Release 0.8.3

This played e.g. a 1264x722 file as 1264x720. There was some code which
dropped the aspect ratio if the video (in original resolution) wasn't
scaled by more than 4 pixels. Commit 5f3c3f8c introduced this (although
I'm not really sure what the code replaced by it did).

Just remove this "feature".

(cherry picked from commit d96fad04be)

.gitignore

@@ -14,6 +14,7 @@
 /input/input_conf.h
 /tags
 /TAGS
+/video/out/gl_video_shaders.h
 /video/out/x11_icon.inc
 /demux/ebml_defs.c
 /demux/ebml_types.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
 

Copyright

@@ -2,15 +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.
+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.
@@ -23,5 +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),
@@ -18,38 +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 0.11.0 ---
- 1.19   - add "GL_MP_D3D_interfaces" pseudo extension to make it possible to
-          use DXVA2 in OpenGL fullscreen mode in some situations
-        - mpv_request_log_messages() now accepts "terminal-default" as parameter
- 1.18   - add MPV_END_FILE_REASON_REDIRECT, and change behavior of
-          MPV_EVENT_END_FILE accordingly
-        - a bunch of interface-changes.rst changes
- 1.17   - mpv_initialize() now blocks SIGPIPE (details see client.h)
- --- mpv 0.9.0 ---
- 1.16   - add mpv_opengl_cb_report_flip()
-        - introduce mpv_opengl_cb_draw() and deprecate mpv_opengl_cb_render()
-        - add MPV_FORMAT_BYTE_ARRAY
- 1.15   - mpv_initialize() will now load config files. This requires setting
-          the "config" and "config-dir" options. In particular, it will load
-          mpv.conf.
-        - minor backwards-compatible change to the "seek" and "screenshot"
-          commands (new flag syntax, old additional args deprecated)
- --- mpv 0.8.0 ---
+ --- 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)
@@ -62,7 +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
@@ -85,7 +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
@@ -94,7 +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://
@@ -106,6 +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

@@ -24,11 +24,6 @@ qml_direct
 Alternative example, which typically avoids a FBO indirection. Might be
 slightly faster, but is less flexible and harder to use.
 
-sdl
----
-
-Show how to embed the mpv OpenGL renderer in SDL.
-
 simple
 ------
 

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>
 
@@ -205,5 +207,5 @@ 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/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/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

@@ -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.

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
 ---------------
@@ -14,9 +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.
-- 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.
@@ -54,9 +51,6 @@ Sending patches
   change in the same commit.
 - If you add a new command line option, document it in options.rst. If you
   add a new input property, document it in input.rst.
-- All new code must be LGPLv2.1+ licensed, or come with the implicit agreement
-  that it will be relicensed to LGPLv2.1+ later (see ``Copyright`` in the
-  repository root directory).
 
 Code formatting
 ---------------

DOCS/interface-changes.rst

@@ -1,90 +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 0.11.0 ---
-    - 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
@@ -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).
@@ -431,20 +601,6 @@ 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 +615,4 @@ Available filters are:
 
     ``o=<string>``
         AVOptions.
+

DOCS/man/ao.rst

@@ -55,14 +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::
 
@@ -107,10 +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
@@ -162,23 +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 effect the selected speaker setup.
-
-    ``exclusive``
-        Use exclusive mode access. This merely redirects to
-        ``coreaudio_exclusive``, but should be preferred over using that AO
-        directly.
-
 ``coreaudio_exclusive`` (Mac OS X only)
     Native Mac OS X audio output driver using direct device access and
     exclusive mode (bypasses the sound server).
 
+    Supports only compressed formats (AC3 and DTS).
+
 ``openal``
     Experimental OpenAL audio output driver
 
@@ -271,10 +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
 
@@ -319,19 +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

@@ -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"``

DOCS/man/input.rst

@@ -65,11 +65,11 @@ C-style escaping can be used.
 
 You can bind multiple commands to one key. For example:
 
-| a show-text "command 1" ; show-text "command 2"
+| a show_text "command 1" ; show_text "command 2"
 
 It's also possible to bind a command to a sequence of keys:
 
-| a-b-c show-text "command run after a, b, c have been pressed"
+| a-b-c show_text "command run after a, b, c have been pressed"
 
 (This is not shown in the general command syntax.)
 
@@ -86,11 +86,11 @@ List of Input Commands
     disabling default bindings, without disabling all bindings with
     ``--no-input-default-bindings``.
 
-``seek <seconds> [relative|absolute|absolute-percent|relative-percent|exact|keyframes]``
+``seek <seconds> [relative|absolute|absolute-percent|- [default-precise|exact|keyframes]]``
     Change the playback position. By default, seeks by a relative amount of
     seconds.
 
-    The second argument consists of flags controlling the seek mode:
+    The second argument sets the seek mode:
 
     relative (default)
         Seek relative to current position (a negative value seeks backwards).
@@ -98,41 +98,36 @@ List of Input Commands
         Seek to a given time.
     absolute-percent
         Seek to a given percent position.
-    relative-percent
-        Seek relative to current position in percent.
-    keyframes
-        Always restart playback at keyframe boundaries (fast).
+
+    The third argument defines how exact the seek is:
+
+    default-precise (default)
+        Follow the default behavior as set by ``--hr-seek``, which by default
+        does imprecise seeks (like ``keyframes``).
     exact
         Always do exact/hr/precise seeks (slow).
+    keyframes
+        Always restart playback at keyframe boundaries (fast).
 
-    Multiple flags can be combined, e.g.: ``absolute+keyframes``.
-
-    By default, ``keyframes`` is used for relative seeks, and ``exact`` is used
-    for absolute seeks.
-
-    Before mpv 0.9, the ``keyframes`` and ``exact`` flags had to be passed as
-    3rd parameter (essentially using a space instead of ``+``). The 3rd
-    parameter is still parsed, but is considered deprecated.
-
-``revert-seek [mode]``
+``revert_seek [mode]``
     Undoes the ``seek`` command, and some other commands that seek (but not
     necessarily all of them). Calling this command once will jump to the
     playback position before the seek. Calling it a second time undoes the
-    ``revert-seek`` command itself. This only works within a single file.
+    ``revert_seek`` command itself. This only works within a single file.
 
     The first argument is optional, and can change the behavior:
 
     mark
-        Mark the current time position. The next normal ``revert-seek`` command
+        Mark the current time position. The next normal ``revert_seek`` command
         will seek back to this point, no matter how many seeks happened since
         last time.
 
     Using it without any arguments gives you the default behavior.
 
-``frame-step``
+``frame_step``
     Play one frame, then pause. Does nothing with audio-only playback.
 
-``frame-back-step``
+``frame_back_step``
     Go back by one frame, then pause. Note that this can be very slow (it tries
     to be precise, not fast), and sometimes fails to behave as expected. How
     well this works depends on whether precise seeking works correctly (e.g.
@@ -175,26 +170,31 @@ List of Input Commands
         Save the contents of the mpv window. Typically scaled, with OSD and
         subtitles. The exact behavior depends on the selected video output, and
         if no support is available, this will act like ``video``.
+
+    Second argument:
+
+    <single> (default)
+        Take a single screenshot.
     <each-frame>
         Take a screenshot each frame. Issue this command again to stop taking
         screenshots. Note that you should disable frame-dropping when using
         this mode - or you might receive duplicate images in cases when a
-        frame was dropped. This flag can be combined with the other flags,
-        e.g. ``video+each-frame``.
+        frame was dropped.
 
-``screenshot-to-file "<filename>" [subtitles|video|window]``
+``screenshot_to_file "<filename>" [subtitles|video|window]``
     Take a screenshot and save it to a given file. The format of the file will
     be guessed by the extension (and ``--screenshot-format`` is ignored - the
     behavior when the extension is missing or unknown is arbitrary).
 
     The second argument is like the first argument to ``screenshot``.
 
-    If the file already exists, it's overwritten.
+    This command tries to never overwrite files. If the file already exists,
+    it fails.
 
     Like all input command parameters, the filename is subject to property
     expansion as described in `Property Expansion`_.
 
-``playlist-next [weak|force]``
+``playlist_next [weak|force]``
     Go to the next entry on the playlist.
 
     weak (default)
@@ -202,7 +202,7 @@ List of Input Commands
     force
         Terminate playback if there are no more files on the playlist.
 
-``playlist-prev [weak|force]``
+``playlist_prev [weak|force]``
     Go to the previous entry on the playlist.
 
     weak (default)
@@ -232,26 +232,22 @@ List of Input Commands
 ``loadlist "<playlist>" [replace|append]``
     Load the given playlist file (like ``--playlist``).
 
-``playlist-clear``
+``playlist_clear``
     Clear the playlist, except the currently played file.
 
-``playlist-remove current|<index>``
+``playlist_remove current|<index>``
     Remove the playlist entry at the given index. Index values start counting
     with 0. The special value ``current`` removes the current entry. Note that
     removing the current entry also stops playback and starts playing the next
     entry.
 
-``playlist-move <index1> <index2>``
+``playlist_move <index1> <index2>``
     Move the playlist entry at index1, so that it takes the place of the
     entry index2. (Paradoxically, the moved playlist entry will not have
     the index value index2 after moving if index1 was lower than index2,
     because index2 refers to the target entry, not the index the entry
     will have after moving.)
 
-``playlist-shuffle``
-    Shuffle the playlist. This is similar to what is done on start if the
-    ``--shuffle`` option is used.
-
 ``run "command" "arg1" "arg2" ...``
     Run the given command. Unlike in MPlayer/mplayer2 and earlier versions of
     mpv (0.2.x and older), this doesn't call the shell. Instead, the command
@@ -277,12 +273,12 @@ List of Input Commands
 ``quit [<code>]``
     Exit the player. If an argument is given, it's used as process exit code.
 
-``quit-watch-later [<code>]``
+``quit_watch_later [<code>]``
     Exit player, and store current playback position. Playing that file later
     will seek to the previous position on start. The (optional) argument is
     exactly as in the ``quit`` command.
 
-``sub-add "<file>" [<flags> [<title> [<lang>]]]``
+``sub_add "<file>" [<flags> [<title> [<lang>]]]``
     Load the given subtitle file. It is selected as current subtitle after
     loading.
 
@@ -309,27 +305,27 @@ List of Input Commands
     The ``lang`` argument sets the track language, and can also influence
     stream selection with ``flags`` set to ``auto``.
 
-``sub-remove [<id>]``
+``sub_remove [<id>]``
     Remove the given subtitle track. If the ``id`` argument is missing, remove
     the current track. (Works on external subtitle files only.)
 
-``sub-reload [<id>]``
+``sub_reload [<id>]``
     Reload the given subtitle tracks. If the ``id`` argument is missing, reload
     the current track. (Works on external subtitle files only.)
 
     This works by unloading and re-adding the subtitle track.
 
-``sub-step <skip>``
+``sub_step <skip>``
     Change subtitle timing such, that the subtitle event after the next
     ``<skip>`` subtitle events is displayed. ``<skip>`` can be negative to step
     backwards.
 
-``sub-seek <skip>``
+``sub_seek <skip>``
     Seek to the next (skip set to 1) or the previous (skip set to -1) subtitle.
-    This is similar to ``sub-step``, except that it seeks video and audio
+    This is similar to ``sub_step``, except that it seeks video and audio
     instead of adjusting the subtitle delay.
 
-    Like with ``sub-step``, this works with external text subtitles only. For
+    Like with ``sub_step``, this works with external text subtitles only. For
     embedded text subtitles (like with Matroska), this works only with subtitle
     events that have already been displayed.
 
@@ -337,11 +333,11 @@ List of Input Commands
     Toggle OSD level. If ``<level>`` is specified, set the OSD mode
     (see ``--osd-level`` for valid values).
 
-``print-text "<string>"``
+``print_text "<string>"``
     Print text to stdout. The string can contain properties (see
     `Property Expansion`_).
 
-``show-text "<string>" [<duration>|- [<level>]]``
+``show_text "<string>" [<duration>|- [<level>]]``
     Show text on the OSD. The string can contain properties, which are expanded
     as described in `Property Expansion`_. This can be used to show playback
     time, filename, and so on.
@@ -353,12 +349,24 @@ List of Input Commands
     <level>
         The minimum OSD level to show the text at (see ``--osd-level``).
 
-``show-progress``
+``show_progress``
     Show the progress bar, the elapsed time and the total duration of the file
     on the OSD.
 
-``write-watch-later-config``
-    Write the resume config file that the ``quit-watch-later`` command writes,
+``discnav "<command>"``
+    Send a menu control command to the DVD/BD menu implementation. The following
+    commands are defined: ``up``, ``down``, ``left``, ``right``,
+    ``menu`` (request to enter menu), ``prev`` (previous screen),
+    ``select`` (activate current button), ``mouse`` (the mouse was clicked),
+    ``mouse_move`` (the mouse cursor changed position).
+
+    ``mouse_move`` will use the current mouse position.
+
+    Note that while the menu is active, the input section ``discnav-menu`` will
+    be enabled, so different key bindings can be mapped for menu mode.
+
+``write_watch_later_config``
+    Write the resume config file that the ``quit_watch_later`` command writes,
     but continue playback normally.
 
 ``stop``
@@ -383,46 +391,14 @@ List of Input Commands
     <double>
         The mouse event represents double-click.
 
-``keypress <key_name>``
-    Send a key event through mpv's input handler, triggering whatever
-    behavior is configured to that key. ``key_name`` uses the ``input.conf``
-    naming scheme for keys and modifiers. Useful for the client API: key events
-    can be sent to libmpv to handle internally.
+``audio_add "<file>" [<flags> [<title> [<lang>]]]``
+    Load the given audio file. See ``sub_add`` command.
 
-``keydown <key_name>``
-    Similar to ``keypress``, but sets the ``KEYDOWN`` flag so that if the key is
-    bound to a repeatable command, it will be run repeatedly with mpv's key
-    repeat timing until the ``keyup`` command is called.
+``audio_remove [<id>]``
+    Remove the given audio track. See ``sub_remove`` command.
 
-``keyup [<key_name>]``
-    Set the ``KEYUP`` flag, stopping any repeated behavior that had been
-    triggered. ``key_name`` is optional. If ``key_name`` is not given or is an
-    empty string, ``KEYUP`` will be set on all keys. Otherwise, ``KEYUP`` will
-    only be set on the key specified by ``key_name``.
-
-``audio-add "<file>" [<flags> [<title> [<lang>]]]``
-    Load the given audio file. See ``sub-add`` command.
-
-``audio-remove [<id>]``
-    Remove the given audio track. See ``sub-remove`` command.
-
-``audio-reload [<id>]``
-    Reload the given audio tracks. See ``sub-reload`` command.
-
-``rescan-external-files [<mode>]``
-    Rescan external files according to the current ``--sub-auto`` and
-    ``--audio-file-auto`` settings. This can be used to auto-load external
-    files *after* the file was loaded.
-
-    The ``mode`` argument is one of the following:
-
-    <reselect> (default)
-        Select the default audio and subtitle streams, which typically selects
-        external files with highest preference. (The implementation is not
-        perfect, and could be improved on request.)
-
-    <keep-selection>
-        Do not change current track selections.
+``audio_reload [<id>]``
+    Reload the given audio tracks. See ``sub_reload`` command.
 
 
 Input Commands that are Possibly Subject to Change
@@ -466,7 +442,7 @@ Input Commands that are Possibly Subject to Change
 
     The ``vf`` command shows the list of requested filters on the OSD after
     changing the filter chain. This is roughly equivalent to
-    ``show-text ${vf}``. Note that auto-inserted filters for format conversion
+    ``show_text ${vf}``. Note that auto-inserted filters for format conversion
     are not shown on the list, only what was requested by the user.
 
     Normally, the commands will check whether the video chain is recreated
@@ -481,7 +457,7 @@ Input Commands that are Possibly Subject to Change
         - ``b vf set ""`` remove all video filters on ``b``
         - ``c vf toggle lavfi=gradfun`` toggle debanding on ``c``
 
-``cycle-values ["!reverse"] <property> "<value1>" "<value2>" ...``
+``cycle_values ["!reverse"] <property> "<value1>" "<value2>" ...``
     Cycle through a list of values. Each invocation of the command will set the
     given property to the next value in the list. The command maintains an
     internal counter which value to pick next, and which is initially 0. It is
@@ -499,7 +475,7 @@ Input Commands that are Possibly Subject to Change
     Note that there is a static limit of (as of this writing) 10 arguments
     (this limit could be raised on demand).
 
-``enable-section "<section>" [flags]``
+``enable_section "<section>" [default|exclusive]``
     Enable all key bindings in the named input section.
 
     The enabled input sections form a stack. Bindings in sections on the top of
@@ -508,45 +484,14 @@ Input Commands that are Possibly Subject to Change
     implicitly removed beforehand. (A section cannot be on the stack more than
     once.)
 
-    The ``flags`` parameter can be a combination (separated by ``+``) of the
-    following flags:
+    If ``exclusive`` is specified as second argument, all sections below the
+    newly enabled section are disabled. They will be re-enabled as soon as
+    all exclusive sections above them are removed.
 
-    <exclusive>
-        All sections enabled before the newly enabled section are disabled.
-        They will be re-enabled as soon as all exclusive sections above them
-        are removed. In other words, the new section shadows all previous
-        sections.
-    <allow-hide-cursor>
-        This feature can't be used through the public API.
-    <allow-vo-dragging>
-        Same.
+``disable_section "<section>"``
+    Disable the named input section. Undoes ``enable_section``.
 
-``disable-section "<section>"``
-    Disable the named input section. Undoes ``enable-section``.
-
-``define-section "<section>" "<contents>" [default|forced]``
-    Create a named input section, or replace the contents of an already existing
-    input section. The ``contents`` parameter uses the same syntax as the
-    ``input.conf`` file (except that using the section syntax in it is not
-    allowed), including the need to separate bindings with a newline character.
-
-    If the ``contents`` parameter is an empty string, the section is removed.
-
-    The section with the name ``default`` is the normal input section.
-
-    In general, input sections have to be enabled with the ``enable-section``
-    command, or they are ignored.
-
-    The last parameter has the following meaning:
-
-    <default> (also used if parameter omitted)
-        Use a key binding defined by this section only if the user hasn't
-        already bound this key to a command.
-    <forced>
-        Always bind a key. (The input section that was made active most recently
-        wins if there are ambiguities.)
-
-``overlay-add <id> <x> <y> "<file>" <offset> "<fmt>" <w> <h> <stride>``
+``overlay_add <id> <x> <y> "<file>" <offset> "<fmt>" <w> <h> <stride>``
     Add an OSD overlay sourced from raw data. This might be useful for scripts
     and applications controlling mpv, and which want to display things on top
     of the video window.
@@ -562,7 +507,7 @@ Input Commands that are Possibly Subject to Change
     ``id`` is an integer between 0 and 63 identifying the overlay element. The
     ID can be used to add multiple overlay parts, update a part by using this
     command with an already existing ID, or to remove a part with
-    ``overlay-remove``. Using a previously unused ID will add a new overlay,
+    ``overlay_remove``. Using a previously unused ID will add a new overlay,
     while reusing an ID will update it. (Future directions: there should be
     something to ensure different programs wanting to create overlays don't
     conflict with each others, should that ever be needed.)
@@ -576,11 +521,11 @@ Input Commands that are Possibly Subject to Change
     vdpau), so no actual copying is involved. Truncating the source file while
     the overlay is active will crash the player. You shouldn't change the data
     while the overlay is active, because the data is essentially accessed at
-    random points. Instead, call ``overlay-add`` again (preferably with a
+    random points. Instead, call ``overlay_add`` again (preferably with a
     different memory region to prevent tearing).
 
     It is also possible to pass a raw memory address for use as bitmap memory
-    by passing a memory address as integer prefixed with an ``&`` character.
+    by passing a memory address as integer prefixed with a ``&`` character.
     Passing the wrong thing here will crash the player. This mode might be
     useful for use with libmpv. The ``offset`` parameter is simply added to the
     memory address (since mpv 0.8.0, ignored before).
@@ -620,22 +565,22 @@ Input Commands that are Possibly Subject to Change
         an overlay's memory at random times whenever it feels the need to do
         so, for example when redrawing the screen.
 
-``overlay-remove <id>``
-    Remove an overlay added with ``overlay-add`` and the same ID. Does nothing
+``overlay_remove <id>``
+    Remove an overlay added with ``overlay_add`` and the same ID. Does nothing
     if no overlay with this ID exists.
 
-``script-message "<arg1>" "<arg2>" ...``
+``script_message "<arg1>" "<arg2>" ...``
     Send a message to all clients, and pass it the following list of arguments.
     What this message means, how many arguments it takes, and what the arguments
     mean is fully up to the receiver and the sender. Every client receives the
     message, so be careful about name clashes (or use ``script_message_to``).
 
-``script-message-to "<target>" "<arg1>" "<arg2>" ...``
+``script_message_to "<target>" "<arg1>" "<arg2>" ...``
     Same as ``script_message``, but send it only to the client named
     ``<target>``. Each client (scripts etc.) has a unique name. For example,
     Lua scripts can get their name via ``mp.get_script_name()``.
 
-``script-binding "<name>"``
+``script_binding "<name>"``
     Invoke a script-provided key binding. This can be used to remap key
     bindings provided by external Lua scripts.
 
@@ -656,40 +601,31 @@ Input Commands that are Possibly Subject to Change
     tracked). The second letter whether the event originates from the mouse,
     either ``m`` (mouse button) or ``-`` (something else).
 
-``ab-loop``
+``ab_loop``
     Cycle through A-B loop states. The first command will set the ``A`` point
     (the ``ab-loop-a`` property); the second the ``B`` point, and the third
     will clear both points.
 
-``vo-cmdline "<args>"``
+``vo_cmdline "<args>"``
     Reset the sub-option of the current VO. Currently works with ``opengl``
     (including ``opengl-hq``). The argument is the sub-option string usually
     passed to the VO on the command line. Not all sub-options can be set, but
     those which can will be reset even if they don't appear in the argument.
     This command might be changed or removed in the future.
 
-``drop-buffers``
+``drop_buffers``
     Drop audio/video/demuxer buffers, and restart from fresh. Might help with
     unseekable streams that are going out of sync.
     This command might be changed or removed in the future.
 
-``screenshot-raw [subtitles|video|window]``
-    Return a screenshot in memory. This can be used only through the client
-    API. The MPV_FORMAT_NODE_MAP returned by this command has the ``w``, ``h``,
-    ``stride`` fields set to obvious contents. A ``format`` field is set to
-    ``bgr0`` by default. This format is organized as ``B8G8R8X8`` (where ``B``
-    is the LSB). The contents of the padding ``X`` is undefined. The ``data``
-    field is of type MPV_FORMAT_BYTE_ARRAY with the actual image data. The image
-    is freed as soon as the result node is freed.
-
-Undocumented commands: ``tv-last-channel`` (TV/DVB only),
-``ao-reload`` (experimental/internal).
+Undocumented commands: ``tv_last_channel`` (TV/DVB only),
+``get_property`` (deprecated), ``ao_reload`` (experimental/internal).
 
 Hooks
 ~~~~~
 
 Hooks are synchronous events between player core and a script or similar. This
-applies to client API (including the Lua scripting interface). Normally,
+applies to the Lua scripting interface and the client API and only. Normally,
 events are supposed to be asynchronous, and the hook API provides an awkward
 and obscure way to handle events that require stricter coordination. There are
 no API stability guarantees made. Not following the protocol exactly can make
@@ -698,7 +634,7 @@ the player freeze randomly. Basically, nobody should use this API.
 There are two special commands involved. Also, the client must listen for
 client messages (``MPV_EVENT_CLIENT_MESSAGE`` in the C API).
 
-``hook-add <hook-name> <id> <priority>``
+``hook_add <hook-name> <id> <priority>``
     Subscribe to the hook identified by the first argument (basically, the
     name of event). The ``id`` argument is an arbitrary integer chosen by the
     user. ``priority`` is used to sort all hook handlers globally across all
@@ -723,9 +659,9 @@ client messages (``MPV_EVENT_CLIENT_MESSAGE`` in the C API).
     typically be stopped.
 
     When the client is done, it must continue the core's hook execution by
-    running the ``hook-ack`` command.
+    running the ``hook_ack`` command.
 
-``hook-ack <string>``
+``hook_ack <string>``
     Run the next hook in the global chain of hooks. The argument is the 3rd
     argument of the client message that starts hook execution for the
     current client.
@@ -798,7 +734,7 @@ Properties
 
 Properties are used to set mpv options during runtime, or to query arbitrary
 information. They can be manipulated with the ``set``/``add``/``cycle``
-commands, and retrieved with ``show-text``, or anything else that uses property
+commands, and retrieved with ``show_text``, or anything else that uses property
 expansion. (See `Property Expansion`_.)
 
 The property name is annotated with RW to indicate whether the property is
@@ -826,16 +762,6 @@ Property list
 ``speed`` (RW)
     See ``--speed``.
 
-``audio-speed-correction``, ``video-speed-correction``
-    Factor multiplied with ``speed`` at which the player attempts to play the
-    file. Usually it's exactly 1. (Display sync mode will make this useful.)
-
-    OSD formatting will display it in the form of ``+1.23456%``, with the number
-    being ``(raw - 1) * 100`` for the given raw property value.
-
-``display-sync-active``
-    Return whether ``--video-sync=display`` is actually active.
-
 ``filename``
     Currently played file, with path stripped. If this is an URL, try to undo
     percent encoding as well. (The result is not necessarily correct, but
@@ -888,14 +814,11 @@ Property list
 ``stream-end``
     Raw end position in bytes in source stream.
 
-``duration``
-    Duration of the current file in seconds. If the duration is unknown, the
+``length``
+    Length of the current file in seconds. If the length is unknown, the
     property is unavailable. Note that the file duration is not always exactly
     known, so this is an estimate.
 
-    This replaces the ``length`` property, which was deprecated after the
-    mpv 0.9 release. (The semantics are the same.)
-
 ``avsync``
     Last A/V synchronization difference. Unavailable if audio or video is
     disabled.
@@ -931,12 +854,10 @@ Property list
     always exactly known, so this is an estimate.
 
 ``playtime-remaining``
-    ``time-remaining`` scaled by the current ``speed``.
+    ``time-remaining`` scaled by the the current ``speed``.
 
-``playback-time`` (RW)
-    The playback time, which is the time relative to playback start. (This can
-    be different from the ``time-pos`` property if the file does not start at
-    position ``0``, in which case ``time-pos`` is the source timestamp.)
+``playback-time``
+    Return the playback time, which is the time difference between start PTS and current PTS.
 
 ``chapter`` (RW)
     Current chapter number. The number of the first chapter is 0.
@@ -979,6 +900,12 @@ Property list
     Current BD/DVD title number. Writing works only for ``dvdnav://`` and
     ``bd://`` (and aliases for these).
 
+``disc-menu-active``
+    Return ``yes`` if the BD/DVD menu is active, or ``no`` on normal video
+    playback. The property is unavailable when playing something that is not
+    a BD or DVD. Use the ``discnav menu`` command to actually enter or leave
+    menu mode.
+
 ``chapters``
     Number of chapters.
 
@@ -1091,9 +1018,6 @@ Property list
     An example of these kind of metadata are the cropping parameters
     added by ``--vf=lavfi=cropdetect``.
 
-``af-metadata/<filter-label>``
-    Equivalent to ``vf-metadata/<filter-label>``, but for audio filters.
-
 ``pause`` (RW)
     Pause status. This is usually ``yes`` or ``no``. See ``--pause``.
 
@@ -1114,13 +1038,11 @@ Property list
     Network cache fill state (0-100.0).
 
 ``cache-size`` (RW)
-    Network cache size in KB. This is similar to ``--cache``. This allows
+    Total network cache size in KB. This is similar to ``--cache``. This allows
     to set the cache size at runtime. Currently, it's not possible to enable
     or disable the cache at runtime using this property, just to resize an
     existing cache.
 
-    This does not include the backbuffer size (changed after mpv 0.10.0).
-
     Note that this tries to keep the cache contents as far as possible. To make
     this easier, the cache resizing code will allocate the new cache while the
     old cache is still allocated.
@@ -1142,11 +1064,6 @@ Property list
     guess is very unreliable, and often the property will not be available
     at all, even if data is buffered.
 
-``demuxer-cache-time``
-    Approximate time of video buffered in the demuxer, in seconds. Same as
-    ``demuxer-cache-duration`` but returns the last timestamp of buffered
-    data in demuxer.
-
 ``demuxer-cache-idle``
     Returns ``yes`` if the demuxer is idle, which means the demuxer cache is
     filled to the requested amount, and is currently not reading more data.
@@ -1178,7 +1095,7 @@ Property list
     See ``--hr-seek``.
 
 ``volume`` (RW)
-    Current volume (see ``--volume`` for details).
+    Current volume (0-100).
 
 ``mute`` (RW)
     Current mute status (``yes``/``no``).
@@ -1186,52 +1103,21 @@ Property list
 ``audio-delay`` (RW)
     See ``--audio-delay``.
 
+``audio-format``
+    Audio format as string.
+
 ``audio-codec``
     Audio codec selected for decoding.
 
-``audio-codec-name``
-    Audio codec.
+``audio-bitrate``
+    Audio bitrate. This is probably a very bad guess in most cases.
 
-``audio-params``
-    Audio format as output by the audio decoder.
-    This has a number of sub-properties:
+``audio-samplerate``
+    Audio samplerate.
 
-    ``audio-params/format``
-        The sample format as string. This uses the same names as used in other
-        places of mpv.
-
-    ``audio-params/samplerate``
-        Samplerate.
-
-    ``audio-params/channels``
-        The channel layout as a string. This is similar to what the
-        ``--audio-channels`` accepts.
-
-    ``audio-params/hr-channels``
-        As ``channels``, but instead of the possibly cryptic actual layout
-        sent to the audio device, return a hopefully more human readable form.
-        (Usually only ``audio-out-params/hr-channels`` makes sense.)
-
-    ``audio-params/channel-count``
-        Number of audio channels. This is redundant to the ``channels`` field
-        described above.
-
-    When querying the property with the client API using ``MPV_FORMAT_NODE``,
-    or with Lua ``mp.get_property_native``, this will return a mpv_node with
-    the following contents:
-
-    ::
-
-        MPV_FORMAT_NODE_MAP
-            "format"            MPV_FORMAT_STRING
-            "samplerate"        MPV_FORMAT_INT64
-            "channels"          MPV_FORMAT_STRING
-            "channel-count"     MPV_FORMAT_INT64
-            "hr-channels"       MPV_FORMAT_STRING
-
-``audio-out-params``
-    Same as ``audio-params``, but the format of the data written to the audio
-    API.
+``audio-channels``
+    Number of audio channels. The OSD value of this property is actually the
+    channel layout, while the raw value returns the number of channels only.
 
 ``aid`` (RW)
     Current audio track (similar to ``--aid``).
@@ -1242,7 +1128,7 @@ Property list
 ``balance`` (RW)
     Audio channel balance. (The implementation of this feature is rather odd.
     It doesn't change the volumes of each channel, but instead sets up a pan
-    matrix to mix the left and right channels.)
+    matrix to mix the the left and right channels.)
 
 ``fullscreen`` (RW)
     See ``--fullscreen``.
@@ -1253,18 +1139,17 @@ Property list
 ``field-dominance`` (RW)
     See ``--field-dominance``
 
-``colormatrix`` (R)
-    Redirects to ``video-params/colormatrix``. This parameter (as well as
-    similar ones) can be overridden with the ``format`` video filter.
+``colormatrix`` (RW)
+    See ``--colormatrix``.
 
-``colormatrix-input-range`` (R)
-    See ``colormatrix``.
+``colormatrix-input-range`` (RW)
+    See ``--colormatrix-input-range``.
 
-``colormatrix-output-range`` (R)
-    See ``colormatrix``.
+``colormatrix-output-range`` (RW)
+    See ``--colormatrix-output-range``.
 
-``colormatrix-primaries`` (R)
-    See ``colormatrix``.
+``colormatrix-primaries`` (RW)
+    See ``--colormatrix-primaries``.
 
 ``ontop`` (RW)
     See ``--ontop``.
@@ -1294,29 +1179,25 @@ Property list
     See ``--hue``.
 
 ``hwdec`` (RW)
-    Reflects the ``--hwdec`` option.
+    Return the current hardware decoder that is used. This uses the same values
+    as the ``--hwdec`` option. If software decoding is active, this returns
+    ``no``. You can write this property. Then the ``--hwdec`` option is set to
+    the new value, and video decoding will be reinitialized (internally, the
+    player will perform a seek to refresh the video properly).
 
-    Writing to it may change the currently used hardware decoder, if possible.
-    (Internally, the player may reinitialize the decoder, and will perform a
-    seek to refresh the video properly.) You can watch the other hwdec
-    properties to see whether this was successful.
+    Note that you don't know the success of the operation immediately after
+    writing this property. It happens with a delay as video is reinitialized.
 
-    Unlike in mpv 0.9.x and before, this does not return the currently active
-    hardware decoder.
+``detected-hwdec``
+    Return the current hardware decoder that was detected and opened. Returns
+    the same values as ``hwdec``.
 
-``hwdec-active``
-    Return ``yes`` or ``no``, depending on whether any type of hardware decoding
-    is actually in use.
-
-``hwdec-detected``
-    If software decoding is active, this returns the hardware decoder in use.
-    Otherwise, it returns either ``no``, or if applicable, the currently loaded
-    hardware decoding API. This is known only once the VO has opened (and
-    possibly later). With some VOs (like ``opengl``), this is never known in
-    advance, but only when the decoder attempted to create the hw decoder
-    successfully. Also, hw decoders with ``-copy`` suffix will return ``no``
-    while no video is being decoded. All this reflects how detecting hw decoders
-    are detected and used internally in mpv.
+    This is known only once the VO has opened (and possibly later). With some
+    VOs (like ``opengl``), this is never known in advance, but only when the
+    decoder attempted to create the hw decoder successfully. Also, hw decoders
+    with ``-copy`` suffix are returned only while hw decoding is active (and
+    unset afterwards). All this reflects how detecting hw decoders are
+    detected and used internally in mpv.
 
 ``panscan`` (RW)
     See ``--panscan``.
@@ -1327,6 +1208,9 @@ Property list
 ``video-codec``
     Video codec selected for decoding.
 
+``video-bitrate``
+    Video bitrate (a bad guess).
+
 ``width``, ``height``
     Video size. This uses the size of the video as decoded, or if no video
     frame has been decoded yet, the (possibly incorrect) container indicated
@@ -1371,38 +1255,32 @@ Property list
     ``video-params/primaries``
         The primaries in use as string. (Exact values subject to change.)
 
-    ``video-params/gamma``
-        The gamma function in use as string. (Exact values subject to change.)
-
     ``video-params/chroma-location``
         Chroma location as string. (Exact values subject to change.)
 
     ``video-params/rotate``
         Intended display rotation in degrees (clockwise).
 
-    ``video-params/stereo-in``
-        Source file stereo 3D mode. (See ``--video-stereo-mode`` option.)
-
     When querying the property with the client API using ``MPV_FORMAT_NODE``,
     or with Lua ``mp.get_property_native``, this will return a mpv_node with
     the following contents:
 
     ::
 
-        MPV_FORMAT_NODE_MAP
-            "pixelformat"       MPV_FORMAT_STRING
-            "w"                 MPV_FORMAT_INT64
-            "h"                 MPV_FORMAT_INT64
-            "dw"                MPV_FORMAT_INT64
-            "dh"                MPV_FORMAT_INT64
-            "aspect"            MPV_FORMAT_DOUBLE
-            "par"               MPV_FORMAT_DOUBLE
-            "colormatrix"       MPV_FORMAT_STRING
-            "colorlevels"       MPV_FORMAT_STRING
-            "primaries"         MPV_FORMAT_STRING
-            "chroma-location"   MPV_FORMAT_STRING
-            "rotate"            MPV_FORMAT_INT64
-            "stereo-in"         MPV_FORMAT_STRING
+        MPV_FORMAT_NODE_ARRAY
+            MPV_FORMAT_NODE_MAP (for each track)
+                "pixelformat"       MPV_FORMAT_STRING
+                "w"                 MPV_FORMAT_INT64
+                "h"                 MPV_FORMAT_INT64
+                "dw"                MPV_FORMAT_INT64
+                "dh"                MPV_FORMAT_INT64
+                "aspect"            MPV_FORMAT_DOUBLE
+                "par"               MPV_FORMAT_DOUBLE
+                "colormatrix"       MPV_FORMAT_STRING
+                "colorlevels"       MPV_FORMAT_STRING
+                "primaries"         MPV_FORMAT_STRING
+                "chroma-location"   MPV_FORMAT_STRING
+                "rotate"            MPV_FORMAT_INT64
 
 ``dwidth``, ``dheight``
     Video display size. This is the video size after filters and aspect scaling
@@ -1447,13 +1325,6 @@ Property list
     Names of the displays that the mpv window covers. On X11, these
     are the xrandr names (LVDS1, HDMI1, DP1, VGA1, etc.).
 
-``display-fps``
-    The refresh rate of the current display. Currently, this is the lowest FPS
-    of any display covered by the video, as retrieved by the underlying system
-    APIs (e.g. xrandr on X11). It is not the measured FPS. It's not necessarily
-    available on all platforms. Note that any of the listed facts may change
-    any time without a warning.
-
 ``video-aspect`` (RW)
     Video aspect, see ``--video-aspect``.
 
@@ -1510,11 +1381,8 @@ Property list
 ``sub-scale`` (RW)
     Subtitle font size multiplier.
 
-``ass-force-margins`` (RW)
-    See ``--ass-force-margins``.
-
-``sub-use-margins`` (RW)
-    See ``--sub-use-margins``.
+``ass-use-margins`` (RW)
+    See ``--ass-use-margins``.
 
 ``ass-vsfilter-aspect-compat`` (RW)
     See ``--ass-vsfilter-aspect-compat``.
@@ -1556,11 +1424,6 @@ Property list
         been unloaded yet; in this case, ``current`` refers to the new
         selection. (Since mpv 0.7.0.)
 
-    ``playlist/N/title``
-        Name of the Nth entry. Only available if the playlist file contains
-        such fields, and only if mpv's parser supports it for the given
-        playlist format.
-
     When querying the property with the client API using ``MPV_FORMAT_NODE``,
     or with Lua ``mp.get_property_native``, this will return a mpv_node with
     the following contents:
@@ -1572,7 +1435,6 @@ Property list
                 "filename"  MPV_FORMAT_STRING
                 "current"   MPV_FORMAT_FLAG (might be missing; since mpv 0.7.0)
                 "playing"   MPV_FORMAT_FLAG (same)
-                "title"     MPV_FORMAT_STRING (optional)
 
 ``track-list``
     List of audio/video/sub tracks, current entry marked. Currently, the raw
@@ -1600,10 +1462,6 @@ Property list
     ``track-list/N/lang``
         Track language as identified by the file. Not always available.
 
-    ``track-list/N/audio-channels``
-        For audio tracks, the number of audio channels in the audio stream.
-        Not always accurate (depends on container hints). Not always available.
-
     ``track-list/N/albumart``
         ``yes`` if this is a video track that consists of a single picture,
         ``no`` or unavailable otherwise. This is used for video tracks that are
@@ -1613,10 +1471,6 @@ Property list
         ``yes`` if the track has the default flag set in the file, ``no``
         otherwise.
 
-    ``track-list/N/forced``
-        ``yes`` if the track has the forced flag set in the file, ``no``
-        otherwise.
-
     ``track-list/N/codec``
         The codec name used by this track, for example ``h264``. Unavailable
         in some rare cases.
@@ -1632,13 +1486,6 @@ Property list
     ``track-list/N/selected``
         ``yes`` if the track is currently decoded, ``no`` otherwise.
 
-    ``track-list/N/ff-index``
-        The stream index as usually used by the FFmpeg utilities. Note that
-        this can be potentially wrong if a demuxer other than libavformat
-        (``--demuxer=lavf``) is used. For mkv files, the index will usually
-        match even if the default (builtin) demuxer is used, but there is
-        no hard guarantee.
-
     When querying the property with the client API using ``MPV_FORMAT_NODE``,
     or with Lua ``mp.get_property_native``, this will return a mpv_node with
     the following contents:
@@ -1652,10 +1499,8 @@ Property list
                 "src-id"            MPV_FORMAT_INT64
                 "title"             MPV_FORMAT_STRING
                 "lang"              MPV_FORMAT_STRING
-                "audio-channels"    MPV_FORMAT_INT64
                 "albumart"          MPV_FORMAT_FLAG
                 "default"           MPV_FORMAT_FLAG
-                "forced"            MPV_FORMAT_FLAG
                 "external"          MPV_FORMAT_FLAG
                 "external-filename" MPV_FORMAT_STRING
                 "codec"             MPV_FORMAT_STRING
@@ -1734,7 +1579,7 @@ Property list
 
 ``osd-sym-cc``
     Inserts the current OSD symbol as opaque OSD control code (cc). This makes
-    sense only with the ``show-text`` command or options which set OSD messages.
+    sense only with the ``show_text`` command or options which set OSD messages.
     The control code is implementation specific and is useless for anything else.
 
 ``osd-ass-cc``
@@ -1742,13 +1587,13 @@ Property list
     ``${osd-ass-cc/1}`` enables it again. By default, ASS sequences are
     escaped to avoid accidental formatting, and this property can disable
     this behavior. Note that the properties return an opaque OSD control
-    code, which only makes sense for the ``show-text`` command or options
+    code, which only makes sense for the ``show_text`` command or options
     which set OSD messages.
 
     .. admonition:: Example
 
         - ``--osd-status-msg='This is ${osd-ass-cc/0}{\\b1}bold text'``
-        - ``show-text "This is ${osd-ass-cc/0}{\b1}bold text"``
+        - ``show_text "This is ${osd-ass-cc/0}{\b1}bold text"``
 
     Any ASS override tags as understood by libass can be used.
 
@@ -1762,7 +1607,7 @@ Property list
     whether the video window is visible. If the ``--force-window`` option is
     used, this is usually always returns ``yes``.
 
-``video-bitrate``, ``audio-bitrate``, ``sub-bitrate``
+``packet-video-bitrate``, ``packet-audio-bitrate``, ``packet-sub-bitrate``
     Bitrate values calculated on the packet level. This works by dividing the
     bit size of all packets between two keyframes by their presentation
     timestamp distance. (This uses the timestamps are stored in the file, so
@@ -1771,29 +1616,8 @@ Property list
     bitrate. To make the property more UI friendly, updates to these properties
     are throttled in a certain way.
 
-    The unit is bits per second. OSD formatting turns these values in kilobits
-    (or megabits, if appropriate), which can be prevented by using the
-    raw property value, e.g. with ``${=video-bitrate}``.
-
-    Note that the accuracy of these properties is influenced by a few factors.
-    If the underlying demuxer rewrites the packets on demuxing (done for some
-    file formats), the bitrate might be slightly off. If timestamps are bad
-    or jittery (like in Matroska), even constant bitrate streams might show
-    fluctuating bitrate.
-
     How exactly these values are calculated might change in the future.
 
-    In earlier versions of mpv, these properties returned a static (but bad)
-    guess using a completely different method.
-
-``packet-video-bitrate``, ``packet-audio-bitrate``, ``packet-sub-bitrate``
-    Old and deprecated properties for ``video-bitrate``, ``audio-bitrate``,
-    ``sub-bitrate``. They behave exactly the same, but return a value in
-    kilobits. Also, they don't have any OSD formatting, though the same can be
-    achieved with e.g. ``${=video-bitrate}``.
-
-    These properties shouldn't be used anymore.
-
 ``audio-device-list``
     Return the list of discovered audio devices. This is mostly for use with
     the client API, and reflects what ``--audio-device=help`` with the command
@@ -1815,13 +1639,6 @@ Property list
     human readable free form text. The description is an empty string if none
     was received.
 
-    The special entry with the name set to ``auto`` selects the default audio
-    output driver and the default device.
-
-    The property can be watched with the property observation mechanism in
-    the client API and in Lua scripts. (Technically, change notification is
-    enabled the first time this property is read.)
-
 ``audio-device`` (RW)
     Set the audio device. This directly reads/writes the ``--audio-device``
     option, but on write accesses, the audio output will be scheduled for
@@ -1845,16 +1662,6 @@ Property list
     Return the audio device selected by the AO driver (only implemented for
     some drivers: currently only ``coreaudio``).
 
-``working-directory``
-    Return the working directory of the mpv process. Can be useful for JSON IPC
-    users, because the command line player usually works with relative paths.
-
-``protocol-list``
-    List of protocol prefixes potentially recognized by the player. They are
-    returned without trailing ``://`` suffix (which is still always required).
-    In some cases, the protocol will not actually be supported (consider
-    ``https`` if ffmpeg is not compiled with TLS support).
-
 ``mpv-version``
     Return the mpv version/copyright string. Depending on how the binary was
     built, it might contain either a release version, or just a git hash.
@@ -1894,16 +1701,11 @@ Property list
         For many complex types, this isn't very accurate.
 
     ``option-info/<name>/set-from-commandline``
+
         Return ``yes`` if the option was set from the mpv command line,
         ``no`` otherwise. What this is set to if the option is e.g. changed
         at runtime is left undefined (meaning it could change in the future).
 
-    ``option-info/<name>/set-locally``
-        Return ``yes`` if the option was set per-file. This is the case with
-        automatically loaded profiles, file-dir configs, and other cases. It
-        means the option value will be restored to the value before playback
-        start when playback ends.
-
     ``option-info/<name>/default-value``
         The default value of the option. May not always be available.
 
@@ -1926,14 +1728,11 @@ Property Expansion
 ------------------
 
 All string arguments to input commands as well as certain options (like
-``--term-playing-msg``) are subject to property expansion. Note that property
-expansion does not work in places where e.g. numeric parameters are expected.
-(For example, the ``add`` command does not do property expansion. The ``set``
-command is an exception and not a general rule.)
+``--term-playing-msg``) are subject to property expansion.
 
 .. admonition:: Example for input.conf
 
-    ``i show-text "Filename: ${filename}"``
+    ``i show_text "Filename: ${filename}"``
         shows the filename of the current file when pressing the ``i`` key
 
 Within ``input.conf``, property expansion can be inhibited by putting the

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

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
@@ -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
@@ -519,7 +496,7 @@ Example implementation::
         optionC = true,
     }
     read_options(options, "myscript")
-    print(options.optionA)
+    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
 ----------------
@@ -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.
@@ -130,17 +132,9 @@ x and z
 l
     Set/clear A-B loop points. See ``ab_loop`` command for details.
 
-L
-    Toggle infinite looping.
-
 Ctrl + and Ctrl -
     Adjust audio delay by +/- 0.1 seconds.
 
-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
@@ -276,19 +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
 
@@ -305,7 +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.)
 
@@ -324,23 +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
 ----------------
@@ -391,8 +368,7 @@ 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 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
@@ -414,10 +390,10 @@ 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 ``"`` and ``'``,
-as well as with the fixed-length syntax (``%n%``) mentioned before. This is like
-passing the exact contents of the quoted string as command line option. C-style
-escapes are currently _not_ interpreted on this level, although some options do
-this manually. (This is a mess and should probably be changed at some point.)
+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
 --------------------------------------------------------
@@ -492,8 +468,11 @@ 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
 ====================
@@ -526,22 +505,15 @@ listed.
   if there is audio "missing", or not enough frames can be dropped. Usually
   this will indicate a problem. (``total-avsync-change`` property.)
 - Encoding state in ``{...}``, only shown in encoding mode.
-- Display sync state. If display sync is active (``display-sync-active``
-  property), this shows ``DS: +0.02598%``, where the number is the speed change
-  factor applied to audio to achieve sync to display, expressed in percent
-  deviation from 1.0 (``audio-speed-correction`` property). In sync modes which
-  don't resample, this will always be ``+0.00000%``.
-- Missed frames, e.g. ``Missed: 4``. (``vo-missed-frame-count`` property.) Shows
-  up in display sync mode only. This is incremented each time a frame took
-  longer to display than intended.
 - Dropped frames, e.g. ``Dropped: 4``. Shows up only if the count is not 0. Can
   grow if the video framerate is higher than that of the display, or if video
   rendering is too slow. Also can be incremented on "hiccups" and when the video
   frame couldn't be displayed on time. (``vo-drop-frame-count`` property.)
   If the decoder drops frames, the number of decoder-dropped frames is appended
-  to the display as well, e.g.: ``Dropped: 4/34``. This happens only if
-  decoder 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
@@ -571,9 +543,13 @@ PROTOCOLS
     you must mount the ISO file as filesystem, and point ``--bluray-device``
     to the mounted directory directly.
 
+``bdnav://[title][/device]``
+    Play a Blu-Ray disc, with navigation features enabled. This feature is
+    permanently experimental.
+
 ``dvd://[title|[starttitle]-endtitle][/device]`` ``--dvd-device=PATH``
-    Play a DVD. 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.
@@ -618,11 +594,6 @@ PROTOCOLS
     ``PATH`` itself should start with a third ``/`` to make the path an
     absolute path.
 
-``fd://123``
-    Read data from the given UNIX FD (for example 123). This is similar to
-    piping data to stdin via ``-``, but can use an arbitrary file descriptor.
-    Will not work correctly on MS Windows.
-
 ``edl://[edl specification as in edl-mpv.rst]``
     Stitch together parts of multiple files and play them.
 
@@ -632,40 +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
@@ -812,8 +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``
@@ -822,15 +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)
@@ -885,14 +816,6 @@ 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

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
@@ -156,7 +152,7 @@ Available filters are:
     number (1.33). Alternatively, you may specify the exact display width and
     height desired. Note that this filter does *not* do any scaling itself; it
     just affects what later scalers (software or hardware) will do when
-    auto-scaling to the correct aspect.
+    auto-scaling to correct aspect.
 
     ``<w>,<h>``
         New display width and height.
@@ -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.
 
@@ -211,148 +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)
-
-    ``<outputlevels>``
-        RGB color levels used with YUV to RGB conversion. Normally, output devices
-        such as PC monitors use full range color levels. However, some TVs and
-        video monitors expect studio RGB levels. Providing full range output to a
-        device expecting studio level input results in crushed blacks and whites,
-        the reverse in dim gray blacks and dim whites.
-
-        The same limitations as with ``<colormatrix>`` apply.
-
-        Available color ranges are:
-
-        :auto:      automatic selection (equals to full range) (default)
-        :limited:   limited range (16-235 per component), studio levels
-        :full:      full range (0-255 per component), PC levels
-
-        .. note::
-
-            It is advisable to use your graphics driver's color range option
-            instead, if available.
-
-    ``<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.
@@ -413,6 +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
@@ -439,6 +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
@@ -470,7 +389,7 @@ Available filters are:
         video. The main purpose of setting ``mp`` to a chroma plane is to reduce
         CPU load and make pullup usable in realtime on slow machines.
 
-``yadif=[mode:interlaced-only]``
+``yadif=[mode[:enabled=yes|no]]``
     Yet another deinterlacing filter
 
     ``<mode>``
@@ -479,10 +398,10 @@ Available filters are:
         :frame-nospatial: Like ``frame`` but skips spatial interlacing check.
         :field-nospatial: Like ``field`` but skips spatial interlacing check.
 
-    ``<interlaced-only>``
-        :no:  Deinterlace all frames (default).
-        :yes: Only deinterlace frames marked as interlaced (default if this
-              filter is inserted via ``deinterlace`` property).
+    ``<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
     other key that toggles the ``deinterlace`` property or when using the
@@ -497,6 +416,28 @@ Available filters are:
     when inserting yadif with ``--vf``, so using the above methods is
     recommended.
 
+``delogo[=x:y:w:h:t:show]``
+    Suppresses a TV station logo by a simple interpolation of the surrounding
+    pixels. Just set a rectangle covering the logo and watch it disappear (and
+    sometimes something even uglier appear - your mileage may vary).
+
+    ``<x>,<y>``
+        top left corner of the logo
+    ``<w>,<h>``
+        width and height of the cleared rectangle
+    ``<t>``
+        Thickness of the fuzzy edge of the rectangle (added to ``w`` and
+        ``h``). When set to -1, a green rectangle is drawn on the screen to
+        simplify finding the right ``x``,``y``,``w``,``h`` parameters.
+    ``show``
+        Draw a rectangle showing the area defined by x/y/w/h.
+
+``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
@@ -504,9 +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
 
@@ -605,8 +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.
@@ -716,7 +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:
@@ -758,10 +697,6 @@ 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
@@ -806,21 +741,12 @@ Available filters are:
     ``pullup``
         Try to apply inverse telecine, needs motion adaptive temporal
         deinterlacing.
-    ``interlaced-only=<yes|no>``
-        If ``yes`` (default), only deinterlace frames marked as interlaced.
     ``hqscaling=<0-9>``
         0
             Use default VDPAU scaling (default).
         1-9
             Apply high quality VDPAU scaling (needs capable hardware).
 
-``vdpaurb``
-    VDPAU video read back. Works with ``--vo=vdpau`` and ``--vo=opengl`` only.
-    This filter will read back frames decoded by VDPAU so that other filters,
-    which are not normally compatible with VDPAU, can be used like normal.
-    This filter must be specified before ``vdpaupp`` in the filter chain if
-    ``vdpaupp`` is used.
-
 ``buffer=<num>``
     Buffer ``<num>`` frames in the filter chain. This filter is probably pretty
     useless, except for debugging. (Note that this won't help smoothing out

DOCS/man/vo.rst

@@ -69,11 +69,11 @@ 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.
+
+    .. note:: This is a fallback only, and should not be normally used.
 
 ``vdpau`` (X11 only)
     Uses the VDPAU interface to display and optionally also decode video.
@@ -290,26 +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.
-
     ``scale=<filter>``
 
         ``bilinear``
@@ -328,75 +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.
 
-        sharpen3, sharpen5
-            Sharpening strength. Increasing this makes the image sharper but
-            adds more ringing and aliasing. Defaults to 0.5.
-
-        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
@@ -408,26 +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``,
-        ``bicubic_fast`` or ``sharpen``.
-
-    ``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.
+        Note that this currently only affects ``ewa_lanczos``.
 
     ``scaler-resizes-only``
         Disable the scaler if the video image is not resized. In that case,
@@ -435,6 +357,21 @@ Available video output drivers are:
         will reproduce the source image perfectly if no scaling is performed.
         Note that this option never affects ``cscale``.
 
+    ``srgb``
+        Convert and color correct the output to sRGB before displaying it on
+        the screen. This option enables ``linear-scaling``.
+
+        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
@@ -475,72 +412,37 @@ Available video output drivers are:
         Unfortunately, this can lead to flicker on LCD displays, since these
         have a high reaction time.
 
-    ``temporal-dither-period=<1-128>``
-        Determines how often the dithering pattern is updated when
-        ``temporal-dither`` is in use. 1 (the default) will update on every
-        video frame, 2 on every other frame, etc.
-
     ``debug``
         Check for OpenGL errors, i.e. call ``glGetError()``. Also request a
         debug OpenGL context (which does nothing with current graphics drivers
         as of this writing).
 
-    ``interpolation``
-        Reduce stuttering caused by mismatches in the video fps and display
-        refresh rate (also known as judder).
-
-        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``).
+    ``cscale=<filter>``
+        As ``scale``, but for interpolating chroma information. If the image
+        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.
 
-    ``dscale=<filter>``
+    ``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.
 
-    ``cscale=<filter>``
-        As ``scale``, but for interpolating chroma information. If the image
-        is not subsampled, this option is ignored entirely.
+    ``cscale-param1``, ``cscale-param2``, ``cscale-radius``, ``cscale-antiring``
+        Set filter parameters and radius for ``cscale``.
 
-    ``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 ``oversample``.
-
-        Note that the maximum supported filter radius is currently 3, due to
-        limitations in the number of video textures that can be loaded
-        simultaneously.
-
-    ``tscale-clamp``
-        Clamp the ``tscale`` filter kernel's value range to [0-1]. This reduces
-        excessive ringing artifacts in the temporal domain (which typically
-        manifest themselves as short flashes or fringes of black, mostly
-        around moving edges) in exchange for potentially adding more blur.
-
-    ``dscale-radius``, ``cscale-radius``, ``tscale-radius``, etc.
-        Set filter parameters for ``dscale``, ``cscale`` and ``tscale``,
-        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.
 
     ``fancy-downscaling``
         When using convolution based filters, extend the filter size
@@ -550,92 +452,9 @@ Available video output drivers are:
         feature doesn't work correctly with different scale factors in
         different directions.
 
-    ``pre-shaders=<files>``, ``post-shaders=<files>``, ``scale-shader=<file>``
-        Custom GLSL fragment shaders.
-
-        pre-shaders (list)
-            These get applied after conversion to RGB and before linearization
-            and upscaling. Operates on non-linear RGB (same as input). This is
-            the best place to put things like sharpen filters.
-        scale-shader
-            This gets used instead of scale/cscale when those options are set
-            to ``custom``. The colorspace it operates on depends on the values
-            of ``linear-scaling`` and ``sigmoid-upscaling``, so no assumptions
-            should be made here.
-        post-shaders (list)
-            These get applied after upscaling and subtitle blending (when
-            ``blend-subtitles`` is enabled), but before color management.
-            Operates on linear RGB if ``linear-scaling`` is in effect,
-            otherwise non-linear RGB. This is the best place for colorspace
-            transformations (eg. saturation mapping).
-
-        These files must define a function with the following signature::
-
-            vec4 sample(sampler2D tex, vec2 pos, vec2 tex_size)
-
-        The meanings of the parameters are as follows:
-
-        sampler2D tex
-            The source texture for the shader.
-        vec2 pos
-            The position to be sampled, in coordinate space [0-1].
-        vec2 tex_size
-            The size of the texture, in pixels. This may differ from image_size,
-            eg. for subsampled content or for post-shaders.
-
-        In addition to these parameters, the following uniforms are also
-        globally available:
-
-        float random
-            A random number in the range [0-1], different per frame.
-        int frame
-            A simple count of frames rendered, increases by one per frame and
-            never resets (regardless of seeks).
-        vec2 image_size
-            The size in pixels of the input image.
-
-        For example, a shader that inverts the colors could look like this::
-
-            vec4 sample(sampler2D tex, vec2 pos, vec2 tex_size)
-            {
-                vec4 color = texture(tex, pos);
-                return vec4(1.0 - color.rgb, color.a);
-            }
-
-    ``deband``
-        Enable the debanding algorithm. This greatly reduces the amount of
-        visible banding, blocking and other quantization artifacts, at the
-        expensive of very slightly blurring some of the finest details. In
-        practice, it's virtually always an improvement - the only reason to
-        disable it would be for performance.
-
-    ``deband-iterations=<1..16>``
-        The number of debanding steps to perform per sample. Each step reduces
-        a bit more banding, but takes time to compute. Note that the strength
-        of each step falls off very quickly, so high numbers are practically
-        useless. (Default 4)
-
-        If the performance hit of debanding is too great, you can reduce this
-        to 2 or 1 with marginal visual quality loss.
-
-    ``deband-threshold=<0..4096>``
-        The debanding filter's cut-off threshold. Higher numbers increase the
-        debanding strength dramatically but progressively diminish image
-        details. (Default 64)
-
-    ``deband-range=<1..64>``
-        The debanding filter's initial radius. The radius increases linearly
-        for each iteration. A higher radius will find more gradients, but
-        a lower radius will smooth more aggressively. (Default 8)
-
-    ``deband-grain=<0..4096>``
-        Add some extra noise to the image. This significantly helps cover up
-        remaining quantization artifacts. Higher numbers add more noise.
-        (Default 48)
-
     ``sigmoid-upscaling``
         When upscaling, use a sigmoidal color transform to avoid emphasizing
-        ringing artifacts. This also implies ``linear-scaling``.
+        ringing artifacts. This also enables ``linear-scaling``.
 
     ``sigmoid-center``
         The center of the sigmoid curve used for ``sigmoid-upscaling``, must
@@ -645,6 +464,10 @@ Available video output drivers are:
         The slope of the sigmoid curve used for ``sigmoid-upscaling``, must
         be a float between 1.0 and 20.0. Defaults to 6.5 if not specified.
 
+    ``no-npot``
+        Force use of power-of-2 texture sizes. For debugging only.
+        Borders will be distorted due to filtering.
+
     ``glfinish``
         Call ``glFinish()`` before and after swapping buffers (default: disabled).
         Slower, but might help getting better results when doing framedropping.
@@ -658,18 +481,6 @@ Available video output drivers are:
 
         X11/GLX only.
 
-    ``dwmflush=<no|windowed|yes>``
-        Calls ``DwmFlush`` after swapping buffers on Windows (default: no).
-        It also sets ``SwapInterval(0)`` to ignore the OpenGL timing. Values
-        are: no (disabled), windowed (only in windowed mode), yes (also in
-        full screen).
-        This may help getting more consistent frame intervals, especially with
-        high-fps clips - which might also reduce dropped frames. Typically a
-        value of ``windowed`` should be enough since full screen may bypass the
-        DWM.
-
-        Windows only.
-
     ``sw``
         Continue even if a software renderer is detected.
 
@@ -684,22 +495,20 @@ Available video output drivers are:
             Cocoa/OS X
         win
             Win32/WGL
-        x11
-            X11/GLX
+        x11, x11es
+            X11/GLX (the ``es`` variant forces GLES)
         wayland
             Wayland/EGL
-        x11egl
-            X11/EGL
-
-    ``es``
-        Force or prefer GLES2/3 over desktop OpenGL, if supported.
+        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: rgba16.
+        Default: rgba.
 
     ``gamma=<0.1..2.0>``
         Set a gamma value (default: 1.0). If gamma is adjusted in other ways
@@ -715,65 +524,12 @@ 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
@@ -781,18 +537,16 @@ Available video output drivers are:
 
         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
@@ -808,28 +562,6 @@ 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``.
-
-        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|yes|no>``
         Decides what to do if the input has an alpha component (default: blend).
 
@@ -845,6 +577,10 @@ 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
@@ -854,16 +590,65 @@ 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:fancy-downscaling:sigmoid-upscaling:pbo:deband
+        --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.
@@ -924,12 +709,6 @@ 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.
 
@@ -1009,52 +788,15 @@ Available video output drivers are:
     ``frame-queue-size=<1..100>``
         The maximum count of frames which the frame queue can hold (default: 1)
 
-    ``frame-drop-mode=<pop|clear|block>``
+    ``frame-drop-mode=<pop|clear>``
         Select the behavior when the frame queue is full.
 
         pop
-            Drop the oldest frame in the frame queue.
+            Drop the oldest frame in the frame queue. (default)
         clear
             Drop all frames in the frame queue.
-        block
-            Wait for a short time, behave like ``clear`` on timeout. (default)
 
-    This also supports many of the suboptions the ``opengl`` VO has. Run
+    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.
-
-``drm`` (Direct Rendering Manager)
-    Video output driver using Kernel Mode Setting / Direct Rendering Manager.
-    Does not support hardware acceleration. Should be used when one doesn't
-    want to install full-blown graphical environment (e.g. no X).
-
-    ``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

@@ -1,39 +1,22 @@
-![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].
 
-## Downloads
-
+Downloads
+---------
 
 For semi-official builds and third-party packages please see
 [mpv.io](http://mpv.io/installation/).
 
-## Compilation
-
+Compilation
+-----------
 
 Compiling with full features requires development files for several
 external libraries. Below is a list of some important requirements.
@@ -57,6 +40,9 @@ 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
@@ -64,10 +50,10 @@ 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)
 - Enca (optional, for subtitle charset detection)
@@ -103,16 +89,16 @@ 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.
@@ -130,21 +116,21 @@ 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
-
+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
@@ -165,15 +151,17 @@ 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.
@@ -182,12 +170,9 @@ only if discretion is required.
 [mpv-build]: https://github.com/mpv-player/mpv-build
 [homebrew-mpv]: https://github.com/mpv-player/homebrew-mpv
 [issue-tracker]:  https://github.com/mpv-player/mpv/issues
+[mpv-users]: https://groups.google.com/forum/?hl=en#!forum/mpv-users
+[mpv-devel]: https://groups.google.com/forum/?hl=en#!forum/mpv-devel
 [ffmpeg_vs_libav]: https://github.com/mpv-player/mpv/wiki/FFmpeg-versus-Libav
 [release-policy]: https://github.com/mpv-player/mpv/blob/master/DOCS/release-policy.md
 [windows_compilation]: https://github.com/mpv-player/mpv/blob/master/DOCS/compile-windows.md
 [mplayer-changes]: https://github.com/mpv-player/mpv/blob/master/DOCS/mplayer-changes.rst
-
-## License 
-
-
-[GPLv2](https://github.com/mpv-player/mpv/blob/master/LICENSE)

RELEASE_NOTES

@@ -1,602 +1,3 @@
-Release 0.11.0
-==============
-
-Features
---------
-
-New
-~~~
-
-- vo_opengl: implement debanding
-
-Removed
-~~~~~~~
-
-- audio/filter: remove center, extrastereo, karaoke, sinesuppress, sub,
-  surround, sweep, ladspa, hrtf, export and bs2b filters (these are either
-  considered useless or have replacements in lavfi)
-- video/filter: remove lavfi wrappers for noise, hqdn3d, unsharp and delogo
-  (these filters remain usable through lavfi)
-
-Behavior
---------
-
-- vo_opengl: require FBOs by default (use dumb-mode suboption for old hardware
-  and broken drivers)
-- vo_opengl: enable debanding by default for the opengl-hq preset
-- audio/out: use new sample format determination code
-- player: prefer logical current directory path (affects logic for resuming
-  playback)
-- vf_vdpaurb: Pass through non-hardware-decoded content
-- player: make force-window=immediate work in auto-profiles
-
-
-Options and Commands
---------------------
-
-Added
-~~~~~
-
-- af_lavrresample: add normalize suboption
-- vo_opengl: add deband, deband-iterations, deband-threshold, deband-range and
-  deband-grain suboptions
-- af_lavfi: implement af-metadata property (like vf-metadata)
-
-Changed
-~~~~~~~
-
-- command: make "add <property> 0" not change the value
-
-Removed
-~~~~~~~
-
-- vo_opengl: remove source-shader suboption
-
-
-Fixes and Minor Enhancements
-----------------------------
-
-- options: fix --no-config
-- cache: do not include backbuffer size in total stream cache size
-- audio/format: actually prefer float over double sample format for
-  int->float conversions
-- audio/format: fix interlaved vs. non-interleaved conversions
-- audio/format: revise format conversion scoring
-- video: make --field-dominance set interlaced flag
-- vf: vf_stereo3d compilation depends on libavfilter
-- vf_yadif: add hack for Libav compatibility
-- player: add opus to list of external audio file extensions
-- build: allow disabling vapoursynth completely
-- libmpv/win32: allow multiple windows at the same time
-
-
-This listing is not complete. Check DOCS/client-api-changes.rst for a history
-of changes to the client API. A complete changelog can be seen by running
-`git log v0.10.0..v0.11.0` in the git repository or by visiting either
-https://github.com/mpv-player/mpv/compare/v0.10.0...v0.11.0 or
-http://git.srsfckn.biz/mpv/log/?qt=range&q=v0.10.0..v0.11.0
-
-
-Release 0.10.0
-==============
-
-
-Features
---------
-
-New
-~~~
-
-- uchardet support
-- Matroska: reading cue sheets embedded in tags
-- Support for VideoToolbox hardware decoding
-- Display sync mode (--display-sync)
-- --force-window=immediate mode
-- fd:// protocol
-- libarchive wrapper for reading compressed archives
-- TOOLS/lua: zones.lua
-- Support for the "new" libavcodec VDPAU API
-- vf_vdpaurb, Add a new filter for reading back VDPAU decoded frames
-- DXVA2: HEVC support
-- Enabled HEVC profiles with VA API
-- HEVC added to whitelist of hwdec codecs
-- vo_null: framerate emulation
-- vo_opengl: support for custom shaders
-- vo_opengl: temporal-dither-period option
-- vo_opengl: tscale-clamp option
-- vo_opengl: option to attach target-prim/target-csp to window screenshots
-- vo_opengl_cb: "block" framedrop mode (now default)
-- vo_opengl_cb: support for interpolation
-- vo_vdpau: rotation support
-- ytdl_hook: support for 'multi_video' results
-
-
-Removed
-~~~~~~~
-
-- af_convert24 (af_lavrresample does this now)
-- af_dummy
-- audio: S8, U16, U24, U32 formats
-- DVD and BD menu support
-- TOOLS: youtube-starttime.lua
-- VA API: compatibility crap (< 0.34.0) and vo_vaapi deinterlacer
-- vo_x11
-
-
-Deprecated
-~~~~~~~~~~
-
-- vf_dlopen
-
-
-Behavior
---------
-
-- audio: softvol scale is now cubic
-- cache: readahead size is limited to half the cache size at the beginning
-- charset_conv: "auto" encoding detection now prefers uchardet
-- demux_playlist: skip hidden directories
-- input.conf: O toggles between 2 states only
-- input.conf: Ctrl+s key binding for window screenshots
-- input.conf: mouse volume control (horizontal scrolling) inverted
-- input.conf: L to toggle infinite looping
-- input.conf: remap d/D keys (dropped framedrop cycle, replaced with deinterlacing toggle)
-- player: disabled seeking on unseekable streams even if the cache is enabled
-- player: parses and exposes m3u playlist titles
-- player: --term-playing-msg is now in a separate log category
-- player: removed automatic DVB channel advancement on no data
-- player: now restores video-aspect on playback resume
-- player: now uses exit code 0 by default for quit, 4 for signals, etc.
-- player: warns against using HLS URLs with --playlist
-- screenshots: changed the default directory in pseudo-gui mode to desktop
-- screenshots: screenshot directory is now created automatically
-- screenshots: default template is now prefixed with "mpv-"
-- TOOLS/lua/autoload: adds all files on start
-- vo: vo_wayland moved up in autoprobe list
-- vo_opengl: enabled pbo by default with opengl-hq
-- vo_opengl: cache dir for ICC profiles is now created automatically
-- w32: shift drag and drop appends
-- x11: shift drag and drop appends
-
-
-Options and Commands
---------------------
-
-Added
-~~~~~
-
-- af_volume: replaygain-fallback option
-- ao_coreaudio: change-physical-format option
-- ao_coreaudio: exclusive option
-- ao_null: channel-layouts option for testing channel layout selection
-- audio: --audio-spdif as new method for enabling passthrough
-- cache: --cache-backbuffer to configure cache backbuffer size
-- command: define-section command for defining input bindings
-- command: audio-params and audio-out-params properties
-- command: keypress, keydown, and keyup commands
-- command: playlist_shuffle command
-- command: option-info/N/set-locally property indicating per-file options
-- command: protocol-list property
-- command: track-list/N/audio-channels property
-- demux: --demuxer-max-packets and --demuxer-max-bytes options to control maximum queue size
-- input: relative percentage seek
-- osc: time display configuration options
-- player: --playlist-pos option
-- screenshots: --screenshot-jpeg-source-chroma option to disable JPEG 4:4:4 output
-- screenshots: --screenshot-high-bit-depth option to allow or disallow 16 bit output
-- screenshots: --screenshot-directory option
-- sub: --stretch-image-subs-to-screen option for stretching image subtitles to screen
-- TOOLS/stats-conv: allow passing regex via command line
-- video: --video-aspect-method option to configure container vs. bitstream aspect ratio
-- vo_drm: mode suboption to set the mode ID to use
-- vo_opengl_cb, vo_opengl: --hwdec-preload option for preloading hwdec context
-- vo_rpi: background disabled by default
-- vo_xv: buffers suboption to configure number of buffers
-- win32: portable config mode
-
-
-Changed
-~~~~~~~
-
-- audio: changed the range of the volume option/property (0 is still silence, and 100 now always means unchanged volume)
-- command: allow changing deinterlace property any time
-- command: allow changing track properties while no file is loaded
-- command: always make video-aspect property accessible
-- command: better choice when to allow playback-related commands
-- command: change OSD symbol for absolute perc. seek
-- command: change the default action for rescan_external_files
-- command: change the hwdec properties
-- command: define-section with empty contents removes a section
-- command: export stereo 3D tags
-- command: make auto-deinterlacing output at field rate
-- command: make deinterlace property use interlaced-only yadif mode
-- command: make property event mask matching more restrictive
-- command: make the playback-time property writable
-- input: allow - as separator between commands, instead of _
-- options: --loop without argument means looping forever
-- options: make keyvalue list parsing less strict
-- player: extend --hls-bitrate option
-- vf_yadif: expose interlaced frame mode
-- video: --video-stereo-mode=no to disable automatic stereo conversion
-- vo_opengl_cb, vo_opengl: --hwdec-preload for preloading hwdec context
-- vo_opengl: replace icc-cache with icc-cache-dir
-- vo_opengl: icc-profile overrides icc-profile-auto
-
-
-Renamed
-~~~~~~~
-
-- command: rename audio-format property to audio-codec-name
-- options: rename --media-title option to --force-media-title
-- vo_opengl: rename use_full_range to use_normalized_range
-
-
-Deprecated
-~~~~~~~~~~
-
-- --ad-spdif-dtshd (use --audio-spdif=dts-hd)
-- audio-samplerate property
-- length property
-
-
-Removed
-~~~~~~~
-
-- get_property command
-- --demuxer-readahead-packets and --demuxer-readahead-bytes
-- image_writer: don't use jpeg baseline, and remove useless jpeg-optimize and jpeg-baseline options
-- --leak-report
-- --slave-broken
-- vo_opengl: npot suboption
-
-
-Fixes and Minor Enhancements
-----------------------------
-
-- ad_spdif: use DTS-HD passthrough only if the audio is really DTS-HD
-- af: fix behavior with filter chains that require a large number of auto-inserted conversion filters
-- af_lavcac3enc: fix A/V sync
-- ao_alsa: accept 7.1 over HDMI
-- ao_alsa: refuse to use spdif if AES flags can't be set
-- ao_wasapi: fix crash on hotplug init error
-- audio: avoid wasting CPU due to continuous wakeup
-- audio: do not exit when loading small files in paused mode
-- audio: fix channel map fallback selection
-- audio: fix crash on uninit
-- audio: fix --end handling
-- audio: fix EOF state with --keep-open
-- audio: fix restoring volume
-- charset_conv: fix switched parameters
-- charset_conv: use our own UTF-8 check with ENCA only
-- cocoa: don't load hardcoded icon if running from bundle
-- cocoa: hide cursor using a blank image instead of a system-wide API
-- command: do not exit playback if the B point of A-B loop is past EOF
-- command: fix audio-out-detected-device property
-- command: fix track property when no file is loaded
-- command: fix video-aspect property update notification
-- command: let track properties return option value in idle mode
-- demux: don't get stuck on some cases of timestamp resets
-- demux: handle Matroska-style replaygain tags as well
-- demux_lavf: do a better job at guessing the vobsub .sub filename
-- demux_mkv: disable ordered chapters if ChapterTimeEnd is missing
-- demux_mkv: discard broken index
-- demux_mkv: fix mpeg2 mapping
-- demux_mkv: ignore deprecated FrameRate, do not assume PAL
-- demux_mkv: improve video duration detection heuristic
-- demux_mkv: parse FLAC channel layouts
-- demux_playlist: make mime type comparison case-insensitive
-- dxva2: fix handling of cropped video
-- idet.sh: Support larger files
-- mp_image: fix vf_vdpaupp references
-- options: fix conversion of flags to strings
-- options: move program name to end of window title
-- options: remove the period at the end of "No file."
-- osc: completely disable if no VO window exists
-- osc: exit tick immediately if disabled
-- osc: reinit on playlist changes
-- osx: add NULL check for input context in a missing case
-- player: fix crashes when adding external tracks before loading main file
-- player: increase tick event update frequency
-- player: make decoding cover art more robust
-- player: raise maximum idle time
-- player: return better guess for playback time during seeks
-- player: show larger cache sizes in MB on status line
-- player: slim down A/V desync warning
-- sd_ass: assume negative durations are unknown durations, and handle them
-- terminal: disable terminal foreground state polling
-- terminal-unix: set terminal mode on init
-- timer: fix a corner case on clock changes
-- TOOLS: make autodeint detect telecine in parallel
-- TOOLS/zsh.pl: die loudly if mpv fails to run
-- vaapi: prefer direct display over copy-back
-- vaapi: fix some videos only showing up green
-- vaapi: treat cropped decoder output slightly more correctly
-- vda: add support for nv12 image formats
-- vd_lavc: fix a hw decoding fallback case
-- vf_stereo3d: drop internal implementation
-- vf_vavpp: don't attempt to deinterlace progressive frames
-- vf_vavpp: fix bob deinterlacing for bottom field first videos
-- vf_vdpaupp: Don't crash when evaluating interlacing of NULL mpi
-- video: always re-probe auto deint filter on filter reconfig
-- video: better heuristic for timestamp resets
-- video: fix panscan in vertical case
-- video: fix VideoToolbox/VDA autodetection
-- video: unbreak EOF with video-only files that have timestamp resets
-- vo_direct3d: fix broken pseudo GUI drag and drop hint
-- vo_drm: fix centering with regard to stride
-- vo_drm: fix crashes with --profile=pseudo-gui
-- vo_drm: fix resolution not restored after exiting
-- vo_drm: fix stride problem for certain devices
-- vo_drm: make VT switching non mandatory
-- vo: free frames before killing VO
-- vo_opengl: avoid broken shader if hwdec fails to provide textures
-- vo_opengl_cb: drop frames eagerly if frames are not rendered
-- vo_opengl: CMS no longer implies linear scaling
-- vo_opengl: fix alpha video in one case
-- vo_opengl: fix dangling pointers with vo_cmdline
-- vo_opengl: fix framestepping/pausing + interpolation
-- vo_opengl: fix "freezes" after seeking with interpolation on
-- vo_opengl: fix scale=oversample's threshold calculations
-- vo_opengl: framebuffers work under GLES 2
-- vo_opengl: improve robustness against PBO failure
-- vo_opengl: reimplement tscale=oversample
-- vo_opengl: reject future images in different formats
-- vo_opengl: X11: don't leak when GL init fails
-- vo: restore frame-drop logic for high-fps clips
-- vo_rpi: fix blackscreen before the first subtitle/OSD is rendered
-- vo_rpi, vo_opengl: do not globally terminate EGL on VO uninit
-- vo_sdl: fix glaring memory leak
-- vo_vdpau: check maximum video size
-- vo_vdpau: limit output surfaces to allowed maximum dimensions
-- win32: fix window resize logic
-- win32: fix crashes when changing system time
-- x11: Handle external fullscreen toggles
-- ytdl: catch bogus extractor info
-- ytdl: do not use deprecated media-title option
-- ytdl: don't print failure warning when youtube-dl was killed by us
-- ytdl: get start_time
-
-This listing is not complete. Check DOCS/client-api-changes.rst for a history
-of changes to the client API. A complete changelog can be seen by running
-`git log v0.9.2..v0.10.0` in the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/v0.9.2...v0.10.0.
-
-
-Release 0.9.2
-=============
-
-Changes
--------
-
-- The Lua check now also checks for `lua52.pc`, as used by Arch Linux testing.
-- (X11) `vo_opengl`'s `icc-profile-auto` now queries the current ICC profile
-  relative to the center of the window.
-- `ao_coreaudio`, `ao_alsa` now support adding dummy padding channels for
-  better compatibility with hardware decoders that only support specific
-  channel counts (e.g. 5.1 now should work on a decoder that only accepts 7.1).
-- Channel fallback (in case the audio device doesn't natively support a given
-  channel layout) has been improved.
-- `vf_vapoursynth` now rejects unaligned video instead of outputting corrupted
-  video.
-- mpv now tries to autoload `.sup` subtitles as well.
-
-Bug fixes
----------
-
-- `vo_opengl`'s default for `fbo-format` is now `rgba16`, to avoid rounding
-  errors when using non-default `cscale` (issue #1918).
-- Improved framedrop behavior when playing video that has higher framerate
-  than the display (issue #1897).
-- Trying to play a directory will no longer spam `Connection lost!` to the
-  console log.
-- (Linux) Several `vo_rpi` bugfixes.
-- (Linux) Several `vo_drm` bugfixes. Pan&Scan is now supported.
-- (X11) Fix fullscreen behavior on certain window managers (issues #1937,
-  #1920).
-- (OSX) The OSD no longer always shows up on startup.
-- (OSX) Several `ao_coreaudio` and `ao_coreaudio_exclusive` bugfixes.
-- (OSX) Fixed potential crash on exit when using Cocoa.
-- (ClientAPI) `vo_opengl_cb` now actually applies options changed at runtime.
-- (OSX, ClientAPI) Cocoa now works when both the `cplayer` (`mpv`) and `libmpv`
-  are built at the same time; however, `libmpv` now always creates an
-  application singleton. Cocoa has to be disabled completely to prevent
-  `libmpv` from creating the singleton.
-
-This listing is not complete. A complete changelog can be seen by running
-`git log v0.9.1..v0.9.2` in the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/v0.9.1...v0.9.2.
-
-Release 0.9.1
-=============
-
-Changes
--------
-
-- mpv's IRC channel moved from #mpv-player to #mpv on chat.freenode.net.
-- Documentation updates.
-- The default value for the `--ytdl-format` option is now `best`, in order to
-  work around `youtube-dl`'s 2015.04.26 release enabling DASH by default, as
-  FFmpeg / Libav do not yet properly support DASH.
-- When seeking, the current timestamp will show the predicted seek timestamp
-  instead, until the final timestamp is resolved. Improves UI responsiveness
-  on slow streams and/or large seeks.
-
-Bug fixes
----------
-
-- Corrected the release marker on DOCS/client-api-changes. The release manager
-  forgot to fix it before release...
-- Fix `vo_vdpau` rendering garbage lines on H.264 video with non-mod16 size
-  (issue #1863).
-- Fix a crash on exit if the "sub_reload" command had run successfully.
-- Fixed seeking with the mouse when `osc-seekbarstyle=bar` is set (issue
-  #1876).
-- (IPC) Fixed encoding of UTF-8 data in JSON (issue #1874).
-
-New features
-------------
-
-This listing is not complete. A complete changelog can be seen by running
-`git log v0.9.0..v0.9.1` in the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/v0.9.0...v0.9.1.
-
-Release 0.9.0
-=============
-
-Changes
--------
-
-Changes that may break users' config files have been annotated with a `(!)`.
-
-- Note: mpv is not compatible with Lua 5.3. Lua 5.1 or 5.2 is required.
-- The minimum required libass version is now 0.12.1 or newer.
-- The minimum required FFmpeg version is now 2.4.0 (equiv. Libav 11) or newer.
-- The internal libmpg123 support was removed. This was already not used by
-  default in the previous release.
-- `(!)` The LIRC support was removed. Configure LIRC remotes as input devices
-  instead.
-- `(!)` The Linux Joystick support was removed.
-- `(!)` `vf_screenshot` was removed, as they are now handled at a VO level and
-  is compatible with all VOs.
-- `(!)` `--ass-use-margins` has been renamed to `--sub-use-margins` and applies
-  only to plain-text (non-ASS) subtitles (enabled by default). The new
-  `--ass-force-margins` option applies only to ASS subtitles (disabled by
-  default). To get the old behaviour back, enable both at the same time.
-- `(!)` The `--sub-scale-with-window` option now only applies to plain text
-  (non-ASS) subtitles (enabled by default). The new `--ass-scale-with-window`
-  option does the same but only with ASS subtitles (disabled by default).
-- `(!)` The range for the `param1` for the `gaussian` `vo_opengl` scaler has
-  been redefined. Instead of being an arbitrary 1-100 range, have a default
-  value of 1.0, and anything higher is blurrier.
-- `(!)` The `seek`, `playlist_next, `playlist_prev`, `loadfile` and `loadlist`
-  parameters no longer accept numerical parameters where symbolic parameter
-  names exist.
-- `(!)` `vo_opengl` changes:
-  - `(!)` The `smoothmotion` suboption has been renamed to `interpolation`.
-    The old name is still supported for now.
-  - `(!)` The `bilinear_slow` scaler has been renamed to `triangle`.
-  - `(!)` `scale-down` has been renamed to `dscale` and now has its own set of
-    config options (e.g. `dscale-radius`).
-  - `(!)` Scaler radius no longer defaults to `3` but to a preferred value
-    that may be different for each filter.
-  - The `scale-radius` option may now go down as low as `0.5`, which is the
-    value used by the `nearest` filter.
-  - `spline36` is the new `cscale` default for `opengl-hq`. This might break
-    setups that use `fbo-format=rgb8`. To work around it, leave `fbo-format`
-    as its default, or set to something higher than 8, or set `cscale=bilinear`,
-    the previous default.
-- `(!)` `vf_format` no longer converts video to YUYV if there is no parameter.
-  Video is now passed unchanged unless a format is specifically requested.
-- `(!)` The `--colormatrix`, `--colormatrix-input-range`,
-  `--colormatrix-output-range` and `--colormatrix-primaries` options have been
-  converted into `vf_format` suboptions. See commit 27715b7 and the manual for
-  details.
-- `vf_mirror`'s implementation was replaced with calling into `libavfilter`'s
-  `vf_hflip` filter, thus depending on `libavfilter` to function.
-- The `device` subption to `ao_wasapi` has been deprecated in favor of
-  `--audio-device`.
-- `--video-rotate` now allows 360 as an argument instead of stopping at 359.
-- Several improvements to `af_scaletempo`.
-- Options that have multiple options and also include a "yes" option now
-  default to that if specified with no arguments.
-- The default value of `--cache-default` is now 150000 (153.6 MB, ~146 MiB).
-- JPEG screenshots now use the same subsampling as the source video. The images
-  are still RGB regardless of source format though.
-
-Bug fixes
----------
-
-- mpv no longer saves position on files that can't be resumed (issue #1701).
-- (X11) Fix the player thinking the mouse has left the window in some WMs /
-  embeddings (issue #1672).
-- mpv no longer freezes on wayland when the compositor stops asking it to draw
-  itself (e.g. when minimized) (issue #249).
-- `.ac3` files are no longer rejected by `--audio-file-auto` (issue #1759).
-- `ao_wasapi` now automatically enables `exclusive` when passthrough is
-  attempted (issue #1742).
-- Attempt to fix flickering on Intel VAAPI drivers with `--hwdec=vaapi` and
-  `--vo=opengl` (issue #1765).
-- `youtube-dl` will no longer download video streams when video playback
-  is disabled with `--no-video`.
-- (Windows) mpv now prevents system sleep when playing a video-only file.
-  Previously, only files with an opened audio track would prevent sleep.
-
-New features
-------------
-
-- `vo_opengl` features:
-  - Added `ewa_ginseng`, `ewa_hanning`, `robidoux`, `robidouxsharp`,
-    `oversample` and `haasnsoft` scalers.
-  - There are now `ewa_lanczossoft` and `ewa_lanczossharp` aliases to
-    `ewa_lanczos` that are tuned to be blurrier and sharper, respectively.
-  - Added `gamma-auto` option that uses ambient light sensors to automatically
-    adjust the video gamma. See commit c028d78 for details.
-  - Added `blend-subtitles` option to draw subtitles directly into the video
-    instead of rendered afterwards. Potentially necessary for correct rendering
-    with files that use ASS subtitles for typesetting in combination with an
-    `icc-profile`. There is a default option for drawing after upscaling,
-    and a `video` option for drawing before upscaling. See details and warning
-    on the manual.
-  - There is now a `tscale` option, used to choose the temporal scaler used
-    in the `interpolation` mode (previously `smoothmotion`).
-  - There is a new `scale-blur` parameter to adjust the amount of blur that
-    most of the filters produce. Deviating from the default may introduce
-    artifacts in EWA filters.
-  - (Windows) There is now a `dwmflush` option that might help improve
-    rendering of high-fps video. Disabled by default. See manual for options.
-- New Linux-only `vo_drm` video output driver. Uses the direct rendering /
-  kernel modesetting drivers to draw directly to the framebuffer, but with
-  no hardware acceleration. See manual for details.
-- New `pseudo-gui` builtin profile, automatically used when launched from
-  `mpv.desktop` by opening `mpv.exe` on windows (`mpv.com` still works as
-  usual), or by opening the `mpv.app` bundle. The `pseudo-gui` tries to make
-  the player window behave closer to what a desktop player would do, by not
-  immediately closing and allowing the user to drag&drop files for playback.
-  See manual for details.
-- mpv can now play directories by automatically playing their contents instead.
-  Works everywhere but on Windows, due to issues with Windows' C runtime.
-- Add support to pitch correct stretched audio with librubberband.
-- Add support for the Raspberry Pi 2's hardware decoder when FFmpeg (or Libav)
-  is built with `--enable-mmal`. See commit 8fff125 for details.
-- The `--cache` option now accepts a "yes" option, that always enables a
-  `--cache-default`-sized cache on all cases a cache can be used, unless
-  `--cache-default` disables caching.
-- `ao_pulse`, `ao_coreaudio` and `ao_wasapi` now support device hotplugging.
-- New `--osd-align-x` and `--osd-align-y` options can be used to align the OSD
-  independently from subtitles.
-- New `--osd-bold` and `--sub-text-bold` options can be enabled to bold all
-  OSD or plain-text subtitle text, respectively.
-- Added a default keybind to the `u` key that enables/disables ASS style
-  overriding. When enabled, is equivalent to `--ass-style-override=force`.
-- There is now a `MOUSE_ENTER` keybind, that is called when the mouse cursor
-  enters the VO from outside.
-- The new `--ytdl-params` option can be set to arguments that are always given
-  to `youtube-dl` invocations. There is no sanity checking, so invalid options
-  can prevent `mpv` from working with http URLs.
-- There's a new `--demuxer-mkv-fix-timestamps` option, enabled by default, that
-  tries to guess more accurate video timestamps by using FPS information, if
-  available. See manual for details.
-- The new `--window-scale` option can be used to scale the video window by the
-  specified multiplier, before other options such as `--autofit` are applied.
-- `vo_direct3d_shaders` now supports NV12 colorspace without using stretchrect,
-  but it seems no drivers actually support that.
-- (Client API) Added a `rescan_external_files` command, as requested on issue
-  #1586.
-- (Client API) If enabled, initializing the Client API will now also load the
-  user's `mpv.conf`.
-- (Client API) There's a new `mpv_opengl_cb_report_flip()` call for API users
-  to call to inform mpv of when exactly a frame was displayed.
-- (Client API) The `mpv_opengl_cb_render()` function was deprecated in favor
-  of the new, simpler `mpv_opengl_cb_draw()` function.
-- (Client API) There is a new `screenshot_raw` command. See `input.rst` for
-  details.
-- The `--input-file` argument may now also be a file descriptor in the form
-  "fd://N", where N is the FD number.
-
-This listing is not complete. A complete changelog can be seen by running
-`git log v0.8.0..v0.9.0` in the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/v0.8.0...v0.9.0 (only 250 commits).
-
 Release 0.8.3
 =============
 
@@ -610,9 +11,6 @@ Bug fixes
 ---------
 
 - (X11) Fix crash on `vo_xv:no-colorkey` with the Overlay adapter. (bug #1629)
-- (X11) `--stop-screensaver` is now implemented by calling into
-  `xdg-screensaver`, fixing some compatibility issues that prevented it from
-  actually stopping screensavers.
 - Make the video equalizer work correctly on some VAAPI drivers. (bug #1647)
 - Prevent OSD from disappearing when clicking on mozplugger. (bug #1672)
 - The new DVB-S2 support code, using S2API, now builds on FreeBSD.
@@ -628,8 +26,7 @@ Bug fixes
 - Fix anamorphic scaling being ignored if it was very minor.
 
 This listing is not complete. A complete changelog can be seen by running
-`git log v0.8.2..v0.8.3` in the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/v0.8.2...v0.8.3.
+`git log v0.8.2..v0.8.3` in the git repository.
 
 Release 0.8.2
 =============
@@ -637,7 +34,7 @@ Release 0.8.2
 Bug fixes
 ---------
 
-- Fix OSD placement accidentally messed up in 0.8.1 (commit 22863d6).
+- Fix OSD placement accidentally messed up in 0.8.1.
 
 Release 0.8.1
 =============
@@ -668,8 +65,7 @@ New features
 - mpv now tries to load '.vtt' subtitles.
 
 This listing is not complete. A complete changelog can be seen by running
-`git log v0.8.0..v0.8.1` in the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/v0.8.0...v0.8.1.
+`git log v0.8.0..v0.8.1` in the git repository.
 
 Release 0.8.0
 =============
@@ -835,224 +231,4 @@ New features
   files.
 
 This listing is not complete. A complete changelog can be seen by running
-`git log v0.7.0..v0.8.0` in the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/v0.7.0...v0.8.0 (only 250 commits).
-
-Release 0.7.3
-=============
-
-Changes
--------
-
-- Several documentation updates and typo fixes.
-- Don't show "0%" position when the stream is infinite / has unknown length.
-- Fixes the config file loading order so that lower priority `mpv.conf` files
-  don't override higher priority `config` files.
-- (OSX) Remove coreaudio_exclusive from the "auto" codec list.
-- Failing to create a GL3 context is now a warning instead of error, to reduce
-  user confusion.
-- The subtitle decoder now gets reset when cycling subtitles. This makes the
-  currently shown subtitle event disappear even if cycling back to the current
-  subtitle track.
-- `--shuffle` and `--merge-files` now affects the contents of playlist files,
-  instead of just the list of files given as arguments to mpv.
-- `./waf install` avoids installing a few additional data/config files if only
-  libmpv was built.
-- Improved channel mapping when the file's channel map doesn't match the AO's
-  available channel maps.
-- (OSX) VDA now gracefully refuses to run on non-OpenGL3-capable systems instead
-  of trying and failing to build its shaders.
-- Add workaround for vf_vapoursynth filters that expect an FPS.
-- The default value for `--screenshot-template` now shows up in
-  `--list-options`.
-- `ao_pulse`'s `latency-hacks` suboption is now off by default, as it causes
-  issues with newer pulseaudio releases. If A/V Sync issues happen, either
-  add the `latency-hacks=on` pulse suboption in mpv.conf, or update the
-  pulseaudio daemon (bug #1430).
-- (Linux) `ao_alsa` now deals better with audio device disconnection.
-- Client API: timestamp properties that have no value return "no" instead of
-  magic number.
-
-Bug fixes
----------
-
-- (Windows, OSX) Many `ao_wasapi` and `ao_coreaudio` fixes.
-- (OSX) Make the window title update correctly on OS X Yosemite.
-- Fixes for handling mono audio on various AOs.
-- (Linux) Fixes resuming from suspend on ao_alsa.
-- Fixes for playlist file parsing.
-- Overly long options in the --list-options output now break the column layout
-  instead of getting truncated; fixes shell completion scripts.
-- Fix rendering resolution of certain DVB subtitles (bug #1425).
-- Fix EDL or --merge-files breaking timestamps with .avi files.
-- Workaround for libavcodec bug with the VP9 codec parser (bug #1448),
-  fixes possible crash.
-- Improve robustness of the matroska parser with broken files (bugs #1457,
-  #1461).
-- Improve 10bit video compatibility with older GPUs (specially Intel) (bug
-  #1383).
-- Fixes flashing the VO window when playing a list of files that includes
-  non-media files (bug #1459).
-- Workaround for window embedding in OpenBox (bug #1235)
-- Fixes for several crashes and lockups (bugs #838, #1389, #1408, #1463, #1473,
-  #1474).
-- Fixes for the zsh completion script (bugs #997)
-
-New features
-------------
-
-- mpv now prints the contents of its config.h when running in verbose mode, to
-  aid with debugging.
-- The `include=` option now accepts `~` to refer to the HOME dir (bug #1406).
-- `af_volume` now prints ReplayGain values in verbose mode.
-- m3u playlists that don't have the `#EXTM3U` header are now accepted if they
-  "look like" ASCII or UTF-8 text and have the .m3u extension.
-- Chapter marks in the seek bar now update when switching files.
-- Supports embedded cover art in MKV files (bug #1374).
-- (Windows) Video window can now be resized even with --no-border.
-- (Windows) Client API: the "run" command now works on Windows too.
-- (Linux) vo_wayland now supports key modifiers (Meta, Alt, Control, Shift)
-- Client API: "display-names" property is now observable.
-
-This listing is not complete. A complete changelog can be seen by running
-`git log v0.7.2..v0.7.3` in the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/v0.7.2...v0.7.3.
-
-Release 0.7.2
-=============
-
-Changes
--------
-
-- Give precedence to the DVD menu navigation keyboard bindings so that user
-  defined LEFT/RIGHT/... bindings don't break DVD menu navigation.
-- Try to fallback to the "default" device if the selected device is busy in the
-  alsa AO.
-- Don't create Dock icon for audio only files on OS X.
-- Save screenshots to desktop when using the app bundle on OS X.
-- Restore ab-loop settings with playback resume.
-- Bump required youtube-dl version to 2014.11.26 and enable the ytdl_hook Lua
-  script by default (now playing videos from YouTube and the like will work
-  out of the box without any configuration change needed).
-
-Bug fixes
----------
-
-- Don't signal an error if --stream-dump is used.
-- Fix removing key bindings from Lua scripts.
-- Reject channel descriptions with too many channels in the coreaudio AO.
-- Don't async redraw when waiting for VO redraw on OS X (this fixes the very
-  annoying glitch where the black bars disappear for a single frame when going
-  fullscreen).
-- Fix mono playback with the also AO.
-- Don't crash if framebuffers are not available in the opengl VO.
-
-New features
-------------
-
-- Try to handle multi-arc videos in the ytdl_hook Lua script.
-
-This listing is not complete. There are many more bug fixes and changes. The
-complete change log can be viewed by running ``git log 47ec404..c7d6b21`` in
-the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/47ec404...c7d6b21.
-
-Release 0.7.1
-=============
-
-Changes
--------
-
-- Don't show the volume neutral marker on the OSD if softvol is disabled.
-- Don't select a subtitle track when executing the sub_add input command in
-  "auto" mode.
-
-Bug fixes
----------
-
-- Fix busy loop when seeking while paused (this fixes a problem with pulseaudio
-  that caused mpv and the pulseaudio daemon to use 100% CPU).
-- Fix Lua function utils.subprocess() in Windows versions older than Vista.
-- Avoid creating a window bigger than the screen on Windows.
-- Don't ignore the last line in m3u playlists.
-- Don't crash if a codec could not be opened.
-- Dynamically allocate audio channel map entries (this should fix a crash in
-  the alsa and coreaudio AOs with audio devices that support more than 20
-  channel maps).
-- Ignore the "srgb" option in the opengl VO if hardware decoding is enabled.
-- Linearize non-RGB sRGB files correctly (eg. JPEG).
-- Fix opening reference URLs (.file/id=) on OS X.
-
-This listing is not complete. There are many more bug fixes and changes. The
-complete change log can be viewed by running ``git log 8d8b36d..6583ad6`` in
-the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/8d8b36d...6583ad6.
-
-Release 0.7.0
-=============
-
-Changes
--------
-
-- Buffer partial log messages in the client API (the client API will now only
-  pass full log messages to clients).
-- Remove ncurses/terminfo/termcap support (it was disabled by default and
-  replaced by new code since v0.6.0).
-- Enable cdda:// support by default again (it was disabled since v0.6.0).
-- Cascade-load input.conf (if there are several input.confs in the set of valid
-  config paths, load them all).
-- Draw the OSD twice in 3D mode (this fixes subtitles display in 3D mode).
-- Make wasapi the default AO on Windows again since many of its problems have
-  been solved.
-- Use "site-functions" subdir to install the zsh completion script instead of
-  the Debian-specific "vendor-completions" (also provide the --zshdir waf
-  configure option for changing this value).
-- Improve synchronization between the Cocoa GUI and the player (this fixes some
-  long standing deadlock issues on Mac OS X).
-- Remove --fs-missioncontrol option (only relevant to Mac OS X).
-
-New features
-------------
-
-- Enable pitch correction by default when playing at higher speeds (this can be
-  controlled with the --audio-pitch-correction option).
-- Open stream and demuxer asynchronously (this should avoid having the player
-  get blocked on network streams).
-- Add cache-buffering-state property for querying the cache fill status until
-  the player unpauses.
-- Add support for listing and selecting the audio device (note that it is not
-  implemented for all AOs, see the --audio-device option for more information).
-- Add support for a JSON-based IPC mechanism (note that this is not currently
-  supported on Windows, see the JSON IPC section in the manpage for more
-  information).
-- Add Lua utility function for starting processes (see utils.subprocess() in the
-  manpage).
-- Add Lua utility function for parsing JSON (see utils.parse_json() in the
-  manpage).
-- Add field-dominance property (see --field-dominance option).
-- Add video-rotate property (see --video-rotate option).
-- Add playback-abort property for querying whether playback is stopped or is to
-  be stopped.
-- Add cursor-autohide property (see --cursor-autohide option).
-- Add vo-configured property for querying whether a window is created.
-- Add support for dxva2 hardware acceleration on Windows.
-- Drop libquvi support (this has been replaced by a built-in Lua script that
-  invokes the youtube-dl tool, which needs to be installed, see the --ytdl
-  option).
-- Add support for loading chapters from an external file (see the
-  --chapters-file option).
-- Add window-minimized property for querying whether the window is minimized
-  (works for X11 only).
-- Make it possible to configure the OSC seekbar style (see the "seekbarstyle"
-  OSC option).
-- Add support for libmpv on Mac OS X (it used to be broken, now it's fully
-  functional and also provides support for embedding the mpv window inside a
-  Cocoa/Qt application).
-- Try to use the audio channel map reported by ALSA in the alsa AO.
-- Add option to disable text OSD rendering completely (useful for working around
-  certain fontconfig issues, see the --use-text-osd option).
-
-This listing is not complete. There are many more bug fixes and changes. The
-complete change log can be viewed by running ``git log 7759c18..9479daa`` in
-the git repository or by visiting
-https://github.com/mpv-player/mpv/compare/7759c18...9479daa (only 250 commits).
+`git log v0.7.0..v0.8.0` in the git repository.

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'
 
@@ -29,51 +18,20 @@ function add_files_at(index, files)
     end
 end
 
-function get_extension(path)
-    return string.match(path, "%.([^%.]+)$" )
-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
@@ -87,7 +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
@@ -96,7 +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.
@@ -104,7 +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)
@@ -116,7 +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/osxbundle.py

@@ -40,7 +40,7 @@ def apply_plist_template(plist_file, version):
         print (line.rstrip().replace('${VERSION}', version))
 
 def main():
-    version = sh("./version.sh").strip()
+    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

@@ -5,10 +5,7 @@ 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.
@@ -29,8 +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')
 
 """
@@ -94,15 +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

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)
@@ -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 @@
-0.11.0
+0.8.3

audio/audio.c

@@ -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;
 }

audio/audio.h

@@ -35,17 +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))
 
-    // --- 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/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"},
@@ -84,11 +78,9 @@ 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"},
@@ -100,7 +92,7 @@ static const char *const std_layout_names[][2] = {
     {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
@@ -133,8 +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;
 }
@@ -149,10 +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.
@@ -176,6 +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;
@@ -195,36 +197,16 @@ 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
@@ -243,8 +225,6 @@ void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels)
 // Set *dst to an unknown layout for the given numbers of channels.
 // If the number of channels is invalid, an invalid map is set, and
 // mp_chmap_is_valid(dst) will return false.
-// A mp_chmap with all entries set to NA is treated specially in some
-// contexts (watch out for mp_chmap_is_unknown()).
 void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels)
 {
     if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
@@ -252,7 +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;
     }
 }
 
@@ -295,7 +275,6 @@ void mp_chmap_remove_useless_channels(struct mp_chmap *map,
 }
 
 // Return the ffmpeg/libav channel layout as in <libavutil/channel_layout.h>.
-// Speakers not representable by ffmpeg/libav are dropped.
 // Warning: this ignores the order of the channels, and will return a channel
 //          mask even if the order is different from libavcodec's.
 uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src)
@@ -305,10 +284,8 @@ uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src)
     if (mp_chmap_is_unknown(&t))
         mp_chmap_from_channels(&t, t.num);
     uint64_t mask = 0;
-    for (int n = 0; n < t.num; n++) {
-        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;
 }
 
@@ -369,43 +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
@@ -508,25 +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

@@ -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.
@@ -100,11 +98,10 @@ 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);
@@ -120,17 +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

@@ -58,29 +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)
@@ -199,10 +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];
@@ -212,6 +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))
@@ -223,95 +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;
-
-    int old_lost = mp_chmap_diffn(req, old);
-    int new_lost = mp_chmap_diffn(req, new);
-
-    // If the situation is equal with replaced speakers, but one of them loses
-    // less if no replacements are performed, pick the better one, even if it
-    // means an upmix. This prefers exact supersets over inexact equivalents.
-    if (new_lost != old_lost)
-        return new_lost < old_lost;
-
-    struct mp_chmap old_p = *old, new_p = *new;
-    mp_chmap_remove_na(&old_p);
-    mp_chmap_remove_na(&new_p);
-
-    // Some kind of upmix. If it's perfect, prefer the smaller one. Even if not,
-    // both have equal loss, so also prefer the smaller one.
-    // Drop padding channels (NA) for the sake of this check, as the number of
-    // padding channels isn't really meaningful.
-    if (new_p.num != old_p.num)
-        return new_p.num < old_p.num;
-
-    // Again, with physical channels (minimizes number of NA channels).
-    return new->num < old->num;
-}
-
-// Determine which channel map to fallback to given a source channel map.
+// 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];
@@ -319,13 +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;

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>
@@ -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,
     },
@@ -116,7 +117,7 @@ static int init(struct dec_audio *da, const char *decoder)
 
     mp_set_avopts(da->log, lavc_context, opts->avopts);
 
-    lavc_context->codec_tag = sh->codec_tag;
+    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;
@@ -126,7 +127,11 @@ static int init(struct dec_audio *da, const char *decoder)
         lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
 
     // demux_mkv
-    mp_lavc_set_extradata(lavc_context, sh->extradata, sh->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 (sh->lav_headers)
         mp_copy_lav_codec_headers(lavc_context, sh->lav_headers);
@@ -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;
 }
 

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;
 
@@ -260,13 +204,9 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
         da->pts        = mpkt->pts;
         da->pts_offset = 0;
     }
-    if (!spdif_ctx->lavf_ctx) {
-        if (init_filter(da, &pkt) < 0)
-            return AD_ERR;
-    }
-    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;
 
@@ -304,8 +244,6 @@ 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

@@ -1,18 +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>
@@ -24,6 +25,7 @@
 
 #include "demux/codec_tags.h"
 
+#include "config.h"
 #include "common/codecs.h"
 #include "common/msg.h"
 #include "misc/bstr.h"
@@ -41,18 +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);
@@ -60,8 +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)
@@ -84,21 +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;
-
     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;
 }
@@ -112,13 +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);
 
@@ -157,8 +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);
 }
@@ -170,9 +164,6 @@ static int decode_new_frame(struct dec_audio *da)
         if (ret < 0)
             return ret;
 
-        if (da->pts == MP_NOPTS_VALUE && da->header->missing_timestamps)
-            da->pts = 0;
-
         if (da->waiting) {
             da->pts_offset += da->waiting->samples;
             da->decode_format = *da->waiting;

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;
     const struct ad_functions *ad_driver;
     struct sh_stream *header;
     struct af_stream *afilter;
@@ -58,7 +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)
@@ -447,23 +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;
-                    break;
-                }
-            }
             goto negotiate_error;
         }
         case AF_DETACH: { // Filter is redundant and wants to be unloaded
@@ -478,8 +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
@@ -594,14 +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))
@@ -610,14 +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)
@@ -682,18 +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)
@@ -723,16 +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)
@@ -772,7 +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>
@@ -74,7 +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);
@@ -155,13 +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;
@@ -172,7 +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;
 }
 
@@ -184,27 +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();
@@ -268,7 +257,6 @@ static int filter_out(struct af_instance *af)
     swap_16((uint16_t *)(buf + header_len), s->pkt.size / 2);
     out->samples = frame_size / out->sstride;
     af_add_output_frame(af, out);
-    update_delay(af);
     return 0;
 }
 

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;
@@ -218,43 +203,16 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
         return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
     }
-    case AF_CONTROL_GET_METADATA:
-        if (p->metadata) {
-            *(struct mp_tags *)arg = *p->metadata;
-            return CONTROL_OK;
-        }
-        return CONTROL_NA;
-    case AF_CONTROL_RESET:
-        reset(af);
-        return AF_OK;
     }
     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)
@@ -297,9 +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;
@@ -307,10 +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;
     }
 
@@ -329,7 +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;

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
@@ -58,33 +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;
@@ -92,151 +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;
@@ -251,103 +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.
-    int fake_out_ch_layout = av_get_default_channel_layout(map_out.num);
-    if (!fake_out_ch_layout)
-        goto error;
+    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);
 
-    // Deplanarize if needed.
-    av_opt_set_int(s->avrctx_out, "in_channel_layout",  fake_out_ch_layout, 0);
-    av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
-    av_opt_set_int(s->avrctx_out, "in_sample_fmt",      out_samplefmtp, 0);
+    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)
@@ -358,23 +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: {
@@ -382,94 +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;
-        int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
-        if (new_rate != s->in_rate && s->avrctx && af->fmt_out.format) {
-            // Before reconfiguring, drain the audio that is still buffered
-            // in the resampler.
-            af->filter_frame(af, NULL);
-            // Reinitialize resampler.
-            configure_lavrr(af, &af->fmt_in, &af->fmt_out, false);
-        }
-        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 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 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);
@@ -477,15 +344,11 @@ static int filter(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;
@@ -498,17 +361,12 @@ static int filter(struct af_instance *af, struct mp_audio *in)
         }
     }
 
-    extra_output_conversion(af, out);
-
     talloc_free(in);
     if (out->samples) {
         af_add_output_frame(af, out);
     } else {
         talloc_free(out);
     }
-
-    af->delay = get_delay(s);
-
     return 0;
 error:
     talloc_free(in);
@@ -527,9 +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
@@ -538,15 +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[]) {
@@ -555,7 +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}
-    },
-};

audio/filter/af_scaletempo.c

@@ -14,20 +14,21 @@
  *
  * Copyright (c) 2007 Robert Juliano
  *
- * 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 <stdlib.h>
@@ -75,9 +76,9 @@ typedef struct af_scaletempo_s
     float ms_stride;
     float percent_overlap;
     float ms_search;
-#define SCALE_TEMPO 1
-#define SCALE_PITCH 2
     int speed_opt;
+    short speed_tempo;
+    short speed_pitch;
 } af_scaletempo_t;
 
 static int fill_queue(struct af_instance *af, struct mp_audio *data, int offset)
@@ -275,22 +276,6 @@ static int filter(struct af_instance *af, struct mp_audio *data)
     return 0;
 }
 
-static void update_speed(struct af_instance *af, float speed)
-{
-    af_scaletempo_t *s = af->priv;
-
-    s->speed = speed;
-
-    double factor = (s->speed_opt & SCALE_PITCH) ? 1.0 / s->speed : s->speed;
-    s->scale = factor * s->scale_nominal;
-
-    s->frames_stride_scaled = s->scale * s->frames_stride;
-    s->frames_stride_error = MPMIN(s->frames_stride_error, s->frames_stride_scaled);
-
-    MP_VERBOSE(af, "%.3f speed * %.3f scale_nominal = %.3f\n",
-               s->speed, s->scale_nominal, s->scale);
-}
-
 // Initialization and runtime control
 static int control(struct af_instance *af, int cmd, void *arg)
 {
@@ -302,9 +287,19 @@ static int control(struct af_instance *af, int cmd, void *arg)
         int nch = data->nch;
         int use_int = 0;
 
+        MP_VERBOSE(af, "%.3f speed * %.3f scale_nominal = %.3f\n",
+               s->speed, s->scale_nominal, s->scale);
+
         mp_audio_force_interleaved_format(data);
         mp_audio_copy_config(af->data, data);
 
+        if (s->scale == 1.0) {
+            if (s->speed_tempo && s->speed_pitch)
+                return AF_DETACH;
+            af->delay = 0;
+            return af_test_output(af, data);
+        }
+
         if (data->format == AF_FORMAT_S16) {
             use_int = 1;
         } else {
@@ -314,10 +309,10 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
         s->frames_stride        = srate * s->ms_stride;
         s->bytes_stride         = s->frames_stride * bps * nch;
+        s->frames_stride_scaled = s->scale * s->frames_stride;
+        s->frames_stride_error  = 0;
         af->delay = 0;
 
-        update_speed(af, s->speed);
-
         int frames_overlap = s->frames_stride * s->percent_overlap;
         if (frames_overlap <= 0) {
             s->bytes_standing   = s->bytes_stride;
@@ -427,14 +422,17 @@ static int control(struct af_instance *af, int cmd, void *arg)
         return af_test_output(af, (struct mp_audio *)arg);
     }
     case AF_CONTROL_SET_PLAYBACK_SPEED: {
-        double speed = *(double *)arg;
-        if (s->speed_opt & SCALE_TEMPO) {
-            if (s->speed_opt & SCALE_PITCH)
+        if (s->speed_tempo) {
+            if (s->speed_pitch)
                 break;
-            update_speed(af, speed);
-        } else if (s->speed_opt & SCALE_PITCH) {
-            update_speed(af, speed);
-            break; // do not signal OK
+            s->speed = *(double *)arg;
+            s->scale = s->speed * s->scale_nominal;
+        } else {
+            if (s->speed_pitch) {
+                s->speed = 1 / *(double *)arg;
+                s->scale = s->speed * s->scale_nominal;
+                break;
+            }
         }
         return AF_OK;
     }
@@ -458,12 +456,22 @@ static void uninit(struct af_instance *af)
     free(s->table_window);
 }
 
+#define SCALE_TEMPO 1
+#define SCALE_PITCH 2
+
 // Allocate memory and set function pointers
 static int af_open(struct af_instance *af)
 {
+    af_scaletempo_t *s = af->priv;
+
     af->control   = control;
     af->uninit    = uninit;
     af->filter_frame = filter;
+
+    s->speed_tempo = !!(s->speed_opt & SCALE_TEMPO);
+    s->speed_pitch = !!(s->speed_opt & SCALE_PITCH);
+
+    s->scale = s->speed * s->scale_nominal;
     return AF_OK;
 }
 

audio/filter/af_sinesuppress.c

@@ -0,0 +1,118 @@
+/*
+ * Copyright (C) 2006 Michael Niedermayer
+ * Copyright (C) 2004 Alex Beregszaszi
+ * based upon af_extrastereo.c by 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 "af.h"
+
+// Data for specific instances of this filter
+typedef struct af_sinesuppress_s
+{
+    double freq;
+    double decay;
+    double real;
+    double imag;
+    double ref;
+    double pos;
+}af_sinesuppress_t;
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+  switch(cmd){
+  case AF_CONTROL_REINIT:{
+    // Sanity check
+    if(!arg) return AF_ERROR;
+
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_num_channels(af->data, 1);
+    mp_audio_set_format(af->data, AF_FORMAT_S16);
+
+    return af_test_output(af,(struct mp_audio*)arg);
+  }
+  }
+  return AF_UNKNOWN;
+}
+
+static int play_s16(struct af_instance *af, struct mp_audio *data)
+{
+  if (!data)
+    return 0;
+  if (af_make_writeable(af, data) < 0) {
+    talloc_free(data);
+    return -1;
+  }
+
+  af_sinesuppress_t *s = af->priv;
+  register int i = 0;
+  int16_t *a = (int16_t*)data->planes[0];       // Audio data
+  int len = data->samples*data->nch;            // Number of samples
+
+  for (i = 0; i < len; i++)
+  {
+    double co= cos(s->pos);
+    double si= sin(s->pos);
+
+    s->real += co * a[i];
+    s->imag += si * a[i];
+    s->ref  += co * co;
+
+    a[i] -= (s->real * co + s->imag * si) / s->ref;
+
+    s->real -= s->real * s->decay;
+    s->imag -= s->imag * s->decay;
+    s->ref  -= s->ref  * s->decay;
+
+    s->pos += 2 * M_PI * s->freq / data->rate;
+  }
+
+   MP_VERBOSE(af, "f:%8.2f: amp:%8.2f\n", s->freq, sqrt(s->real*s->real + s->imag*s->imag) / s->ref);
+
+  af_add_output_frame(af, data);
+  return 0;
+}
+
+// Allocate memory and set function pointers
+static int af_open(struct af_instance* af){
+  af->control=control;
+  af->filter_frame = play_s16;
+  return AF_OK;
+}
+
+#define OPT_BASE_STRUCT af_sinesuppress_t
+const struct af_info af_info_sinesuppress = {
+    .info = "Sine Suppress",
+    .name = "sinesuppress",
+    .open = af_open,
+    .priv_size = sizeof(af_sinesuppress_t),
+    .options = (const struct m_option[]) {
+        OPT_DOUBLE("freq", freq, 0, OPTDEF_DOUBLE(50.0)),
+        OPT_DOUBLE("decay", decay, 0, OPTDEF_DOUBLE(0.0001)),
+        {0}
+    },
+};

audio/filter/af_sub.c

@@ -0,0 +1,149 @@
+/*
+ * Copyright (C) 2002 Anders Johansson ajh@watri.uwa.edu.au
+ *
+ * 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.
+ */
+
+/* This filter adds a sub-woofer channels to the audio stream by
+   averaging the left and right channel and low-pass filter them. The
+   low-pass filter is implemented as a 4th order IIR Butterworth
+   filter, with a variable cutoff frequency between 10 and 300 Hz. The
+   filter gives 24dB/octave attenuation. There are two runtime
+   controls one for setting which channel to insert the sub-audio into
+   called AF_CONTROL_SUB_CH and one for setting the cutoff frequency
+   called AF_CONTROL_SUB_FC.
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "common/common.h"
+#include "af.h"
+#include "dsp.h"
+
+// Q value for low-pass filter
+#define Q 1.0
+
+// Analog domain biquad section
+typedef struct{
+  float a[3];           // Numerator coefficients
+  float b[3];           // Denominator coefficients
+} biquad_t;
+
+// S-parameters for designing 4th order Butterworth filter
+static const biquad_t sp[2] = {{{1.0,0.0,0.0},{1.0,0.765367,1.0}},
+                               {{1.0,0.0,0.0},{1.0,1.847759,1.0}}};
+
+// Data for specific instances of this filter
+typedef struct af_sub_s
+{
+  float w[2][4];        // Filter taps for low-pass filter
+  float q[2][2];        // Circular queues
+  float fc;             // Cutoff frequency [Hz] for low-pass filter
+  float k;              // Filter gain;
+  int ch;               // Channel number which to insert the filtered data
+
+}af_sub_t;
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+  af_sub_t* s   = af->priv;
+
+  switch(cmd){
+  case AF_CONTROL_REINIT:{
+    // Sanity check
+    if(!arg) return AF_ERROR;
+
+    af->data->rate   = ((struct mp_audio*)arg)->rate;
+    mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
+
+    // Design low-pass filter
+    s->k = 1.0;
+    if((-1 == af_filter_szxform(sp[0].a, sp[0].b, Q, s->fc,
+       (float)af->data->rate, &s->k, s->w[0])) ||
+       (-1 == af_filter_szxform(sp[1].a, sp[1].b, Q, s->fc,
+       (float)af->data->rate, &s->k, s->w[1])))
+      return AF_ERROR;
+    return af_test_output(af,(struct mp_audio*)arg);
+  }
+  }
+  return AF_UNKNOWN;
+}
+
+#ifndef IIR
+#define IIR(in,w,q,out) { \
+  float h0 = (q)[0]; \
+  float h1 = (q)[1]; \
+  float hn = (in) - h0 * (w)[0] - h1 * (w)[1];  \
+  out = hn + h0 * (w)[2] + h1 * (w)[3];  \
+  (q)[1] = h0; \
+  (q)[0] = hn; \
+}
+#endif
+
+static int filter_frame(struct af_instance *af, struct mp_audio *data)
+{
+  if (!data)
+    return 0;
+  if (af_make_writeable(af, data) < 0) {
+    talloc_free(data);
+    return -1;
+  }
+
+  struct mp_audio*    c   = data;        // Current working data
+  af_sub_t*     s   = af->priv; // Setup for this instance
+  float*        a   = c->planes[0];      // Audio data
+  int           len = c->samples*c->nch;         // Number of samples in current audio block
+  int           nch = c->nch;    // Number of channels
+  int           ch  = s->ch;     // Channel in which to insert the sub audio
+  register int  i;
+
+  // Run filter
+  for(i=0;i<len;i+=nch){
+    // Average left and right
+    register float x = 0.5 * (a[i] + a[i+1]);
+    IIR(x * s->k, s->w[0], s->q[0], x);
+    IIR(x , s->w[1], s->q[1], a[i+ch]);
+  }
+
+  af_add_output_frame(af, data);
+  return 0;
+}
+
+// Allocate memory and set function pointers
+static int af_open(struct af_instance* af){
+  af->control=control;
+  af->filter_frame = filter_frame;
+  return AF_OK;
+}
+
+#define OPT_BASE_STRUCT af_sub_t
+const struct af_info af_info_sub = {
+    .info = "Audio filter for adding a sub-base channel",
+    .name = "sub",
+    .flags = AF_FLAGS_NOT_REENTRANT,
+    .open = af_open,
+    .priv_size = sizeof(af_sub_t),
+    .options = (const struct m_option[]) {
+        OPT_FLOATRANGE("fc", fc, 0, 20, 300, OPTDEF_FLOAT(60.0)),
+        OPT_INTRANGE("ch", ch, 0, 0, AF_NCH - 1, OPTDEF_INT(5)),
+        {0}
+    },
+};

audio/filter/af_surround.c

@@ -0,0 +1,247 @@
+/*
+ * Filter to do simple decoding of matrixed surround sound.
+ * This will provide a (basic) surround-sound effect from
+ * audio encoded for Dolby Surround, Pro Logic etc.
+ *
+ * original author: Steve Davies <steve@daviesfam.org>
+ *
+ * This file is part of 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.
+ */
+
+/* The principle:  Make rear channels by extracting anti-phase data
+   from the front channels, delay by 20ms and feed to rear in anti-phase
+*/
+
+
+/* SPLITREAR: Define to decode two distinct rear channels - this
+     doesn't work so well in practice because separation in a passive
+     matrix is not high. C (dialogue) to Ls and Rs 14dB or so - so
+     dialogue leaks to the rear.  Still - give it a try and send
+     feedback. Comment this define for old behavior of a single
+     surround sent to rear in anti-phase */
+#define SPLITREAR 1
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "af.h"
+#include "dsp.h"
+
+#define L  32    // Length of fir filter
+#define LD 65536 // Length of delay buffer
+
+// 32 Tap fir filter loop unrolled
+#define FIR(x,w,y) \
+  y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3]  \
+      + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7]  \
+      + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
+      + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \
+      + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \
+      + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \
+      + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \
+      + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31])
+
+// Add to circular queue macro + update index
+#ifdef SPLITREAR
+#define ADDQUE(qi,rq,lq,r,l)\
+  lq[qi]=lq[qi+L]=(l);\
+  rq[qi]=rq[qi+L]=(r);\
+  qi=(qi-1)&(L-1);
+#else
+#define ADDQUE(qi,lq,l)\
+  lq[qi]=lq[qi+L]=(l);\
+  qi=(qi-1)&(L-1);
+#endif
+
+// Macro for updating queue index in delay queues
+#define UPDATEQI(qi) qi=(qi+1)&(LD-1)
+
+// instance data
+typedef struct af_surround_s
+{
+  float lq[2*L]; // Circular queue for filtering left rear channel
+  float rq[2*L]; // Circular queue for filtering right rear channel
+  float w[L];    // FIR filter coefficients for surround sound 7kHz low-pass
+  float* dr;     // Delay queue right rear channel
+  float* dl;     // Delay queue left rear channel
+  float  d;      // Delay time
+  int i;         // Position in circular buffer
+  int wi;        // Write index for delay queue
+  int ri;        // Read index for delay queue
+}af_surround_t;
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+  af_surround_t *s = af->priv;
+  switch(cmd){
+  case AF_CONTROL_REINIT:{
+    struct mp_audio *in = arg;
+    float fc;
+    if (!mp_chmap_is_stereo(&in->channels)) {
+        MP_ERR(af, "Only stereo input is supported.\n");
+        return AF_DETACH;
+    }
+
+    mp_audio_set_format(in, AF_FORMAT_FLOAT);
+    mp_audio_copy_config(af->data, in);
+    mp_audio_set_channels_old(af->data, in->nch * 2);
+
+    // Surround filer coefficients
+    fc = 2.0 * 7000.0/(float)af->data->rate;
+    if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
+      MP_ERR(af, "Unable to design low-pass filter.\n");
+      return AF_ERROR;
+    }
+
+    // Free previous delay queues
+    free(s->dl);
+    free(s->dr);
+    // Allocate new delay queues
+    s->dl = calloc(LD,af->data->bps);
+    s->dr = calloc(LD,af->data->bps);
+    if((NULL == s->dl) || (NULL == s->dr))
+      MP_FATAL(af, "Out of memory\n");
+
+    // Initialize delay queue index
+    if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
+      return AF_ERROR;
+//    printf("%i\n",s->wi);
+    s->ri = 0;
+
+    return AF_OK;
+  }
+  }
+  return AF_UNKNOWN;
+}
+
+// The beginnings of an active matrix...
+static const float steering_matrix[][12] = {
+//      LL      RL      LR      RR      LS      RS
+//      LLs     RLs     LRs     RRs     LC      RC
+       {.707,   .0,     .0,     .707,   .5,     -.5,
+        .5878,  -.3928, .3928,  -.5878, .5,     .5},
+};
+
+// Experimental moving average dominance
+//static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
+
+static int filter_frame(struct af_instance *af, struct mp_audio *data)
+{
+  if (!data)
+    return 0;
+  struct mp_audio *outframe =
+    mp_audio_pool_get(af->out_pool, &af->fmt_out, data->samples);
+  if (!outframe) {
+    talloc_free(data);
+    return -1;
+  }
+  mp_audio_copy_attributes(outframe, data);
+
+  af_surround_t* s   = (af_surround_t*)af->priv;
+  const float*   m   = steering_matrix[0];
+  float*         in  = data->planes[0];         // Input audio data
+  float*         out = outframe->planes[0];     // Output audio data
+  float*         end = in + data->samples * data->nch;
+  int            i   = s->i;    // Filter queue index
+  int            ri  = s->ri;   // Read index for delay queue
+  int            wi  = s->wi;   // Write index for delay queue
+
+  while(in < end){
+    /* Dominance:
+       abs(in[0])  abs(in[1]);
+       abs(in[0]+in[1])  abs(in[0]-in[1]);
+       10 * log( abs(in[0]) / (abs(in[1])|1) );
+       10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */
+
+    /* About volume balancing...
+       Surround encoding does the following:
+           Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
+       So S should be extracted as:
+           (Lt-Rt)
+       But we are splitting the S to two output channels, so we
+       must take 3dB off as we split it:
+           Ls=Rs=.707*(Lt-Rt)
+       Trouble is, Lt could be +1, Rt -1, so possibility that S will
+       overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by
+       6dB (/2). This keeps the overall balance, but guarantees no
+       overflow. */
+
+    // Output front left and right
+    out[0] = m[0]*in[0] + m[1]*in[1];
+    out[1] = m[2]*in[0] + m[3]*in[1];
+
+    // Low-pass output @ 7kHz
+    FIR((&s->lq[i]), s->w, s->dl[wi]);
+
+    // Delay output by d ms
+    out[2] = s->dl[ri];
+
+#ifdef SPLITREAR
+    // Low-pass output @ 7kHz
+    FIR((&s->rq[i]), s->w, s->dr[wi]);
+
+    // Delay output by d ms
+    out[3] = s->dr[ri];
+#else
+    out[3] = -out[2];
+#endif
+
+    // Update delay queues indexes
+    UPDATEQI(ri);
+    UPDATEQI(wi);
+
+    // Calculate and save surround in circular queue
+#ifdef SPLITREAR
+    ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
+#else
+    ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
+#endif
+
+    // Next sample...
+    in = &in[data->nch];
+    out = &out[af->data->nch];
+  }
+
+  // Save indexes
+  s->i  = i; s->ri = ri; s->wi = wi;
+
+  talloc_free(data);
+  af_add_output_frame(af, outframe);
+  return 0;
+}
+
+static int af_open(struct af_instance* af){
+  af->control=control;
+  af->filter_frame = filter_frame;
+  return AF_OK;
+}
+
+#define OPT_BASE_STRUCT af_surround_t
+const struct af_info af_info_surround =
+{
+    .info = "Surround decoder filter",
+    .name = "surround",
+    .flags = AF_FLAGS_NOT_REENTRANT,
+    .open = af_open,
+    .priv_size = sizeof(af_surround_t),
+    .options = (const struct m_option[]) {
+        OPT_FLOATRANGE("d", d, 0, 0, 1000, OPTDEF_FLOAT(20.0)),
+        {0}
+    },
+};

audio/filter/af_sweep.c

@@ -0,0 +1,93 @@
+/*
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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 "config.h"
+#include "af.h"
+
+typedef struct af_sweep_s{
+    double x;
+    double delta;
+}af_sweept;
+
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+  struct mp_audio *data= (struct mp_audio*)arg;
+
+  switch(cmd){
+  case AF_CONTROL_REINIT:
+    mp_audio_copy_config(af->data, data);
+    mp_audio_set_format(af->data, AF_FORMAT_S16);
+
+    return af_test_output(af, data);
+  }
+  return AF_UNKNOWN;
+}
+
+static int filter_frame(struct af_instance *af, struct mp_audio *data)
+{
+  if (!data)
+    return 0;
+  if (af_make_writeable(af, data) < 0) {
+      talloc_free(data);
+    return 0;
+  }
+
+  af_sweept *s = af->priv;
+  int i, j;
+  int16_t *in = (int16_t*)data->planes[0];
+  int chans   = data->nch;
+  int in_len  = data->samples;
+
+  for(i=0; i<in_len; i++){
+      for(j=0; j<chans; j++)
+          in[i*chans+j]= 32000*sin(s->x*s->x);
+      s->x += s->delta;
+      if(2*s->x*s->delta >= 3.141592) s->x=0;
+  }
+
+  af_add_output_frame(af, data);
+  return 0;
+}
+
+static int af_open(struct af_instance* af){
+  af->control=control;
+  af->filter_frame = filter_frame;
+  return AF_OK;
+}
+
+#define OPT_BASE_STRUCT af_sweept
+const struct af_info af_info_sweep = {
+    .info = "sine sweep",
+    .name = "sweep",
+    .open = af_open,
+    .priv_size = sizeof(af_sweept),
+    .options = (const struct m_option[]) {
+        OPT_DOUBLE("delta", delta, 0),
+        {0}
+    },
+};

audio/filter/af_volume.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>
@@ -30,28 +31,18 @@
 #include "demux/demux.h"
 
 struct priv {
-    float vol;                  // User-specified non-linear volume
     float level;                // User-specified gain level for each channel
     float rgain;                // Replaygain level
     int rgain_track;            // Enable/disable track based replaygain
     int rgain_album;            // Enable/disable album based replaygain
     float rgain_preamp;         // Set replaygain pre-amplification
     int rgain_clip;             // Enable/disable clipping prevention
-    float replaygain_fallback;
     int soft;                   // Enable/disable soft clipping
     int fast;                   // Use fix-point volume control
     int detach;                 // Detach if gain volume is neutral
     float cfg_volume;
 };
 
-// Convert to gain value from dB. input <= -200dB will become 0 gain.
-static float from_dB(float in, float k, float mi, float ma)
-{
-    if (in <= -200)
-        return 0.0;
-    return pow(10.0, MPCLAMP(in, mi, ma) / k);
-}
-
 static int control(struct af_instance *af, int cmd, void *arg)
 {
     struct priv *s = af->priv;
@@ -83,7 +74,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
             }
 
             gain += s->rgain_preamp;
-            s->rgain = from_dB(gain, 20.0, -200.0, 60.0);
+            af_from_dB(1, &gain, &s->rgain, 20.0, -200.0, 60.0);
 
             MP_VERBOSE(af, "Applying replay-gain: %f\n", s->rgain);
 
@@ -91,21 +82,16 @@ static int control(struct af_instance *af, int cmd, void *arg)
                 s->rgain = MPMIN(s->rgain, 1.0 / peak);
                 MP_VERBOSE(af, "...with clipping prevention: %f\n", s->rgain);
             }
-        } else if (s->replaygain_fallback) {
-            s->rgain = from_dB(s->replaygain_fallback, 20.0, -200.0, 60.0);
-            MP_VERBOSE(af, "Applying fallback gain: %f\n", s->rgain);
         }
         if (s->detach && fabs(s->level * s->rgain - 1.0) < 0.00001)
             return AF_DETACH;
         return af_test_output(af, in);
     }
     case AF_CONTROL_SET_VOLUME:
-        s->vol = *(float *)arg;
-        s->level = pow(s->vol, 3);
-        MP_VERBOSE(af, "volume gain: %f\n", s->level);
+        s->level = *(float *)arg;
         return AF_OK;
     case AF_CONTROL_GET_VOLUME:
-        *(float *)arg = s->vol;
+        *(float *)arg = s->level;
         return AF_OK;
     }
     return AF_UNKNOWN;
@@ -158,7 +144,7 @@ static int af_open(struct af_instance *af)
     struct priv *s = af->priv;
     af->control = control;
     af->filter_frame = filter;
-    s->level = from_dB(s->cfg_volume, 20.0, -200.0, 60.0);
+    af_from_dB(1, &s->cfg_volume, &s->level, 20.0, -200.0, 60.0);
     return AF_OK;
 }
 
@@ -168,6 +154,7 @@ static int af_open(struct af_instance *af)
 const struct af_info af_info_volume = {
     .info = "Volume control audio filter",
     .name = "volume",
+    .flags = AF_FLAGS_NOT_REENTRANT,
     .open = af_open,
     .priv_size = sizeof(struct priv),
     .options = (const struct m_option[]) {
@@ -176,7 +163,6 @@ const struct af_info af_info_volume = {
         OPT_FLAG("replaygain-album", rgain_album, 0),
         OPT_FLOATRANGE("replaygain-preamp", rgain_preamp, 0, -15, 15),
         OPT_FLAG("replaygain-clip", rgain_clip, 0),
-        OPT_FLOATRANGE("replaygain-fallback", replaygain_fallback, 0, -200, 60),
         OPT_FLAG("softclip", soft, 0),
         OPT_FLAG("s16", fast, 0),
         OPT_FLAG("detach", detach, 0),

audio/filter/dsp.h

@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2002 Anders Johansson ajh@atri.curtin.edu.au
+ *
+ * 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_DSP_H
+#define MPLAYER_DSP_H
+
+/* Implementation of routines used for DSP */
+
+/* Size of floating point type used in routines */
+#define FLOAT_TYPE float
+
+#include "window.h"
+#include "filter.h"
+
+#endif /* MPLAYER_DSP_H */

audio/filter/equalizer.h

@@ -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.
  */
 
 #ifndef MPLAYER_EQUALIZER_H

audio/filter/filter.c

@@ -0,0 +1,360 @@
+/*
+ * design and implementation of different types of digital filters
+ *
+ * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
+ *
+ * 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 <string.h>
+#include <math.h>
+#include "dsp.h"
+
+/******************************************************************************
+*  FIR filter implementations
+******************************************************************************/
+
+/* C implementation of FIR filter y=w*x
+
+   n number of filter taps, where mod(n,4)==0
+   w filter taps
+   x input signal must be a circular buffer which is indexed backwards
+*/
+inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w,
+                                const FLOAT_TYPE* x)
+{
+  register FLOAT_TYPE y; // Output
+  y = 0.0;
+  do{
+    n--;
+    y+=w[n]*x[n];
+  }while(n != 0);
+  return y;
+}
+
+/******************************************************************************
+*  FIR filter design
+******************************************************************************/
+
+/* Design FIR filter using the Window method
+
+   n     filter length must be odd for HP and BS filters
+   w     buffer for the filter taps (must be n long)
+   fc    cutoff frequencies (1 for LP and HP, 2 for BP and BS)
+         0 < fc < 1 where 1 <=> Fs/2
+   flags window and filter type as defined in filter.h
+         variables are ored together: i.e. LP|HAMMING will give a
+         low pass filter designed using a hamming window
+   opt   beta constant used only when designing using kaiser windows
+
+   returns 0 if OK, -1 if fail
+*/
+int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
+                         unsigned int flags, FLOAT_TYPE opt)
+{
+  unsigned int  o   = n & 1;            // Indicator for odd filter length
+  unsigned int  end = ((n + 1) >> 1) - o;       // Loop end
+  unsigned int  i;                      // Loop index
+
+  FLOAT_TYPE k1 = 2 * M_PI;             // 2*pi*fc1
+  FLOAT_TYPE k2 = 0.5 * (FLOAT_TYPE)(1 - o);// Constant used if the filter has even length
+  FLOAT_TYPE k3;                        // 2*pi*fc2 Constant used in BP and BS design
+  FLOAT_TYPE g  = 0.0;                  // Gain
+  FLOAT_TYPE t1,t2,t3;                  // Temporary variables
+  FLOAT_TYPE fc1,fc2;                   // Cutoff frequencies
+
+  // Sanity check
+  if(!w || (n == 0)) return -1;
+
+  // Get window coefficients
+  switch(flags & WINDOW_MASK){
+  case(BOXCAR):
+    af_window_boxcar(n,w); break;
+  case(TRIANG):
+    af_window_triang(n,w); break;
+  case(HAMMING):
+    af_window_hamming(n,w); break;
+  case(HANNING):
+    af_window_hanning(n,w); break;
+  case(BLACKMAN):
+    af_window_blackman(n,w); break;
+  case(FLATTOP):
+    af_window_flattop(n,w); break;
+  case(KAISER):
+    af_window_kaiser(n,w,opt); break;
+  default:
+    return -1;
+  }
+
+  if(flags & (LP | HP)){
+    fc1=*fc;
+    // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2
+    fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
+    k1 *= fc1;
+
+    if(flags & LP){ // Low pass filter
+
+      // If the filter length is odd, there is one point which is exactly
+      // in the middle. The value at this point is 2*fCutoff*sin(x)/x,
+      // where x is zero. To make sure nothing strange happens, we set this
+      // value separately.
+      if (o){
+        w[end] = fc1 * w[end] * 2.0;
+        g=w[end];
+      }
+
+      // Create filter
+      for (i=0 ; i<end ; i++){
+        t1 = (FLOAT_TYPE)(i+1) - k2;
+        w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
+        g += 2*w[end-i-1]; // Total gain in filter
+      }
+    }
+    else{ // High pass filter
+      if (!o) // High pass filters must have odd length
+        return -1;
+      w[end] = 1.0 - (fc1 * w[end] * 2.0);
+      g= w[end];
+
+      // Create filter
+      for (i=0 ; i<end ; i++){
+        t1 = (FLOAT_TYPE)(i+1);
+        w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
+        g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter
+      }
+    }
+  }
+
+  if(flags & (BP | BS)){
+    fc1=fc[0];
+    fc2=fc[1];
+    // Cutoff frequencies must be < 1.0 where 1.0 <=> Fs/2
+    fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
+    fc2 = ((fc2 <= 1.0) && (fc2 > 0.0)) ? fc2/2 : 0.25;
+    k3  = k1 * fc2; // 2*pi*fc2
+    k1 *= fc1;      // 2*pi*fc1
+
+    if(flags & BP){ // Band pass
+      // Calculate center tap
+      if (o){
+        g=w[end]*(fc1+fc2);
+        w[end] = (fc2 - fc1) * w[end] * 2.0;
+      }
+
+      // Create filter
+      for (i=0 ; i<end ; i++){
+        t1 = (FLOAT_TYPE)(i+1) - k2;
+        t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
+        t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
+        g += w[end-i-1] * (t3 + t2);   // Total gain in filter
+        w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
+      }
+    }
+    else{ // Band stop
+      if (!o) // Band stop filters must have odd length
+        return -1;
+      w[end] = 1.0 - (fc2 - fc1) * w[end] * 2.0;
+      g= w[end];
+
+      // Create filter
+      for (i=0 ; i<end ; i++){
+        t1 = (FLOAT_TYPE)(i+1);
+        t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
+        t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
+        w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
+        g += 2*w[end-i-1]; // Total gain in filter
+      }
+    }
+  }
+
+  // Normalize gain
+  g=1/g;
+  for (i=0; i<n; i++)
+    w[i] *= g;
+
+  return 0;
+}
+
+/******************************************************************************
+*  IIR filter design
+******************************************************************************/
+
+/* Helper functions for the bilinear transform */
+
+/* Pre-warp the coefficients of a numerator or denominator.
+   Note that a0 is assumed to be 1, so there is no wrapping
+   of it.
+*/
+static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs)
+{
+  FLOAT_TYPE wp;
+  wp = 2.0 * fs * tan(M_PI * fc / fs);
+  a[2] = a[2]/(wp * wp);
+  a[1] = a[1]/wp;
+}
+
+/* Transform the numerator and denominator coefficients of s-domain
+   biquad section into corresponding z-domain coefficients.
+
+   The transfer function for z-domain is:
+
+          1 + alpha1 * z^(-1) + alpha2 * z^(-2)
+   H(z) = -------------------------------------
+          1 + beta1 * z^(-1) + beta2 * z^(-2)
+
+   Store the 4 IIR coefficients in array pointed by coef in following
+   order:
+   beta1, beta2    (denominator)
+   alpha1, alpha2  (numerator)
+
+   Arguments:
+   a       - s-domain numerator coefficients
+   b       - s-domain denominator coefficients
+   k       - filter gain factor. Initially set to 1 and modified by each
+             biquad section in such a way, as to make it the
+             coefficient by which to multiply the overall filter gain
+             in order to achieve a desired overall filter gain,
+             specified in initial value of k.
+   fs      - sampling rate (Hz)
+   coef    - array of z-domain coefficients to be filled in.
+
+   Return: On return, set coef z-domain coefficients and k to the gain
+   required to maintain overall gain = 1.0;
+*/
+static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE* k,
+                               FLOAT_TYPE fs, FLOAT_TYPE *coef)
+{
+  FLOAT_TYPE ad, bd;
+
+  /* alpha (Numerator in s-domain) */
+  ad = 4. * a[2] * fs * fs + 2. * a[1] * fs + a[0];
+  /* beta (Denominator in s-domain) */
+  bd = 4. * b[2] * fs * fs + 2. * b[1] * fs + b[0];
+
+  /* Update gain constant for this section */
+  *k *= ad/bd;
+
+  /* Denominator */
+  *coef++ = (2. * b[0] - 8. * b[2] * fs * fs)/bd; /* beta1 */
+  *coef++ = (4. * b[2] * fs * fs - 2. * b[1] * fs + b[0])/bd; /* beta2 */
+
+  /* Numerator */
+  *coef++ = (2. * a[0] - 8. * a[2] * fs * fs)/ad; /* alpha1 */
+  *coef   = (4. * a[2] * fs * fs - 2. * a[1] * fs + a[0])/ad;   /* alpha2 */
+}
+
+
+
+/* IIR filter design using bilinear transform and prewarp. Transforms
+   2nd order s domain analog filter into a digital IIR biquad link. To
+   create a filter fill in a, b, Q and fs and make space for coef and k.
+
+
+   Example Butterworth design:
+
+   Below are Butterworth polynomials, arranged as a series of 2nd
+   order sections:
+
+   Note: n is filter order.
+
+   n  Polynomials
+   -------------------------------------------------------------------
+   2  s^2 + 1.4142s + 1
+   4  (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1)
+   6  (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1)
+
+   For n=4 we have following equation for the filter transfer function:
+                       1                              1
+   T(s) = --------------------------- * ----------------------------
+          s^2 + (1/Q) * 0.765367s + 1   s^2 + (1/Q) * 1.847759s + 1
+
+   The filter consists of two 2nd order sections since highest s power
+   is 2.  Now we can take the coefficients, or the numbers by which s
+   is multiplied and plug them into a standard formula to be used by
+   bilinear transform.
+
+   Our standard form for each 2nd order section is:
+
+          a2 * s^2 + a1 * s + a0
+   H(s) = ----------------------
+          b2 * s^2 + b1 * s + b0
+
+   Note that Butterworth numerator is 1 for all filter sections, which
+   means s^2 = 0 and s^1 = 0
+
+   Let's convert standard Butterworth polynomials into this form:
+
+             0 + 0 + 1                  0 + 0 + 1
+   --------------------------- * --------------------------
+   1 + ((1/Q) * 0.765367) + 1   1 + ((1/Q) * 1.847759) + 1
+
+   Section 1:
+   a2 = 0; a1 = 0; a0 = 1;
+   b2 = 1; b1 = 0.765367; b0 = 1;
+
+   Section 2:
+   a2 = 0; a1 = 0; a0 = 1;
+   b2 = 1; b1 = 1.847759; b0 = 1;
+
+   Q is filter quality factor or resonance, in the range of 1 to
+   1000. The overall filter Q is a product of all 2nd order stages.
+   For example, the 6th order filter (3 stages, or biquads) with
+   individual Q of 2 will have filter Q = 2 * 2 * 2 = 8.
+
+
+   Arguments:
+   a       - s-domain numerator coefficients, a[1] is always assumed to be 1.0
+   b       - s-domain denominator coefficients
+   Q       - Q value for the filter
+   k       - filter gain factor. Initially set to 1 and modified by each
+             biquad section in such a way, as to make it the
+             coefficient by which to multiply the overall filter gain
+             in order to achieve a desired overall filter gain,
+             specified in initial value of k.
+   fs      - sampling rate (Hz)
+   coef    - array of z-domain coefficients to be filled in.
+
+   Note: Upon return from each call, the k argument will be set to a
+   value, by which to multiply our actual signal in order for the gain
+   to be one. On second call to szxform() we provide k that was
+   changed by the previous section. During actual audio filtering
+   k can be used for gain compensation.
+
+   return -1 if fail 0 if success.
+*/
+int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q, FLOAT_TYPE fc,
+                      FLOAT_TYPE fs, FLOAT_TYPE *k, FLOAT_TYPE *coef)
+{
+  FLOAT_TYPE at[3];
+  FLOAT_TYPE bt[3];
+
+  if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0))
+    return -1;
+
+  memcpy(at,a,3*sizeof(FLOAT_TYPE));
+  memcpy(bt,b,3*sizeof(FLOAT_TYPE));
+
+  bt[1]/=Q;
+
+  /* Calculate a and b and overwrite the original values */
+  af_filter_prewarp(at, fc, fs);
+  af_filter_prewarp(bt, fc, fs);
+  /* Execute bilinear transform */
+  af_filter_bilinear(at, bt, k, fs, coef);
+
+  return 0;
+}

audio/filter/filter.h

@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
+ *
+ * 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.
+ */
+
+#if !defined MPLAYER_DSP_H
+# error Never use filter.h directly; include dsp.h instead.
+#endif
+
+#ifndef MPLAYER_FILTER_H
+#define MPLAYER_FILTER_H
+
+
+// Design and implementation of different types of digital filters
+
+
+// Flags used for filter design
+
+// Filter characteristics
+#define LP          0x00010000 // Low pass
+#define HP          0x00020000 // High pass
+#define BP          0x00040000 // Band pass
+#define BS          0x00080000 // Band stop
+#define TYPE_MASK   0x000F0000
+
+// Window types
+#define BOXCAR      0x00000001
+#define TRIANG      0x00000002
+#define HAMMING     0x00000004
+#define HANNING     0x00000008
+#define BLACKMAN    0x00000010
+#define FLATTOP     0x00000011
+#define KAISER      0x00000012
+#define WINDOW_MASK 0x0000001F
+
+// Parallel filter design
+#define FWD         0x00000001 // Forward indexing of polyphase filter
+#define REW         0x00000002 // Reverse indexing of polyphase filter
+#define ODD         0x00000010 // Make filter HP
+
+// Exported functions
+FLOAT_TYPE af_filter_fir(unsigned int n, const FLOAT_TYPE* w, const FLOAT_TYPE* x);
+
+int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
+                         unsigned int flags, FLOAT_TYPE opt);
+
+int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q,
+                      FLOAT_TYPE fc, FLOAT_TYPE fs, FLOAT_TYPE *k,
+                      FLOAT_TYPE *coef);
+
+/* Add new data to circular queue designed to be used with a FIR
+   filter. xq is the circular queue, in pointing at the new sample, xi
+   current index for xq and n the length of the filter. xq must be n*2
+   long.
+*/
+#define af_filter_updateq(n,xi,xq,in)\
+  xq[xi]=(xq)[(xi)+(n)]=*(in);\
+  xi=(++(xi))&((n)-1);
+
+#endif /* MPLAYER_FILTER_H */

audio/filter/tools.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 <math.h>
@@ -21,6 +22,42 @@
 #include "common/common.h"
 #include "af.h"
 
+/* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
+ * fail. input <= -200dB will become 0 gain. */
+int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
+{
+  int i = 0;
+  // Sanity check
+  if(!in || !out)
+    return AF_ERROR;
+
+  for(i=0;i<n;i++){
+    if(in[i]<=-200)
+      out[i]=0.0;
+    else
+      out[i]=pow(10.0,MPCLAMP(in[i],mi,ma)/k);
+  }
+  return AF_OK;
+}
+
+/* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if
+ * fail. gain=0 will become -200 dB. k is just a multiplier. */
+int af_to_dB(int n, float* in, float* out, float k)
+{
+  int i = 0;
+  // Sanity check
+  if(!in || !out)
+    return AF_ERROR;
+
+  for(i=0;i<n;i++){
+    if(in[i] == 0.0)
+      out[i]=-200.0;
+    else
+      out[i]=k*log10(in[i]);
+  }
+  return AF_OK;
+}
+
 /* Convert from ms to sample time */
 int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
 {
@@ -35,6 +72,20 @@ int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
   return AF_OK;
 }
 
+/* Convert from sample time to ms */
+int af_to_ms(int n, int* in, float* out, int rate)
+{
+  int i = 0;
+  // Sanity check
+  if(!in || !out || !rate)
+    return AF_ERROR;
+
+  for(i=0;i<n;i++)
+    out[i]=1000.0 * (float)in[i]/((float)rate);
+
+  return AF_OK;
+}
+
 /*
  * test if output format matches
  *  af: audio filter

audio/filter/window.c

@@ -0,0 +1,213 @@
+/*
+ * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
+ *
+ * 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.
+ */
+
+/* Calculates a number of window functions. The following window
+   functions are currently implemented: Boxcar, Triang, Hanning,
+   Hamming, Blackman, Flattop and Kaiser. In the function call n is
+   the number of filter taps and w the buffer in which the filter
+   coefficients will be stored.
+*/
+
+#include <math.h>
+#include "dsp.h"
+
+/*
+// Boxcar
+//
+// n window length
+// w buffer for the window parameters
+*/
+void af_window_boxcar(int n, FLOAT_TYPE* w)
+{
+  int i;
+  // Calculate window coefficients
+  for (i=0 ; i<n ; i++)
+    w[i] = 1.0;
+}
+
+
+/*
+// Triang a.k.a Bartlett
+//
+//               |    (N-1)|
+//           2 * |k - -----|
+//               |      2  |
+// w = 1.0 - ---------------
+//                    N+1
+// n window length
+// w buffer for the window parameters
+*/
+void af_window_triang(int n, FLOAT_TYPE* w)
+{
+  FLOAT_TYPE k1  = (FLOAT_TYPE)(n & 1);
+  FLOAT_TYPE k2  = 1/((FLOAT_TYPE)n + k1);
+  int      end = (n + 1) >> 1;
+  int      i;
+
+  // Calculate window coefficients
+  for (i=0 ; i<end ; i++)
+    w[i] = w[n-i-1] = (2.0*((FLOAT_TYPE)(i+1))-(1.0-k1))*k2;
+}
+
+
+/*
+// Hanning
+//                   2*pi*k
+// w = 0.5 - 0.5*cos(------), where 0 < k <= N
+//                    N+1
+// n window length
+// w buffer for the window parameters
+*/
+void af_window_hanning(int n, FLOAT_TYPE* w)
+{
+  int      i;
+  FLOAT_TYPE k = 2*M_PI/((FLOAT_TYPE)(n+1)); // 2*pi/(N+1)
+
+  // Calculate window coefficients
+  for (i=0; i<n; i++)
+    *w++ = 0.5*(1.0 - cos(k*(FLOAT_TYPE)(i+1)));
+}
+
+/*
+// Hamming
+//                        2*pi*k
+// w(k) = 0.54 - 0.46*cos(------), where 0 <= k < N
+//                         N-1
+//
+// n window length
+// w buffer for the window parameters
+*/
+void af_window_hamming(int n,FLOAT_TYPE* w)
+{
+  int      i;
+  FLOAT_TYPE k = 2*M_PI/((FLOAT_TYPE)(n-1)); // 2*pi/(N-1)
+
+  // Calculate window coefficients
+  for (i=0; i<n; i++)
+    *w++ = 0.54 - 0.46*cos(k*(FLOAT_TYPE)i);
+}
+
+/*
+// Blackman
+//                       2*pi*k             4*pi*k
+// w(k) = 0.42 - 0.5*cos(------) + 0.08*cos(------), where 0 <= k < N
+//                        N-1                 N-1
+//
+// n window length
+// w buffer for the window parameters
+*/
+void af_window_blackman(int n,FLOAT_TYPE* w)
+{
+  int      i;
+  FLOAT_TYPE k1 = 2*M_PI/((FLOAT_TYPE)(n-1)); // 2*pi/(N-1)
+  FLOAT_TYPE k2 = 2*k1; // 4*pi/(N-1)
+
+  // Calculate window coefficients
+  for (i=0; i<n; i++)
+    *w++ = 0.42 - 0.50*cos(k1*(FLOAT_TYPE)i) + 0.08*cos(k2*(FLOAT_TYPE)i);
+}
+
+/*
+// Flattop
+//                                        2*pi*k                     4*pi*k
+// w(k) = 0.2810638602 - 0.5208971735*cos(------) + 0.1980389663*cos(------), where 0 <= k < N
+//                                          N-1                        N-1
+//
+// n window length
+// w buffer for the window parameters
+*/
+void af_window_flattop(int n,FLOAT_TYPE* w)
+{
+  int      i;
+  FLOAT_TYPE k1 = 2*M_PI/((FLOAT_TYPE)(n-1)); // 2*pi/(N-1)
+  FLOAT_TYPE k2 = 2*k1;                   // 4*pi/(N-1)
+
+  // Calculate window coefficients
+  for (i=0; i<n; i++)
+    *w++ = 0.2810638602 - 0.5208971735*cos(k1*(FLOAT_TYPE)i)
+                        + 0.1980389663*cos(k2*(FLOAT_TYPE)i);
+}
+
+/* Computes the 0th order modified Bessel function of the first kind.
+// (Needed to compute Kaiser window)
+//
+// y = sum( (x/(2*n))^2 )
+//      n
+*/
+#define BIZ_EPSILON 1E-21 // Max error acceptable
+
+static FLOAT_TYPE besselizero(FLOAT_TYPE x)
+{
+  FLOAT_TYPE temp;
+  FLOAT_TYPE sum   = 1.0;
+  FLOAT_TYPE u     = 1.0;
+  FLOAT_TYPE halfx = x/2.0;
+  int      n     = 1;
+
+  do {
+    temp = halfx/(FLOAT_TYPE)n;
+    u *=temp * temp;
+    sum += u;
+    n++;
+  } while (u >= BIZ_EPSILON * sum);
+  return sum;
+}
+
+/*
+// Kaiser
+//
+// n window length
+// w buffer for the window parameters
+// b beta parameter of Kaiser window, Beta >= 1
+//
+// Beta trades the rejection of the low pass filter against the
+// transition width from passband to stop band.  Larger Beta means a
+// slower transition and greater stop band rejection.  See Rabiner and
+// Gold (Theory and Application of DSP) under Kaiser windows for more
+// about Beta.  The following table from Rabiner and Gold gives some
+// feel for the effect of Beta:
+//
+// All ripples in dB, width of transition band = D*N where N = window
+// length
+//
+// BETA    D       PB RIP   SB RIP
+// 2.120   1.50  +-0.27      -30
+// 3.384   2.23    0.0864    -40
+// 4.538   2.93    0.0274    -50
+// 5.658   3.62    0.00868   -60
+// 6.764   4.32    0.00275   -70
+// 7.865   5.0     0.000868  -80
+// 8.960   5.7     0.000275  -90
+// 10.056  6.4     0.000087  -100
+*/
+void af_window_kaiser(int n, FLOAT_TYPE* w, FLOAT_TYPE b)
+{
+  FLOAT_TYPE tmp;
+  FLOAT_TYPE k1  = 1.0/besselizero(b);
+  int      k2  = 1 - (n & 1);
+  int      end = (n + 1) >> 1;
+  int      i;
+
+  // Calculate window coefficients
+  for (i=0 ; i<end ; i++){
+    tmp = (FLOAT_TYPE)(2*i + k2) / ((FLOAT_TYPE)n - 1.0);
+    w[end-(1&(!k2))+i] = w[end-1-i] = k1 * besselizero(b*sqrt(1.0 - tmp*tmp));
+  }
+}

audio/filter/window.h

@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
+ *
+ * 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.
+ */
+
+/* Calculates a number of window functions. The following window
+   functions are currently implemented: Boxcar, Triang, Hanning,
+   Hamming, Blackman, Flattop and Kaiser. In the function call n is
+   the number of filter taps and w the buffer in which the filter
+   coefficients will be stored.
+*/
+
+#if !defined MPLAYER_DSP_H
+# error Never use window.h directly; include dsp.h instead.
+#endif
+
+#ifndef MPLAYER_WINDOW_H
+#define MPLAYER_WINDOW_H
+
+void af_window_boxcar(int n, FLOAT_TYPE* w);
+void af_window_triang(int n, FLOAT_TYPE* w);
+void af_window_hanning(int n, FLOAT_TYPE* w);
+void af_window_hamming(int n, FLOAT_TYPE* w);
+void af_window_blackman(int n, FLOAT_TYPE* w);
+void af_window_flattop(int n, FLOAT_TYPE* w);
+void af_window_kaiser(int n, FLOAT_TYPE* w, FLOAT_TYPE b);
+
+#endif /* MPLAYER_WINDOW_H */

audio/fmt-conversion.c

@@ -1,19 +1,20 @@
 /*
- * 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 <libavutil/avutil.h>
 #include <libavutil/samplefmt.h>