (!buf)
+    for (int n = 0; n < AV_NUM_DATA_POINTERS; n++) {
+        if (!avframe->buf[n])
             break;
         if (n >= MP_NUM_CHANNELS)
             goto fail;
-        new->allocated[n] = av_buffer_ref(buf);
+        new->allocated[n] = av_buffer_ref(avframe->buf[n]);
         if (!new->allocated[n])
             goto fail;
     }
 
     for (int n = 0; n < new->num_planes; n++)
-        new->planes[n] = avframe->extended_data[n];
+        new->planes[n] = avframe->data[n];
     new->samples = avframe->nb_samples;
 
     return new;

audio/audio.h

@@ -35,19 +35,15 @@ 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_fmt_to_bytes(format))
+    int bps;            // size of sub-samples (af_fmt2bps(format))
 
-    double pts;         // currently invalid within the filter chain
-
-    // --- 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.
+    // private
     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);

audio/audio_buffer.c

@@ -119,15 +119,6 @@ void mp_audio_buffer_prepend_silence(struct mp_audio_buffer *ab, int samples)
     mp_audio_fill_silence(ab->buffer, 0, samples);
 }
 
-void mp_audio_buffer_duplicate(struct mp_audio_buffer *ab, int samples)
-{
-    assert(samples >= 0 && samples <= ab->buffer->samples);
-    int oldlen = ab->buffer->samples;
-    ab->buffer->samples += samples;
-    mp_audio_realloc_min(ab->buffer, ab->buffer->samples);
-    mp_audio_copy(ab->buffer, oldlen, ab->buffer, oldlen - samples, samples);
-}
-
 // Get the start of the current readable buffer.
 void mp_audio_buffer_peek(struct mp_audio_buffer *ab, struct mp_audio *out_mpa)
 {

audio/audio_buffer.h

@@ -35,7 +35,6 @@ void mp_audio_buffer_get_write_buffer(struct mp_audio_buffer *ab, int minsamples
 void mp_audio_buffer_finish_write(struct mp_audio_buffer *ab, int samples);
 void mp_audio_buffer_append(struct mp_audio_buffer *ab, struct mp_audio *mpa);
 void mp_audio_buffer_prepend_silence(struct mp_audio_buffer *ab, int samples);
-void mp_audio_buffer_duplicate(struct mp_audio_buffer *ab, int samples);
 void mp_audio_buffer_peek(struct mp_audio_buffer *ab, struct mp_audio *out_mpa);
 void mp_audio_buffer_skip(struct mp_audio_buffer *ab, int samples);
 void mp_audio_buffer_clear(struct mp_audio_buffer *ab);

audio/chmap.c

@@ -18,8 +18,6 @@
 #include <stdlib.h>
 #include <assert.h>
 
-#include <libavutil/common.h>
-
 #include "common/common.h"
 #include "common/msg.h"
 #include "chmap.h"
@@ -52,7 +50,6 @@ 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.)
@@ -60,9 +57,7 @@ static const char *const speaker_names[MP_SPEAKER_ID_COUNT][2] = {
 static const char *const std_layout_names[][2] = {
     {"empty",           ""}, // not in lavc
     {"mono",            "fc"},
-    {"1.0",             "fc"}, // not in lavc
     {"stereo",          "fl-fr"},
-    {"2.0",             "fl-fr"}, // not in lavc
     {"2.1",             "fl-fr-lfe"},
     {"3.0",             "fl-fr-fc"},
     {"3.0(back)",       "fl-fr-bc"},
@@ -70,7 +65,6 @@ 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"},
@@ -84,23 +78,21 @@ 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"},
     {"7.1(wide-side)",  "fl-fr-fc-lfe-flc-frc-sl-sr"},
-    {"7.1(rear)",       "fl-fr-fc-lfe-bl-br-sdl-sdr"}, // not in lavc
+    {"7.1(rear)",       "fl-fr-fc-lfe-bl-br-sdl-sdr"},
     {"octagonal",       "fl-fr-fc-bl-br-bc-sl-sr"},
     {"downmix",         "dl-dr"},
     {"auto",            ""}, // not in lavc
     {0}
 };
 
-static const struct mp_chmap default_layouts[] = {
+static const struct mp_chmap default_layouts[MP_NUM_CHANNELS + 1] = {
     {0},                                        // empty
     MP_CHMAP_INIT_MONO,                         // mono
     MP_CHMAP2(FL, FR),                          // stereo
@@ -112,6 +104,19 @@ static const struct mp_chmap default_layouts[] = {
     MP_CHMAP8(FL, FR, FC, LFE, BL, BR, SL, SR), // 7.1
 };
 
+// The channel order was lavc/waveex, but differs from lavc for 5, 6 and 8
+// channels. 3 and 7 channels were likely undefined (no ALSA support).
+// I'm not sure about the 4 channel case: ALSA uses "quad", while the ffmpeg
+// default layout is "4.0".
+static const char *const mplayer_layouts[MP_NUM_CHANNELS + 1] = {
+    [1] = "mono",
+    [2] = "stereo",
+    [4] = "quad",
+    [5] = "5.0(alsa)",
+    [6] = "5.1(alsa)",
+    [8] = "7.1(alsa)",
+};
+
 // Returns true if speakers are mapped uniquely, and there's at least 1 channel.
 bool mp_chmap_is_valid(const struct mp_chmap *src)
 {
@@ -120,8 +125,7 @@ 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;
-        if (sp != MP_SPEAKER_ID_NA)
-            mapped[sp] = true;
+        mapped[sp] = true;
     }
     return src->num > 0;
 }
@@ -136,10 +140,12 @@ 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++) {
-        if (src->speaker[n] != MP_SPEAKER_ID_NA)
-            return false;
+        int speaker = src->speaker[n];
+        if (speaker >= MP_SPEAKER_ID_UNKNOWN0 &&
+            speaker <= MP_SPEAKER_ID_UNKNOWN_LAST)
+            return true;
     }
-    return mp_chmap_is_valid(src);
+    return false;
 }
 
 // Note: empty channel maps compare as equal. Invalid ones can equal too.
@@ -163,6 +169,15 @@ 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;
@@ -182,43 +197,34 @@ 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)
 {
-    *dst = (struct mp_chmap) {0};
-    if (num_channels >= 0 && num_channels < MP_ARRAY_SIZE(default_layouts))
+    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
+        *dst = (struct mp_chmap) {0};
+    } else {
         *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
+// introduced, i.e. get the old default channel order.
+void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels)
+{
+    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
+        *dst = (struct mp_chmap) {0};
+    } else {
+        mp_chmap_from_str(dst, bstr0(mplayer_layouts[num_channels]));
+        if (!dst->num)
+            mp_chmap_from_channels(dst, 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) {
@@ -226,7 +232,7 @@ 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_NA;
+            dst->speaker[n] = MP_SPEAKER_ID_UNKNOWN0 + n;
     }
 }
 
@@ -269,26 +275,17 @@ 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.
-//          Also, "unknown" channel maps are translated to non-sense channel
-//          maps with the same number of channels.
 uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src)
 {
+    // lavc has no concept for unknown layouts yet, so pick a default
     struct mp_chmap t = *src;
-    if (t.num > 64)
-        return 0;
-    // lavc has no concept for unknown layouts yet, so pick something that does
-    // the job of signaling the number of channels, even if it makes no sense
-    // as a proper layout.
     if (mp_chmap_is_unknown(&t))
-        return t.num == 64 ? (uint64_t)-1 : (1ULL << t.num) - 1;
+        mp_chmap_from_channels(&t, t.num);
     uint64_t mask = 0;
-    for (int n = 0; n < t.num; n++) {
-        if (t.speaker[n] < 64) // ignore MP_SPEAKER_ID_NA etc.
-            mask |= 1ULL << t.speaker[n];
-    }
+    for (int n = 0; n < t.num; n++)
+        mask |= 1ULL << t.speaker[n];
     return mask;
 }
 
@@ -339,8 +336,6 @@ bool mp_chmap_is_lavc(const struct mp_chmap *src)
     return true;
 }
 
-// Warning: for "unknown" channel maps, this returns something that may not
-//          make sense. Invalid channel maps are not changed.
 void mp_chmap_reorder_to_lavc(struct mp_chmap *map)
 {
     if (!mp_chmap_is_valid(map))
@@ -351,43 +346,56 @@ 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, 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,
+// 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,
                           const struct mp_chmap *to)
 {
-    for (int n = 0; n < MP_NUM_CHANNELS; n++)
-        src[n] = -1;
-
+    assert(from->num == to->num);
     if (mp_chmap_is_unknown(from) || mp_chmap_is_unknown(to)) {
-        for (int n = 0; n < to->num; n++)
-            src[n] = n < from->num ? n : -1;
+        for (int n = 0; n < from->num; n++)
+            dst[n] = n;
         return;
     }
-
-    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;
+    // 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;
                 break;
             }
         }
+        assert(dst[n] != -1);
     }
-
-    for (int n = 0; n < to->num; n++)
-        assert(src[n] < 0 || (to->speaker[n] == from->speaker[src[n]]));
+    for (int n = 0; n < from->num; n++)
+        assert(to->speaker[dst[n]] == from->speaker[n]);
 }
 
-// Return the number of channels only in a.
-int mp_chmap_diffn(const struct mp_chmap *a, const struct mp_chmap *b)
+// 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)
 {
     uint64_t a_mask = mp_chmap_to_lavc_unchecked(a);
     uint64_t b_mask = mp_chmap_to_lavc_unchecked(b);
-    return av_popcount64((a_mask ^ b_mask) & a_mask);
+    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;
 }
 
 // Returns something like "fl-fr-fc". If there's a standard layout in lavc
@@ -490,25 +498,6 @@ 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");

audio/chmap.h

@@ -22,7 +22,7 @@
 #include <stdbool.h>
 #include "misc/bstr.h"
 
-#define MP_NUM_CHANNELS 16
+#define MP_NUM_CHANNELS 8
 
 // Speaker a channel can be assigned to.
 // This corresponds to WAVEFORMATEXTENSIBLE channel mask bit indexes.
@@ -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
-     // Unofficial/libav* extensions
+     // Inofficial/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
@@ -56,12 +56,10 @@ enum mp_speaker_id {
     MP_SPEAKER_ID_SDR,          // SURROUND_DIRECT_RIGHT
     MP_SPEAKER_ID_LFE2,         // LOW_FREQUENCY_2
 
-    // 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,
+    // 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,
 
     // Including the unassigned IDs in between. This is not a valid ID anymore,
     // but is still within uint8_t.
@@ -100,14 +98,14 @@ 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);
+void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels);
 
 void mp_chmap_remove_useless_channels(struct mp_chmap *map,
                                       const struct mp_chmap *requested);
@@ -119,17 +117,17 @@ 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 src[MP_NUM_CHANNELS], const struct mp_chmap *from,
+void mp_chmap_get_reorder(int dst[MP_NUM_CHANNELS], const struct mp_chmap *from,
                           const struct mp_chmap *to);
 
-int mp_chmap_diffn(const struct mp_chmap *a, const struct mp_chmap *b);
+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);
 
 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;

audio/chmap_sel.c

@@ -20,7 +20,6 @@
 #include <limits.h>
 
 #include "common/common.h"
-#include "common/msg.h"
 #include "chmap_sel.h"
 
 static const struct mp_chmap speaker_replacements[][2] = {
@@ -59,29 +58,6 @@ 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)
@@ -95,6 +71,17 @@ void mp_chmap_sel_add_waveext(struct mp_chmap_sel *s)
     s->allow_waveext = true;
 }
 
+// Classic ALSA-based MPlayer layouts.
+void mp_chmap_sel_add_alsa_def(struct mp_chmap_sel *s)
+{
+    for (int n = 0; n < MP_NUM_CHANNELS; n++) {
+        struct mp_chmap t;
+        mp_chmap_from_channels_alsa(&t, n);
+        if (t.num)
+            mp_chmap_sel_add_map(s, &t);
+    }
+}
+
 // Add a channel map that should be allowed.
 void mp_chmap_sel_add_map(struct mp_chmap_sel *s, const struct mp_chmap *map)
 {
@@ -115,7 +102,7 @@ void mp_chmap_sel_add_map(struct mp_chmap_sel *s, const struct mp_chmap *map)
 // Allow all waveext formats in default order.
 void mp_chmap_sel_add_waveext_def(struct mp_chmap_sel *s)
 {
-    for (int n = 1; n <= MP_NUM_CHANNELS; n++) {
+    for (int n = 1; n < MP_NUM_CHANNELS; n++) {
         struct mp_chmap map;
         mp_chmap_from_channels(&map, n);
         mp_chmap_sel_add_map(s, &map);
@@ -189,10 +176,6 @@ 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];
@@ -202,6 +185,9 @@ 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))
@@ -213,101 +199,64 @@ bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map)
     return false;
 }
 
-// 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)
-{
-    int mindiff = INT_MAX;
+#define UPMIX_IDX 0
+#define DOWNMIX_IDX 1
+#define FALLBACK_IDX 2
 
-    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;
+static bool test_fallbacks(struct mp_chmap *a, struct mp_chmap *b,
+                           int best_diffs[2], struct mp_chmap best[2])
+{
+    struct mp_chmap diff1, diff2;
+
+    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;
     }
 
-    // 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, b) && mp_chmap_equals(&stereo, a))
-        mindiff = 0;
+    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;
+    }
 
-    return mindiff;
+    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;
 }
 
-// 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;
-
-    struct mp_chmap old_p = *old, new_p = *new;
-    mp_chmap_remove_na(&old_p);
-    mp_chmap_remove_na(&new_p);
-
-    // If the situation is equal with replaced speakers, but the replacement is
-    // perfect for only one of them, let the better one win. This prefers
-    // inexact equivalents over exact supersets.
-    bool perfect_r_new = !new_lost_r && new_p.num <= old_p.num;
-    bool perfect_r_old = !old_lost_r && old_p.num <= new_p.num;
-    if (perfect_r_new != perfect_r_old)
-        return perfect_r_new;
-
-    int old_lost = mp_chmap_diffn(req, old);
-    int new_lost = mp_chmap_diffn(req, new);
-    // If the situation is equal with replaced speakers, pick the better one,
-    // even if it means an upmix.
-    if (new_lost != old_lost)
-        return new_lost < old_lost;
-
-    // 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.
+// 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)
 {
-    struct mp_chmap best = {0};
+    // special case: if possible always fallback mono to stereo (instead of
+    // looking for a multichannel upmix)
+    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;
+    }
+
+    int best_diffs[] = { INT_MAX, INT_MAX, INT_MAX };
+    struct mp_chmap best[] = { {0}, {0}, {0} };
 
     for (int n = 0; n < s->num_chmaps; n++) {
         struct mp_chmap e = s->chmaps[n];
@@ -315,13 +264,25 @@ bool mp_chmap_sel_fallback(const struct mp_chmap_sel *s, struct mp_chmap *map)
         if (mp_chmap_is_unknown(&e))
             continue;
 
-        if (mp_chmap_is_better(map, &best, &e))
-            best = e;
+        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 (best.num) {
-        *map = best;
-        return true;
+    for (int i = UPMIX_IDX; i < MP_ARRAY_SIZE(best); i++) {
+        if (best_diffs[i] < INT_MAX) {
+            *map = best[i];
+            return true;
+        }
     }
 
     return false;
@@ -353,24 +314,3 @@ bool mp_chmap_sel_get_def(const struct mp_chmap_sel *s, struct mp_chmap *map,
     }
     return map->num > 0;
 }
-
-// Print the set of allowed channel layouts.
-void mp_chmal_sel_log(const struct mp_chmap_sel *s, struct mp_log *log, int lev)
-{
-    if (!mp_msg_test(log, lev))
-        return;
-
-    for (int i = 0; i < s->num_chmaps; i++)
-        mp_msg(log, lev, " - %s\n", mp_chmap_to_str(&s->chmaps[i]));
-    for (int i = 0; i < MP_SPEAKER_ID_COUNT; i++) {
-        if (!s->speakers[i])
-            continue;
-        struct mp_chmap l = {.num = 1, .speaker = { i }};
-        mp_msg(log, lev, " - #%s\n",
-                    i == MP_SPEAKER_ID_FC ? "fc" : mp_chmap_to_str_hr(&l));
-    }
-    if (s->allow_waveext)
-        mp_msg(log, lev, " - waveext\n");
-    if (s->allow_any)
-        mp_msg(log, lev, " - anything\n");
-}

audio/chmap_sel.h

@@ -37,6 +37,7 @@ struct mp_chmap_sel {
 void mp_chmap_sel_add_any(struct mp_chmap_sel *s);
 void mp_chmap_sel_add_waveext(struct mp_chmap_sel *s);
 void mp_chmap_sel_add_waveext_def(struct mp_chmap_sel *s);
+void mp_chmap_sel_add_alsa_def(struct mp_chmap_sel *s);
 void mp_chmap_sel_add_map(struct mp_chmap_sel *s, const struct mp_chmap *map);
 void mp_chmap_sel_add_speaker(struct mp_chmap_sel *s, int id);
 bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map);
@@ -44,7 +45,4 @@ bool mp_chmap_sel_fallback(const struct mp_chmap_sel *s, struct mp_chmap *map);
 bool mp_chmap_sel_get_def(const struct mp_chmap_sel *s, struct mp_chmap *map,
                           int num);
 
-struct mp_log;
-void mp_chmal_sel_log(const struct mp_chmap_sel *s, struct mp_log *log, int lev);
-
 #endif

audio/decode/ad.h

@@ -1,18 +1,19 @@
 /*
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #ifndef MPLAYER_AD_H

audio/decode/ad_lavc.c

@@ -1,18 +1,19 @@
 /*
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <stdio.h>
@@ -26,7 +27,7 @@
 #include <libavutil/common.h>
 #include <libavutil/intreadwrite.h>
 
-#include "mpv_talloc.h"
+#include "talloc.h"
 
 #include "config.h"
 #include "common/av_common.h"
@@ -58,7 +59,7 @@ struct ad_lavc_params {
 
 const struct m_sub_options ad_lavc_conf = {
     .opts = (const m_option_t[]) {
-        OPT_FLOATRANGE("ac3drc", ac3drc, 0, 0, 6),
+        OPT_FLOATRANGE("ac3drc", ac3drc, 0, 0, 2),
         OPT_FLAG("downmix", downmix, 0),
         OPT_INTRANGE("threads", threads, 0, 1, 16),
         OPT_KEYVALUELIST("o", avopts, 0),
@@ -66,7 +67,7 @@ const struct m_sub_options ad_lavc_conf = {
     },
     .size = sizeof(struct ad_lavc_params),
     .defaults = &(const struct ad_lavc_params){
-        .ac3drc = 0,
+        .ac3drc = 1.,
         .downmix = 1,
         .threads = 1,
     },
@@ -79,12 +80,12 @@ static int init(struct dec_audio *da, const char *decoder)
     AVCodecContext *lavc_context;
     AVCodec *lavc_codec;
     struct sh_stream *sh = da->header;
-    struct mp_codec_params *c = sh->codec;
+    struct sh_audio *sh_audio = sh->audio;
 
     struct priv *ctx = talloc_zero(NULL, struct priv);
     da->priv = ctx;
 
-    ctx->force_channel_map = c->force_channels;
+    ctx->force_channel_map = sh_audio->force_channels;
 
     lavc_codec = avcodec_find_decoder_by_name(decoder);
     if (!lavc_codec) {
@@ -116,20 +117,24 @@ static int init(struct dec_audio *da, const char *decoder)
 
     mp_set_avopts(da->log, lavc_context, opts->avopts);
 
-    lavc_context->codec_tag = c->codec_tag;
-    lavc_context->sample_rate = c->samplerate;
-    lavc_context->bit_rate = c->bitrate;
-    lavc_context->block_align = c->block_align;
-    lavc_context->bits_per_coded_sample = c->bits_per_coded_sample;
-    lavc_context->channels = c->channels.num;
-    if (!mp_chmap_is_unknown(&c->channels))
-        lavc_context->channel_layout = mp_chmap_to_lavc(&c->channels);
+    lavc_context->codec_tag = sh->format;
+    lavc_context->sample_rate = sh_audio->samplerate;
+    lavc_context->bit_rate = sh_audio->bitrate;
+    lavc_context->block_align = sh_audio->block_align;
+    lavc_context->bits_per_coded_sample = sh_audio->bits_per_coded_sample;
+    lavc_context->channels = sh_audio->channels.num;
+    if (!mp_chmap_is_unknown(&sh_audio->channels))
+        lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
 
     // demux_mkv
-    mp_lavc_set_extradata(lavc_context, c->extradata, c->extradata_size);
+    if (sh_audio->codecdata_len && sh_audio->codecdata &&
+            !lavc_context->extradata) {
+        mp_lavc_set_extradata(lavc_context, sh_audio->codecdata,
+                              sh_audio->codecdata_len);
+    }
 
-    if (c->lav_headers)
-        mp_copy_lav_codec_headers(lavc_context, c->lav_headers);
+    if (sh->lav_headers)
+        mp_copy_lav_codec_headers(lavc_context, sh->lav_headers);
 
     mp_set_avcodec_threads(da->log, lavc_context, opts->threads);
 
@@ -140,6 +145,9 @@ static int init(struct dec_audio *da, const char *decoder)
         return 0;
     }
 
+    if (lavc_context->bit_rate != 0)
+        da->bitrate = lavc_context->bit_rate;
+
     return 1;
 }
 
@@ -191,6 +199,12 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
     AVPacket pkt;
     mp_set_av_packet(&pkt, mpkt, NULL);
 
+    // If we don't have a PTS yet, use the first packet PTS we can get.
+    if (da->pts == MP_NOPTS_VALUE && mpkt && mpkt->pts != MP_NOPTS_VALUE) {
+        da->pts = mpkt->pts;
+        da->pts_offset = 0;
+    }
+
     int got_frame = 0;
     av_frame_unref(priv->avframe);
     int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
@@ -219,6 +233,10 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
         return mpkt ? AD_OK : AD_EOF;
 
     double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
+    if (out_pts != MP_NOPTS_VALUE) {
+        da->pts = out_pts;
+        da->pts_offset = 0;
+    }
 
     struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
     if (!mpframe)
@@ -228,13 +246,12 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
     if (lavc_chmap.num != avctx->channels)
         mp_chmap_from_channels(&lavc_chmap, avctx->channels);
     if (priv->force_channel_map) {
-        if (lavc_chmap.num == da->header->codec->channels.num)
-            lavc_chmap = da->header->codec->channels;
+        struct sh_audio *sh_audio = da->header->audio;
+        if (lavc_chmap.num == sh_audio->channels.num)
+            lavc_chmap = sh_audio->channels;
     }
     mp_audio_set_channels(mpframe, &lavc_chmap);
 
-    mpframe->pts = out_pts;
-
 #if HAVE_AVFRAME_SKIP_SAMPLES
     AVFrameSideData *sd =
         av_frame_get_side_data(priv->avframe, AV_FRAME_DATA_SKIP_SAMPLES);
@@ -245,8 +262,7 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
         uint32_t skip = MPMIN(priv->skip_samples, mpframe->samples);
         if (skip) {
             mp_audio_skip_samples(mpframe, skip);
-            if (mpframe->pts != MP_NOPTS_VALUE)
-                mpframe->pts += skip / (double)mpframe->rate;
+            da->pts_offset += skip;
             priv->skip_samples -= skip;
         }
         if (pad <= mpframe->samples)

audio/decode/ad_mpg123.c

@@ -0,0 +1,304 @@
+/*
+ * MPEG 1.0/2.0/2.5 audio layer I, II, III decoding with libmpg123
+ *
+ * Copyright (C) 2010-2013 Thomas Orgis <thomas@orgis.org>
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+
+#include "ad.h"
+#include "common/msg.h"
+
+#include <mpg123.h>
+
+#if (defined MPG123_API_VERSION) && (MPG123_API_VERSION < 33)
+#error "This should not happen"
+#endif
+
+struct ad_mpg123_context {
+    mpg123_handle *handle;
+    int sample_size;
+    /* Running mean for bit rate, stream length estimation. */
+    float mean_rate;
+    unsigned int mean_count;
+    /* Time delay for updates. */
+    short delay;
+    /* If the stream is actually VBR. */
+    char vbr;
+    struct mp_audio frame;
+};
+
+static void uninit(struct dec_audio *da)
+{
+    struct ad_mpg123_context *con = da->priv;
+
+    mpg123_close(con->handle);
+    mpg123_delete(con->handle);
+    mpg123_exit();
+}
+
+/* This initializes libmpg123 and prepares the handle, including funky
+ * parameters. */
+static int init(struct dec_audio *da, const char *decoder)
+{
+    int err;
+    struct ad_mpg123_context *con;
+    /* Assumption: You always call preinit + init + uninit, on every file.
+     * But you stop at preinit in case it fails.
+     * If that is not true, one must ensure not to call mpg123_init / exit
+     * twice in a row. */
+    if (mpg123_init() != MPG123_OK)
+        return 0;
+
+    da->priv = talloc_zero(NULL, struct ad_mpg123_context);
+    con = da->priv;
+    /* Auto-choice of optimized decoder (first argument NULL). */
+    con->handle = mpg123_new(NULL, &err);
+    if (!con->handle)
+        goto bad_end;
+
+    /* Basic settings.
+     * Don't spill messages, enable better resync with non-seekable streams.
+     * Give both flags individually without error checking to keep going with
+     * old libmpg123. Generally, it is not fatal if the flags are not
+     * honored */
+    mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_QUIET, 0.0);
+    /* Do not bail out on malformed streams at all.
+     * MPlayer does not handle a decoder throwing the towel on crappy input. */
+    mpg123_param(con->handle, MPG123_RESYNC_LIMIT, -1, 0.0);
+
+    /* Open decisions: Configure libmpg123 to force encoding (or stay open about
+     * library builds that support only float or int32 output), (de)configure
+     * gapless decoding (won't work with seeking in MPlayer, though).
+     * Don't forget to eventually enable ReplayGain/RVA support, too.
+     * Let's try to run with the default for now. */
+
+    /* That would produce floating point output.
+     * You can get 32 and 24 bit ints, even 8 bit via format matrix.
+     * If wanting a specific encoding here, configure format matrix and
+     * make sure it is in set_format(). */
+    /* mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_FORCE_FLOAT, 0.); */
+
+    /* Example for RVA choice (available since libmpg123 1.0.0):
+    mpg123_param(con->handle, MPG123_RVA, MPG123_RVA_MIX, 0.0) */
+
+    /* Prevent funky automatic resampling.
+     * This way, we can be sure that one frame will never produce
+     * more than 1152 stereo samples.
+     * Background:
+     * Going to decode directly to the output buffer. It is important to have
+     * MPG123_AUTO_RESAMPLE disabled for the buffer size being an all-time
+     * limit.
+     * We need at least 1152 samples. dec_audio.c normally guarantees this. */
+    mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0.);
+
+    err = mpg123_open_feed(con->handle);
+    if (err != MPG123_OK)
+        goto bad_end;
+
+    return 1;
+
+  bad_end:
+    if (!con->handle) {
+        MP_ERR(da, "mpg123 preinit error: %s\n", mpg123_plain_strerror(err));
+    } else {
+        MP_ERR(da, "mpg123 preinit error: %s\n", mpg123_strerror(con->handle));
+    }
+
+    uninit(da);
+    return 0;
+}
+
+static int mpg123_format_to_af(int mpg123_encoding)
+{
+    /* Without external force, mpg123 will always choose signed encoding,
+     * and non-16-bit only on builds that don't support it.
+     * Be reminded that it doesn't matter to the MPEG file what encoding
+     * is produced from it. */
+    switch (mpg123_encoding) {
+    case MPG123_ENC_SIGNED_8:   return AF_FORMAT_S8;
+    case MPG123_ENC_SIGNED_16:  return AF_FORMAT_S16;
+    case MPG123_ENC_SIGNED_32:  return AF_FORMAT_S32;
+    case MPG123_ENC_FLOAT_32:   return AF_FORMAT_FLOAT;
+    }
+    return 0;
+}
+
+/* Compute bitrate from frame size. */
+static int compute_bitrate(struct mpg123_frameinfo *i)
+{
+    static const int samples_per_frame[4][4] = {
+        {-1, 384, 1152, 1152},  /* MPEG 1 */
+        {-1, 384, 1152,  576},  /* MPEG 2 */
+        {-1, 384, 1152,  576},  /* MPEG 2.5 */
+        {-1,  -1,   -1,   -1},  /* Unknown */
+    };
+    return (int) ((i->framesize + 4) * 8 * i->rate /
+                  samples_per_frame[i->version][i->layer] + 0.5);
+}
+
+/* Update mean bitrate. This could be dropped if accurate time display
+ * on audio file playback is not desired. */
+static void update_info(struct dec_audio *da)
+{
+    struct ad_mpg123_context *con = da->priv;
+    struct mpg123_frameinfo finfo;
+    if (mpg123_info(con->handle, &finfo) != MPG123_OK)
+        return;
+
+    /* finfo.bitrate is expressed in kilobits */
+    const int bitrate = finfo.bitrate * 1000;
+
+    if (finfo.vbr != MPG123_CBR) {
+        if (--con->delay < 1) {
+            if (++con->mean_count > ((unsigned int) -1) / 2)
+                con->mean_count = ((unsigned int) -1) / 4;
+
+            /* Might not be numerically optimal, but works fine enough. */
+            con->mean_rate = ((con->mean_count - 1) * con->mean_rate +
+                              bitrate) / con->mean_count;
+            da->bitrate = (int) (con->mean_rate + 0.5);
+
+            con->delay = 10;
+        }
+    } else {
+        da->bitrate = bitrate ? bitrate : compute_bitrate(&finfo);
+        con->delay      = 1;
+        con->mean_rate  = 0.;
+        con->mean_count = 0;
+    }
+}
+
+/* libmpg123 has a new format ready; query and store, return return value
+   of mpg123_getformat() */
+static int set_format(struct dec_audio *da)
+{
+    struct ad_mpg123_context *con = da->priv;
+    int ret;
+    long rate;
+    int channels;
+    int encoding;
+    ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
+    if (ret == MPG123_OK) {
+        mp_audio_set_num_channels(&con->frame, channels);
+        con->frame.rate = rate;
+        int af = mpg123_format_to_af(encoding);
+        if (!af) {
+            /* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
+            MP_ERR(da, "Bad encoding from mpg123: %i.\n", encoding);
+            return MPG123_ERR;
+        }
+        mp_audio_set_format(&con->frame, af);
+        con->sample_size = channels * af_fmt2bps(af);
+    }
+    return ret;
+}
+
+static int decode_packet(struct dec_audio *da, struct mp_audio **out)
+{
+    struct ad_mpg123_context *con = da->priv;
+    int ret;
+
+    struct demux_packet *pkt;
+    if (demux_read_packet_async(da->header, &pkt) == 0)
+        return AD_WAIT;
+    if (!pkt)
+        return AD_EOF;
+
+    /* Next bytes from that presentation time. */
+    if (pkt->pts != MP_NOPTS_VALUE) {
+        da->pts        = pkt->pts;
+        da->pts_offset = 0;
+    }
+
+    /* Have to use mpg123_feed() to avoid decoding here. */
+    ret = mpg123_feed(con->handle, pkt->buffer, pkt->len);
+    talloc_free(pkt);
+
+    if (ret != MPG123_OK)
+        goto mpg123_fail;
+
+    unsigned char *audio = NULL;
+    size_t bytes = 0;
+    ret = mpg123_decode_frame(con->handle, NULL, &audio, &bytes);
+
+    if (ret == MPG123_NEED_MORE)
+        return 0;
+
+    if (ret != MPG123_OK && ret != MPG123_DONE && ret != MPG123_NEW_FORMAT)
+        goto mpg123_fail;
+
+    ret = set_format(da);
+    if (ret != MPG123_OK)
+        goto mpg123_fail;
+
+    if (con->sample_size < 1) {
+        MP_ERR(da, "no sample size\n");
+        return AD_ERR;
+    }
+
+    int got_samples = bytes / con->sample_size;
+    *out = mp_audio_pool_get(da->pool, &con->frame, got_samples);
+    if (!*out)
+        return AD_ERR;
+
+    memcpy((*out)->planes[0], audio, bytes);
+
+    update_info(da);
+    return 0;
+
+mpg123_fail:
+    MP_ERR(da, "mpg123 decoding error: %s\n", mpg123_strerror(con->handle));
+    return AD_ERR;
+}
+
+static int control(struct dec_audio *da, int cmd, void *arg)
+{
+    struct ad_mpg123_context *con = da->priv;
+
+    switch (cmd) {
+    case ADCTRL_RESET:
+        mpg123_close(con->handle);
+
+        if (mpg123_open_feed(con->handle) != MPG123_OK) {
+            MP_ERR(da, "mpg123 failed to reopen stream: %s\n",
+                   mpg123_strerror(con->handle));
+            return CONTROL_FALSE;
+        }
+        return CONTROL_TRUE;
+    }
+    return CONTROL_UNKNOWN;
+}
+
+static void add_decoders(struct mp_decoder_list *list)
+{
+    mp_add_decoder(list, "mpg123", "mp3", "mp3",
+                   "High-performance decoder using libmpg123");
+}
+
+const struct ad_functions ad_mpg123 = {
+    .name = "mpg123",
+    .add_decoders = add_decoders,
+    .init = init,
+    .uninit = uninit,
+    .control = control,
+    .decode_packet = decode_packet,
+};

audio/decode/ad_spdif.c

@@ -1,20 +1,21 @@
 /*
+ * This file is part of MPlayer.
+ *
  * Copyright (C) 2012 Naoya OYAMA
  *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <string.h>
@@ -34,12 +35,11 @@
 
 struct spdifContext {
     struct mp_log   *log;
-    enum AVCodecID   codec_id;
     AVFormatContext *lavf_ctx;
+    int              iec61937_packet_size;
     int              out_buffer_len;
     uint8_t          out_buffer[OUTBUF_SIZE];
     bool             need_close;
-    bool             use_dts_hd;
     struct mp_audio  fmt;
 };
 
@@ -78,78 +78,17 @@ 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();
@@ -169,50 +108,54 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
     if (!stream)
         goto fail;
 
-    stream->codec->codec_id = spdif_ctx->codec_id;
+    stream->codec->codec_id = mp_codec_to_av_codec_id(decoder);
 
     AVDictionary *format_opts = NULL;
 
     int num_channels = 0;
     int sample_format = 0;
     int samplerate = 0;
-    switch (spdif_ctx->codec_id) {
+    switch (stream->codec->codec_id) {
     case AV_CODEC_ID_AAC:
+        spdif_ctx->iec61937_packet_size = 16384;
         sample_format                   = AF_FORMAT_S_AAC;
         samplerate                      = 48000;
         num_channels                    = 2;
         break;
     case AV_CODEC_ID_AC3:
+        spdif_ctx->iec61937_packet_size = 6144;
         sample_format                   = AF_FORMAT_S_AC3;
         samplerate                      = 48000;
         num_channels                    = 2;
         break;
-    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) {
+    case AV_CODEC_ID_DTS:
+        if (da->opts->dtshd) {
             av_dict_set(&format_opts, "dtshd_rate", "768000", 0); // 4*192000
-            sample_format               = AF_FORMAT_S_DTSHD;
-            samplerate                  = 192000;
-            num_channels                = 2*4;
+            spdif_ctx->iec61937_packet_size = 32768;
+            sample_format                   = AF_FORMAT_S_DTSHD;
+            samplerate                      = 192000;
+            num_channels                    = 2*4;
         } else {
-            sample_format               = AF_FORMAT_S_DTS;
-            samplerate                  = 48000;
-            num_channels                = 2;
+            spdif_ctx->iec61937_packet_size = 32768;
+            sample_format                   = AF_FORMAT_S_DTS;
+            samplerate                      = 48000;
+            num_channels                    = 2;
         }
         break;
-    }
     case AV_CODEC_ID_EAC3:
+        spdif_ctx->iec61937_packet_size = 24576;
         sample_format                   = AF_FORMAT_S_EAC3;
         samplerate                      = 192000;
         num_channels                    = 2;
         break;
     case AV_CODEC_ID_MP3:
+        spdif_ctx->iec61937_packet_size = 4608;
         sample_format                   = AF_FORMAT_S_MP3;
         samplerate                      = 48000;
         num_channels                    = 2;
         break;
     case AV_CODEC_ID_TRUEHD:
+        spdif_ctx->iec61937_packet_size = 61440;
         sample_format                   = AF_FORMAT_S_TRUEHD;
         samplerate                      = 192000;
         num_channels                    = 8;
@@ -233,16 +176,17 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
 
     spdif_ctx->need_close = true;
 
-    return 0;
+    return 1;
 
 fail:
     uninit(da);
-    return -1;
+    return 0;
 }
 
 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;
 
@@ -253,18 +197,17 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
     if (!mpkt)
         return AD_EOF;
 
-    double pts = mpkt->pts;
-
     AVPacket pkt;
     mp_set_av_packet(&pkt, mpkt, NULL);
     pkt.pts = pkt.dts = 0;
-    if (!spdif_ctx->lavf_ctx) {
-        if (init_filter(da, &pkt) < 0)
-            return AD_ERR;
+    MP_VERBOSE(da, "spdif packet, size=%d\n", pkt.size);
+    if (mpkt->pts != MP_NOPTS_VALUE) {
+        da->pts        = mpkt->pts;
+        da->pts_offset = 0;
     }
-    int ret = av_write_frame(spdif_ctx->lavf_ctx, &pkt);
+    int ret = av_write_frame(lavf_ctx, &pkt);
     talloc_free(mpkt);
-    avio_flush(spdif_ctx->lavf_ctx->pb);
+    avio_flush(lavf_ctx->pb);
     if (ret < 0)
         return AD_ERR;
 
@@ -274,7 +217,6 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
         return AD_ERR;
 
     memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
-    (*out)->pts = pts;
 
     return 0;
 }
@@ -303,8 +245,6 @@ 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 = {

audio/decode/dec_audio.c

@@ -1,30 +1,31 @@
 /*
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
-#include <math.h>
 #include <assert.h>
 
 #include <libavutil/mem.h>
 
 #include "demux/codec_tags.h"
 
+#include "config.h"
 #include "common/codecs.h"
 #include "common/msg.h"
 #include "misc/bstr.h"
@@ -42,18 +43,21 @@
 
 #include "audio/filter/af.h"
 
+extern const struct ad_functions ad_mpg123;
 extern const struct ad_functions ad_lavc;
 extern const struct ad_functions ad_spdif;
 
 static const struct ad_functions * const ad_drivers[] = {
     &ad_lavc,
+#if HAVE_MPG123
+    &ad_mpg123,
+#endif
     &ad_spdif,
     NULL
 };
 
 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);
@@ -61,8 +65,6 @@ 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)
@@ -85,21 +87,11 @@ struct mp_decoder_list *audio_decoder_list(void)
     return list;
 }
 
-static struct mp_decoder_list *audio_select_decoders(struct dec_audio *d_audio)
+static struct mp_decoder_list *audio_select_decoders(const char *codec,
+                                                     char *selection)
 {
-    struct MPOpts *opts = d_audio->opts;
-    const char *codec = d_audio->header->codec->codec;
-
     struct mp_decoder_list *list = audio_decoder_list();
-    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;
-    }
+    struct mp_decoder_list *new = mp_select_decoders(list, codec, selection);
     talloc_free(list);
     return new;
 }
@@ -113,13 +105,14 @@ static const struct ad_functions *find_driver(const char *name)
     return NULL;
 }
 
-int audio_init_best_codec(struct dec_audio *d_audio)
+int audio_init_best_codec(struct dec_audio *d_audio, char *audio_decoders)
 {
-    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);
+    struct mp_decoder_list *list =
+        audio_select_decoders(d_audio->header->codec, audio_decoders);
 
     mp_print_decoders(d_audio->log, MSGL_V, "Codec list:", list);
 
@@ -146,7 +139,7 @@ int audio_init_best_codec(struct dec_audio *d_audio)
         MP_VERBOSE(d_audio, "Selected audio codec: %s\n", d_audio->decoder_desc);
     } else {
         MP_ERR(d_audio, "Failed to initialize an audio decoder for codec '%s'.\n",
-               d_audio->header->codec->codec);
+               d_audio->header->codec ? d_audio->header->codec : "<unknown>");
     }
 
     talloc_free(list);
@@ -158,8 +151,8 @@ void audio_uninit(struct dec_audio *d_audio)
     if (!d_audio)
         return;
     MP_VERBOSE(d_audio, "Uninit audio filters...\n");
-    uninit_decoder(d_audio);
     af_destroy(d_audio->afilter);
+    uninit_decoder(d_audio);
     talloc_free(d_audio->waiting);
     talloc_free(d_audio);
 }
@@ -172,37 +165,10 @@ static int decode_new_frame(struct dec_audio *da)
             return ret;
 
         if (da->waiting) {
-            if (da->waiting->pts != MP_NOPTS_VALUE) {
-                if (da->pts != MP_NOPTS_VALUE) {
-                    da->pts += da->pts_offset / (double)da->waiting->rate;
-                    da->pts_offset = 0;
-                }
-                double newpts = da->waiting->pts;
-                // Keep the interpolated timestamp if it doesn't deviate more
-                // than 1 ms from the real one. (MKV rounded timestamps.)
-                if (da->pts == MP_NOPTS_VALUE || da->pts_offset != 0 ||
-                    fabs(da->pts - newpts) > 0.001)
-                {
-                    // Attempt to detect jumps in PTS. Even for the lowest
-                    // sample rates and with worst container rounded timestamp,
-                    // this should be a margin more than enough.
-                    if (da->pts != MP_NOPTS_VALUE && fabs(newpts - da->pts) > 0.1)
-                    {
-                        MP_WARN(da, "Invalid audio PTS: %f -> %f\n",
-                                da->pts, newpts);
-                        da->pts_reset = true;
-                    }
-                    da->pts = da->waiting->pts;
-                    da->pts_offset = 0;
-                }
-            }
             da->pts_offset += da->waiting->samples;
             da->decode_format = *da->waiting;
             mp_audio_set_null_data(&da->decode_format);
         }
-
-        if (da->pts == MP_NOPTS_VALUE && da->header->missing_timestamps)
-            da->pts = 0;
     }
     return mp_audio_config_valid(da->waiting) ? AD_OK : AD_ERR;
 }
@@ -284,7 +250,6 @@ void audio_reset_decoding(struct dec_audio *d_audio)
     af_seek_reset(d_audio->afilter);
     d_audio->pts = MP_NOPTS_VALUE;
     d_audio->pts_offset = 0;
-    d_audio->pts_reset = false;
     if (d_audio->waiting) {
         talloc_free(d_audio->waiting);
         d_audio->waiting = NULL;

audio/decode/dec_audio.h

@@ -1,18 +1,19 @@
 /*
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #ifndef MPLAYER_DEC_AUDIO_H
@@ -30,7 +31,6 @@ struct dec_audio {
     struct mp_log *log;
     struct MPOpts *opts;
     struct mpv_global *global;
-    bool spdif_passthrough, spdif_failed;
     const struct ad_functions *ad_driver;
     struct sh_stream *header;
     struct af_stream *afilter;
@@ -45,8 +45,6 @@ struct dec_audio {
     double pts;
     // number of samples output by decoder after last known pts
     int pts_offset;
-    // set every time a jump in timestamps is encountered
-    bool pts_reset;
     // For free use by the ad_driver
     void *priv;
 };
@@ -60,7 +58,7 @@ enum {
 };
 
 struct mp_decoder_list *audio_decoder_list(void);
-int audio_init_best_codec(struct dec_audio *d_audio);
+int audio_init_best_codec(struct dec_audio *d_audio, char *audio_decoders);
 int audio_decode(struct dec_audio *d_audio, struct mp_audio_buffer *outbuf,
                  int minsamples);
 int initial_audio_decode(struct dec_audio *d_audio);

audio/filter/af.c

@@ -1,18 +1,19 @@
 /*
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include "config.h"
@@ -31,37 +32,68 @@
 #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_forcespeed;
 extern const struct af_info af_info_bs2b;
 extern const struct af_info af_info_lavfi;
-extern const struct af_info af_info_rubberband;
+extern const struct af_info af_info_convert24;
+extern const struct af_info af_info_convertsignendian;
 
 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,
-#if HAVE_RUBBERBAND
-    &af_info_rubberband,
+    &af_info_sweep,
+    &af_info_hrtf,
+#if HAVE_LADSPA
+    &af_info_ladspa,
 #endif
+    &af_info_center,
+    &af_info_sinesuppress,
+    &af_info_karaoke,
+    &af_info_forcespeed,
     &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
 };
 
@@ -159,11 +191,24 @@ 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 = desc.p,
+        .info = info,
         .data = talloc_zero(af, struct mp_audio),
         .log = mp_log_new(af, s->log, name),
         .replaygain_data = s->replaygain_data,
@@ -288,9 +333,89 @@ 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 && strcmp(af->info->name, "lavrresample") == 0;
+    return af && af->info->test_conversion != NULL;
 }
 
 // in is what af can take as input - insert a conversion filter if the actual
@@ -310,24 +435,21 @@ static int af_fix_format_conversion(struct af_stream *s,
     if (actual.format == in.format)
         return AF_FALSE;
     int dstfmt = in.format;
-    char *filter = "lavrresample";
-    if (!af_lavrresample_test_conversion(actual.format, dstfmt))
+    char *filter = af_find_conversion_filter(actual.format, &dstfmt);
+    if (!filter)
         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))) {
-        af_remove(s, new);
+    if (AF_OK != (rv = new->control(new, AF_CONTROL_SET_FORMAT, &dstfmt)))
         return rv;
-    }
     *p_af = new;
     return AF_OK;
 }
@@ -381,15 +503,6 @@ static int af_fix_rate(struct af_stream *s, struct af_instance **p_af,
     return AF_OK;
 }
 
-static void reset_formats(struct af_stream *s)
-{
-    for (struct af_instance *af = s->first; af; af = af->next) {
-        af->control(af, AF_CONTROL_SET_RESAMPLE_RATE, &(int){0});
-        af->control(af, AF_CONTROL_SET_CHANNELS, &(struct mp_chmap){0});
-        af->control(af, AF_CONTROL_SET_FORMAT, &(int){0});
-    }
-}
-
 // Return AF_OK on success or AF_ERROR on failure.
 // Warning:
 // A failed af_reinit() leaves the audio chain behind in a useless, broken
@@ -400,13 +513,12 @@ static int af_reinit(struct af_stream *s)
 {
     remove_auto_inserted_filters(s);
     af_chain_forget_frames(s);
-    reset_formats(s);
     s->first->fmt_in = s->first->fmt_out = s->input;
     // 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 4 retries per filter (channel, rate, format conversions)
-    int max_retry = 4;
+    // Up to 7 retries per filter (channel, rate, 5x format conversions)
+    int max_retry = af_count_filters(s) * 7;
     int retry = 0;
     while (af) {
         if (retry >= max_retry)
@@ -447,24 +559,6 @@ static int af_reinit(struct af_stream *s)
                 retry++;
                 continue;
             }
-            // If the format conversion is (probably) caused by spdif, then
-            // (as a feature) drop the filter, instead of failing hard.
-            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_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);
-                    struct af_instance *aft = af->prev;
-                    af_remove(s, af);
-                    af = aft->next;
-                    retry++;
-                    continue;
-                }
-            }
             goto negotiate_error;
         }
         case AF_DETACH: { // Filter is redundant and wants to be unloaded
@@ -479,8 +573,6 @@ 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
@@ -595,14 +687,8 @@ int af_init(struct af_stream *s)
    to the stream s. The filter will be inserted somewhere nice in the
    list of filters. The return value is a pointer to the new filter,
    If the filter couldn't be added the return value is NULL. */
-struct af_instance *af_add(struct af_stream *s, char *name, char *label,
-                           char **args)
+struct af_instance *af_add(struct af_stream *s, char *name, char **args)
 {
-    assert(label);
-
-    if (af_find_by_label(s, label))
-        return NULL;
-
     struct af_instance *new;
     // Insert the filter somewhere nice
     if (af_is_conversion_filter(s->first->next))
@@ -611,14 +697,17 @@ struct af_instance *af_add(struct af_stream *s, char *name, char *label,
         new = af_prepend(s, s->first->next, name, args);
     if (!new)
         return NULL;
-    new->label = talloc_strdup(new, label);
 
-    // Reinitialize the filter list
+    // Reinitalize the filter list
     if (af_reinit(s) != AF_OK) {
-        af_remove_by_label(s, label);
+        af_remove(s, new);
+        if (af_reinit(s) != AF_OK) {
+            af_uninit(s);
+            af_init(s);
+        }
         return NULL;
     }
-    return af_find_by_label(s, label);
+    return new;
 }
 
 struct af_instance *af_find_by_label(struct af_stream *s, char *label)
@@ -683,18 +772,6 @@ 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)
@@ -724,16 +801,6 @@ 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)
@@ -773,7 +840,6 @@ 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;

audio/filter/af.h

@@ -1,18 +1,19 @@
 /*
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #ifndef MPLAYER_AF_H
@@ -27,7 +28,6 @@
 #include "audio/chmap.h"
 #include "audio/audio.h"
 #include "common/msg.h"
-#include "common/common.h"
 
 struct af_instance;
 struct mpv_global;
@@ -35,6 +35,10 @@ 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
 
@@ -43,7 +47,9 @@ 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;
@@ -100,12 +106,13 @@ struct af_stream {
 };
 
 // Return values
-#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
+#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
 
 // Parameters for af_control_*
 enum af_control {
@@ -122,7 +129,6 @@ 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
@@ -135,13 +141,11 @@ struct af_stream *af_new(struct mpv_global *global);
 void af_destroy(struct af_stream *s);
 int af_init(struct af_stream *s);
 void af_uninit(struct af_stream *s);
-struct af_instance *af_add(struct af_stream *s, char *name, char *label,
-                           char **args);
+struct af_instance *af_add(struct af_stream *s, char *name, char **args);
 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);
@@ -154,9 +158,10 @@ 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_to_dB(int n, float *in, float *out, float k);
 int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma);
+int af_to_ms(int n, int *in, float *out, int rate);
 float af_softclip(float a);
 
-bool af_lavrresample_test_conversion(int src_format, int dst_format);
-
 #endif /* MPLAYER_AF_H */

audio/filter/af_bs2b.c

@@ -0,0 +1,172 @@
+/*
+ * The Bauer stereophonic-to-binaural DSP using bs2b library:
+ * http://bs2b.sourceforge.net/
+ *
+ * Copyright (c) 2009 Andrew Savchenko
+ *
+ * This file is part of MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#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,
+                   ({"default", BS2B_DEFAULT_CLEVEL},
+                    {"cmoy", BS2B_CMOY_CLEVEL},
+                    {"jmeier", BS2B_JMEIER_CLEVEL})),
+        {0}
+    },
+};

audio/filter/af_center.c

@@ -0,0 +1,105 @@
+/*
+ * 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 MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#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}
+    },
+};

audio/filter/af_channels.c

@@ -5,20 +5,21 @@
  *
  * Original author: Anders
  *
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <stdio.h>

audio/filter/af_convert24.c

@@ -0,0 +1,122 @@
+/*
+ * 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,
+};

audio/filter/af_convertsignendian.c

@@ -0,0 +1,107 @@
+/*
+ * 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,
+};

audio/filter/af_delay.c

@@ -5,20 +5,21 @@
  *
  * Original author: Anders
  *
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <stdio.h>

audio/filter/af_drc.c

@@ -1,20 +1,21 @@
 /*
  * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard
  *
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <stdio.h>
@@ -131,7 +132,7 @@ static void method1_int16(af_drc_t *s, struct mp_audio *c)
     data[i] = tmp;
   }
 
-  // Evaluation of newavg (not 100% accurate because of values clamping)
+  // Evaulation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -168,7 +169,7 @@ static void method1_float(af_drc_t *s, struct mp_audio *c)
   for (i = 0; i < len; i++)
     data[i] *= s->mul;
 
-  // Evaluation of newavg (not 100% accurate because of values clamping)
+  // Evaulation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -216,7 +217,7 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c)
     data[i] = tmp;
   }
 
-  // Evaluation of newavg (not 100% accurate because of values clamping)
+  // Evaulation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -262,7 +263,7 @@ static void method2_float(af_drc_t *s, struct mp_audio *c)
   for (i = 0; i < len; i++)
     data[i] *= s->mul;
 
-  // Evaluation of newavg (not 100% accurate because of values clamping)
+  // Evaulation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -324,6 +325,7 @@ 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[]) {

audio/filter/af_dummy.c

@@ -0,0 +1,62 @@
+/*
+ * 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 MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#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,
+};

audio/filter/af_equalizer.c

@@ -6,20 +6,21 @@
  *
  * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
  *
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <stdio.h>
@@ -203,6 +204,7 @@ 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[]) {

audio/filter/af_export.c

@@ -0,0 +1,238 @@
+/*
+ * 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 MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#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}
+    },
+};

audio/filter/af_extrastereo.c

@@ -0,0 +1,133 @@
+/*
+ * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard
+ *
+ * This file is part of MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#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}
+    },
+};

audio/filter/af_forcespeed.c

@@ -0,0 +1,71 @@
+/*
+ * 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 "af.h"
+
+struct priv {
+    double speed;
+};
+
+static int control(struct af_instance *af, int cmd, void *arg)
+{
+    struct priv *priv = af->priv;
+
+    switch (cmd) {
+    case AF_CONTROL_REINIT: {
+        struct mp_audio *in = arg;
+        struct mp_audio orig_in = *in;
+        struct mp_audio *out = af->data;
+
+        mp_audio_copy_config(out, in);
+        out->rate = in->rate * priv->speed;
+
+        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
+    }
+    case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
+        priv->speed = *(double *)arg;
+        return AF_OK;
+    }
+    }
+    return AF_UNKNOWN;
+}
+
+static int filter(struct af_instance *af, struct mp_audio *data)
+{
+    if (data)
+        mp_audio_copy_config(data, af->data);
+    af_add_output_frame(af, data);
+    return 0;
+}
+
+static int af_open(struct af_instance *af)
+{
+    struct priv *priv = af->priv;
+    af->control = control;
+    af->filter_frame = filter;
+    priv->speed = 1.0;
+    return AF_OK;
+}
+
+#define OPT_BASE_STRUCT struct priv
+
+const struct af_info af_info_forcespeed = {
+    .info = "Force audio speed",
+    .name = "forcespeed",
+    .open = af_open,
+    .priv_size = sizeof(struct priv),
+};

audio/filter/af_hrtf.c

@@ -0,0 +1,671 @@
+/*
+ * 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 MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+//#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}
+    },
+};

audio/filter/af_hrtf.h

@@ -0,0 +1,511 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#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 */

audio/filter/af_karaoke.c

@@ -0,0 +1,87 @@
+/*
+ * 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 MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#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,
+};

audio/filter/af_ladspa.c

@@ -0,0 +1,852 @@
+/*
+ * LADSPA plugin loader
+ *
+ * Written by Ivo van Poorten <ivop@euronet.nl>
+ * Copyright (C) 2004, 2005
+ *
+ * This file is part of MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+/* ------------------------------------------------------------------------- */
+
+/* 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;
+}
+
+/* ------------------------------------------------------------------------- */

audio/filter/af_lavcac3enc.c

@@ -3,20 +3,21 @@
  *
  * Copyright (C) 2007 Ulion <ulion A gmail P com>
  *
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <stdio.h>
@@ -26,6 +27,7 @@
 #include <assert.h>
 
 #include <libavcodec/avcodec.h>
+#include <libavutil/audioconvert.h>
 #include <libavutil/intreadwrite.h>
 #include <libavutil/common.h>
 #include <libavutil/bswap.h>
@@ -49,6 +51,7 @@ const uint16_t ac3_bitrate_tab[19] = {
 typedef struct af_ac3enc_s {
     struct AVCodec        *lavc_acodec;
     struct AVCodecContext *lavc_actx;
+    AVPacket pkt;
     int bit_rate;
     struct mp_audio *input;     // frame passed to libavcodec
     struct mp_audio *pending;   // unconsumed input data
@@ -73,7 +76,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
         struct mp_audio *in = arg;
         struct mp_audio orig_in = *in;
 
-        if (!af_fmt_is_pcm(in->format) || in->nch < s->cfg_min_channel_num)
+        if (AF_FORMAT_IS_SPECIAL(in->format) || in->nch < s->cfg_min_channel_num)
             return AF_DETACH;
 
         mp_audio_set_format(in, s->in_sampleformat);
@@ -145,6 +148,7 @@ static void uninit(struct af_instance* af)
     af_ac3enc_t *s = af->priv;
 
     if (s) {
+        av_free_packet(&s->pkt);
         if(s->lavc_actx) {
             avcodec_close(s->lavc_actx);
             av_free(s->lavc_actx);
@@ -153,13 +157,6 @@ 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;
@@ -170,7 +167,6 @@ static int filter_frame(struct af_instance *af, struct mp_audio *audio)
 
     talloc_free(s->pending);
     s->pending = audio;
-    update_delay(af);
     return 0;
 }
 
@@ -182,27 +178,22 @@ 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 frame is complete.
-static bool fill_buffer(struct af_instance *af)
+// Return true if the
+static bool fill_buffer(af_ac3enc_t *s)
 {
-    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(af))
+    if (!fill_buffer(s))
         return 0; // need more input
 
     AVFrame *frame = av_frame_alloc();
@@ -210,8 +201,6 @@ static int filter_out(struct af_instance *af)
         MP_FATAL(af, "Could not allocate memory \n");
         return -1;
     }
-    int err = -1;
-
     frame->nb_samples = s->in_samples;
     frame->format = s->lavc_actx->sample_fmt;
     frame->channel_layout = s->lavc_actx->channel_layout;
@@ -221,33 +210,30 @@ static int filter_out(struct af_instance *af)
         frame->data[n] = s->input->planes[n];
     frame->linesize[0] = s->input->samples * s->input->sstride;
 
-    AVPacket pkt = {0};
-    av_init_packet(&pkt);
-
     int ok;
-    int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &pkt, frame, &ok);
+    int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
     av_frame_free(&frame);
     s->input->samples = 0;
     if (lavc_ret < 0 || !ok) {
         MP_FATAL(af, "Encode failed.\n");
-        goto done;
+        return -1;
     }
 
     MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
-            pkt.size, s->pending->samples);
+            s->pkt.size, s->pending->samples);
 
     struct mp_audio *out =
         mp_audio_pool_get(af->out_pool, af->data, s->out_samples);
     if (!out)
-        goto done;
+        return -1;
     mp_audio_copy_attributes(out, s->pending);
 
-    int frame_size = pkt.size;
+    int frame_size = s->pkt.size;
     int header_len = 0;
     char hdr[8];
 
-    if (s->cfg_add_iec61937_header && pkt.size > 5) {
-        int bsmod = pkt.data[5] & 0x7;
+    if (s->cfg_add_iec61937_header && s->pkt.size > 5) {
+        int bsmod = s->pkt.data[5] & 0x7;
         int len = frame_size;
 
         frame_size = AC3_FRAME_SIZE * 2 * 2;
@@ -265,18 +251,13 @@ static int filter_out(struct af_instance *af)
 
     char *buf = (char *)out->planes[0];
     memcpy(buf, hdr, header_len);
-    memcpy(buf + header_len, pkt.data, pkt.size);
-    memset(buf + header_len + pkt.size, 0,
-           frame_size - (header_len + pkt.size));
-    swap_16((uint16_t *)(buf + header_len), pkt.size / 2);
+    memcpy(buf + header_len, s->pkt.data, s->pkt.size);
+    memset(buf + header_len + s->pkt.size, 0,
+            frame_size - (header_len + s->pkt.size));
+    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);
-
-    err = 0;
-done:
-    av_packet_unref(&pkt);
-    return err;
+    return 0;
 }
 
 static int af_open(struct af_instance* af){
@@ -314,6 +295,8 @@ static int af_open(struct af_instance* af){
     MP_VERBOSE(af, "[af_lavcac3enc]: in sample format: %s\n",
            af_fmt_to_str(s->in_sampleformat));
 
+    av_init_packet(&s->pkt);
+
     s->input = talloc_zero(s, struct mp_audio);
 
     if (s->cfg_bit_rate) {

audio/filter/af_lavfi.c

@@ -34,14 +34,11 @@
 #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"
 
@@ -64,10 +61,6 @@ struct priv {
 
     AVRational timebase_out;
 
-    bool eof;
-
-    struct mp_tags *metadata;
-
     // options
     char *cfg_graph;
     char **cfg_avopts;
@@ -78,8 +71,6 @@ static void destroy_graph(struct af_instance *af)
     struct priv *p = af->priv;
     avfilter_graph_free(&p->graph);
     p->in = p->out = NULL;
-    p->samples_in = 0;
-    p->eof = false;
 }
 
 static bool recreate_graph(struct af_instance *af, struct mp_audio *config)
@@ -176,12 +167,6 @@ error:
     return false;
 }
 
-static void reset(struct af_instance *af)
-{
-    if (!recreate_graph(af, &af->fmt_in))
-        MP_FATAL(af, "Can't recreate libavfilter filter after a seek reset.\n");
-}
-
 static int control(struct af_instance *af, int cmd, void *arg)
 {
     struct priv *p = af->priv;
@@ -195,10 +180,6 @@ 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);
 
-        // Removing this requires fixing AVFrame.data vs. AVFrame.extended_data
-        if (in->channels.num > AV_NUM_DATA_POINTERS)
-            return AF_ERROR;
-
         if (!mp_chmap_is_lavc(&in->channels))
             mp_chmap_reorder_to_lavc(&in->channels); // will always work
 
@@ -215,50 +196,23 @@ static int control(struct af_instance *af, int cmd, void *arg)
         mp_chmap_from_lavc(&out_cm, l_out->channel_layout);
         mp_audio_set_channels(out, &out_cm);
 
-        if (!mp_audio_config_valid(out) || out->channels.num > AV_NUM_DATA_POINTERS)
+        if (!mp_audio_config_valid(out))
             return AF_ERROR;
 
         p->timebase_out = l_out->time_base;
 
         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;
     }
     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;
-    AVFrame *frame = NULL;
-
-    if (p->eof && data)
-        reset(af);
-
-    if (!p->graph)
-        goto error;
-
     AVFilterLink *l_in = p->in->outputs[0];
 
+    AVFrame *frame = NULL;
     if (data) {
         frame = av_frame_alloc();
         if (!frame)
@@ -301,9 +255,6 @@ static int filter_out(struct af_instance *af)
 {
     struct priv *p = af->priv;
 
-    if (!p->graph)
-        goto error;
-
     AVFrame *frame = av_frame_alloc();
     if (!frame)
         goto error;
@@ -311,10 +262,7 @@ static int filter_out(struct af_instance *af)
     int err = av_buffersink_get_frame(p->out, frame);
     if (err == AVERROR(EAGAIN) || err == AVERROR_EOF) {
         // Not an error situation - no more output buffers in queue.
-        // AVERROR_EOF means we shouldn't even give the filter more
-        // input, but we don't handle that completely correctly.
         av_frame_free(&frame);
-        p->eof |= err == AVERROR_EOF;
         return 0;
     }
 
@@ -333,7 +281,6 @@ 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;
@@ -353,6 +300,8 @@ static int af_open(struct af_instance *af)
     af->uninit = uninit;
     af->filter_frame = filter_frame;
     af->filter_out = filter_out;
+    // Removing this requires fixing AVFrame.data vs. AVFrame.extended_data
+    assert(MP_NUM_CHANNELS <= AV_NUM_DATA_POINTERS);
     return AF_OK;
 }
 

audio/filter/af_lavrresample.c

@@ -2,38 +2,37 @@
  * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
  * Copyright (c) 2013 Stefano Pigozzi <stefano.pigozzi@gmail.com>
  *
+ * This file is part of mpv.
  * Based on Michael Niedermayer's lavcresample.
  *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #include <inttypes.h>
-#include <math.h>
 #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"
+#include "talloc.h"
 #include "config.h"
 
 #if HAVE_LIBAVRESAMPLE
@@ -49,7 +48,6 @@
 #define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \
     swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples)
 #define avresample_set_channel_mapping swr_set_channel_mapping
-#define avresample_set_compensation swr_set_compensation
 #else
 #error "config.h broken or no resampler found"
 #endif
@@ -59,33 +57,14 @@
 #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
+    int in_rate;
     int in_format;
     struct mp_chmap in_channels;
     int out_rate;
@@ -93,151 +72,100 @@ struct af_resample {
     struct mp_chmap out_channels;
 };
 
+struct af_resample {
+    int allow_detach;
+    char **avopts;
+    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 double get_delay(struct af_resample *s)
+static int get_delay(struct af_resample *s)
 {
-    return avresample_get_delay(s->avrctx) / (double)s->in_rate +
-           avresample_available(s->avrctx) / (double)s->out_rate;
+    return avresample_get_delay(s->avrctx);
 }
 static void drop_all_output(struct af_resample *s)
 {
     while (avresample_read(s->avrctx, NULL, 1000) > 0) {}
 }
-static int get_out_samples(struct af_resample *s, int in_samples)
-{
-    return avresample_get_out_samples(s->avrctx, in_samples);
-}
 #else
-static double get_delay(struct af_resample *s)
+static int get_delay(struct af_resample *s)
 {
-    int64_t base = s->in_rate * (int64_t)s->out_rate;
-    return swr_get_delay(s->avrctx, base) / (double)base;
+    return swr_get_delay(s->avrctx, s->ctx.in_rate);
 }
 static void drop_all_output(struct af_resample *s)
 {
     while (swr_drop_output(s->avrctx, 1000) > 0) {}
 }
-static int get_out_samples(struct af_resample *s, int in_samples)
-{
-#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)
-{
-    return avresample_convert(r,
-        out ? (uint8_t **)out->planes : NULL,
-        out ? mp_audio_get_allocated_size(out) : 0,
-        out ? out->samples : 0,
-        in ? (uint8_t **)in->planes : NULL,
-        in ? mp_audio_get_allocated_size(in) : 0,
-        in ? in->samples : 0);
-}
 
 static double af_resample_default_cutoff(int filter_size)
 {
     return FFMAX(1.0 - 6.5 / (filter_size + 8), 0.80);
 }
 
-static int rate_from_speed(int rate, double speed)
+static bool needs_lavrctx_reconfigure(struct af_resample *s,
+                                      struct mp_audio *in,
+                                      struct mp_audio *out)
 {
-    return lrint(rate * speed);
+    return s->ctx.in_rate     != 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;
+
 }
 
-// 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)
-{
-    if (mp_format == AF_FORMAT_S24)
-        return AV_SAMPLE_FMT_S32;
-    return af_to_avformat(mp_format);
-}
-
-bool af_lavrresample_test_conversion(int src_format, int dst_format)
+static bool test_conversion(int src_format, int dst_format)
 {
     return af_to_avformat(src_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));
+           af_to_avformat(dst_format) != AV_SAMPLE_FMT_NONE;
 }
 
 static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
-                           struct mp_audio *out, bool verbose)
+                           struct mp_audio *out)
 {
     struct af_resample *s = af->priv;
 
-    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 = check_output_conversion(out->format);
-    enum AVSampleFormat out_samplefmtp = av_get_planar_sample_fmt(out_samplefmt);
+    enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
 
-    if (in_samplefmt == AV_SAMPLE_FMT_NONE ||
-        out_samplefmt == AV_SAMPLE_FMT_NONE ||
-        out_samplefmtp == AV_SAMPLE_FMT_NONE)
-        goto error;
+    if (in_samplefmt == AV_SAMPLE_FMT_NONE || out_samplefmt == AV_SAMPLE_FMT_NONE)
+        return AF_ERROR;
 
-    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;
+    avresample_close(s->avrctx);
+    avresample_close(s->avrctx_out);
 
-    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     = in->rate;
+    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_double(s->avrctx, "cutoff",          s->opts.cutoff, 0);
+    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);
 
-#if HAVE_LIBSWRESAMPLE
-    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
+    av_opt_set_double(s->avrctx, "cutoff",          s->ctx.cutoff, 0);
 
     if (mp_set_avopts(af->log, s->avrctx, s->avopts) < 0)
-        goto error;
+        return AF_ERROR;
 
     struct mp_chmap map_in = in->channels;
     struct mp_chmap map_out = out->channels;
@@ -252,104 +180,61 @@ 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);
 
-    struct mp_chmap in_lavc, out_lavc;
-    mp_chmap_from_lavc(&in_lavc, in_ch_layout);
-    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);
-
-    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_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_samplefmtp, 0);
+    av_opt_set_int(s->avrctx, "out_sample_fmt",     out_samplefmt, 0);
 
-    // Just needs the correct number of channels for deplanarization.
-    struct mp_chmap fake_chmap;
-    mp_chmap_set_unknown(&fake_chmap, map_out.num);
-    uint64_t fake_out_ch_layout = mp_chmap_to_lavc_unchecked(&fake_chmap);
-    if (!fake_out_ch_layout)
-        goto error;
-    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);
+    struct mp_chmap in_lavc;
+    mp_chmap_from_lavc(&in_lavc, in_ch_layout);
+    mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
 
-    av_opt_set_int(s->avrctx_out, "in_sample_fmt",      out_samplefmtp, 0);
+    struct mp_chmap out_lavc;
+    mp_chmap_from_lavc(&out_lavc, out_ch_layout);
+    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);
     av_opt_set_int(s->avrctx_out, "out_sample_fmt",     out_samplefmt, 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);
+    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);
 
     // 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");
-        goto error;
+        return AF_ERROR;
     }
     return AF_OK;
-
-error:
-    close_lavrr(af);
-    return AF_ERROR;
 }
 
 
 static int control(struct af_instance *af, int cmd, void *arg)
 {
-    struct af_resample *s = af->priv;
+    struct af_resample *s = (struct af_resample *) af->priv;
+    struct mp_audio *in   = (struct mp_audio *) arg;
+    struct mp_audio *out  = (struct mp_audio *) af->data;
 
     switch (cmd) {
     case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio *out = af->data;
         struct mp_audio orig_in = *in;
 
         if (((out->rate    == in->rate) || (out->rate == 0)) &&
             (out->format   == in->format) &&
             (mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
-            s->allow_detach && s->playback_speed == 1.0)
+            s->allow_detach)
             return AF_DETACH;
 
         if (out->rate == 0)
@@ -360,23 +245,22 @@ 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 (check_output_conversion(out->format) == AV_SAMPLE_FMT_NONE)
+        if (af_to_avformat(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)
-            r = configure_lavrr(af, in, out, true);
+        if (r == AF_OK && needs_lavrctx_reconfigure(s, in, out))
+            r = configure_lavrr(af, in, out);
         return r;
     }
     case AF_CONTROL_SET_FORMAT: {
-        int format = *(int *)arg;
-        if (format && check_output_conversion(format) == AV_SAMPLE_FMT_NONE)
+        if (af_to_avformat(*(int*)arg) == AV_SAMPLE_FMT_NONE)
             return AF_FALSE;
 
-        mp_audio_set_format(af->data, format);
+        mp_audio_set_format(af->data, *(int*)arg);
         return AF_OK;
     }
     case AF_CONTROL_SET_CHANNELS: {
@@ -384,98 +268,75 @@ static int control(struct af_instance *af, int cmd, void *arg)
         return AF_OK;
     }
     case AF_CONTROL_SET_RESAMPLE_RATE:
-        af->data->rate = *(int *)arg;
+        out->rate = *(int *)arg;
         return AF_OK;
-    case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
-        s->playback_speed = *(double *)arg;
-        return AF_OK;
-    }
     case AF_CONTROL_RESET:
-        if (s->avrctx)
-            drop_all_output(s);
+        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)
 {
-    close_lavrr(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);
 }
 
-// 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)
+static bool needs_reorder(int *reorder, int num_ch)
 {
-    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);
-    }
-
-    for (int p = 0; p < mpa->num_planes; p++) {
-        void *ptr = mpa->planes[p];
-        int total = mpa->samples * mpa->spf;
-        if (af_fmt_from_planar(mpa->format) == AF_FORMAT_FLOAT) {
-            for (int s = 0; s < total; s++)
-                ((float *)ptr)[s] = av_clipf(((float *)ptr)[s], -1.0f, 1.0f);
-        } else if (af_fmt_from_planar(mpa->format) == AF_FORMAT_DOUBLE) {
-            for (int s = 0; s < total; s++)
-                ((double *)ptr)[s] = MPCLAMP(((double *)ptr)[s], -1.0, 1.0);
-        }
+    for (int n = 0; n < num_ch; n++) {
+        if (reorder[n] != n)
+            return true;
     }
+    return false;
 }
 
-// This relies on the tricky way mpa was allocated.
-static void reorder_planes(struct mp_audio *mpa, int *reorder,
-                           struct mp_chmap *newmap)
+static void reorder_planes(struct mp_audio *mpa, int *reorder)
 {
     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++) {
-        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);
-        }
+        assert(reorder[n] >= 0 && reorder[n] < mpa->num_planes);
+        mpa->planes[n] = prev.planes[reorder[n]];
     }
 }
 
-static int filter_resample(struct af_instance *af, struct mp_audio *in)
+static int resample_frame(struct AVAudioResampleContext *r,
+                          struct mp_audio *out, struct mp_audio *in)
+{
+    return avresample_convert(r,
+        out ? (uint8_t **)out->planes : NULL,
+        out ? mp_audio_get_allocated_size(out) : 0,
+        out ? out->samples : 0,
+        in ? (uint8_t **)in->planes : NULL,
+        in ? mp_audio_get_allocated_size(in) : 0,
+        in ? in->samples : 0);
+}
+
+static int filter(struct af_instance *af, struct mp_audio *in)
 {
     struct af_resample *s = af->priv;
 
-    int samples = get_out_samples(s, in ? in->samples : 0);
-
-    struct mp_audio out_format = s->pool_fmt;
-    struct mp_audio *out = mp_audio_pool_get(af->out_pool, &out_format, samples);
+    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);
     if (!out)
         goto error;
     if (in)
         mp_audio_copy_attributes(out, in);
 
-    if (!s->avrctx)
-        goto error;
+    mp_audio_realloc_min(out, out->samples);
+
+    af->delay = get_delay(s) / (double)s->ctx.in_rate;
 
     if (out->samples) {
         out->samples = resample_frame(s->avrctx, out, in);
@@ -483,15 +344,11 @@ static int filter_resample(struct af_instance *af, struct mp_audio *in)
             goto error;
     }
 
-    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,
+    if (needs_reorder(s->reorder_out, out->nch)) {
+        if (af_fmt_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,
                                                      out->samples);
             if (!new)
                 goto error;
@@ -504,17 +361,12 @@ static int filter_resample(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);
@@ -522,39 +374,6 @@ error:
     return -1;
 }
 
-static int filter(struct af_instance *af, struct mp_audio *in)
-{
-    struct af_resample *s = af->priv;
-
-    int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
-    bool need_reinit = fabs(new_rate / (double)s->in_rate - 1) > 0.01;
-
-    if (s->avrctx) {
-        AVRational r = av_d2q(s->playback_speed * s->in_rate_af / s->in_rate,
-                              INT_MAX / 2);
-        // Essentially, swr/avresample_set_compensation() does 2 things:
-        // - adjust output sample rate by sample_delta/compensation_distance
-        // - reset the adjustment after compensation_distance output samples
-        // Increase the compensation_distance to avoid undesired reset
-        // semantics - we want to keep the ratio for the whole frame we're
-        // feeding it, until the next filter() call.
-        int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1);
-        r = (AVRational){ r.num * mult, r.den * mult };
-        if (avresample_set_compensation(s->avrctx, r.den - r.num, r.den) < 0)
-            need_reinit = true;
-    }
-
-    if (need_reinit && new_rate != s->in_rate) {
-        // Before reconfiguring, drain the audio that is still buffered
-        // in the resampler.
-        filter_resample(af, NULL);
-        // Reinitialize resampler.
-        configure_lavrr(af, &af->fmt_in, &af->fmt_out, false);
-    }
-
-    return filter_resample(af, in);
-}
-
 static int af_open(struct af_instance *af)
 {
     struct af_resample *s = af->priv;
@@ -566,9 +385,17 @@ 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);
 
-    return AF_OK;
+    if (s->avrctx && s->avrctx_out) {
+        return AF_OK;
+    } else {
+        MP_ERR(af, "Cannot initialize Libavresample Context. \n");
+        uninit(af);
+        return AF_ERROR;
+    }
 }
 
 #define OPT_BASE_STRUCT struct af_resample
@@ -577,15 +404,14 @@ 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,
     },
     .options = (const struct m_option[]) {
@@ -594,7 +420,6 @@ 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}
     },

audio/filter/af_pan.c

@@ -1,20 +1,21 @@
 /*
  * Copyright (C) 2002 Anders Johansson ajh@atri.curtin.edu.au
  *
- * This file is part of mpv.
+ * This file is part of MPlayer.
  *
- * mpv is free software; you can redistribute it and/or modify
+ * MPlayer 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,
+ * MPlayer 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/>.
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
 #include <stdio.h>

audio/filter/af_rubberband.c

@@ -1,225 +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 <rubberband/rubberband-c.h>
-
-#include "common/common.h"
-#include "af.h"
-
-struct priv {
-    RubberBandState rubber;
-    double speed;
-    struct mp_audio *pending;
-    bool needs_reset;
-    // Estimate how much librubberband has buffered internally.
-    // I could not find a way to do this with the librubberband API.
-    double rubber_delay;
-    // command line options
-    int opt_transients, opt_detector, opt_phase, opt_window,
-        opt_smoothing, opt_formant, opt_pitch, opt_channels;
-};
-
-static void update_speed(struct af_instance *af, double new_speed)
-{
-    struct priv *p = af->priv;
-
-    p->speed = new_speed;
-    rubberband_set_time_ratio(p->rubber, 1.0 / p->speed);
-}
-
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    struct priv *p = af->priv;
-
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-        struct mp_audio *out = af->data;
-
-        in->format = AF_FORMAT_FLOATP;
-        mp_audio_copy_config(out, in);
-
-        if (p->rubber)
-            rubberband_delete(p->rubber);
-
-        int opts = p->opt_transients | p->opt_detector | p->opt_phase |
-                   p->opt_window | p->opt_smoothing | p->opt_formant |
-                   p->opt_pitch | p-> opt_channels |
-                   RubberBandOptionProcessRealTime;
-
-        p->rubber = rubberband_new(in->rate, in->channels.num, opts, 1.0, 1.0);
-        if (!p->rubber) {
-            MP_FATAL(af, "librubberband initialization failed.\n");
-            return AF_ERROR;
-        }
-
-        update_speed(af, p->speed);
-        control(af, AF_CONTROL_RESET, NULL);
-
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
-    }
-    case AF_CONTROL_SET_PLAYBACK_SPEED: {
-        update_speed(af, *(double *)arg);
-        return AF_OK;
-    }
-    case AF_CONTROL_RESET:
-        if (p->rubber)
-            rubberband_reset(p->rubber);
-        talloc_free(p->pending);
-        p->pending = NULL;
-        p->rubber_delay = 0;
-        return AF_OK;
-    }
-    return AF_UNKNOWN;
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *data)
-{
-    struct priv *p = af->priv;
-
-    talloc_free(p->pending);
-    p->pending = data;
-
-    return 0;
-}
-
-static int filter_out(struct af_instance *af)
-{
-    struct priv *p = af->priv;
-
-    while (rubberband_available(p->rubber) <= 0) {
-        const float *dummy[MP_NUM_CHANNELS] = {0};
-        const float **in_data = dummy;
-        size_t in_samples = 0;
-        if (p->pending) {
-            if (!p->pending->samples)
-                break;
-
-            // recover from previous EOF
-            if (p->needs_reset) {
-                rubberband_reset(p->rubber);
-                p->rubber_delay = 0;
-            }
-            p->needs_reset = false;
-
-            size_t needs = rubberband_get_samples_required(p->rubber);
-            in_data = (void *)&p->pending->planes;
-            in_samples = MPMIN(p->pending->samples, needs);
-        }
-
-        if (p->needs_reset)
-            break; // previous EOF
-        p->needs_reset = !p->pending; // EOF
-
-        rubberband_process(p->rubber, in_data, in_samples, p->needs_reset);
-        p->rubber_delay += in_samples;
-
-        if (!p->pending)
-            break;
-        mp_audio_skip_samples(p->pending, in_samples);
-    }
-
-    int out_samples = rubberband_available(p->rubber);
-    if (out_samples > 0) {
-        struct mp_audio *out =
-            mp_audio_pool_get(af->out_pool, af->data, out_samples);
-        if (!out)
-            return -1;
-        if (p->pending)
-            mp_audio_copy_config(out, p->pending);
-
-        float **out_data = (void *)&out->planes;
-        out->samples = rubberband_retrieve(p->rubber, out_data, out->samples);
-        p->rubber_delay -= out->samples * p->speed;
-
-        af_add_output_frame(af, out);
-    }
-
-    int delay_samples = p->rubber_delay;
-    if (p->pending)
-        delay_samples += p->pending->samples;
-    af->delay = delay_samples / (af->data->rate * p->speed);
-
-    return 0;
-}
-
-static void uninit(struct af_instance *af)
-{
-    struct priv *p = af->priv;
-
-    if (p->rubber)
-        rubberband_delete(p->rubber);
-    talloc_free(p->pending);
-}
-
-static int af_open(struct af_instance *af)
-{
-    af->control = control;
-    af->filter_frame = filter_frame;
-    af->filter_out = filter_out;
-    af->uninit = uninit;
-    return AF_OK;
-}
-
-#define OPT_BASE_STRUCT struct priv
-const struct af_info af_info_rubberband = {
-    .info = "Pitch conversion with librubberband",
-    .name = "rubberband",
-    .open = af_open,
-    .priv_size = sizeof(struct priv),
-    .priv_defaults = &(const struct priv) {
-        .speed = 1.0,
-        .opt_pitch = RubberBandOptionPitchHighConsistency,
-        .opt_transients = RubberBandOptionTransientsMixed,
-        .opt_formant = RubberBandOptionFormantPreserved,
-    },
-    .options = (const struct m_option[]) {
-        OPT_CHOICE("transients", opt_transients, 0,
-                   ({"crisp", RubberBandOptionTransientsCrisp},
-                    {"mixed", RubberBandOptionTransientsMixed},
-                    {"smooth", RubberBandOptionTransientsSmooth})),
-        OPT_CHOICE("detector", opt_detector, 0,
-                   ({"compound", RubberBandOptionDetectorCompound},
-                    {"percussive", RubberBandOptionDetectorPercussive},
-                    {"soft", RubberBandOptionDetectorSoft})),
-        OPT_CHOICE("phase", opt_phase, 0,
-                   ({"laminar", RubberBandOptionPhaseLaminar},
-                    {"independent", RubberBandOptionPhaseIndependent})),
-        OPT_CHOICE("window", opt_window, 0,
-                   ({"standard", RubberBandOptionWindowStandard},
-                    {"short", RubberBandOptionWindowShort},
-                    {"long", RubberBandOptionWindowLong})),
-        OPT_CHOICE("smoothing", opt_smoothing, 0,
-                   ({"off", RubberBandOptionSmoothingOff},
-                    {"on", RubberBandOptionSmoothingOn})),
-        OPT_CHOICE("formant", opt_formant, 0,
-                   ({"shifted", RubberBandOptionFormantShifted},
-                    {"preserved", RubberBandOptionFormantPreserved})),
-        OPT_CHOICE("pitch", opt_pitch, 0,
-                   ({"quality", RubberBandOptionPitchHighQuality},
-                    {"speed", RubberBandOptionPitchHighSpeed},
-                    {"consistency", RubberBandOptionPitchHighConsistency})),
-        OPT_CHOICE("channels", opt_channels, 0,
-                   ({"apart", RubberBandOptionChannelsApart},
-                    {"together", RubberBandOptionChannelsTogether})),
-        {0}
-    },
-};