Release 0.9.2

It sometimes happens on exit, and it's probably a bad idea. If the
process hangs on exit (possibly due to stupid hardcoded timeouts it's
doing), mpv will also hang now, unfortunately.

(cherry picked from commit be9bf4cc7c)

.github/ISSUE_TEMPLATE

@@ -1,23 +0,0 @@
-### mpv version and platform
-
-If you're not using git master or the latest release, update.
-
-### Reproduction steps
-
-### Expected behavior
-
-### Actual behavior
-
-### Log file
-
-Make a log file made with -v or --log-file=output.txt, paste it to
-http://sprunge.us or a similar site, and replace this text with a link to it.
-
-Providing a log file is strongly encouraged. It is very helpful for reproducing
-bugs or getting important technical context. Without a log it might not be
-possible to analyze and fix certain bugs.
-
-### Sample files
-
-Sample files needed to reproduce this issue can be uploaded to https://0x0.st/
-or similar sites. (Only needed if the issue cannot be reproduced without it.)

.github/PULL_REQUEST_TEMPLATE

@@ -1,7 +0,0 @@
-Ideally, you shouldn't need enter any text here, and your commit messages should
-explain your changes sufficiently (especially why they are needed). Read
-https://github.com/mpv-player/mpv/blob/master/DOCS/contribute.md for coding
-style and development conventions. Remove this text block, but if you haven't
-agreed to it before, leave the following sentence in place:
-
-I agree that my changes can be relicensed to LGPL 2.1 or later.

.gitignore

@@ -19,7 +19,6 @@
 /demux/ebml_types.h
 /sub/osd_font.h
 /player/lua/*.inc
-/player/builtin_conf.inc
 /DOCS/man/mpv.1
 /DOCS/man/mpv.aux
 /DOCS/man/mpv.log

.travis.yml

@@ -12,8 +12,6 @@ env:
   global:
     # Coverity token
     - secure: "H21mSRlMhk4BKS0xHZvCFGJxteCP0hRVUxTuNfM2Z9HBsyutuLEYMtViLO86VtM+Tqla3xXPzUdS4ozLwI72Ax/5ZUDXACROj73yW6QhFB5D6rLut12+FjqC7M33Qv2hl0xwgNBmR5dsm1ToP37+Wn+ecJQNvN8fkTXF+HVzOEw="
-    # Travis token for mpv.io
-    - secure: "nlTVLJK6kRhtXvhKCoJ3YdFGHuKaq/eHowfPw25hqRWuBOZd+HjHY5KIYjV7SxuKFDpJE4GpNcvA3Q31nsqomxpkLYgrwjg6TSazN7ZP+x85ZgV1QGFebrPfGm2n5UR5CAPAwFoeF3pZheLi4bajVzwq1fWW+x3grS188P9OZso="
 compiler:
   - clang
   - gcc
@@ -39,12 +37,11 @@ matrix:
     - os: linux
       compiler: clang
 
-before_install: TOOLS/travis-deps libass-stable $LIBAV
+before_install: ./travis-deps libass-stable $LIBAV
 script:
   - ./bootstrap.py
   - ./waf configure
   - ./waf build
-after_script: TOOLS/travis-rebuild-website
 
 notifications:
   email: false

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),
@@ -18,87 +20,21 @@ 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 0.21.0 ---
- 1.23   - deprecate setting "no-" options via mpv_set_option*(). For example,
-          instead of "no-video=" you should set "video=no".
-        - do not override the SIGPIPE signal handler anymore. This was done as
-          workaround for the FFmpeg TLS code, which has been fixed long ago.
-        - deprecate mpv_suspend() and mpv_resume(). They will be stubbed out
-          in mpv 0.22.0.
-        - make mpv_set_property() work to some degree before mpv_initialize().
-          It can now be used instead of mpv_set_option().
-        - semi-deprecate mpv_set_option()/mpv_set_option_string(). You should
-          use mpv_set_property() instead. There are some deprecated properties
-          which conflict with some options (see client.h remarks on
-          mpv_set_option()), for which mpv_set_option() might still be required.
-          In future mpv releases, the conflicting deprecated options/properties
-          will be removed, and mpv_set_option() will internally translate API
-          calls to mpv_set_property().
-        - qthelper.hpp: deprecate get_property_variant, set_property_variant,
-          mpv_set_option, command_variant, and replace them with get_property,
-          set_property, command.
- --- mpv 0.19.0 ---
- 1.22   - add stream_cb API for custom protocols
- --- mpv 0.18.1 ---
- ----   - remove "status" log level from mpv_request_log_messages() docs. This
-          is 100% equivalent to "v". The behavior is still the same, thus no
-          actual API change.
- --- mpv 0.18.0 ---
- 1.21   - mpv_set_property() changes behavior with MPV_FORMAT_NODE. Before this
-          change it rejected mpv_nodes with format==MPV_FORMAT_STRING if the
-          property was not a string or did not have special mechanisms in place
-          the function failed. Now it always invokes the option string parser,
-          and mpv_node with a basic data type works exactly as if the function
-          is invoked with that type directly. This new behavior is equivalent
-          to mpv_set_option().
-          This also affects the mp.set_property_native() Lua function.
-        - generally, setting choice options/properties with "yes"/"no" options
-          can now be set as MPV_FORMAT_FLAG
-        - reading a choice property as MPV_FORMAT_NODE will now return a
-          MPV_FORMAT_FLAG value if the choice is "yes" (true) or "no" (false)
-          This implicitly affects Lua and JSON IPC interfaces as well.
-        - big changes to vo-cmdline on vo_opengl and vo_opengl_hq (but not
-          vo_opengl_cb): options are now normally not reset, but applied on top
-          of the current options. The special undocumented value "-" still
-          works, but now resets all options to before any vo-cmdline command
-          has been called.
- --- mpv 0.12.0 ---
- 1.20   - deprecate "GL_MP_D3D_interfaces"/"glMPGetD3DInterface", and introduce
-          "GL_MP_MPGetNativeDisplay"/"glMPGetNativeDisplay" (this is a
-          backwards-compatible rename)
- --- mpv 0.11.0 ---
- --- mpv 0.10.0 ---
- 1.19   - add "GL_MP_D3D_interfaces" pseudo extension to make it possible to
-          use DXVA2 in OpenGL fullscreen mode in some situations
-        - mpv_request_log_messages() now accepts "terminal-default" as parameter
- 1.18   - add MPV_END_FILE_REASON_REDIRECT, and change behavior of
-          MPV_EVENT_END_FILE accordingly
-        - a bunch of interface-changes.rst changes
- 1.17   - mpv_initialize() now blocks SIGPIPE (details see client.h)
- --- mpv 0.9.0 ---
+ --- mpv 0.9.0 is released ---
+ 1.17   - add MPV_FORMAT_BYTE_ARRAY
  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)
@@ -111,7 +47,7 @@ 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
@@ -134,7 +70,7 @@ 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
@@ -143,7 +79,7 @@ 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://
@@ -155,6 +91,6 @@ 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/Copyright

@@ -0,0 +1,14 @@
+All examples in this directory and its sub-directories are licensed
+under one of the following licenses:
+
+    WTFPL, ISC, Ms-PL, AGPLv3, BSD (any)
+
+Pick any license of your liking, and disregard the others.
+
+(The full text of each license is available on this website:
+  http://opensource.org/licenses/alphabetical )
+
+Additionally, you may consider the example code to be public domain.
+You are free to use any of the example code without further
+requirements or need for attribution.
+

DOCS/client_api_examples/README.md

@@ -0,0 +1,30 @@
+Client API examples
+===================
+
+All these examples use the mpv client API through libmpv.
+
+cocoa
+-----
+
+Shows how to embed the mpv video window in Objective-C/Cocoa.
+
+qt
+--
+
+Shows how to embed the mpv video window in Qt (using normal desktop widgets).
+
+qml
+---
+
+Shows how to use mpv's OpenGL video renderer in QtQuick2 with QML.
+
+qml_direct
+----------
+
+Alternative example, which typically avoids a FBO indirection. Might be
+slightly faster, but is less flexible and harder to use.
+
+simple
+------
+
+Very primitive terminal-only example. Shows some most basic API usage.

DOCS/client_api_examples/cocoa/cocoabasic.m

@@ -0,0 +1,211 @@
+// 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`
+
+#include <mpv/client.h>
+
+#include <stdio.h>
+#include <stdlib.h>
+
+static inline void check_error(int status)
+{
+    if (status < 0) {
+        printf("mpv API error: %s\n", mpv_error_string(status));
+        exit(1);
+    }
+}
+
+#import <Cocoa/Cocoa.h>
+
+@interface CocoaWindow : NSWindow
+@end
+
+@implementation CocoaWindow
+- (BOOL)canBecomeMainWindow { return YES; }
+- (BOOL)canBecomeKeyWindow { return YES; }
+@end
+
+@interface AppDelegate : NSObject <NSApplicationDelegate>
+{
+    mpv_handle *mpv;
+    dispatch_queue_t queue;
+    NSWindow *w;
+    NSView *wrapper;
+}
+@end
+
+static void wakeup(void *);
+
+@implementation AppDelegate
+
+- (void)createWindow {
+
+    int mask = NSTitledWindowMask|NSClosableWindowMask|
+               NSMiniaturizableWindowMask|NSResizableWindowMask;
+
+    self->w = [[CocoaWindow alloc]
+        initWithContentRect:NSMakeRect(0,0, 1280, 720)
+                  styleMask:mask
+                    backing:NSBackingStoreBuffered
+                      defer:NO];
+
+    [self->w setTitle:@"cocoabasic example"];
+    [self->w makeMainWindow];
+    [self->w makeKeyAndOrderFront:nil];
+
+    NSRect frame = [[self->w contentView] bounds];
+    self->wrapper = [[NSView alloc] initWithFrame:frame];
+    [self->wrapper setAutoresizingMask:NSViewWidthSizable|NSViewHeightSizable];
+    [[self->w contentView] addSubview:self->wrapper];
+    [self->wrapper release];
+
+    NSMenu *m = [[NSMenu alloc] initWithTitle:@"AMainMenu"];
+    NSMenuItem *item = [m addItemWithTitle:@"Apple" action:nil keyEquivalent:@""];
+    NSMenu *sm = [[NSMenu alloc] initWithTitle:@"Apple"];
+    [m setSubmenu:sm forItem:item];
+    [sm addItemWithTitle: @"mpv_command('stop')" action:@selector(mpv_stop) keyEquivalent:@""];
+    [sm addItemWithTitle: @"mpv_command('quit')" action:@selector(mpv_quit) keyEquivalent:@""];
+    [sm addItemWithTitle: @"quit" action:@selector(terminate:) keyEquivalent:@"q"];
+    [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];
+
+    // Deal with MPV in the background.
+    queue = dispatch_queue_create("mpv", DISPATCH_QUEUE_SERIAL);
+    dispatch_async(queue, ^{
+
+        mpv = mpv_create();
+        if (!mpv) {
+            printf("failed creating context\n");
+            exit(1);
+        }
+
+        int64_t wid = (intptr_t) self->wrapper;
+        check_error(mpv_set_option(mpv, "wid", MPV_FORMAT_INT64, &wid));
+
+        // Maybe set some options here, like default key bindings.
+        // NOTE: Interaction with the window seems to be broken for now.
+        check_error(mpv_set_option_string(mpv, "input-default-bindings", "yes"));
+
+        // for testing!
+        check_error(mpv_set_option_string(mpv, "input-media-keys", "yes"));
+        check_error(mpv_set_option_string(mpv, "input-cursor", "no"));
+        check_error(mpv_set_option_string(mpv, "input-vo-keyboard", "yes"));
+
+        // request important errors
+        check_error(mpv_request_log_messages(mpv, "warn"));
+
+        check_error(mpv_initialize(mpv));
+
+        // 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));
+    });
+}
+
+- (void) handleEvent:(mpv_event *)event
+{
+    switch (event->event_id) {
+    case MPV_EVENT_SHUTDOWN: {
+        mpv_detach_destroy(mpv);
+        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);
+    }
+
+    case MPV_EVENT_VIDEO_RECONFIG: {
+        dispatch_async(dispatch_get_main_queue(), ^{
+            NSArray *subviews = [self->wrapper subviews];
+            if ([subviews count] > 0) {
+                // mpv's events view
+                NSView *eview = [self->wrapper subviews][0];
+                [self->w makeFirstResponder:eview];
+            }
+        });
+    }
+
+    default:
+        printf("event: %s\n", mpv_event_name(event->event_id));
+    }
+}
+
+- (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];
+}
+
+// Ostensibly, mpv's window would be hooked up to this.
+- (BOOL) windowShouldClose:(id)sender
+{
+    return NO;
+}
+
+- (void) mpv_stop
+{
+    if (mpv) {
+        const char *args[] = {"stop", NULL};
+        mpv_command(mpv, args);
+    }
+}
+
+- (void) mpv_quit
+{
+    if (mpv) {
+        const char *args[] = {"quit", NULL};
+        mpv_command(mpv, args);
+    }
+}
+@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 EXIT_SUCCESS;
+}

DOCS/client_api_examples/qml/main.cpp

@@ -0,0 +1,136 @@
+#include "main.h"
+
+#include <stdexcept>
+#include <clocale>
+
+#include <QObject>
+#include <QtGlobal>
+#include <QOpenGLContext>
+#include <QGuiApplication>
+
+#include <QtGui/QOpenGLFramebufferObject>
+
+#include <QtQuick/QQuickWindow>
+#include <QtQuick/QQuickView>
+
+class MpvRenderer : public QQuickFramebufferObject::Renderer
+{
+    static void *get_proc_address(void *ctx, const char *name) {
+        (void)ctx;
+        QOpenGLContext *glctx = QOpenGLContext::currentContext();
+        if (!glctx)
+            return NULL;
+        return (void *)glctx->getProcAddress(QByteArray(name));
+    }
+
+    mpv::qt::Handle mpv;
+    QQuickWindow *window;
+    mpv_opengl_cb_context *mpv_gl;
+public:
+    MpvRenderer(const MpvObject *obj)
+        : mpv(obj->mpv), window(obj->window()), mpv_gl(obj->mpv_gl)
+    {
+        int r = mpv_opengl_cb_init_gl(mpv_gl, NULL, get_proc_address, NULL);
+        if (r < 0)
+            throw std::runtime_error("could not initialize OpenGL");
+    }
+
+    virtual ~MpvRenderer()
+    {
+        // Until this call is done, we need to make sure the player remains
+        // alive. This is done implicitly with the mpv::qt::Handle instance
+        // in this class.
+        mpv_opengl_cb_uninit_gl(mpv_gl);
+    }
+
+    void render()
+    {
+        QOpenGLFramebufferObject *fbo = framebufferObject();
+        window->resetOpenGLState();
+        mpv_opengl_cb_draw(mpv_gl, fbo->handle(), fbo->width(), fbo->height());
+        window->resetOpenGLState();
+    }
+};
+
+MpvObject::MpvObject(QQuickItem * parent)
+    : QQuickFramebufferObject(parent), mpv_gl(0)
+{
+    mpv = mpv::qt::Handle::FromRawHandle(mpv_create());
+    if (!mpv)
+        throw std::runtime_error("could not create mpv context");
+
+    mpv_set_option_string(mpv, "terminal", "yes");
+    mpv_set_option_string(mpv, "msg-level", "all=v");
+
+    if (mpv_initialize(mpv) < 0)
+        throw std::runtime_error("could not initialize mpv context");
+
+    // Make use of the MPV_SUB_API_OPENGL_CB API.
+    mpv::qt::set_option_variant(mpv, "vo", "opengl-cb");
+
+    // Request hw decoding, just for testing.
+    mpv::qt::set_option_variant(mpv, "hwdec", "auto");
+
+    // Setup the callback that will make QtQuick update and redraw if there
+    // is a new video frame. Use a queued connection: this makes sure the
+    // doUpdate() function is run on the GUI thread.
+    mpv_gl = (mpv_opengl_cb_context *)mpv_get_sub_api(mpv, MPV_SUB_API_OPENGL_CB);
+    if (!mpv_gl)
+        throw std::runtime_error("OpenGL not compiled in");
+    mpv_opengl_cb_set_update_callback(mpv_gl, MpvObject::on_update, (void *)this);
+    connect(this, &MpvObject::onUpdate, this, &MpvObject::doUpdate,
+            Qt::QueuedConnection);
+}
+
+MpvObject::~MpvObject()
+{
+    if (mpv_gl)
+        mpv_opengl_cb_set_update_callback(mpv_gl, NULL, NULL);
+}
+
+void MpvObject::on_update(void *ctx)
+{
+    MpvObject *self = (MpvObject *)ctx;
+    emit self->onUpdate();
+}
+
+// connected to onUpdate(); signal makes sure it runs on the GUI thread
+void MpvObject::doUpdate()
+{
+    update();
+}
+
+void MpvObject::command(const QVariant& params)
+{
+    mpv::qt::command_variant(mpv, params);
+}
+
+void MpvObject::setProperty(const QString& name, const QVariant& value)
+{
+    mpv::qt::set_property_variant(mpv, name, value);
+}
+
+QQuickFramebufferObject::Renderer *MpvObject::createRenderer() const
+{
+    window()->setPersistentOpenGLContext(true);
+    window()->setPersistentSceneGraph(true);
+    return new MpvRenderer(this);
+}
+
+int main(int argc, char **argv)
+{
+    QGuiApplication app(argc, argv);
+
+    // Qt sets the locale in the QGuiApplication constructor, but libmpv
+    // requires the LC_NUMERIC category to be set to "C", so change it back.
+    std::setlocale(LC_NUMERIC, "C");
+
+    qmlRegisterType<MpvObject>("mpvtest", 1, 0, "MpvObject");
+
+    QQuickView view;
+    view.setResizeMode(QQuickView::SizeRootObjectToView);
+    view.setSource(QUrl("qrc:///mpvtest/main.qml"));
+    view.show();
+
+    return app.exec();
+}

DOCS/client_api_examples/qml/main.h

@@ -0,0 +1,36 @@
+#ifndef MPVRENDERER_H_
+#define MPVRENDERER_H_
+
+#include <QtQuick/QQuickFramebufferObject>
+
+#include <mpv/client.h>
+#include <mpv/opengl_cb.h>
+#include <mpv/qthelper.hpp>
+
+class MpvRenderer;
+
+class MpvObject : public QQuickFramebufferObject
+{
+    Q_OBJECT
+
+    mpv::qt::Handle mpv;
+    mpv_opengl_cb_context *mpv_gl;
+
+    friend class MpvRenderer;
+
+public:
+    MpvObject(QQuickItem * parent = 0);
+    virtual ~MpvObject();
+    virtual Renderer *createRenderer() const;
+public slots:
+    void command(const QVariant& params);
+    void setProperty(const QString& name, const QVariant& value);
+signals:
+    void onUpdate();
+private slots:
+    void doUpdate();
+private:
+    static void on_update(void *ctx);
+};
+
+#endif

DOCS/client_api_examples/qml/main.qml

@@ -0,0 +1,71 @@
+import QtQuick 2.0
+import QtQuick.Controls 1.0
+
+import mpvtest 1.0
+
+Item {
+    width: 1280
+    height: 720
+
+    MpvObject {
+        id: renderer
+        anchors.fill: parent
+
+        MouseArea {
+            anchors.fill: parent
+            onClicked: renderer.command(["loadfile", "../../../test.mkv"])
+        }
+    }
+
+    Rectangle {
+        id: labelFrame
+        anchors.margins: -50
+        radius: 5
+        color: "white"
+        border.color: "black"
+        opacity: 0.8
+        anchors.fill: box
+    }
+
+    Row {
+        id: box
+        anchors.bottom: renderer.bottom
+        anchors.left: renderer.left
+        anchors.right: renderer.right
+        anchors.margins: 100
+
+        Text {
+            anchors.margins: 10
+            wrapMode: Text.WordWrap
+            text: "QtQuick and mpv are both rendering stuff.\n
+                   Click to load ../../../test.mkv"
+        }
+
+        // Don't take these controls too seriously. They're for testing.
+        Column {
+            CheckBox {
+                id: checkbox
+                anchors.margins: 10
+                // Heavily filtered means good, right?
+                text: "Make video look like on a Smart TV"
+                onClicked: {
+                    if (checkbox.checked) {
+                        renderer.command(["vo_cmdline", "scale=sharpen3:scale-param1=5.0"])
+                    } else {
+                        renderer.command(["vo_cmdline", ""])
+                    }
+                }
+            }
+            Slider {
+                id: slider
+                anchors.margins: 10
+                anchors.left: checkbox.left
+                anchors.right: checkbox.right
+                minimumValue: -100
+                maximumValue:  100
+                value: 0
+                onValueChanged: renderer.setProperty("gamma", slider.value | 0)
+            }
+        }
+    }
+}

DOCS/client_api_examples/qml/mpvtest.pro

@@ -0,0 +1,12 @@
+QT += qml quick
+
+HEADERS += main.h
+SOURCES += main.cpp
+
+QT_CONFIG -= no-pkg-config
+CONFIG += link_pkgconfig debug
+PKGCONFIG += mpv
+
+RESOURCES += mpvtest.qrc
+
+OTHER_FILES += main.qml

DOCS/client_api_examples/qml/mpvtest.qrc

@@ -0,0 +1,5 @@
+<RCC>
+    <qresource prefix="/mpvtest">
+        <file>main.qml</file>
+    </qresource>
+</RCC>

DOCS/client_api_examples/qml_direct/main.cpp

@@ -0,0 +1,159 @@
+#include "main.h"
+
+#include <stdexcept>
+#include <clocale>
+
+#include <QObject>
+#include <QtGlobal>
+#include <QOpenGLContext>
+
+#include <QGuiApplication>
+#include <QtQuick/QQuickWindow>
+#include <QtQuick/QQuickView>
+
+static void *get_proc_address(void *ctx, const char *name) {
+    (void)ctx;
+    QOpenGLContext *glctx = QOpenGLContext::currentContext();
+    if (!glctx)
+        return NULL;
+    return (void *)glctx->getProcAddress(QByteArray(name));
+}
+
+MpvRenderer::MpvRenderer(mpv::qt::Handle a_mpv, mpv_opengl_cb_context *a_mpv_gl)
+    : mpv(a_mpv), mpv_gl(a_mpv_gl), window(0), size()
+{
+    int r = mpv_opengl_cb_init_gl(mpv_gl, NULL, get_proc_address, NULL);
+    if (r < 0)
+        throw std::runtime_error("could not initialize OpenGL");
+}
+
+MpvRenderer::~MpvRenderer()
+{
+    // Until this call is done, we need to make sure the player remains
+    // alive. This is done implicitly with the mpv::qt::Handle instance
+    // in this class.
+    mpv_opengl_cb_uninit_gl(mpv_gl);
+}
+
+void MpvRenderer::paint()
+{
+    window->resetOpenGLState();
+
+    // 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());
+
+    window->resetOpenGLState();
+}
+
+MpvObject::MpvObject(QQuickItem * parent)
+    : QQuickItem(parent), mpv_gl(0), renderer(0)
+{
+    mpv = mpv::qt::Handle::FromRawHandle(mpv_create());
+    if (!mpv)
+        throw std::runtime_error("could not create mpv context");
+
+    mpv_set_option_string(mpv, "terminal", "yes");
+    mpv_set_option_string(mpv, "msg-level", "all=v");
+
+    if (mpv_initialize(mpv) < 0)
+        throw std::runtime_error("could not initialize mpv context");
+
+    // Make use of the MPV_SUB_API_OPENGL_CB API.
+    mpv::qt::set_option_variant(mpv, "vo", "opengl-cb");
+
+    // Setup the callback that will make QtQuick update and redraw if there
+    // is a new video frame. Use a queued connection: this makes sure the
+    // doUpdate() function is run on the GUI thread.
+    mpv_gl = (mpv_opengl_cb_context *)mpv_get_sub_api(mpv, MPV_SUB_API_OPENGL_CB);
+    if (!mpv_gl)
+        throw std::runtime_error("OpenGL not compiled in");
+    mpv_opengl_cb_set_update_callback(mpv_gl, MpvObject::on_update, (void *)this);
+    connect(this, &MpvObject::onUpdate, this, &MpvObject::doUpdate,
+            Qt::QueuedConnection);
+
+    connect(this, &QQuickItem::windowChanged,
+            this, &MpvObject::handleWindowChanged);
+}
+
+MpvObject::~MpvObject()
+{
+    if (mpv_gl)
+        mpv_opengl_cb_set_update_callback(mpv_gl, NULL, NULL);
+}
+
+void MpvObject::handleWindowChanged(QQuickWindow *win)
+{
+    if (!win)
+        return;
+    connect(win, &QQuickWindow::beforeSynchronizing,
+            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);
+}
+
+void MpvObject::sync()
+{
+    if (!renderer) {
+        renderer = new MpvRenderer(mpv, mpv_gl);
+        connect(window(), &QQuickWindow::beforeRendering,
+                renderer, &MpvRenderer::paint, Qt::DirectConnection);
+    }
+    renderer->window = window();
+    renderer->size = window()->size() * window()->devicePixelRatio();
+}
+
+void MpvObject::swapped()
+{
+    mpv_opengl_cb_report_flip(mpv_gl, 0);
+}
+
+void MpvObject::cleanup()
+{
+    if (renderer) {
+        delete renderer;
+        renderer = 0;
+    }
+}
+
+void MpvObject::on_update(void *ctx)
+{
+    MpvObject *self = (MpvObject *)ctx;
+    emit self->onUpdate();
+}
+
+// connected to onUpdate(); signal makes sure it runs on the GUI thread
+void MpvObject::doUpdate()
+{
+    window()->update();
+}
+
+void MpvObject::command(const QVariant& params)
+{
+    mpv::qt::command_variant(mpv, params);
+}
+
+int main(int argc, char **argv)
+{
+    QGuiApplication app(argc, argv);
+
+    // Qt sets the locale in the QGuiApplication constructor, but libmpv
+    // requires the LC_NUMERIC category to be set to "C", so change it back.
+    std::setlocale(LC_NUMERIC, "C");
+
+    qmlRegisterType<MpvObject>("mpvtest", 1, 0, "MpvObject");
+
+    QQuickView view;
+    view.setResizeMode(QQuickView::SizeRootObjectToView);
+    view.setSource(QUrl("qrc:///mpvtest/main.qml"));
+    view.show();
+
+    return app.exec();
+}

DOCS/client_api_examples/qml_direct/main.h

@@ -0,0 +1,51 @@
+#ifndef MPVRENDERER_H_
+#define MPVRENDERER_H_
+
+#include <QtQuick/QQuickItem>
+
+#include <mpv/client.h>
+#include <mpv/opengl_cb.h>
+#include <mpv/qthelper.hpp>
+
+class MpvRenderer : public QObject
+{
+    Q_OBJECT
+    mpv::qt::Handle mpv;
+    mpv_opengl_cb_context *mpv_gl;
+    QQuickWindow *window;
+    QSize size;
+
+    friend class MpvObject;
+public:
+    MpvRenderer(mpv::qt::Handle a_mpv, mpv_opengl_cb_context *a_mpv_gl);
+    virtual ~MpvRenderer();
+public slots:
+    void paint();
+};
+
+class MpvObject : public QQuickItem
+{
+    Q_OBJECT
+
+    mpv::qt::Handle mpv;
+    mpv_opengl_cb_context *mpv_gl;
+    MpvRenderer *renderer;
+
+public:
+    MpvObject(QQuickItem * parent = 0);
+    virtual ~MpvObject();
+public slots:
+    void command(const QVariant& params);
+    void sync();
+    void swapped();
+    void cleanup();
+signals:
+    void onUpdate();
+private slots:
+    void doUpdate();
+    void handleWindowChanged(QQuickWindow *win);
+private:
+    static void on_update(void *ctx);
+};
+
+#endif

DOCS/client_api_examples/qml_direct/main.qml

@@ -0,0 +1,49 @@
+import QtQuick 2.0
+import QtQuick.Controls 1.0
+
+import mpvtest 1.0
+
+Item {
+    width: 1280
+    height: 720
+
+    MpvObject {
+        id: renderer
+
+        // This object isn't real and not visible; it just renders into the
+        // background of the containing Window.
+        width: 0
+        height: 0
+    }
+
+    MouseArea {
+        anchors.fill: parent
+        onClicked: renderer.command(["loadfile", "../../../test.mkv"])
+    }
+
+    Rectangle {
+        id: labelFrame
+        anchors.margins: -50
+        radius: 5
+        color: "white"
+        border.color: "black"
+        opacity: 0.8
+        anchors.fill: box
+    }
+
+    Row {
+        id: box
+        anchors.bottom: parent.bottom
+        anchors.left: parent.left
+        anchors.right: parent.right
+        anchors.margins: 100
+
+        Text {
+            anchors.margins: 10
+            wrapMode: Text.WordWrap
+            text: "QtQuick and mpv are both rendering stuff.\n
+                   In this example, mpv is always in the background.\n
+                   Click to load ../../../test.mkv"
+        }
+    }
+}

DOCS/client_api_examples/qml_direct/mpvtest.pro

@@ -0,0 +1,12 @@
+QT += qml quick
+
+HEADERS += main.h
+SOURCES += main.cpp
+
+QT_CONFIG -= no-pkg-config
+CONFIG += link_pkgconfig debug
+PKGCONFIG += mpv
+
+RESOURCES += mpvtest.qrc
+
+OTHER_FILES += main.qml

DOCS/client_api_examples/qml_direct/mpvtest.qrc

@@ -0,0 +1,5 @@
+<RCC>
+    <qresource prefix="/mpvtest">
+        <file>main.qml</file>
+    </qresource>
+</RCC>

DOCS/client_api_examples/qt/qtexample.cpp

@@ -0,0 +1,225 @@
+// This example can be built with: qmake && make
+
+#include <clocale>
+#include <sstream>
+#include <stdexcept>
+
+#include <QtGlobal>
+#include <QFileDialog>
+#include <QStatusBar>
+#include <QMenuBar>
+#include <QMenu>
+#include <QGridLayout>
+#include <QApplication>
+#include <QTextEdit>
+
+#if QT_VERSION >= 0x050000
+#include <QJsonDocument>
+#endif
+
+#include <mpv/qthelper.hpp>
+
+#include "qtexample.h"
+
+static void wakeup(void *ctx)
+{
+    // This callback is invoked from any mpv thread (but possibly also
+    // recursively from a thread that is calling the mpv API). Just notify
+    // the Qt GUI thread to wake up (so that it can process events with
+    // mpv_wait_event()), and return as quickly as possible.
+    MainWindow *mainwindow = (MainWindow *)ctx;
+    emit mainwindow->mpv_events();
+}
+
+MainWindow::MainWindow(QWidget *parent) :
+    QMainWindow(parent)
+{
+    setWindowTitle("Qt embedding demo");
+    setMinimumSize(640, 480);
+
+    QMenu *menu = menuBar()->addMenu(tr("&File"));
+    QAction *on_open = new QAction(tr("&Open"), this);
+    on_open->setShortcuts(QKeySequence::Open);
+    on_open->setStatusTip(tr("Open a file"));
+    connect(on_open, &QAction::triggered, this, &MainWindow::on_file_open);
+    menu->addAction(on_open);
+
+    QAction *on_new = new QAction(tr("&New window"), this);
+    connect(on_new, &QAction::triggered, this, &MainWindow::on_new_window);
+    menu->addAction(on_new);
+
+    statusBar();
+
+    QMainWindow *log_window = new QMainWindow(this);
+    log = new QTextEdit(log_window);
+    log->setReadOnly(true);
+    log_window->setCentralWidget(log);
+    log_window->setWindowTitle("mpv log window");
+    log_window->setMinimumSize(500, 50);
+    log_window->show();
+
+    mpv = mpv_create();
+    if (!mpv)
+        throw std::runtime_error("can't create mpv instance");
+
+    // Create a video child window. Force Qt to create a native window, and
+    // pass the window ID to the mpv wid option. Works on: X11, win32, Cocoa
+    mpv_container = new QWidget(this);
+    setCentralWidget(mpv_container);
+    mpv_container->setAttribute(Qt::WA_DontCreateNativeAncestors);
+    mpv_container->setAttribute(Qt::WA_NativeWindow);
+    // If you have a HWND, use: int64_t wid = (intptr_t)hwnd;
+    int64_t wid = mpv_container->winId();
+    mpv_set_option(mpv, "wid", MPV_FORMAT_INT64, &wid);
+
+    // Enable default bindings, because we're lazy. Normally, a player using
+    // mpv as backend would implement its own key bindings.
+    mpv_set_option_string(mpv, "input-default-bindings", "yes");
+
+    // Enable keyboard input on the X11 window. For the messy details, see
+    // --input-vo-keyboard on the manpage.
+    mpv_set_option_string(mpv, "input-vo-keyboard", "yes");
+
+    // Let us receive property change events with MPV_EVENT_PROPERTY_CHANGE if
+    // this property changes.
+    mpv_observe_property(mpv, 0, "time-pos", MPV_FORMAT_DOUBLE);
+
+    mpv_observe_property(mpv, 0, "track-list", MPV_FORMAT_NODE);
+    mpv_observe_property(mpv, 0, "chapter-list", MPV_FORMAT_NODE);
+
+    // Request log messages with level "info" or higher.
+    // They are received as MPV_EVENT_LOG_MESSAGE.
+    mpv_request_log_messages(mpv, "info");
+
+    // From this point on, the wakeup function will be called. The callback
+    // can come from any thread, so we use the QueuedConnection mechanism to
+    // relay the wakeup in a thread-safe way.
+    connect(this, &MainWindow::mpv_events, this, &MainWindow::on_mpv_events,
+            Qt::QueuedConnection);
+    mpv_set_wakeup_callback(mpv, wakeup, this);
+
+    if (mpv_initialize(mpv) < 0)
+        throw std::runtime_error("mpv failed to initialize");
+}
+
+void MainWindow::handle_mpv_event(mpv_event *event)
+{
+    switch (event->event_id) {
+    case MPV_EVENT_PROPERTY_CHANGE: {
+        mpv_event_property *prop = (mpv_event_property *)event->data;
+        if (strcmp(prop->name, "time-pos") == 0) {
+            if (prop->format == MPV_FORMAT_DOUBLE) {
+                double time = *(double *)prop->data;
+                std::stringstream ss;
+                ss << "At: " << time;
+                statusBar()->showMessage(QString::fromStdString(ss.str()));
+            } else if (prop->format == MPV_FORMAT_NONE) {
+                // The property is unavailable, which probably means playback
+                // was stopped.
+                statusBar()->showMessage("");
+            }
+        } else if (strcmp(prop->name, "chapter-list") == 0 ||
+                   strcmp(prop->name, "track-list") == 0)
+        {
+            // Dump the properties as JSON for demo purposes.
+#if QT_VERSION >= 0x050000
+            if (prop->format == MPV_FORMAT_NODE) {
+                QVariant v = mpv::qt::node_to_variant((mpv_node *)prop->data);
+                // Abuse JSON support for easily printing the mpv_node contents.
+                QJsonDocument d = QJsonDocument::fromVariant(v);
+                append_log("Change property " + QString(prop->name) + ":\n");
+                append_log(d.toJson().data());
+            }
+#endif
+        }
+        break;
+    }
+    case MPV_EVENT_VIDEO_RECONFIG: {
+        // Retrieve the new video size.
+        int64_t w, h;
+        if (mpv_get_property(mpv, "dwidth", MPV_FORMAT_INT64, &w) >= 0 &&
+            mpv_get_property(mpv, "dheight", MPV_FORMAT_INT64, &h) >= 0 &&
+            w > 0 && h > 0)
+        {
+            // Note that the MPV_EVENT_VIDEO_RECONFIG event doesn't necessarily
+            // imply a resize, and you should check yourself if the video
+            // dimensions really changed.
+            // mpv itself will scale/letter box the video to the container size
+            // if the video doesn't fit.
+            std::stringstream ss;
+            ss << "Reconfig: " << w << " " << h;
+            statusBar()->showMessage(QString::fromStdString(ss.str()));
+        }
+        break;
+    }
+    case MPV_EVENT_LOG_MESSAGE: {
+        struct mpv_event_log_message *msg = (struct mpv_event_log_message *)event->data;
+        std::stringstream ss;
+        ss << "[" << msg->prefix << "] " << msg->level << ": " << msg->text;
+        append_log(QString::fromStdString(ss.str()));
+        break;
+    }
+    case MPV_EVENT_SHUTDOWN: {
+        mpv_terminate_destroy(mpv);
+        mpv = NULL;
+        break;
+    }
+    default: ;
+        // Ignore uninteresting or unknown events.
+    }
+}
+
+// This slot is invoked by wakeup() (through the mpv_events signal).
+void MainWindow::on_mpv_events()
+{
+    // Process all events, until the event queue is empty.
+    while (mpv) {
+        mpv_event *event = mpv_wait_event(mpv, 0);
+        if (event->event_id == MPV_EVENT_NONE)
+            break;
+        handle_mpv_event(event);
+    }
+}
+
+void MainWindow::on_file_open()
+{
+    QString filename = QFileDialog::getOpenFileName(this, "Open file");
+    if (mpv) {
+        const QByteArray c_filename = filename.toUtf8();
+        const char *args[] = {"loadfile", c_filename.data(), NULL};
+        mpv_command_async(mpv, 0, args);
+    }
+}
+
+void MainWindow::on_new_window()
+{
+    (new MainWindow())->show();
+}
+
+void MainWindow::append_log(const QString &text)
+{
+    QTextCursor cursor = log->textCursor();
+    cursor.movePosition(QTextCursor::End);
+    cursor.insertText(text);
+    log->setTextCursor(cursor);
+}
+
+MainWindow::~MainWindow()
+{
+    if (mpv)
+        mpv_terminate_destroy(mpv);
+}
+
+int main(int argc, char *argv[])
+{
+    QApplication a(argc, argv);
+
+    // Qt sets the locale in the QApplication constructor, but libmpv requires
+    // the LC_NUMERIC category to be set to "C", so change it back.
+    std::setlocale(LC_NUMERIC, "C");
+
+    MainWindow w;
+    w.show();
+
+    return a.exec();
+}

DOCS/client_api_examples/qt/qtexample.h

@@ -0,0 +1,37 @@
+#ifndef QTEXAMPLE_H
+#define QTEXAMPLE_H
+
+#include <QMainWindow>
+
+#include <mpv/client.h>
+
+class QTextEdit;
+
+class MainWindow : public QMainWindow
+{
+    Q_OBJECT
+
+public:
+    explicit MainWindow(QWidget *parent = 0);
+    ~MainWindow();
+
+private slots:
+    void on_file_open();
+    void on_new_window();
+    void on_mpv_events();
+
+signals:
+    void mpv_events();
+
+private:
+    QWidget *mpv_container;
+    mpv_handle *mpv;
+    QTextEdit *log;
+
+    void append_log(const QString &text);
+
+    void create_player();
+    void handle_mpv_event(mpv_event *event);
+};
+
+#endif // QTEXAMPLE_H

DOCS/client_api_examples/qt/qtexample.pro

@@ -0,0 +1,13 @@
+QT       += core gui
+
+greaterThan(QT_MAJOR_VERSION, 4): QT += widgets
+
+TARGET = qtexample
+TEMPLATE = app
+
+QT_CONFIG -= no-pkg-config
+CONFIG += link_pkgconfig debug
+PKGCONFIG += mpv
+
+SOURCES += qtexample.cpp
+HEADERS += qtexample.h

DOCS/client_api_examples/simple/simple.c

@@ -0,0 +1,54 @@
+// Build with: gcc -o simple simple.c `pkg-config --libs --cflags mpv`
+
+#include <stddef.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <mpv/client.h>
+
+static inline void check_error(int status)
+{
+    if (status < 0) {
+        printf("mpv API error: %s\n", mpv_error_string(status));
+        exit(1);
+    }
+}
+
+int main(int argc, char *argv[])
+{
+    if (argc != 2) {
+        printf("pass a single media file as argument\n");
+        return 1;
+    }
+
+    mpv_handle *ctx = mpv_create();
+    if (!ctx) {
+        printf("failed creating context\n");
+        return 1;
+    }
+
+    // Enable default key bindings, so the user can actually interact with
+    // the player (and e.g. close the window).
+    check_error(mpv_set_option_string(ctx, "input-default-bindings", "yes"));
+    mpv_set_option_string(ctx, "input-vo-keyboard", "yes");
+    int val = 1;
+    check_error(mpv_set_option(ctx, "osc", MPV_FORMAT_FLAG, &val));
+
+    // Done setting up options.
+    check_error(mpv_initialize(ctx));
+
+    // Play this file.
+    const char *cmd[] = {"loadfile", argv[1], NULL};
+    check_error(mpv_command(ctx, cmd));
+
+    // Let it play, and wait until the user quits.
+    while (1) {
+        mpv_event *event = mpv_wait_event(ctx, 10000);
+        printf("event: %s\n", mpv_event_name(event->event_id));
+        if (event->event_id == MPV_EVENT_SHUTDOWN)
+            break;
+    }
+
+    mpv_terminate_destroy(ctx);
+    return 0;
+}

DOCS/compile-windows.md

@@ -68,7 +68,7 @@ echo "MXE_TARGETS := i686-w64-mingw32.static" >> settings.mk
 # Build required packages. The following provide a minimum required to build
 # a reasonable mpv binary (though not an absolute minimum).
 
-make gcc ffmpeg libass jpeg lua
+make gcc ffmpeg libass jpeg pthreads lua
 
 # Add MXE binaries to $PATH
 export PATH=/opt/mxe/usr/bin/:$PATH
@@ -104,21 +104,22 @@ Installing MSYS2
    It doesn't matter whether the i686 or the x86_64 version is used. Both can
    build 32-bit and 64-bit binaries when running on a 64-bit version of Windows.
 
-2. Start a MinGW-w64 shell (``mingw64.exe``). Note that this is different
-   from the MSYS2 shell that is started from the final installation dialog.
+2. Add ``C:\msys64\mingw64\bin`` to your ``%PATH%``. mpv will depend on several
+   DLLs in this folder.
 
-   For a 32-bit build, use ``mingw32.exe``.
+3. Start a MinGW-w64 shell (``mingw64_shell.bat``). For a 32-bit build, use
+   ``mingw32_shell.bat``.
 
 Updating MSYS2
 --------------
 
-To prevent errors during post-install, the MSYS2 core runtime must be updated
-separately.
+To prevent errors during post-install, ``msys2-runtime`` and ``pacman`` must be
+updated first.
 
 ```bash
-# Check for core updates. If instructed, close the shell window and reopen it
-# before continuing.
-pacman -Syu
+# Check for updates to msys2-runtime and pacman. If there were updates, restart
+# MSYS2 before continuing.
+pacman -Sy --needed msys2-runtime pacman
 
 # Update everything else
 pacman -Su
@@ -129,14 +130,14 @@ Installing mpv dependencies
 
 ```bash
 # Install MSYS2 build dependencies and a MinGW-w64 compiler
-pacman -S git mingw-w64-x86_64-pkg-config python mingw-w64-x86_64-gcc
+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 mingw-w64-x86_64-angleproject-git
+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.
@@ -156,14 +157,14 @@ Finally, compile and install mpv. Binaries will be installed to
 
 ```bash
 # For a 32-bit build, use --prefix=/mingw32 instead
-./waf configure CC=gcc.exe --check-c-compiler=gcc --prefix=/mingw64
+./waf configure CC=gcc --prefix=/mingw64
 ./waf install
 ```
 
 Or, compile and install both libmpv and mpv:
 
 ```bash
-./waf configure CC=gcc.exe --check-c-compiler=gcc --enable-libmpv-shared --prefix=/mingw64
+./waf configure CC=gcc --enable-libmpv-shared --prefix=/mingw64
 ./waf install
 
 # waf installs libmpv to the wrong directory, so fix it up
@@ -173,46 +174,39 @@ sed -i 's_/mingw64/bin_/mingw64/lib_' /mingw64/lib/pkgconfig/mpv.pc
 rmdir /mingw64/bin/pkgconfig
 ```
 
-Linking libmpv with MSVC programs
----------------------------------
+Additional dependencies
+=======================
 
-You can build C++ programs in Visual Studio and link them with libmpv. To do
-this, you need a Visual Studio which supports ``stdint.h`` (recent ones do),
-and you need to create a import library for the mpv DLL:
+pthreads
+--------
+
+mpv will use a pthreads wrapper by default. Either pthreads-win32 or
+winpthreads should work. The latter is packaged with most MinGW-w64
+environments, including MSYS2, so it shouldn't be a problem. If you don't have
+a pthreads wrapper or you want to build mpv without one, configure with:
 
 ```bash
-lib /def:mpv.def /name:mpv-1.dll /out:mpv.lib /MACHINE:X64
+./waf configure --enable-win32-internal-pthreads
 ```
 
-The string in the ``/name:`` parameter must match the filename of the DLL (this
-is simply the filename the MSVC linker will use). The ``mpv.def`` can be
-retrieved from the mpv build directory, or can be produced by MingGW's
-gendef.exe helper from the mpv DLL.
+libwaio
+-------
 
-Static linking is not possible.
+If you want to use ``--input-file``, you need libwaio. It's available from
+git://midipix.org/waio
 
-Running mpv
------------
-
-If you want to run mpv from the MinGW-w64 shell, you will find the experience
-much more pleasant if you use the ``winpty`` utility
+To compile libwaio in MSYS2, run:
 
 ```bash
-pacman -S winpty
-winpty mpv.com ToS-4k-1920.mov
+git clone git://midipix.org/waio && cd waio
+
+# 32-bit build, run from mingw32_shell.bat
+./build-mingw-nt32 lib-static CC=gcc AR=ar
+cp -r include/waio /mingw32/include
+cp lib32/libwaio.a /mingw32/lib
+
+# 64-bit build, run from mingw64_shell.bat
+./build-mingw-nt64 lib-static CC=gcc AR=ar
+cp -r include/waio /mingw64/include
+cp lib64/libwaio.a /mingw64/lib
 ```
-
-If you want to move / copy ``mpv.exe`` and ``mpv.com`` to somewhere other than
-``/mingw64/bin/`` for use outside the MinGW-w64 shell, they will still depend on
-DLLs in that folder. The simplest solution is to add ``C:\msys64\mingw64\bin``
-to the windows system ``%PATH%``. Beware though that this can cause problems or
-confusion in Cygwin if that is also installed on the machine.
-
-Use of the ANGLE OpenGL backend requires a copy of ``d3dcompiler_43.dll`` (yes,
-exactly 43) in the path or in the same folder as mpv. It must be of the same
-architecture (x86_64 / i686) as the mpv you compiled. You can find a copy in the
-official mpv builds:
-
-https://mpv.srsfckn.biz/mpv-x86_64-20160118.7z
-
-https://mpv.srsfckn.biz/mpv-i686-20160118.7z

DOCS/contribute.md

@@ -14,11 +14,9 @@ Sending patches
   ``git format-patch``. diffs posted as pastebins (especially if the http link
   returns HTML) just cause extra work for everyone, because they lack commit
   message and authorship information.
-- All new code must be LGPLv2.1+ licensed, or come with the implicit agreement
-  that it will be relicensed to LGPLv2.1+ later (see ``Copyright`` in the
-  repository root directory).
-- When creating pull requests, be sure to test your changes. If you didn't,
-  please say so in the pull request message.
+- When creating pull requests, be sure to test your changes. If you didn't, please
+  say so in the pull request message. 
+
 - Write informative commit messages. Use present tense to describe the
   situation with the patch applied, and past tense for the situation before
   the change.
@@ -29,8 +27,8 @@ Sending patches
 
   For example, you fixed a crash in af_volume.c:
 
-  - Bad: ``fixed the bug (wtf?)``
-  - Good: ``af_volume: fix crash due to null pointer access``
+  Bad: ``fixed the bug (wtf?)``
+  Good: ``af_volume: fix crash due to null pointer access``
 
   Having a prefix gives context, and is especially useful when trying to find
   a specific change by looking at the history, or when running ``git blame``.
@@ -55,10 +53,7 @@ Sending patches
   something like reformatting a whole file, and hiding an actual functional
   change in the same commit.
 - If you add a new command line option, document it in options.rst. If you
-  add a new input property, document it in input.rst. Changes to the user
-  interface (options, properties, commands) should be documented in
-  interface-changes.rst. Changes to libmpv should be documented in
-  client-api-changes.rst.
+  add a new input property, document it in input.rst.
 
 Code formatting
 ---------------

DOCS/edl-mpv.rst

@@ -31,8 +31,9 @@ estimated remaining duration of the source file is used.
 
 Note::
 
-    Usage of relative or absolute paths as well as any protocol prefixes may be
-    prevented for security reasons.
+    mpv can't use ordered chapter files in EDL entries. Usage of relative or
+    absolute paths as well as any protocol prefixes is prevented for security
+    reasons.
 
 
 Syntax of mpv EDL files

DOCS/interface-changes.rst

@@ -1,260 +0,0 @@
-Introduction
-============
-
-mpv provides access to its internals 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 0.21.0 ---
-    - setting certain options at runtime will now take care of updating them
-      property (see for example issue #3281). On the other hand, it will also
-      do runtime verification and reject option changes that do not work
-      (example: setting the "vf" option to a filter during playback, which fails
-      to initialize - the option value will remain at its old value). In general,
-      "set name value" should be mostly equivalent to "set options/name value"
-      in cases where the "name" property is not deprecated and "options/name"
-      exists - deviations from this are either bugs, or documented as caveats
-      in the "Inconsistencies between options and properties" manpage section.
-    - deprecate _all_ --vo and --ao suboptions. Generally, all suboptions are
-      replaced by global options, which do exactly the same. For example,
-      "--vo=opengl:scale=nearest" turns into "--scale=nearest". In some cases,
-      the global option is prefixed, e.g. "--vo=opengl:pbo" turns into
-      "--opengl-pbo".
-      Most of the exact replacements are documented here:
-        https://github.com/mpv-player/mpv/wiki/Option-replacement-list
-    - remove --vo=opengl-hq. Set --profile=opengl-hq instead. Note that this
-      profile does not force the VO. This means if you use the --vo option to
-      set another VO, it won't work. But this also means it can be used with
-      opengl-cb.
-    - remove the --vo=opengl "pre-shaders", "post-shaders" and "scale-shader"
-      sub-options: they were deprecated in favor of "user-shaders"
-    - deprecate --vo-defaults (no replacement)
-    - remove the vo-cmdline command. You can set OpenGL renderer options
-      directly via properties instead.
-    - deprecate the device/sink options on all AOs. Use --audio-device instead.
-    - deprecate "--ao=wasapi:exclusive" and "--ao=coreaudio:exclusive",
-      use --audio-exclusive instead.
-    - subtle changes in how "--no-..." options are treated mean that they are
-      not accessible under "options/..." anymore (instead, these are resolved
-      at parsing time). This does not affect options which start with "--no-",
-      but do not use the mechanism for negation options.
-      (Also see client API change for API version 1.23.)
-    - rename the following properties
-        - "demuxer" -> "current-demuxer"
-        - "fps" -> "container-fps"
-        - "idle" -> "idle-active"
-        - "cache" -> "cache-percent"
-      the old names are deprecated and will change behavior in mpv 0.22.0.
-    - remove deprecated "hwdec-active" and "hwdec-detected" properties
-    - deprecate the ao and vo auto-profiles (they never made any sense)
-    - deprecate "--vo=direct3d_shaders" - use "--vo=direct3d" instead.
-      Change "--vo=direct3d" to always use shaders by default.
-    - deprecate --playlist-pos option, renamed to --playlist-start
-    - deprecate the --chapter option, as it is redundant with --start/--end,
-      and conflicts with the semantics of the "chapter" property
-    - rename --sub-text-* to --sub-* and --ass-* to --sub-ass-* (old options
-      deprecated)
-    - incompatible change to cdda:// protocol options: the part after cdda://
-      now always sets the device, not the span or speed to be played. No
-      separating extra "/" is needed. The hidden --cdda-device options is also
-      deleted (it was redundant with the documented --cdrom-device).
-    - deprecate --vo=rpi. It will be removed in mpv 0.22.0. Its functionality
-      was folded into --vo=opengl, which now uses RPI hardware decoding by
-      treating it as a hardware overlay (without applying GL filtering). Also
-      to be changed in 0.22.0: the --fs flag will be reset to "no" by default
-      (like on the other platforms).
-    - deprecate --mute=auto (informally has been since 0.18.1)
-    - deprecate "resume" and "suspend" IPC commands. They will be completely
-      removed in 0.22.0.
-    - deprecate mp.suspend(), mp.resume(), mp.resume_all() Lua scripting
-      commands, as well as setting mp.use_suspend. They will be completely
-      removed in 0.22.0.
-    - the "seek" command's absolute seek mode will now interpret negative
-      seek times as relative from the end of the file (and clamps seeks that
-      still go before 0)
-    - add almost all options to the property list, meaning you can change
-      options without adding "options/" to the property name (a new section
-      has been added to the manpage describing some conflicting behavior
-      between options and properties)
-    - implement changing sub-speed during playback
-    - make many previously fixed options changeable at runtime (for example
-      --terminal, --osc, --ytdl, can all be enable/disabled after
-      mpv_initialize() - this can be extended to other still fixed options
-      on user requests)
- --- mpv 0.20.0 ---
-    - add --image-display-duration option - this also means that image duration
-      is not influenced by --mf-fps anymore in the general case (this is an
-      incompatible change)
- --- mpv 0.19.0 ---
-    - deprecate "balance" option/property (no replacement)
- --- mpv 0.18.1 ---
-    - deprecate --heartbeat-cmd
-    - remove --softvol=no capability:
-        - deprecate --softvol, it now does nothing
-        - --volume, --mute, and the corresponding properties now always control
-          softvol, and behave as expected without surprises (e.g. you can set
-          them normally while no audio is initialized)
-        - rename --softvol-max to --volume-max (deprecated alias is added)
-        - the --volume-restore-data option and property are removed without
-          replacement. They were _always_ internal, and used for watch-later
-          resume/restore. Now --volume/--mute are saved directly instead.
-        - the previous point means resuming files with older watch-later configs
-          will print an error about missing --volume-restore-data (which you can
-          ignore), and will not restore the previous value
-        - as a consequence, volume controls will no longer control PulseAudio
-          per-application value, or use the system mixer's per-application
-          volume processing
-        - system or per-application volume can still be controlled with the
-          ao-volume and ao-mute properties (there are no command line options)
- --- mpv 0.18.0 ---
-    - now ab-loops are active even if one of the "ab-loop-a"/"-b" properties is
-      unset ("no"), in which case the start of the file is used if the A loop
-      point is unset, and the end of the file for an unset B loop point
-    - deprecate --sub-ass=no option by --ass-style-override=strip
-      (also needs --embeddedfonts=no)
-    - add "hwdec-interop" and "hwdec-current" properties
-    - deprecated "hwdec-active" and "hwdec-detected" properties (to be removed
-      in mpv 0.20.0)
-    - choice option/property values that are "yes" or "no" will now be returned
-      as booleans when using the mpv_node functions in the client API, the
-      "native" property accessors in Lua, and the JSON API. They can be set as
-      such as well.
-    - the VO opengl fbo-format sub-option does not accept "rgb" or "rgba"
-      anymore
-    - all VO opengl prescalers have been removed (replaced by user scripts)
- --- mpv 0.17.0 ---
-    - deprecate "track-list/N/audio-channels" property (use
-      "track-list/N/demux-channel-count" instead)
-    - remove write access to "stream-pos", and change semantics for read access
-    - Lua scripts now don't suspend mpv by default while script code is run
-    - add "cache-speed" property
-    - rename --input-unix-socket to --input-ipc-server, and make it work on
-      Windows too
-    - change the exact behavior of the "video-zoom" property
-    - --video-unscaled no longer disables --video-zoom and --video-aspect
-      To force the old behavior, set --video-zoom=0 and --video-aspect=0
- --- mpv 0.16.0 ---
-    - change --audio-channels default to stereo (use --audio-channels=auto to
-      get the old default)
-    - add --audio-normalize-downmix
-    - change the default downmix behavior (--audio-normalize-downmix=yes to get
-      the old default)
-    - VO opengl custom shaders must now use "sample_pixel" as function name,
-      instead of "sample"
-    - change VO opengl scaler-resizes-only default to enabled
-    - add VO opengl "interpolation-threshold" suboption (introduces new default
-      behavior, which can change e.g. ``--video-sync=display-vdrop`` to the
-      worse, but is usually what you want)
-    - make "volume" and "mute" properties changeable even if no audio output is
-      active (this gives not-ideal behavior if --softvol=no is used)
-    - add "volume-max" and "mixer-active" properties
-    - ignore --input-cursor option for events injected by input commands like
-      "mouse", "keydown", etc.
- --- mpv 0.15.0 ---
-    - change "yadif" video filter defaults
- --- mpv 0.14.0 ---
-    - vo_opengl interpolation now requires --video-sync=display-... to be set
-    - change some vo_opengl defaults (including changing tscale)
-    - add "vsync-ratio", "estimated-display-fps" properties
-    - add --rebase-start-time option
-      This is a breaking change to start time handling. Instead of making start
-      time handling an aspect of different options and properties (like
-      "time-pos" vs. "playback-time"), make it dependent on the new option. For
-      compatibility, the "time-start" property now always returns 0, so code
-      which attempted to handle rebasing manually will not break.
- --- mpv 0.13.0 ---
-    - remove VO opengl-cb frame queue suboptions (no replacement)
- --- mpv 0.12.0 ---
-    - remove --use-text-osd (useless; fontconfig isn't a requirement anymore,
-      and text rendering is also lazily initialized)
-    - some time properties (at least "playback-time", "time-pos",
-      "time-remaining", "playtime-remaining") now are unavailable if the time
-      is unknown, instead of just assuming that the internal playback position
-      is 0
-    - add --audio-fallback-to-null option
-    - replace vf_format outputlevels suboption with "video-output-levels" global
-      property/option; also remove "colormatrix-output-range" property
-    - vo_opengl: remove sharpen3/sharpen5 scale filters, add sharpen sub-option
- --- mpv 0.11.0 ---
-    - add "af-metadata" property
- --- mpv 0.10.0 ---
-    - add --video-aspect-method option
-    - add --playlist-pos option
-    - add --video-sync* options
-      "display-sync-active" property
-      "vo-missed-frame-count" property
-      "audio-speed-correction" and "video-speed-correction" properties
-    - remove --demuxer-readahead-packets and --demuxer-readahead-bytes
-      add --demuxer-max-packets and --demuxer-max-bytes
-      (the new options are not replacement and have very different semantics)
-    - change "video-aspect" property: always settable, even if no video is
-      running; always return the override - if no override is set, return
-      the video's aspect ratio
-    - remove disc-nav (DVD, BD) related properties and commands
-    - add "option-info/<name>/set-locally" property
-    - add --cache-backbuffer; change --cache-default default to 75MB
-      the new total cache size is the sum of backbuffer and the cache size
-      specified by --cache-default or --cache
-    - add ``track-list/N/audio-channels`` property
-    - change --screenshot-tag-colorspace default value
-    - add --stretch-image-subs-to-screen
-    - add "playlist/N/title" property
-    - add --video-stereo-mode=no to disable auto-conversions
-    - add --force-seekable, and change default seekability in some cases
-    - add vf yadif/vavpp/vdpaupp interlaced-only suboptions
-      Also, the option is enabled by default (Except vf_yadif, which has
-      it enabled only if it's inserted by the deinterlace property.)
-    - add --hwdec-preload
-    - add ao coreaudio exclusive suboption
-    - add ``track-list/N/forced`` property
-    - add audio-params/channel-count and ``audio-params-out/channel-count props.
-    - add af volume replaygain-fallback suboption
-    - add video-params/stereo-in property
-    - add "keypress", "keydown", and "keyup" commands
-    - deprecate --ad-spdif-dtshd and enabling passthrough via --ad
-      add --audio-spdif as replacement
-    - remove "get_property" command
-    - remove --slave-broken
-    - add vo opengl custom shader suboptions (source-shader, scale-shader,
-      pre-shaders, post-shaders)
-    - completely change how the hwdec properties work:
-        - "hwdec" now reflects the --hwdec option
-        - "hwdec-detected" does partially what the old "hwdec" property did
-          (and also, "detected-hwdec" is removed)
-        - "hwdec-active" is added
-    - add protocol-list property
-    - deprecate audio-samplerate and audio-channels properties
-      (audio-params sub-properties are the replacement)
-    - add audio-params and audio-out-params properties
-    - deprecate "audio-format" property, replaced with "audio-codec-name"
-    - deprecate --media-title, replaced with --force-media-title
-    - deprecate "length" property, replaced with "duration"
-    - change volume property:
-        - the value 100 is now always "unchanged volume" - with softvol, the
-          range is 0 to --softvol-max, without it is 0-100
-        - the minimum value of --softvol-max is raised to 100
-    - remove vo opengl npot suboption
-    - add relative seeking by percentage to "seek" command
-    - add playlist_shuffle command
-    - add --force-window=immediate
-    - add ao coreaudio change-physical-format suboption
-    - remove vo opengl icc-cache suboption, add icc-cache-dir suboption
-    - add --screenshot-directory
-    - add --screenshot-high-bit-depth
-    - add --screenshot-jpeg-source-chroma
-    - default action for "rescan_external_files" command changes
- --- mpv 0.9.0 ---

DOCS/man/af.rst

@@ -11,10 +11,6 @@ syntax is:
 
     To get a full list of available audio filters, see ``--af=help``.
 
-    Also, keep in mind that most actual filters are available via the ``lavfi``
-    wrapper, which gives you access to most of libavfilter's filters. This
-    includes all filters that have been ported from MPlayer to libavfilter.
-
 You can also set defaults for each filter. The defaults are applied before the
 normal filter parameters.
 
@@ -63,14 +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|auto>``
-        Whether to normalize when remixing channel layouts (default: auto).
-        ``auto`` uses the value set by ``--audio-normalize-downmix``.
     ``o=<string>``
         Set AVOptions on the SwrContext or AVAudioResampleContext. These should
         be documented by FFmpeg or Libav.
 
-``lavcac3enc[=options]``
+``lavcac3enc[=tospdif[:bitrate[:minch]]]``
     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
@@ -103,9 +96,58 @@ Available filters are:
         If the input channel number is less than ``<minch>``, the filter will
         detach itself (default: 3).
 
-    ``encoder=<name>``
-        Select the libavcodec encoder used. Currently, this should be an AC-3
-        encoder, and using another codec will fail horribly.
+``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
@@ -164,7 +206,7 @@ Available filters are:
 
     .. admonition:: Examples
 
-        ``mpv --af=channels=4:[0-1,1-0,2-2,3-3] media.avi``
+        ``mpv --af=channels=4:[0-1,1-0,0-2,1-3] media.avi``
             Would change the number of channels to 4 and set up 4 routes that
             swap channel 0 and channel 1 and leave channel 2 and 3 intact.
             Observe that if media containing two channels were played back,
@@ -221,6 +263,13 @@ Available filters are:
     used to do conversion itself, unlike this one which lets the filter system
     handle the conversion.
 
+``convert24``
+    Filter for internal use only. Converts between 24-bit and 32-bit sample
+    formats.
+
+``convertsign``
+    Filter for internal use only. Converts between signed/unsigned formats.
+
 ``volume[=<volumedb>[:...]]``
     Implements software volume control. Use this filter with caution since it
     can reduce the signal to noise ratio of the sound. In most cases it is
@@ -245,10 +294,6 @@ 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
@@ -312,12 +357,121 @@ 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.
 
-    This filter supports the following ``af-command`` commands:
+``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.
 
-    ``set-matrix``
-        Set the ``<matrix>`` argument dynamically. This can be used to change
-        the mixing matrix at runtime, without reinitializing the entire filter
-        chain.
+    .. 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
+    only useful if you have more than 2 loudspeakers.
+
+    ``[ch1,ch2,...]``
+        The delay in ms that should be imposed on each channel (floating point
+        number between 0 and 1000).
+
+    To calculate the required delay for the different channels, do as follows:
+
+    1. Measure the distance to the loudspeakers in meters in relation to your
+       listening position, giving you the distances s1 to s5 (for a 5.1
+       system). There is no point in compensating for the subwoofer (you will
+       not hear the difference anyway).
+
+    2. Subtract the distances s1 to s5 from the maximum distance, i.e.
+       ``s[i] = max(s) - s[i]; i = 1...5``.
+
+    3. Calculate the required delays in ms as ``d[i] = 1000*s[i]/342; i =
+       1...5``.
+
+    .. admonition:: Example
+
+        ``mpv --af=delay=[10.5,10.5,0,0,7,0] media.avi``
+            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
@@ -342,6 +496,45 @@ 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).
@@ -401,6 +594,10 @@ Available filters are:
         ``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
             Would tweak the quality and performance parameters.
 
+        ``mpv --af=format=float,scaletempo media.ogg``
+            Would make scaletempo use float code. Maybe faster on some
+            platforms.
+
         ``mpv --af=scaletempo=scale=1.2:speed=pitch audio.ogg``
             Would play media at 1.2x normal speed, with audio at normal pitch.
             Changing playback speed would change pitch, leaving audio tempo at
@@ -409,13 +606,9 @@ Available filters are:
 ``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. It can also be used to adjust audio pitch
-    without changing playback speed.
+    at speed different from normal.
 
-    ``<pitch-scale>``
-        Sets the pitch scaling factor. Frequencies are multiplied by this value.
-
-    This filter has a number of additional sub-options. You can list them with
+    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
@@ -424,13 +617,6 @@ Available filters are:
     (The mapping of the mpv rubberband filter sub-option names and values to
     those of librubberband follows a simple pattern: ``"Option" + Name + Value``.)
 
-    This filter supports the following ``af-command`` commands:
-
-    ``set-pitch``
-        Set the ``<pitch-scale>`` argument dynamically. This can be used to
-        change the playback pitch at runtime. Note that speed is controlled
-        using the standard ``speed`` property, not ``af-command``.
-
 ``lavfi=graph``
     Filter audio using FFmpeg's libavfilter.
 

DOCS/man/ao.rst

@@ -4,37 +4,108 @@ AUDIO OUTPUT DRIVERS
 Audio output drivers are interfaces to different audio output facilities. The
 syntax is:
 
-``--ao=<driver1,driver2,...[,]>``
+``--ao=<driver1[:suboption1[=value]:...],driver2,...[,]>``
     Specify a priority list of audio output drivers to be used.
 
 If the list has a trailing ',', mpv will fall back on drivers not contained
-in the list.
+in the list. Suboptions are optional and can mostly be omitted.
+
+You can also set defaults for each driver. The defaults are applied before the
+normal driver parameters.
 
 ``--ao-defaults=<driver1[:parameter1:parameter2:...],driver2,...>``
     Set defaults for each driver.
 
-    Deprecated. No replacement.
-
 .. note::
 
     See ``--ao=help`` for a list of compiled-in audio output drivers. The
     driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
-    where PulseAudio is used. On BSD systems, ``--ao=oss`` or ``--ao=sndio``
-    may work (the latter being experimental).
+    where PulseAudio is used. On Windows, ``--ao=wasapi`` is preferred,
+    though it might cause trouble sometimes, in which case ``--ao=dsound``
+    should be used. On BSD systems, ``--ao=oss`` or `--ao=sndio`` may work
+    (the latter being experimental). On OS X systems, use ``--ao=coreaudio``.
+
+.. admonition:: Examples
+
+    - ``--ao=alsa,oss,`` Try the ALSA driver, then the OSS driver, then others.
+    - ``--ao=alsa:resample=yes:device=[plughw:0,3]`` Lets ALSA resample and
+      sets the device-name as first card, fourth device.
 
 Available audio output drivers are:
 
 ``alsa`` (Linux only)
     ALSA audio output driver
 
-    See `ALSA audio output options`_ for options specific to this AO.
+    ``device=<device>``
+        Sets the device name. For ac3 output via S/PDIF, use an "iec958" or
+        "spdif" device, unless you really know how to set it correctly.
+    ``resample=yes``
+        Enable ALSA resampling plugin. (This is disabled by default, because
+        some drivers report incorrect audio delay in some cases.)
+    ``mixer-device=<device>``
+        Set the mixer device used with ``--no-softvol`` (default: ``default``).
+    ``mixer-name=<name>``
+        Set the name of the mixer element (default: ``Master``). This is for
+        example ``PCM`` or ``Master``.
+    ``mixer-index=<number>``
+        Set the index of the mixer channel (default: 0). Consider the output of
+        "``amixer scontrols``", then the index is the number that follows the
+        name of the element.
+    ``non-interleaved``
+        Allow output of non-interleaved formats (if the audio decoder uses
+        this format). Currently disabled by default, because some popular
+        ALSA plugins are utterly broken with non-interleaved formats.
+    ``ingore-chmap``
+        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::
+
+        MPlayer and mplayer2 required you to replace any ',' with '.' and
+        any ':' with '=' in the ALSA device name. mpv does not do this anymore.
+        Instead, quote the device name:
+
+            ``--ao=alsa:device=[plug:surround50]``
+
+        Note that the ``[`` and ``]`` simply quote the device name. With some
+        shells (like zsh), you have to quote the option string to prevent the
+        shell from interpreting the brackets instead of passing them to mpv.
+
+        Actually, you should use the ``--audio-device`` option, instead of
+        setting the device directly.
 
     .. warning::
 
-        To get multichannel/surround audio, use ``--audio-channels=auto``. The
-        default for this option is ``auto-safe``, which makes this audio otuput
-        explicitly reject multichannel output, as there is no way to detect
-        whether a certain channel layout is actually supported.
+        Handling of multichannel/surround audio changed in mpv 0.8.0 from the
+        behavior in MPlayer/mplayer2 and older versions of mpv.
+
+        The old behavior is that the player always downmixed to stereo by
+        default. The ``--audio-channels`` (or ``--channels`` before that) option
+        had to be set to get multichannel audio. Then playing stereo would
+        use the ``default`` device (which typically allows multiple programs
+        to play audio at the same time via dmix), while playing anything with
+        more channels would open one of the hardware devices, e.g. via the
+        ``surround51`` alias (typically with exclusive access). Whether the
+        player would use exclusive access or not would depend on the file
+        being played.
+
+        The new behavior since mpv 0.8.0 always enables multichannel audio,
+        i.e. ``--audio-channels=auto`` is the default. However, since ALSA
+        provides no good way to play multichannel audio in a non-exclusive
+        way (without blocking other applications from using audio), the player
+        is restricted to the capabilities of the ``default`` device by default,
+        which means it supports only stereo and mono (at least with current
+        typical ALSA configurations). But if a hardware device is selected,
+        then multichannel audio will typically work.
+
+        The short story is: if you want multichannel audio with ALSA, use
+        ``--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
@@ -44,42 +115,40 @@ Available audio output drivers are:
 ``oss``
     OSS audio output driver
 
-    The following global options are supported by this audio output:
-
-    ``--oss-device``
+    ``<dsp-device>``
         Sets the audio output device (default: ``/dev/dsp``).
-        Deprecated, use ``--audio-device``.
-    ``--oss-mixer-device``
+    ``<mixer-device>``
         Sets the audio mixer device (default: ``/dev/mixer``).
-    ``--oss-mixer-channel``
+    ``<mixer-channel>``
         Sets the audio mixer channel (default: ``pcm``). Other valid values
         include **vol, pcm, line**. For a complete list of options look for
         ``SOUND_DEVICE_NAMES`` in ``/usr/include/linux/soundcard.h``.
 
 ``jack``
-    JACK (Jack Audio Connection Kit) audio output driver.
+    JACK (Jack Audio Connection Kit) audio output driver
 
-    The following global options are supported by this audio output:
-
-    ``--jack-port=<name>``
+    ``port=<name>``
         Connects to the ports with the given name (default: physical ports).
-    ``--jack-name=<client>``
+    ``name=<client>``
         Client name that is passed to JACK (default: ``mpv``). Useful
         if you want to have certain connections established automatically.
-    ``--jack-autostart=<yes|no>``
+    ``(no-)autostart``
         Automatically start jackd if necessary (default: disabled). Note that
         this tends to be unreliable and will flood stdout with server messages.
-    ``--jack-connect=<yes|no>``
+    ``(no-)connect``
         Automatically create connections to output ports (default: enabled).
         When enabled, the maximum number of output channels will be limited to
         the number of available output ports.
-    ``--jack-std-channel-layout=<waveext|any>``
-        Select the standard channel layout (default: waveext). JACK itself has no
+    ``std-channel-layout=alsa|waveext|any``
+        Select the standard channel layout (default: alsa). JACK itself has no
         notion of channel layouts (i.e. assigning which speaker a given
         channel is supposed to map to) - it just takes whatever the application
         outputs, and reroutes it to whatever the user defines. This means the
         user and the application are in charge of dealing with the channel
-        layout. ``waveext`` uses WAVE_FORMAT_EXTENSIBLE order, which, even
+        layout. ``alsa`` uses the old MPlayer layout, which is inspired by
+        ALSA's standard layouts. In this mode, ao_jack will refuse to play 3
+        or 7 channels (because these do not really have a defined meaning in
+        MPlayer). ``waveext`` uses WAVE_FORMAT_EXTENSIBLE order, which, even
         though it was defined by Microsoft, is the standard on many systems.
         The value ``any`` makes JACK accept whatever comes from the audio
         filter chain, regardless of channel layout and without reordering. This
@@ -93,27 +162,12 @@ Available audio output drivers are:
     Automatically redirects to ``coreaudio_exclusive`` when playing compressed
     formats.
 
-    The following global options are supported by this audio output:
-
-    ``--coreaudio-change-physical-format=<yes|no>``
-        Change the physical format to one similar to the requested audio format
-        (default: no). This has the advantage that multichannel audio output
-        will actually work. The disadvantage is that it will change the
-        system-wide audio settings. This is equivalent to changing the ``Format``
-        setting in the ``Audio Devices`` dialog in the ``Audio MIDI Setup``
-        utility. Note that this does not affect the selected speaker setup.
-
-    ``--coreaudio-exclusive``
-        Deprecated, use ``--audio-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
 
@@ -123,21 +177,18 @@ Available audio output drivers are:
 ``pulse``
     PulseAudio audio output driver
 
-    The following global options are supported by this audio output:
-
-    ``--pulse-host=<host>``, ``--pulse-sink=<sink>``
+    ``[<host>][:<output sink>]``
         Specify the host and optionally output sink to use. An empty <host>
         string uses a local connection, "localhost" uses network transfer
         (most likely not what you want).
-        Deprecated, use ``--audio-device``.
 
-    ``--pulse-buffer=<1-2000|native>``
+    ``buffer=<1-2000|native>``
         Set the audio buffer size in milliseconds. A higher value buffers
         more data, and has a lower probability of buffer underruns. A smaller
         value makes the audio stream react faster, e.g. to playback speed
         changes. Default: 250.
 
-    ``--pulse-latency-hacks=<yes|no>``
+    ``latency-hacks=<yes|no>``
         Enable hacks to workaround PulseAudio timing bugs (default: no). If
         enabled, mpv will do elaborate latency calculations on its own. If
         disabled, it will use PulseAudio automatically updated timing
@@ -146,7 +197,19 @@ Available audio output drivers are:
         (it used to be required to get good behavior on old PulseAudio versions).
 
         If you have stuttering video when using pulse, try to enable this
-        option. (Or try to update PulseAudio.)
+        option. (Or alternatively, try to update PulseAudio.)
+
+``dsound`` (Windows only)
+    DirectX DirectSound audio output driver
+
+    .. note:: This driver is for compatibility with old systems.
+
+    ``device=<devicenum>``
+        Sets the device number to use. Playing a file with ``-v`` will show a
+        list of available devices.
+
+    ``buffersize=<ms>``
+        DirectSound buffer size in milliseconds (default: 200).
 
 ``sdl``
     SDL 1.2+ audio output driver. Should work on any platform supported by SDL
@@ -157,67 +220,57 @@ Available audio output drivers are:
               environments, such as systems where none of the other drivers
               are available.
 
-    The following global options are supported by this audio output:
-
-    ``--sdl-buflen=<length>``
+    ``buflen=<length>``
         Sets the audio buffer length in seconds. Is used only as a hint by the
         sound system. Playing a file with ``-v`` will show the requested and
         obtained exact buffer size. A value of 0 selects the sound system
         default.
 
-    ``--sdl-bufcnt=<count>``
+    ``bufcnt=<count>``
         Sets the number of extra audio buffers in mpv. Usually needs not be
         changed.
 
 ``null``
-    Produces no audio output but maintains video playback speed. You can use
-    ``--ao=null --ao-null-untimed`` for benchmarking.
+    Produces no audio output but maintains video playback speed. Use
+    ``--ao=null:untimed`` for benchmarking.
 
-    The following global options are supported by this audio output:
-
-    ``--ao-null-untimed``
+    ``untimed``
         Do not simulate timing of a perfect audio device. This means audio
         decoding will go as fast as possible, instead of timing it to the
         system clock.
 
-    ``--ao-null-buffer``
+    ``buffer``
         Simulated buffer length in seconds.
 
-    ``--ao-null-outburst``
+    ``outburst``
         Simulated chunk size in samples.
 
-    ``--ao-null-speed``
+    ``speed``
         Simulated audio playback speed as a multiplier. Usually, a real audio
         device will not go exactly as fast as the system clock. It will deviate
-        just a little, and this option helps to simulate this.
+        just a little, and this option helps simulating this.
 
-    ``--ao-null-latency``
+    ``latency``
         Simulated device latency. This is additional to EOF.
 
-    ``--ao-null-broken-eof``
+    ``broken-eof``
         Simulate broken audio drivers, which always add the fixed device
         latency to the reported audio playback position.
 
-    ``--ao-null-broken-delay``
+    ``broken-delay``
         Simulate broken audio drivers, which don't report latency correctly.
 
-    ``--ao-null-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
 
-    The following global options are supported by this audio output:
-
-    ``--ao-pcm-waveheader=<yes|no>``
+    ``(no-)waveheader``
         Include or do not include the WAVE header (default: included). When
         not included, raw PCM will be generated.
-    ``--ao-pcm-file=<filename>``
+    ``file=<filename>``
         Write the sound to ``<filename>`` instead of the default
         ``audiodump.wav``. If ``no-waveheader`` is specified, the default is
         ``audiodump.pcm``.
-    ``--ao-pcm-append=<yes|no>``
+    ``(no-)append``
         Append to the file, instead of overwriting it. Always use this with the
         ``no-waveheader`` option - with ``waveheader`` it's broken, because
         it will write a WAVE header every time the file is opened.
@@ -229,19 +282,14 @@ Available audio output drivers are:
               confused with RoarAudio, which is something completely
               different.
 
-    The following global options are supported by this audio output:
-
-    ``--rsound-host=<name/path>``
+    ``host=<name/path>``
         Set the address of the server (default: localhost).  Can be either a
         network hostname for TCP connections or a Unix domain socket path
         starting with '/'.
-    ``--rsound-port=<number>``
+    ``port=<number>``
         Set the TCP port used for connecting to the server (default: 12345).
         Not used if connecting to a Unix domain socket.
 
-    These options are deprecated. If anyone cares enough, their functionality
-    can be added back using ``--audio-device``.
-
 ``sndio``
     Audio output to the OpenBSD sndio sound system
 
@@ -250,28 +298,25 @@ Available audio output drivers are:
     (Note: only supports mono, stereo, 4.0, 5.1 and 7.1 channel
     layouts.)
 
-    The following global options are supported by this audio output:
-
-    ``--ao-sndio-device=<device>``
+    ``device=<device>``
         sndio device to use (default: ``$AUDIODEVICE``, resp. ``snd0``).
-        Deprecated, use ``--audio-device``.
 
 ``wasapi``
     Audio output to the Windows Audio Session API.
 
-    The following global options are supported by this audio output:
-
-    ``--ao-wasapi-exclusive``
-        Deprecated, use ``--audio-exclusive``.
+    ``exclusive``
         Requests exclusive, direct hardware access. By definition prevents
         sound playback of any other program until mpv exits.
-    ``--ao-wasapi-device=<id>``
-        Deprecated, use ``--audio-device``.
-
+    ``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
+        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.

DOCS/man/changes.rst

@@ -8,11 +8,6 @@ There is no real changelog, but you can look at the following things:
 
   https://github.com/mpv-player/mpv/releases
 * The git log, which is the "real" changelog
-* The files ``client-api-changes.rst`` and ``interface-changes.rst`` in the
-  ``DOCS`` sub directoryon the git repository, which document API and user
-  interface changes (the latter usually documents breaking changes only, rather
-  than additions).
 * The file ``mplayer-changes.rst`` in the ``DOCS`` sub directory on the git
   repository, which used to be in place of this section. It documents some
-  changes that happened since mplayer2 forked off MPlayer. (Not updated
-  anymore.)
+  changes that happened since mplayer2 forked off MPlayer.

DOCS/man/encode.rst

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

DOCS/man/ipc.rst

@@ -1,10 +1,10 @@
 JSON IPC
 ========
 
-mpv can be controlled by external programs using the JSON-based IPC protocol.
-It can be enabled by specifying the path to a unix socket or a named pipe using
-the option ``--input-ipc-server``. Clients can connect to this socket and send
-commands to the player or receive events from it.
+mpv can be controlled by external programs using the JSON-based IPC protocol. It
+can be enabled by specifying the path to a unix socket using the option
+``--input-unix-socket``. Clients can connect to this socket and send commands to
+the player or receive events from it.
 
 .. warning::
 
@@ -17,12 +17,12 @@ commands to the player or receive events from it.
 Socat example
 -------------
 
-You can use the ``socat`` tool to send commands (and receive replies) from the
+You can use the ``socat`` tool to send commands (and receive reply) from the
 shell. Assuming mpv was started with:
 
 ::
 
-    mpv file.mkv --input-ipc-server=/tmp/mpvsocket
+    mpv file.mkv --input-unix-socket=/tmp/mpvsocket
 
 Then you can control it using socat:
 
@@ -41,36 +41,11 @@ It's also possible to send input.conf style text-only commands:
 
 ::
 
-    > echo 'show-text ${playback-time}' | socat - /tmp/mpvsocket
+    > echo 'show_text ${playback-time}' | socat - /tmp/mpvsocket
 
 But you won't get a reply over the socket. (This particular command shows the
 playback time on the player's OSD.)
 
-Command Prompt example
-----------------------
-
-Unfortunately, it's not as easy to test the IPC protocol on Windows, since
-Windows ports of socat (in Cygwin and MSYS2) don't understand named pipes. In
-the absence of a simple tool to send and receive from bidirectional pipes, the
-``echo`` command can be used to send commands, but not receive replies from the
-command prompt.
-
-Assuming mpv was started with:
-
-::
-
-    mpv file.mkv --input-ipc-server=\\.\pipe\mpvsocket
-
-You can send commands from a command prompt:
-
-::
-
-    echo show-text ${playback-time} >\\.\pipe\mpvsocket
-
-To be able to simultaneously read and write from the IPC pipe, like on Linux,
-it's necessary to write an external program that uses overlapped file I/O (or
-some wrapper like .NET's NamedPipeClientStream.)
-
 Protocol
 --------
 
@@ -104,30 +79,12 @@ 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, lines starting
+(or ``mpv_command_string()`` in the client API). Additionally, line starting
 with ``#`` and empty lines are ignored.
 
 Currently, embedded 0 bytes terminate the current line, but you should not
@@ -136,7 +93,7 @@ rely on this.
 Commands
 --------
 
-In addition to the commands described in `List of Input Commands`_, a few
+Additionally to  the commands described in `List of Input Commands`_, a few
 extra commands can also be used as part of the protocol:
 
 ``client_name``
@@ -214,7 +171,7 @@ extra commands can also be used as part of the protocol:
 
 ``unobserve_property``
     Undo ``observe_property`` or ``observe_property_string``. This requires the
-    numeric id passed to the observed command as argument.
+    numeric id passed to the observe command as argument.
 
     Example:
 
@@ -242,24 +199,18 @@ extra commands can also be used as part of the protocol:
     command.
 
 ``suspend``
-    Deprecated, will be removed completely in 0.21.0.
-
     Suspend the mpv main loop. There is a long-winded explanation of this in
     the C API function ``mpv_suspend()``. In short, this prevents the player
     from displaying the next video frame, so that you don't get blocked when
     trying to access the player.
 
 ``resume``
-    Deprecated, will be removed completely in 0.21.0.
-
     Undo one ``suspend`` call. ``suspend`` increments an internal counter, and
     ``resume`` decrements it. When 0 is reached, the player is actually resumed.
 
 ``get_version``
     Returns the client API version the C API of the remote mpv instance
-    provides.
-
-    See also: ``DOCS/client-api-changes.rst``.
+    provides. (Also see ``DOCS/client-api-changes.rst``.)
 
 UTF-8
 -----

DOCS/man/libmpv.rst

@@ -1,16 +0,0 @@
-EMBEDDING INTO OTHER PROGRAMS (LIBMPV)
-======================================
-
-mpv can be embedded into other programs as video/audio playback backend. The
-recommended way to do so is using libmpv. See ``libmpv/client.h`` in the mpv
-source code repository. This provides a C API. Bindings for other languages
-might be available (see wiki).
-
-Since libmpv merely allows access to underlying mechanisms that can control
-mpv, further documentation is spread over a few places:
-
-- https://github.com/mpv-player/mpv/blob/master/libmpv/client.h
-- http://mpv.io/manual/master/#options
-- http://mpv.io/manual/master/#list-of-input-commands
-- http://mpv.io/manual/master/#properties
-- https://github.com/mpv-player/mpv-examples/tree/master/libmpv

DOCS/man/lua.rst

@@ -99,7 +99,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
     functions.
 
     Unlike ``mp.command``, this will not use OSD by default either (except
-    for some OSD-specific commands).
+    for some OSd-specific commands).
 
 ``mp.command_native(table [,def])``
     Similar to ``mp.commandv``, but pass the argument list as table. This has
@@ -175,7 +175,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
     Similar to ``mp.set_property``, but set the given property using its native
     type.
 
-    Since there are several data types which cannot represented natively in
+    Since there are several data types which can not represented natively in
     Lua, this might not always work as expected. For example, while the Lua
     wrapper can do some guesswork to decide whether a Lua table is an array
     or a map, this would fail with empty tables. Also, there are not many
@@ -192,14 +192,13 @@ The ``mp`` module is preloaded, although it can be loaded manually with
     Register callback to be run on a key binding. The binding will be mapped to
     the given ``key``, which is a string describing the physical key. This uses
     the same key names as in input.conf, and also allows combinations
-    (e.g. ``ctrl+a``). If the key is empty or ``nil``, no physical key is
-    registered, but the user still can create own bindings (see below).
+    (e.g. ``ctrl+a``).
 
     After calling this function, key presses will cause the function ``fn`` to
     be called (unless the user remapped the key with another binding).
 
     The ``name`` argument should be a short symbolic string. It allows the user
-    to remap the key binding via input.conf using the ``script-message``
+    to remap the key binding via input.conf using the ``script_message``
     command, and the name of the key binding (see below for
     an example). The name should be unique across other bindings in the same
     script - if not, the previous binding with the same name will be
@@ -221,8 +220,8 @@ The ``mp`` module is preloaded, although it can be loaded manually with
             has an ``is_mouse`` entry, which tells whether the event was caused
             by a mouse button.
 
-    Internally, key bindings are dispatched via the ``script-message-to`` or
-    ``script-binding`` input commands and ``mp.register_script_message``.
+    Internally, key bindings are dispatched via the ``script_message_to`` or
+    ``script_binding`` input commands and ``mp.register_script_message``.
 
     Trying to map multiple commands to a key will essentially prefer a random
     binding, while the other bindings are not called. It is guaranteed that
@@ -245,7 +244,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 
     ::
 
-        y script-binding something
+        y script_binding something
 
 
     This will print the message when the key ``y`` is pressed. (``x`` will
@@ -256,7 +255,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 
     ::
 
-        y script-binding fooscript/something
+        y script_binding fooscript.something
 
 ``mp.add_forced_key_binding(...)``
     This works almost the same as ``mp.add_key_binding``, but registers the
@@ -350,10 +349,6 @@ The ``mp`` module is preloaded, although it can be loaded manually with
             with ``add_timeout()``), this starts the timer from the beginning,
             using the initially configured timeout.
 
-        ``is_enabled()``
-            Whether the timer is currently enabled or was previously disabled
-            (e.g. by ``stop()`` or ``kill()``).
-
         ``timeout`` (RW)
             This field contains the current timeout period. This value is not
             updated as time progresses. It's only used to calculate when the
@@ -394,7 +389,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 ``mp.get_script_name()``
     Return the name of the current script. The name is usually made of the
     filename of the script, with directory and file extension removed. If
-    there are several scripts which would have the same name, it's made unique
+    there are several script which would have the same name, it's made unique
     by appending a number.
 
     .. admonition:: Example
@@ -412,28 +407,26 @@ These also live in the ``mp`` module, but are documented separately as they
 are useful only in special situations.
 
 ``mp.suspend()``
-    This function has been deprecated in mpv 0.21.0 (no replacement).
-
     Suspend the mpv main loop. There is a long-winded explanation of this in
     the C API function ``mpv_suspend()``. In short, this prevents the player
     from displaying the next video frame, so that you don't get blocked when
     trying to access the player.
 
-    Before mpv 0.17.0, this was automatically called by the event handler.
+    This is automatically called by the event handler.
 
 ``mp.resume()``
-    This function has been deprecated in mpv 0.21.0 (no replacement).
-
     Undo one ``mp.suspend()`` call. ``mp.suspend()`` increments an internal
     counter, and ``mp.resume()`` decrements it. When 0 is reached, the player
     is actually resumed.
 
 ``mp.resume_all()``
-    This function has been deprecated in mpv 0.21.0 (no replacement).
-
     This resets the internal suspend counter and resumes the player. (It's
     like calling ``mp.resume()`` until the player is actually resumed.)
 
+    You might want to call this if you're about to do something that takes a
+    long time, but doesn't really need access to the player (like a network
+    operation). Note that you still can access the player at any time.
+
 ``mp.get_wakeup_pipe()``
     Calls ``mpv_get_wakeup_pipe()`` and returns the read end of the wakeup
     pipe. (See ``client.h`` for details.)
@@ -456,12 +449,7 @@ are useful only in special situations.
     ``mp.get_wakeup_pipe()`` if you're interested in properly working
     notification of new events and working timers.
 
-``mp.register_idle(fn)``
-    Register an event loop idle handler. Idle handlers are called before the
-    script goes to sleep after handling all new events. This can be used for
-    example to delay processing of property change events: if you're observing
-    multiple properties at once, you might not want to act on each property
-    change, but only when all change notifications have been received.
+    This function calls ``mp.suspend()`` and ``mp.resume_all()`` on its own.
 
 ``mp.enable_messages(level)``
     Set the minimum log level of which mpv message output to receive. These
@@ -471,9 +459,9 @@ are useful only in special situations.
     The level is a string, see ``msg.log`` for allowed log levels.
 
 ``mp.register_script_message(name, fn)``
-    This is a helper to dispatch ``script-message`` or ``script-message-to``
-    invocations to Lua functions. ``fn`` is called if ``script-message`` or
-    ``script-message-to`` (with this script as destination) is run
+    This is a helper to dispatch ``script_message`` or ``script_message_to``
+    invocations to Lua functions. ``fn`` is called if ``script_message`` or
+    ``script_message_to`` (with this script as destination) is run
     with ``name`` as first parameter. The other parameters are passed to ``fn``.
     If a message with the given name is already registered, it's overwritten.
 
@@ -531,7 +519,7 @@ 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
@@ -554,8 +542,8 @@ Example command-line::
      --script-opts=myscript-optionA=TEST,myscript-optionB=0,myscript-optionC=yes
 
 
-mp.utils functions
-------------------
+mp.utils options
+----------------
 
 This built-in module provides generic helper functions for Lua, and have
 strictly speaking nothing to do with mpv or video/audio playback. They are
@@ -642,20 +630,7 @@ 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``.
-
-``utils.subprocess_detached(t)``
-    Runs an external process and detaches it from mpv's control.
-
-    The parameter ``t`` is a table. The function reads the following entries:
-
-        ``args``
-            Array of strings of the same semantics as the ``args`` used in the
-            ``subprocess`` function.
-
-    The function returns ``nil``.
+    In all cases, ``mp.resume_all()`` is implicitly called.
 
 ``utils.parse_json(str [, trail])``
     Parses the given string argument as JSON, and returns it as a Lua table. On
@@ -715,33 +690,6 @@ 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

@@ -37,10 +37,6 @@ LIRC support - configure remotes as input devices instead).
 
 See the ``--input-`` options for ways to customize it.
 
-The following listings are not necessarily complete. See ``etc/input.conf`` for
-a list of default bindings. User ``input.conf`` files and Lua scripts can
-define additional key bindings.
-
 Keyboard Control
 ----------------
 
@@ -54,7 +50,7 @@ UP and DOWN
 
 Ctrl+LEFT and Ctrl+RIGHT
     Seek to the previous/next subtitle. Subject to some restrictions and
-    might not always work; see ``sub-seek`` command.
+    might not work always; see ``sub_seek`` command.
 
 [ and ]
     Decrease/increase current playback speed by 10%.
@@ -120,7 +116,11 @@ 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.
@@ -132,10 +132,7 @@ x and z
     Adjust subtitle delay by +/- 0.1 seconds.
 
 l
-    Set/clear A-B loop points. See ``ab-loop`` command for details.
-
-L
-    Toggle infinite looping.
+    Set/clear A-B loop points. See ``ab_loop`` command for details.
 
 Ctrl + and Ctrl -
     Adjust audio delay by +/- 0.1 seconds.
@@ -143,11 +140,11 @@ Ctrl + and Ctrl -
 u
     Switch between applying no style overrides to SSA/ASS subtitles, and
     overriding them almost completely with the normal subtitle style. See
-    ``--sub-ass-style-override`` for more info.
+    ``--ass-style-override`` for more info.
 
 V
     Toggle subtitle VSFilter aspect compatibility mode. See
-    ``--sub-ass-vsfilter-aspect-compat`` for more info.
+    ``--ass-vsfilter-aspect-compat`` for more info.
 
 r and t
     Move subtitles up/down.
@@ -159,10 +156,6 @@ 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.
 
@@ -175,7 +168,7 @@ 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
@@ -196,19 +189,23 @@ corresponding adjustment, or the software equalizer (``--vf=eq``).)
 7 and 8
     Adjust saturation.
 
-Alt+0 (and command+0 on OSX)
-    Resize video window to half its original size.
+(The following keys are valid only on OS X.)
 
-Alt+1 (and command+1 on OSX)
+command + 0
+    Resize video window to half its original size.
+    (On other platforms, you can bind keys to change the ``window-scale``
+    property.)
+
+command + 1
     Resize video window to its original size.
 
-Alt+2 (and command+2 on OSX)
+command + 2
     Resize video window to double its original size.
 
-command + f (OSX only)
+command + f
     Toggle fullscreen (see also ``--fs``).
 
-command + [ and command + ] (OSX only)
+command + [ and command + ]
     Set video window alpha.
 
 (The following keys are valid if you have a keyboard with multimedia keys.)
@@ -226,10 +223,7 @@ PREVIOUS and NEXT
 support.)
 
 h and k
-    Select previous/next tv-channel.
-
-H and K
-    Select previous/next dvb-channel.
+    Select previous/next channel.
 
 Mouse Control
 -------------
@@ -265,22 +259,21 @@ parser puts several options into a single string, and passes them to a
 component at once, instead of using multiple options on the level of the
 command line.
 
-The suboption parser can quote strings with ``"`` and ``[...]``.
+The suboption parser can quote strings with ``"``, ``'``, and ``[...]``.
 Additionally, there is a special form of quoting with ``%n%`` described below.
 
-For example, assume the hypothetical ``foo`` filter can take multiple options:
+For example, the ``opengl`` VO can take multiple options:
 
-    ``mpv test.mkv --vf=foo:option1=value1:option2:option3=value3,bar``
+    ``mpv test.mkv --vo=opengl:scale=lanczos:icc-profile=file.icc,xv``
 
-This passes ``option1`` and ``option3`` to the ``foo`` filter, with ``option2``
-as flag (implicitly ``option2=yes``), and adds a ``bar`` filter after that. If
-an option contains spaces or characters like ``,`` or ``:``, you need to quote
-them:
+This passes ``scale=lanczos`` and ``icc-profile=file.icc`` to ``opengl``,
+and also specifies ``xv`` as fallback VO. If the icc-profile path contains
+spaces or characters like ``,`` or ``:``, you need to quote them:
 
-    ``mpv '--vf=foo:option1="option value with spaces",bar'``
+    ``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
@@ -298,19 +291,20 @@ It is started with ``%`` and has the following format::
 
 .. admonition:: Examples
 
-    ``mpv '--vf=foo:option1=%11%quoted text' test.avi``
+    ``mpv --ao=pcm:file=%10%C:test.wav test.avi``
 
     Or in a script:
 
-    ``mpv --vf=foo:option1=%`expr length "$NAME"`%"$NAME" test.avi``
+    ``mpv --ao=pcm:file=%`expr length "$NAME"`%"$NAME" test.avi``
 
 Suboptions passed to the client API are also subject to escaping. Using
 ``mpv_set_option_string()`` is exactly like passing ``--name=data`` to the
 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
-of maps and arrays.
+``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.)
 
 Paths
 -----
@@ -320,22 +314,11 @@ example, paths starting with ``-`` will be interpreted as options. Likewise,
 if a path contains the sequence ``://``, the string before that might be
 interpreted as protocol prefix, even though ``://`` can be part of a legal
 UNIX path. To avoid problems with arbitrary paths, you should be sure that
-absolute paths passed to mpv start with ``/``, and prefix relative paths with
-``./``.
-
-Using the ``file://`` pseudo-protocol is discouraged, because it involves
-strange URL unescaping rules.
+absolute paths passed to mpv start with ``/``, and relative paths with ``./``.
 
 The name ``-`` itself is interpreted as stdin, and will cause mpv to disable
 console controls. (Which makes it suitable for playing data piped to stdin.)
 
-The special argument ``--`` can be used to stop mpv from interpreting the
-following arguments as options.
-
-When using the client API, you should strictly avoid using ``mpv_command_string``
-for invoking the ``loadfile`` command, and instead prefer e.g. ``mpv_command``
-to avoid the need for filename escaping.
-
 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``
@@ -344,17 +327,7 @@ additionally wrapped in the fixed-length syntax, e.g. ``%n%string_of_length_n``
 Some mpv options interpret paths starting with ``~``. Currently, the prefix
 ``~~/`` expands to the mpv configuration directory (usually ``~/.config/mpv/``).
 ``~/`` expands to the user's home directory. (The trailing ``/`` is always
-required.) 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
 ----------------
@@ -396,45 +369,6 @@ file stops playing. If option ``--c`` is changed during playback of
 ``file2.mkv``, it is reset when advancing to ``file3.mkv``. This only affects
 file-local options. The option ``--a`` is never reset here.
 
-
-Playing DVDs
-------------
-
-DVDs can be played with the ``dvd://[title]`` syntax. The optional
-title specifier is a number which selects between separate video
-streams on the DVD. If no title is given (``dvd://``) then the longest
-title is selected automatically by the library. This is usually what
-you want. mpv does not support DVD menus.
-
-DVDs which have been copied on to a hard drive or other mounted
-filesystem (by e.g. the ``dvdbackup`` tool) are accommodated by
-specifying the path to the local copy: ``--dvd-device=PATH``.
-Alternatively, running ``mpv PATH`` should auto-detect a DVD directory
-tree and play the longest title.
-
-.. note::
-
-    mpv uses a different default DVD library than MPlayer. MPlayer
-    uses libdvdread by default, and mpv uses libdvdnav by default.
-    Both libraries are developed in parallel, but libdvdnav is
-    intended to support more sophisticated DVD features such as menus
-    and multi-angle playback. mpv uses libdvdnav for files specified
-    as either ``dvd://...`` or ``dvdnav://...``. To use libdvdread,
-    which will produce behavior more like MPlayer, specify
-    ``dvdread://...`` instead. Some users have experienced problems
-    when using libdvdnav, in which playback gets stuck in a DVD menu
-    stream. These problems are reported to go away when auto-selecting
-    the title (``dvd://`` rather than ``dvd://1``) or when using
-    libdvdread (e.g. ``dvdread://0``).
-
-    DVDs use image-based subtitles. Image subtitles are implemented as
-    a bitmap video stream which can be superimposed over the main
-    movie. mpv's subtitle styling and positioning options and keyboard
-    shortcuts generally do not work with image-based subtitles.
-    Exceptions include options like ``--stretch-dvd-subs`` and 
-    ``--stretch-image-subs-to-screen``.
-
-
 CONFIGURATION FILES
 ===================
 
@@ -445,11 +379,10 @@ You can put all of the options in configuration files which will be read every
 time mpv is run. The system-wide configuration file 'mpv.conf' is in your
 configuration directory (e.g. ``/etc/mpv`` or ``/usr/local/etc/mpv``), the
 user-specific one is ``~/.config/mpv/mpv.conf``. For details and platform
-specifics (in particular Windows paths) see the `FILES`_ section.
-
+specifics see the `FILES`_ section.
 User-specific options override system-wide options and options given on the
 command line override either. The syntax of the configuration files is
-``option=value``. Everything after a *#* is considered a comment. Options
+``option=<value>``; everything after a *#* is considered a comment. Options
 that work without values can be enabled by setting them to *yes* and disabled by
 setting them to *no*. Even suboptions can be specified in this way.
 
@@ -467,10 +400,10 @@ 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 ``"``,
+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
+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
@@ -514,41 +447,13 @@ description (shown by ``--profile=help``) can be defined with the
 ``profile-desc`` option. To end the profile, start another one or use the
 profile name ``default`` to continue with normal options.
 
-.. admonition:: Example mpv config file with profiles
+.. admonition:: Example mpv profile
 
     ::
 
-        # normal top-level option
-        fullscreen=yes
-
-        # a profile that can be enabled with --profile=big-cache
-        [big-cache]
-        cache=123400
-        demuxer-readahead-secs=20
-
-        [slow]
-        profile-desc="some profile name"
-        # reference a builtin profile
-        profile=opengl-hq
-
-        [fast]
-        vo=vdpau
-
-        # using a profile again extends it
-        [slow]
-        framedrop=no
-        # you can also include other profiles
-        profile=big-cache
-
-
-Auto profiles
--------------
-
-Some profiles are loaded automatically. The following example demonstrates this:
-
-.. admonition:: Auto profile loading
-
-    ::
+        [vo.vdpau]
+        # Use hardware decoding
+        hwdec=vdpau
 
         [protocol.dvd]
         profile-desc="profile for dvd:// streams"
@@ -558,21 +463,17 @@ Some profiles are loaded automatically. The following example demonstrates this:
         profile-desc="profile for .flv files"
         vf=flip
 
-The profile name follows the schema ``type.name``, where type can be
-``protocol`` for the input/output protocol in use (see ``--list-protocols``),
-and ``extension`` for the extension of the path of the currently played file
-(*not* the file format).
+        [ao.alsa]
+        device=spdif
 
-This feature is very limited, and there are no other auto profiles.
 
 TAKING SCREENSHOTS
 ==================
 
 Screenshots of the currently played file can be taken using the 'screenshot'
 input mode command, which is by default bound to the ``s`` key. Files named
-``mpv-shotNNNN.jpg`` will be saved in the working directory, using the first
-available number - no files will be overwritten. In pseudo-GUI mode, the
-screenshot will be saved somewhere else. See `PSEUDO GUI MODE`_.
+``shotNNNN.jpg`` will be saved in the working directory, using the first
+available number - no files will be overwritten.
 
 A screenshot will usually contain the unscaled video contents at the end of the
 video filter chain and subtitles. By default, ``S`` takes screenshots without
@@ -612,21 +513,13 @@ listed.
   if there is audio "missing", or not enough frames can be dropped. Usually
   this will indicate a problem. (``total-avsync-change`` property.)
 - Encoding state in ``{...}``, only shown in encoding mode.
-- Display sync state. If display sync is active (``display-sync-active``
-  property), this shows ``DS: 2.500/13``, where the first number is average
-  number of vsyncs per video frame (e.g. 2.5 when playing 24Hz videos on 60Hz
-  screens), which might jitter if the ratio doesn't round off, or there are
-  mistimed frames (``vsync-ratio``), and the second number of estimated number
-  of vsyncs which took too long (``vo-delayed-frame-count`` property). The
-  latter is a heuristic, as it's generally not possible to determine this with
-  certainty.
 - Dropped frames, e.g. ``Dropped: 4``. Shows up only if the count is not 0. Can
   grow if the video framerate is higher than that of the display, or if video
-  rendering is too slow. May also be incremented on "hiccups" and when the video
+  rendering is too slow. Also can be incremented on "hiccups" and when the video
   frame couldn't be displayed on time. (``vo-drop-frame-count`` property.)
   If the decoder drops frames, the number of decoder-dropped frames is appended
   to the display as well, e.g.: ``Dropped: 4/34``. This happens only if
-  decoder frame dropping is enabled with the ``--framedrop`` options.
+  decoder-framedropping is enabled with the ``--framedrop`` options.
   (``drop-frame-count`` property.)
 - Cache state, e.g. ``Cache:  2s+134KB``. Visible if the stream cache is enabled.
   The first value shows the amount of video buffered in the demuxer in seconds,
@@ -653,13 +546,17 @@ PROTOCOLS
     Play a path from  Samba share.
 
 ``bd://[title][/device]`` ``--bluray-device=PATH``
-    Play a Blu-ray disc. Currently, this does not accept ISO files. Instead,
+    Play a Blu-Ray disc. Currently, this does not accept ISO files. Instead,
     you must mount the ISO file as filesystem, and point ``--bluray-device``
     to the mounted directory directly.
 
+``bdnav://[title][/device]``
+    Play a Blu-Ray disc, with navigation features enabled. This feature is
+    permanently experimental.
+
 ``dvd://[title|[starttitle]-endtitle][/device]`` ``--dvd-device=PATH``
-    Play a DVD. DVD menus are not supported. If no title is given, the longest
-    title is auto-selected.
+    Play a DVD. If you want dvdnav menus, use ``dvd://menu``. If no title
+    is given, the longest title is auto-selected.
 
     ``dvdnav://`` is an old alias for ``dvd://`` and does exactly the same
     thing.
@@ -681,7 +578,7 @@ PROTOCOLS
 ``mf://[filemask|@listfile]`` ``--mf-...``
     Play a series of images as video.
 
-``cdda://[device]`` ``--cdrom-device=PATH`` ``--cdda-...``
+``cdda://track[-endtrack][:speed][/device]`` ``--cdrom-device=PATH`` ``--cdda-...``
     Play CD.
 
 ``lavf://...``
@@ -704,24 +601,15 @@ PROTOCOLS
     ``PATH`` itself should start with a third ``/`` to make the path an
     absolute path.
 
-``fd://123``
-    Read data from the given file descriptor (for example 123). This is similar
-    to piping data to stdin via ``-``, but can use an arbitrary file descriptor.
-
 ``edl://[edl specification as in edl-mpv.rst]``
     Stitch together parts of multiple files and play them.
 
 ``null://``
-    Simulate an empty file. If opened for writing, it will discard all data.
-    The ``null`` demuxer will specifically pass autoprobing if this protocol
-    is used (while it's not automatically invoked for empty files).
+    Simulate an empty file.
 
 ``memory://data``
     Use the ``data`` part as source data.
 
-``hex://data``
-    Like ``memory://``, but the string is interpreted as hexdump.
-
 PSEUDO GUI MODE
 ===============
 
@@ -736,32 +624,23 @@ Currently this happens only in the following cases:
   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)
-- started out of the bundle on OSX
-- if you manually use ``--player-operation-mode=pseudo-gui`` on the command line
+- manually adding ``--profile=pseudo-gui`` to the command line
 
-This mode applies options from the builtin profile ``builtin-pseudo-gui``, but
-only if these haven't been set in the user's config file or on the command line.
-Also, for compatibility with the old pseudo-gui behavior, the options in the
-``pseudo-gui`` profile are applied unconditionally. In addition, the profile
-makes sure to enable the pseudo-GUI mode, so that ``--profile=pseudo-gui``
-works like in older mpv releases. The profiles are currently defined as follows:
+This mode implicitly adds ``--profile=pseudo-gui`` to the command line, with
+the ``pseudo-gui`` profile being predefined with the following contents:
 
 ::
 
-    [builtin-pseudo-gui]
+    [pseudo-gui]
     terminal=no
     force-window=yes
     idle=once
-    screenshot-directory=~~desktop/
-    [pseudo-gui]
-    player-operation-mode=pseudo-gui
 
-.. warning::
+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.
 
-    Currently, you can extend the ``pseudo-gui`` profile in the config file the
-    normal way. This is deprecated. In future mpv releases, the behavior might
-    change, and not apply your additional settings, and/or use a different
-    profile name.
+The profile always overrides other settings in ``mpv.conf``.
 
 
 .. include:: options.rst
@@ -786,8 +665,6 @@ works like in older mpv releases. The profiles are currently defined as follows:
 
 .. include:: changes.rst
 
-.. include:: libmpv.rst
-
 ENVIRONMENT VARIABLES
 =====================
 
@@ -805,6 +682,9 @@ behavior of mpv.
     If set, XDG-style system configuration directories are used. Otherwise,
     the UNIX convention (``PREFIX/etc/mpv/``) is used.
 
+``TERM``
+    Used to determine terminal type.
+
 ``MPV_HOME``
     Directory where mpv looks for user settings. Overrides ``HOME``, and mpv
     will try to load the config file as ``$MPV_HOME/mpv.conf``.
@@ -815,7 +695,8 @@ behavior of mpv.
     of ``--v`` options passed to the command line.
 
 ``MPV_LEAK_REPORT``
-    If set to ``1``, enable internal talloc leak reporting.
+    If set to ``1``, enable internal talloc leak reporting. Note that this can
+    cause trouble with multithreading, so only developers should use this.
 
 ``LADSPA_PATH``
     Specifies the search path for LADSPA plugins. If it is unset, fully
@@ -855,7 +736,7 @@ libdvdcss:
 
         key
            is the default method. libdvdcss will use a set of calculated
-           player keys to try to get the disc key. This can fail if the drive
+           player keys to try and get the disc key. This can fail if the drive
            does not recognize any of the player keys.
 
         disc
@@ -908,8 +789,6 @@ 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``
@@ -940,7 +819,7 @@ For Windows-specifics, see `FILES ON WINDOWS`_ section.
 ``~/.config/mpv/watch_later/``
     Contains temporary config files needed for resuming playback of files with
     the watch later feature. See for example the ``Q`` key binding, or the
-    ``quit-watch-later`` input command.
+    ``quit_watch_later`` input command.
 
     Each file is a small config file which is loaded if the corresponding media
     file is loaded. It contains the playback position and some (not necessarily
@@ -971,7 +850,7 @@ locations are different. They are generally located under ``%APPDATA%/mpv/``.
 For example, the path to mpv.conf is ``%APPDATA%/mpv/mpv.conf``, which maps to
 a system and user-specific path, for example
 
-    ``C:\users\USERNAME\AppData\Roaming\mpv\mpv.conf``
+    ``C:\users\USERNAME\Application Data\mpv\mpv.conf``
 
 You can find the exact path by running ``echo %APPDATA%\mpv\mpv.conf`` in cmd.exe.
 
@@ -981,23 +860,61 @@ Other config files (such as ``input.conf``) are in the same directory. See the
 The environment variable ``$MPV_HOME`` completely overrides these, like on
 UNIX.
 
-If a directory named ``portable_config`` next to the mpv.exe exists, all
-config will be loaded from this directory only. Watch later config files are
-written to this directory as well. (This exists on Windows only and is redundant
-with ``$MPV_HOME``. However, since Windows is very scripting unfriendly, a
-wrapper script just setting ``$MPV_HOME``, like you could do it on other
-systems, won't work. ``portable_config`` is provided for convenience to get
-around this restriction.)
-
 Config files located in the same directory as ``mpv.exe`` are loaded with
 lower priority. Some config files are loaded only once, which means that
 e.g. of 2 ``input.conf`` files located in two config directories, only the
 one from the directory with higher priority will be loaded.
 
-A third config directory with the lowest priority is the directory named ``mpv``
-in the same directory as ``mpv.exe``. This used to be the directory with the
+A third config directory with lowest priority is the directory named ``mpv``
+in the same directory as ``mpv.exe``. This used to be the directory with
 highest priority, but is now discouraged to use and might be removed in the
 future.
 
 Note that mpv likes to mix ``/`` and ``\`` path separators for simplicity.
 kernel32.dll accepts this, but cmd.exe does not.
+
+EXAMPLES OF MPV USAGE
+=====================
+
+Blu-ray playback:
+    - ``mpv bd:////path/to/disc``
+    - ``mpv bd:// --bluray-device=/path/to/disc``
+
+Play in Japanese with English subtitles:
+    ``mpv dvd://1 --alang=ja --slang=en``
+
+Play only chapters 5, 6, 7:
+    ``mpv dvd://1 --chapter=5-7``
+
+Play only titles 5, 6, 7:
+    ``mpv dvd://5-7``
+
+Play a multi-angle DVD:
+    ``mpv dvd://1 --dvd-angle=2``
+
+Play from a different DVD device:
+    ``mpv dvd://1 --dvd-device=/dev/dvd2``
+
+Play DVD video from a directory with VOB files:
+    ``mpv dvd://1 --dvd-device=/path/to/directory/``
+
+Stream from HTTP:
+    ``mpv http://example.com/example.avi``
+
+Stream using RTSP:
+    ``mpv rtsp://server.example.com/streamName``
+
+Play a libavfilter graph:
+    ``mpv avdevice://lavfi:mandelbrot``
+
+AUTHORS
+=======
+
+mpv is a MPlayer fork based on mplayer2, which in turn is a fork of MPlayer.
+
+MPlayer was initially written by Arpad Gereoffy. See the ``AUTHORS`` file for
+a list of some of the many other contributors.
+
+MPlayer is (C) 2000-2013 The MPlayer Team
+
+This man page was written mainly by Gabucino, Jonas Jermann and Diego Biurrun.

DOCS/man/osc.rst

@@ -5,7 +5,7 @@ The On Screen Controller (short: OSC) is a minimal GUI integrated with mpv to
 offer basic mouse-controllability. It is intended to make interaction easier
 for new users and to enable precise and direct seeking.
 
-The OSC is enabled by default if mpv was compiled with Lua support. It can be
+The OSC is enabled by default if mpv was compiled with lua support. It can be
 disabled entirely using the ``--osc=no`` option.
 
 Using the OSC
@@ -20,25 +20,22 @@ The Interface
 
 ::
 
-    +---------+----------+----------------------------+-------------+
-    | pl prev | pl next  |  title                     |       cache |
-    +------+--+---+------+---------+-----------+------+-------+-----+
-    | play | skip | skip | time    |  seekbar  | time | audio | sub |
-    |      | back | frwd | elapsed |           | left |       |     |
-    +------+------+------+---------+-----------+------+-------+-----+
+    +------------------+-----------+--------------------+
+    | playlist prev    |   title   |      playlist next |
+    +-------+------+---+--+------+-+----+------+--------+
+    | audio | skip | seek |      | seek | skip |  full  |
+    +-------+ back | back | play | frwd | frwd | screen |
+    | sub   |      |      |      |      |      |        |
+    +-------+------+------+------+------+------+--------+
+    |                     seekbar                       |
+    +----------------+--------------+-------------------+
+    | time passed    | cache status |    time remaining |
+    +----------------+--------------+-------------------+
 
 
-pl prev
+playlist prev
     =============   ================================================
     left-click      play previous file in playlist
-    right-click     show playlist
-    shift+L-click   show playlist
-    =============   ================================================
-
-pl next
-    =============   ================================================
-    left-click      play next file in playlist
-    right-click     show playlist
     shift+L-click   show playlist
     =============   ================================================
 
@@ -50,47 +47,10 @@ title
     right-click     show filename
     =============   ================================================
 
-cache
-    | Shows current cache fill status
-
-play
+playlist next
     =============   ================================================
-    left-click      toggle play/pause
-    =============   ================================================
-
-skip back
-    =============   ================================================
-    left-click      go to beginning of chapter / previous chapter
-    right-click     show chapters
-    shift+L-click   show chapters
-    =============   ================================================
-
-skip frwd
-    =============   ================================================
-    left-click      go to next chapter
-    right-click     show chapters
-    shift+L-click   show chapters
-    =============   ================================================
-
-time elapsed
-    | Shows current playback position timestamp
-
-    =============   ================================================
-    left-click      toggle displaying timecodes with milliseconds
-    =============   ================================================
-
-seekbar
-    | Indicates current playback position and position of chapters
-
-    =============   ================================================
-    left-click      seek to position
-    =============   ================================================
-
-time left
-    | Shows remaining playback time timestamp
-
-    =============   ================================================
-    left-click      toggle between total and remaining time
+    left-click      play next file in playlist
+    shift+L-click   show playlist
     =============   ================================================
 
 audio and sub
@@ -102,6 +62,66 @@ audio and sub
     shift+L-click   show available audio/sub tracks
     =============   ================================================
 
+skip back
+    =============   ================================================
+    left-click      go to beginning of chapter / previous chapter
+    shift+L-click   show chapters
+    =============   ================================================
+
+seek back
+    =============   ================================================
+    left-click      skip back  5 seconds
+    right-click     skip back 30 seconds
+    shift-L-click   skip back  1 frame
+    =============   ================================================
+
+play
+    =============   ================================================
+    left-click      toggle play/pause
+    =============   ================================================
+
+seek frwd
+    =============   ================================================
+    left-click      skip forward 10 seconds
+    right-click     skip forward 60 seconds
+    shift-L-click   skip forward  1 frame
+    =============   ================================================
+
+skip frwd
+    =============   ================================================
+    left-click      go to next chapter
+    shift+L-click   show chapters
+    =============   ================================================
+
+fullscreen
+    =============   ================================================
+    left-click      toggle fullscreen
+    =============   ================================================
+
+seekbar
+    | Indicates current playback position and position of chapters
+
+    =============   ================================================
+    left-click      seek to position
+    =============   ================================================
+
+time passed
+    | Shows current playback position timestamp
+
+    =============   ================================================
+    left-click      toggle displaying timecodes with milliseconds
+    =============   ================================================
+
+cache status
+    | Shows current cache fill status (only visible when below 45%)
+
+time remaining
+    | Shows remaining playback time timestamp
+
+    =============   ================================================
+    left-click      toggle between total and remaining time
+    =============   ================================================
+
 Key Bindings
 ~~~~~~~~~~~~
 
@@ -110,7 +130,7 @@ these keys. In case of collision, the function needs to be bound to a
 different key. See the `Script Commands`_ section.
 
 =============   ================================================
-del             Cycles visibility between never / auto (mouse-move) / always
+del             Hide the OSC permanently until mpv is restarted.
 =============   ================================================
 
 Configuration
@@ -156,11 +176,11 @@ Configurable Options
     | Enable the OSC when fullscreen
 
 ``scalewindowed``
-    | Default: 1.5
+    | Default: 1.0
     | Scale factor of the OSC when windowed
 
 ``scalefullscreen``
-    | Default: 1.5
+    | Default: 1.0
     | Scale factor of the OSC when fullscreen
 
 ``scaleforcedwindow``
@@ -180,82 +200,63 @@ Configurable Options
     | Default: 0.0
     | Horizontal alignment, -1 (left) to 1 (right)
 
-``barmargin``
-    | Default: 0
-    | Margin from bottom (bottombar) or top (topbar), in pixels
-
 ``boxalpha``
     | Default: 80
     | Alpha of the background box, 0 (opaque) to 255 (fully transparent)
 
 ``hidetimeout``
     | Default: 500
-    | Duration in ms until the OSC hides if no mouse movement, must not be
-      negative
+    | Duration in ms until the OSC hides if no mouse movement, negative value
+      disables auto-hide
 
 ``fadeduration``
     | Default: 200
     | Duration of fade out in ms, 0 = no fade
 
 ``deadzonesize``
-    | Default: 1
+    | Default: 0
     | Size of the deadzone. The deadzone is an area that makes the mouse act
       like leaving the window. Movement there won't make the OSC show up and
       it will hide immediately if the mouse enters it. The deadzone starts
       at the window border opposite to the OSC and the size controls how much
-      of the window it will span. Values between 0 and 1, where 0 means the
-      OSC will always popup with mouse movement in the window, and 1 means the
-      OSC will only show up when the mouse hovers it.
+      of the window it will span. Values between 0 and 1.
 
 ``minmousemove``
-    | Default: 0
+    | Default: 3
     | Minimum amount of pixels the mouse has to move between ticks to make
       the OSC show up
 
 ``layout``
-    | Default: bottombar
+    | Default: box
     | The layout for the OSC. Currently available are: box, slimbox,
       bottombar and topbar.
 
 ``seekbarstyle``
-    | Default: bar
+    | Default: slider
     | Sets the style of the seekbar, slider (diamond marker) or bar (fill)
 
-``tooltipborder``
-    | Default: 1
-    | Size of the tooltip outline when using bottombar or topbar layouts
-
-``timetotal``
-    | Default: no
-    | Show total time instead of time remaining
-
-``timems``
-    | Default: no
-    | Display timecodes with milliseconds
-
-``visibility``
-    | Default: auto (auto hide/show on mouse move)
-    | Also supports ``never`` and ``always``
-
 Script Commands
 ~~~~~~~~~~~~~~~
 
 The OSC script listens to certain script commands. These commands can bound
 in ``input.conf``, or sent by other scripts.
 
+``enable-osc``
+    Undoes ``disable-osc`` or the effect of the ``del`` key.
+
+``disable-osc``
+    Hide the OSC permanently. This is also what the ``del`` key does.
+
 ``osc-message``
     Show a message on screen using the OSC. First argument is the message,
     second the duration in seconds.
 
-``osc-visibility``
-    Controls visibility mode ``never`` / ``auto`` (on mouse move) / ``always``
-    and also ``cycle`` to cycle between the modes
 
 Example
 
 You could put this into ``input.conf`` to hide the OSC with the ``a`` key and
-to set auto mode (the default) with ``b``::
+to unhide it with ``b``::
 
-    a script-message osc-visibility never
-    b script-message osc-visibility auto
+    a script_message disable-osc
+    b script_message enable-osc
 

DOCS/man/vf.rst

@@ -17,10 +17,6 @@ normal filter parameters.
 
     To get a full list of available video filters, see ``--vf=help``.
 
-    Also, keep in mind that most actual filters are available via the ``lavfi``
-    wrapper, which gives you access to most of libavfilter's filters. This
-    includes all filters that have been ported from MPlayer to libavfilter.
-
 Video filters are managed in lists. There are a few commands to manage the
 filter list.
 
@@ -101,7 +97,7 @@ Available filters are:
 
     All parameters are optional.
 
-    ``<w>:<h>``
+    ``<w>,<h>``
         scaled width/height (default: original width/height)
 
         :0:      scaled d_width/d_height
@@ -151,15 +147,15 @@ Available filters are:
         :yes: Enable accurate rounding.
 
 ``dsize[=w:h:aspect-method:r:aspect]``
-    Changes the intended display aspect at an arbitrary point in the
+    Changes the intended display size/aspect at an arbitrary point in the
     filter chain. Aspect can be given as a fraction (4/3) or floating point
-    number (1.33). Note that this filter does *not* do any scaling itself; it
+    number (1.33). Alternatively, you may specify the exact display width and
+    height desired. Note that this filter does *not* do any scaling itself; it
     just affects what later scalers (software or hardware) will do when
-    auto-scaling to the correct aspect.
+    auto-scaling to correct aspect.
 
     ``<w>,<h>``
-        New aspect ratio given by a display width and height. Unlike older mpv
-        versions or MPlayer, this does not set the display size.
+        New display width and height.
 
         Can also be these special values:
 
@@ -214,7 +210,7 @@ Available filters are:
         Format name, e.g. rgb15, bgr24, 420p, etc. (default: don't change).
     ``<outfmt>``
         Format name that should be substituted for the output. If they do not
-        have the same bytes per pixel and chroma subsampling, it will fail.
+        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
@@ -259,6 +255,26 @@ Available filters are:
         :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
@@ -288,41 +304,28 @@ Available filters are:
         :adobe:        Adobe RGB (1998)
         :prophoto:     ProPhoto RGB (ROMM)
         :cie1931:      CIE 1931 RGB
-        :dci-p3:       DCI-P3 (Digital Cinema)
-        :v-gamut:      Panasonic V-Gamut primaries
 
-    ``<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.
+     ``<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.
+        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.
+        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:
+        Available gamma functions are:
 
-       :auto:         automatic selection (default)
-       :bt.1886:      ITU-R BT.1886 (EOTF corresponding to BT.601/BT.709/BT.2020)
-       :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
-       :st2084:       SMPTE ST2084 (HDR) curve
-       :std-b67:      ARIB STD-B67 (Hybrid Log-gamma) curve
-       :v-log:        Panasonic V-Log transfer curve
-
-    ``<peak>``
-        Reference peak illumination for the video file. This is mostly
-        interesting for HDR, but it can also be used tone map SDR content
-        to a darker or brighter exposure.
-
-        The default of 0.0 will default to the display's reference brightness
-        for SDR and the source's reference brightness for HDR.
+        :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
@@ -406,6 +409,41 @@ 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
@@ -432,6 +470,23 @@ 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
@@ -451,8 +506,9 @@ Available filters are:
         generating an occasional mismatched frame, but it may also cause an
         excessive number of frames to be dropped during high motion sequences.
         Conversely, setting it to -1 will make ``pullup`` match fields more
-        easily. This may help process video with slight blurring between the
-        fields, but may also cause interlaced frames in the output.
+        easily. This may help processing of video where there is slight
+        blurring between the fields, but may also cause there to be interlaced
+        frames in the output.
 
     ``mp`` (metric plane)
         This option may be set to ``u`` or ``v`` to use a chroma plane instead of the
@@ -462,32 +518,49 @@ Available filters are:
         video. The main purpose of setting ``mp`` to a chroma plane is to reduce
         CPU load and make pullup usable in realtime on slow machines.
 
-``yadif=[mode:interlaced-only]``
+``yadif=[mode[:enabled=yes|no]]``
     Yet another deinterlacing filter
 
     ``<mode>``
         :frame: Output 1 frame for each frame.
-        :field: Output 1 frame for each field (default).
+        :field: Output 1 frame for each field.
         :frame-nospatial: Like ``frame`` but skips spatial interlacing check.
         :field-nospatial: Like ``field`` but skips spatial interlacing check.
 
-    ``<interlaced-only>``
-        :no:  Deinterlace all frames.
-        :yes: Only deinterlace frames marked as interlaced (default).
+    ``<enabled>``
+        :yes: Filter is active (default).
+        :no:  Filter is not active, but can be activated with the ``D`` key
+              (or any other key that toggles the ``deinterlace`` property).
 
-    This filter is automatically inserted when using the ``d`` key (or any
+    This filter, is automatically inserted when using the ``D`` key (or any
     other key that toggles the ``deinterlace`` property or when using the
     ``--deinterlace`` switch), assuming the video output does not have native
     deinterlacing support.
 
     If you just want to set the default mode, put this filter and its options
-    into ``--vf-defaults`` instead, and enable deinterlacing with ``d`` or
+    into ``--vf-defaults`` instead, and enable deinterlacing with ``D`` or
     ``--deinterlace``.
 
-    Also, note that the ``d`` key is stupid enough to insert a deinterlacer twice
+    Also note that the ``D`` key is stupid enough to insert an interlacer twice
     when inserting yadif with ``--vf``, so using the above methods is
     recommended.
 
+``delogo[=x:y:w:h:t:show]``
+    Suppresses a TV station logo by a simple interpolation of the surrounding
+    pixels. Just set a rectangle covering the logo and watch it disappear (and
+    sometimes something even uglier appear - your mileage may vary).
+
+    ``<x>,<y>``
+        top left corner of the logo
+    ``<w>,<h>``
+        width and height of the cleared rectangle
+    ``<t>``
+        Thickness of the fuzzy edge of the rectangle (added to ``w`` and
+        ``h``). When set to -1, a green rectangle is drawn on the screen to
+        simplify finding the right ``x``,``y``,``w``,``h`` parameters.
+    ``show``
+        Draw a rectangle showing the area defined by x/y/w/h.
+
 ``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
@@ -596,8 +669,6 @@ 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.
@@ -656,11 +727,10 @@ Available filters are:
     ``buffered-frames``
         Maximum number of decoded video frames that should be buffered before
         the filter (default: 4). This specifies the maximum number of frames
-        the script can request in reverse direction.
-        E.g. if ``buffered-frames=5``, and the script just requested frame 15,
-        it can still request frame 10, but frame 9 is not available anymore.
-        If it requests frame 30, mpv will decode 15 more frames, and keep only
-        frames 25-30.
+        the script can requests backwards. E.g. if ``buffered-frames=5``, and
+        the script just requested frame 15, it can still request frame 10, but
+        frame 9 is not available anymore. If it requests frame 30, mpv will
+        decode 15 more frames, and keep only frames 25-30.
 
         The actual number of buffered frames also depends on the value of the
         ``concurrent-frames`` option. Currently, both option values are
@@ -708,7 +778,7 @@ 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:
@@ -733,7 +803,7 @@ Available filters are:
 ``vavpp``
     VA-AP-API video post processing. Works with ``--vo=vaapi`` and ``--vo=opengl``
     only. Currently deinterlaces. This filter is automatically inserted if
-    deinterlacing is requested (either using the ``d`` key, by default mapped to
+    deinterlacing is requested (either using the ``D`` key, by default mapped to
     the command ``cycle deinterlace``, or the ``--deinterlace`` option).
 
     ``deint=<method>``
@@ -750,14 +820,10 @@ Available filters are:
             depends on the GPU hardware, the GPU drivers, driver bugs, and
             mpv bugs.
 
-    ``<interlaced-only>``
-        :no:  Deinterlace all frames.
-        :yes: Only deinterlace frames marked as interlaced (default).
-
 ``vdpaupp``
     VDPAU video post processing. Works with ``--vo=vdpau`` and ``--vo=opengl``
     only. This filter is automatically inserted if deinterlacing is requested
-    (either using the ``d`` key, by default mapped to the command
+    (either using the ``D`` key, by default mapped to the command
     ``cycle deinterlace``, or the ``--deinterlace`` option). When enabling
     deinterlacing, it is always preferred over software deinterlacer filters
     if the ``vdpau`` VO is used, and also if ``opengl`` is used and hardware
@@ -798,38 +864,14 @@ Available filters are:
     ``pullup``
         Try to apply inverse telecine, needs motion adaptive temporal
         deinterlacing.
-    ``interlaced-only=<yes|no>``
-        If ``yes`` (default), only deinterlace frames marked as interlaced.
     ``hqscaling=<0-9>``
         0
             Use default VDPAU scaling (default).
         1-9
             Apply high quality VDPAU scaling (needs capable hardware).
 
-``vdpaurb``
-    VDPAU video read back. Works with ``--vo=vdpau`` and ``--vo=opengl`` only.
-    This filter will read back frames decoded by VDPAU so that other filters,
-    which are not normally compatible with VDPAU, can be used like normal.
-    This filter must be specified before ``vdpaupp`` in the filter chain if
-    ``vdpaupp`` is used.
-
-``d3d11vpp``
-    Direct3D 11 video post processing. Currently requires D3D11 hardware
-    decoding for use.
-
-    ``deint=<yes|no>``
-        Whether deinterlacing is enabled (default: no).
-    ``interlaced-only=<yes|no>``
-        If ``yes`` (default), only deinterlace frames marked as interlaced.
-    ``mode=<blend|bob|adaptive|mocomp|ivctc|none>``
-        Tries to select a video processor with the given processing capability.
-        If a video processor supports multiple capabilities, it is not clear
-        which algorithm is actually selected. ``none`` always falls back. On
-        most if not all hardware, this option will probably do nothing, because
-        a video processor usually supports all modes or none.
-
 ``buffer=<num>``
     Buffer ``<num>`` frames in the filter chain. This filter is probably pretty
-    useless, except for debugging. (Note that this won't help to smooth out
+    useless, except for debugging. (Note that this won't help smoothing out
     latencies with decoding, because the filter will never output a frame if
     the buffer isn't full, except on EOF.)

DOCS/man/vo.rst

@@ -4,25 +4,29 @@ VIDEO OUTPUT DRIVERS
 Video output drivers are interfaces to different video output facilities. The
 syntax is:
 
-``--vo=<driver1,driver2,...[,]>``
+``--vo=<driver1[:suboption1[=value]:...],driver2,...[,]>``
     Specify a priority list of video output drivers to be used.
 
-If the list has a trailing ``,``, mpv will fall back on drivers not contained
-in the list.
+If the list has a trailing ',', mpv will fall back on drivers not contained
+in the list. Suboptions are optional and can mostly be omitted.
+
+You can also set defaults for each driver. The defaults are applied before the
+normal driver parameters.
 
 ``--vo-defaults=<driver1[:parameter1:parameter2:...],driver2,...>``
     Set defaults for each driver.
 
-    Deprecated. No replacement.
-
 .. note::
 
     See ``--vo=help`` for a list of compiled-in video output drivers.
 
-    The recommended output driver is ``--vo=opengl``, which is the default. All
-    other drivers are for compatibility or special purposes. If the default
-    does not work, it will fallback to other drivers (in the same order as
-    listed by ``--vo=help``).
+    The recommended output drivers are ``--vo=vdpau`` and ``--vo=opengl-hq``.
+    All other drivers are just for compatibility or special purposes.
+
+.. admonition:: Example
+
+    ``--vo=opengl,xv,``
+        Try the ``opengl`` driver, then the ``xv`` driver, then others.
 
 Available video output drivers are:
 
@@ -33,13 +37,11 @@ Available video output drivers are:
 
     .. note:: This driver is for compatibility with old systems.
 
-    The following global options are supported by this video output:
-
-    ``--xv-adaptor=<number>``
+    ``adaptor=<number>``
         Select a specific XVideo adapter (check xvinfo results).
-    ``--xv-port=<number>``
+    ``port=<number>``
         Select a specific XVideo port.
-    ``--xv-ck=<cur|use|set>``
+    ``ck=<cur|use|set>``
         Select the source from which the color key is taken (default: cur).
 
         cur
@@ -50,11 +52,9 @@ Available video output drivers are:
         set
           Same as use but also sets the supplied color key.
 
-    ``--xv-ck-method=<none|man|bg|auto>``
+    ``ck-method=<man|bg|auto>``
         Sets the color key drawing method (default: man).
 
-        none
-          Disables color-keying.
         man
           Draw the color key manually (reduces flicker in some cases).
         bg
@@ -62,15 +62,12 @@ Available video output drivers are:
         auto
           Let Xv draw the color key.
 
-    ``--xv-colorkey=<number>``
+    ``colorkey=<number>``
         Changes the color key to an RGB value of your choice. ``0x000000`` is
         black and ``0xffffff`` is white.
 
-    ``--xv-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.
+    ``no-colorkey``
+        Disables color-keying.
 
 ``x11`` (X11 only)
     Shared memory video output driver without hardware acceleration that works
@@ -89,25 +86,23 @@ Available video output drivers are:
         ``chroma-deint``, ``pullup``, ``hqscaling``. These sub-options are
         deprecated, and you should use the ``vdpaupp`` video filter instead.
 
-    The following global options are supported by this video output:
-
-    ``--vo-vdpau-sharpen=<-1-1>``
+    ``sharpen=<-1-1>``
         (Deprecated. See note about ``vdpaupp``.)
 
         For positive values, apply a sharpening algorithm to the video, for
         negative values a blurring algorithm (default: 0).
-    ``--vo-vdpau-denoise=<0-1>``
+    ``denoise=<0-1>``
         (Deprecated. See note about ``vdpaupp``.)
 
         Apply a noise reduction algorithm to the video (default: 0; no noise
         reduction).
-    ``--vo-vdpau-deint=<-4-4>``
+    ``deint=<-4-4>``
         (Deprecated. See note about ``vdpaupp``.)
 
         Select deinterlacing mode (default: 0). In older versions (as well as
         MPlayer/mplayer2) you could use this option to enable deinterlacing.
         This doesn't work anymore, and deinterlacing is enabled with either
-        the ``d`` key (by default mapped to the command ``cycle deinterlace``),
+        the ``D`` key (by default mapped to the command ``cycle deinterlace``),
         or the ``--deinterlace`` option. Also, to select the default deint mode,
         you should use something like ``--vf-defaults=vdpaupp:deint-mode=temporal``
         instead of this sub-option.
@@ -124,32 +119,32 @@ Available video output drivers are:
         4
             Motion-adaptive temporal deinterlacing with edge-guided spatial
             interpolation. Needs fast video hardware.
-    ``--vo-vdpau-chroma-deint``
+    ``chroma-deint``
         (Deprecated. See note about ``vdpaupp``.)
 
         Makes temporal deinterlacers operate both on luma and chroma (default).
         Use no-chroma-deint to solely use luma and speed up advanced
         deinterlacing. Useful with slow video memory.
-    ``--vo-vdpau-pullup``
+    ``pullup``
         (Deprecated. See note about ``vdpaupp``.)
 
         Try to apply inverse telecine, needs motion adaptive temporal
         deinterlacing.
-    ``--vo-vdpau-hqscaling=<0-9>``
+    ``hqscaling=<0-9>``
         (Deprecated. See note about ``vdpaupp``.)
 
         0
             Use default VDPAU scaling (default).
         1-9
             Apply high quality VDPAU scaling (needs capable hardware).
-    ``--vo-vdpau-fps=<number>``
+    ``fps=<number>``
         Override autodetected display refresh rate value (the value is needed
         for framedrop to allow video playback rates higher than display
         refresh rate, and for vsync-aware frame timing adjustments). Default 0
         means use autodetected value. A positive value is interpreted as a
         refresh rate in Hz and overrides the autodetected value. A negative
         value disables all timing adjustment and framedrop logic.
-    ``--vo-vdpau-composite-detect``
+    ``composite-detect``
         NVIDIA's current VDPAU implementation behaves somewhat differently
         under a compositing window manager and does not give accurate frame
         timing information. With this option enabled, the player tries to
@@ -160,20 +155,20 @@ Available video output drivers are:
         with the composited mode behavior of the NVIDIA driver, there is no
         hard playback speed limit even without the disabled logic. Enabled by
         default, use ``no-composite-detect`` to disable.
-    ``--vo-vdpau-queuetime-windowed=<number>`` and ``queuetime-fs=<number>``
+    ``queuetime_windowed=<number>`` and ``queuetime_fs=<number>``
         Use VDPAU's presentation queue functionality to queue future video
         frame changes at most this many milliseconds in advance (default: 50).
         See below for additional information.
-    ``--vo-vdpau-output-surfaces=<2-15>``
+    ``output_surfaces=<2-15>``
         Allocate this many output surfaces to display video frames (default:
         3). See below for additional information.
-    ``--vo-vdpau-colorkey=<#RRGGBB|#AARRGGBB>``
+    ``colorkey=<#RRGGBB|#AARRGGBB>``
         Set the VDPAU presentation queue background color, which in practice
         is the colorkey used if VDPAU operates in overlay mode (default:
         ``#020507``, some shade of black). If the alpha component of this value
         is 0, the default VDPAU colorkey will be used instead (which is usually
         green).
-    ``--vo-vdpau-force-yuv``
+    ``force-yuv``
         Never accept RGBA input. This means mpv will insert a filter to convert
         to a YUV format before the VO. Sometimes useful to force availability
         of certain YUV-only features, like video equalizer or deinterlacing.
@@ -200,40 +195,31 @@ Available video output drivers are:
     driver implementation may also have limits on the length of maximum
     queuing time or number of queued surfaces that work well or at all.
 
-``direct3d`` (Windows only)
+``direct3d_shaders`` (Windows only)
     Video output driver that uses the Direct3D interface.
 
     .. note:: This driver is for compatibility with systems that don't provide
-              proper OpenGL drivers, and where ANGLE does not perform well.
+              proper OpenGL drivers.
 
-    .. note:: Before to 0.21.0, ``direct3d_shaders`` and ``direct3d`` were
-              different, with ``direct3d`` not using shader by default. Now
-              both use shaders by default, and ``direct3d_shaders`` is a
-              deprecated alias. Use the ``--vo-direct3d-prefer-stretchrect``
-              or the ``--vo-direct3d-disable-shaders`` options to get the old
-              behavior of ``direct3d``.
-
-    The following global options are supported by this video output:
-
-    ``--vo-direct3d-prefer-stretchrect``
+    ``prefer-stretchrect``
         Use ``IDirect3DDevice9::StretchRect`` over other methods if possible.
 
-    ``--vo-direct3d-disable-stretchrect``
+    ``disable-stretchrect``
         Never render the video using ``IDirect3DDevice9::StretchRect``.
 
-    ``--vo-direct3d-disable-textures``
+    ``disable-textures``
         Never render the video using D3D texture rendering. Rendering with
         textures + shader will still be allowed. Add ``disable-shaders`` to
         completely disable video rendering with textures.
 
-    ``--vo-direct3d-disable-shaders``
+    ``disable-shaders``
         Never use shaders when rendering video.
 
-    ``--vo-direct3d-only-8bit``
+    ``only-8bit``
         Never render YUV video with more than 8 bits per component.
         Using this flag will force software conversion to 8-bit.
 
-    ``--vo-direct3d-disable-texture-align``
+    ``disable-texture-align``
         Normally texture sizes are always aligned to 16. With this option
         enabled, the video texture will always have exactly the same size as
         the video itself.
@@ -243,11 +229,11 @@ Available video output drivers are:
     might crash, might cause slow downs, etc. Contact the developers if you
     actually need any of these for performance or proper operation.
 
-    ``--vo-direct3d-force-power-of-2``
+    ``force-power-of-2``
         Always force textures to power of 2, even if the device reports
         non-power-of-2 texture sizes as supported.
 
-    ``--vo-direct3d-texture-memory=<mode>``
+    ``texture-memory=<mode>``
         Only affects operation with shaders/texturing enabled, and (E)OSD.
         Possible values:
 
@@ -271,24 +257,26 @@ Available video output drivers are:
             Use ``D3DPOOL_SCRATCH``, with a ``D3DPOOL_SYSTEMMEM`` texture for
             locking.
 
-    ``--vo-direct3d-swap-discard``
+    ``swap-discard``
         Use ``D3DSWAPEFFECT_DISCARD``, which might be faster.
         Might be slower too, as it must(?) clear every frame.
 
-    ``--vo-direct3d-exact-backbuffer``
+    ``exact-backbuffer``
         Always resize the backbuffer to window size.
 
+``direct3d`` (Windows only)
+    Same as ``direct3d_shaders``, but with the options ``disable-textures``
+    and ``disable-shaders`` forced.
+
+    .. note:: This driver is for compatibility with old systems.
+
 ``opengl``
     OpenGL video output driver. It supports extended scaling methods, dithering
     and color management.
 
-    See `OpenGL renderer options`_ for options specific to this VO.
-
-    By default, it tries to use fast and fail-safe settings. Use the
-    ``opengl-hq`` profile to use this driver with defaults set to high
-    quality rendering. (This profile is also the replacement for
-    ``--vo=opengl-hq``.) The profile can be applied with ``--profile=opengl-hq``
-    and its contents can be viewed with ``--show-profile=opengl-hq``.
+    By default, it tries to use fast and fail-safe settings. Use the alias
+    ``opengl-hq`` to use this driver with defaults set to high quality
+    rendering.
 
     Requires at least OpenGL 2.1.
 
@@ -302,13 +290,471 @@ 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 ``--opengl-fbo-format`` option 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 enabling the
-    ``--opengl-dumb-mode=yes`` option.
+    ``scale=<filter>``
+
+        ``bilinear``
+            Bilinear hardware texture filtering (fastest, very low quality).
+            This is the default for compatibility reasons.
+
+        ``spline36``
+            Mid quality and speed. This is the default when using ``opengl-hq``.
+
+        ``lanczos``
+            Lanczos scaling. Provides mid quality and speed. Generally worse
+            than ``spline36``, but it results in a slightly sharper image
+            which is good for some content types. The number of taps can be
+            controlled with ``scale-radius``, but is best left unchanged.
+
+            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
+            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.
+
+        ``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``).
+
+        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:
+
+        bcspline
+            Spline parameters (``B`` and ``C``). Defaults to 0.5 for both.
+
+        gaussian
+            Scale parameter (``t``). Increasing this makes the result blurrier.
+            Defaults to 1.
+
+        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``
+
+        Note that depending on filter implementation details and video scaling
+        ratio, the radius that actually being used might be different
+        (most likely being increased a bit).
+
+    ``scale-antiring=<value>``
+        Set the antiringing strength. This tries to eliminate ringing, but can
+        introduce other artifacts in the process. Must be a float number
+        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.
+
+    ``scaler-resizes-only``
+        Disable the scaler if the video image is not resized. In that case,
+        ``bilinear`` is used instead whatever is set with ``scale``. Bilinear
+        will reproduce the source image perfectly if no scaling is performed.
+        Note that this option never affects ``cscale``.
+
+    ``pbo``
+        Enable use of PBOs. This is slightly faster, but can sometimes lead to
+        sporadic and temporary image corruption (in theory, because reupload
+        is not retried when it fails), and perhaps actually triggers slower
+        paths with drivers that don't support PBOs properly.
+
+    ``dither-depth=<N|no|auto>``
+        Set dither target depth to N. Default: no.
+
+        no
+            Disable any dithering done by mpv.
+        auto
+            Automatic selection. If output bit depth cannot be detected,
+            8 bits per component are assumed.
+        8
+            Dither to 8 bit output.
+
+        Note that the depth of the connected video display device can not be
+        detected. Often, LCD panels will do dithering on their own, which
+        conflicts with ``opengl``'s dithering and leads to ugly output.
+
+    ``dither-size-fruit=<2-8>``
+        Set the size of the dither matrix (default: 6). The actual size of
+        the matrix is ``(2^N) x (2^N)`` for an option value of ``N``, so a
+        value of 6 gives a size of 64x64. The matrix is generated at startup
+        time, and a large matrix can take rather long to compute (seconds).
+
+        Used in ``dither=fruit`` mode only.
+
+    ``dither=<fruit|ordered|no>``
+        Select dithering algorithm (default: fruit). (Normally, the
+        ``dither-depth`` option controls whether dithering is enabled.)
+
+    ``temporal-dither``
+        Enable temporal dithering. (Only active if dithering is enabled in
+        general.) This changes between 8 different dithering pattern on each
+        frame by changing the orientation of the tiled dithering matrix.
+        Unfortunately, this can lead to flicker on LCD displays, since these
+        have a high reaction time.
+
+    ``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``).
+
+    ``dscale=<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.
+
+    ``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, and that
+        using filters with larger radius may introduce issues when pausing or
+        framestepping, proportional to the radius used. It is recommended to
+        stick to a radius of 1 or 2.
+
+    ``dscale-radius``, ``cscale-radius``, ``tscale-radius``, etc.
+        Set filter parameters for ``dscale``, ``cscale`` and ``tscale``,
+        respectively.
+
+        See the corresponding options for ``scale``.
+
+    ``linear-scaling``
+        Scale in linear light. This is automatically enabled if
+        ``target-prim``, ``target-trc``, ``icc-profile`` or
+        ``sigmoid-upscaling`` is set. It should only be used with a
+        ``fbo-format`` that has at least 16 bit precision.
+
+    ``fancy-downscaling``
+        When using convolution based filters, extend the filter size
+        when downscaling. Trades quality for reduced downscaling performance.
+
+        This is automatically disabled for anamorphic video, because this
+        feature doesn't work correctly with different scale factors in
+        different directions.
+
+    ``sigmoid-upscaling``
+        When upscaling, use a sigmoidal color transform to avoid emphasizing
+        ringing artifacts. This also enables ``linear-scaling``.
+
+    ``sigmoid-center``
+        The center of the sigmoid curve used for ``sigmoid-upscaling``, must
+        be a float between 0.0 and 1.0. Defaults to 0.75 if not specified.
+
+    ``sigmoid-slope``
+        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.
+        Can completely ruin performance. The details depend entirely on the
+        OpenGL driver.
+
+    ``waitvsync``
+        Call ``glXWaitVideoSyncSGI`` after each buffer swap (default: disabled).
+        This may or may not help with video timing accuracy and frame drop. It's
+        possible that this makes video output slower, or has no effect at all.
+
+        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 1 should be enough since full screen may bypass the DWM.
+
+        Windows only.
+
+    ``sw``
+        Continue even if a software renderer is detected.
+
+    ``backend=<sys>``
+        The value ``auto`` (the default) selects the windowing backend. You
+        can also pass ``help`` to get a complete list of compiled in backends
+        (sorted by autoprobe order).
+
+        auto
+            auto-select (default)
+        cocoa
+            Cocoa/OS X
+        win
+            Win32/WGL
+        x11, x11es
+            X11/GLX (the ``es`` variant forces GLES)
+        wayland
+            Wayland/EGL
+        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. (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.
+
+    ``gamma=<0.1..2.0>``
+        Set a gamma value (default: 1.0). If gamma is adjusted in other ways
+        (like with the ``--gamma`` option or key bindings and the ``gamma``
+        property), the value is multiplied with the other gamma value.
+
+        Recommended values based on the environmental brightness:
+
+        1.0
+            Brightly illuminated (default)
+        0.9
+            Slightly dim
+        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`` and ``target-trc`` options. It also enables
+        ``linear-scaling``.
+
+    ``icc-profile-auto``
+        Automatically select the ICC display profile currently specified by
+        the display settings of the operating system.
+
+        NOTE: Only implemented on OS X and X11
+
+    ``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 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``).
+
+        0
+            perceptual
+        1
+            relative colorimetric (default)
+        2
+            saturation
+        3
+            absolute colorimetric
+
+    ``3dlut-size=<r>x<g>x<b>``
+        Size of the 3D LUT generated from the ICC profile in each dimension.
+        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 ``linear-scaling``.
+        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).
+
+        blend
+            Blend the frame against a black background.
+        yes
+            Try to create a framebuffer with alpha component. This only makes sense
+            if the video contains alpha information (which is extremely rare). May
+            not be supported on all platforms. If alpha framebuffers are
+            unavailable, it silently falls back on a normal framebuffer. Note
+            that if you set the ``fbo-format`` option to a non-default value,
+            a format with alpha must be specified, or this won't work.
+        no
+            Ignore alpha component.
+
+    ``rectangle-textures``
+        Force use of rectangle textures (default: no). Normally this shouldn't
+        have any advantages over normal textures. Note that hardware decoding
+        overrides this flag.
+
+    ``background=<color>``
+        Color used to draw parts of the mpv window not covered by video.
+        See ``--osd-color`` option how colors are defined.
+
+``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
+
+    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
@@ -318,12 +764,10 @@ Available video output drivers are:
     .. note:: This driver is for compatibility with systems that don't provide
               proper graphics drivers, or which support GLES only.
 
-    The following global options are supported by this video output:
-
-    ``--sdl-sw``
+    ``sw``
         Continue even if a software renderer is detected.
 
-    ``--sdl-switch-mode``
+    ``switch-mode``
         Instruct SDL to switch the monitor video mode when going fullscreen.
 
 ``vaapi``
@@ -335,9 +779,7 @@ Available video output drivers are:
     .. note:: This driver is for compatibility with crappy systems. You can
               use vaapi hardware decoding with ``--vo=opengl`` too.
 
-    The following global options are supported by this video output:
-
-    ``--vo-vaapi-scaling=<algorithm>``
+    ``scaling=<algorithm>``
         default
             Driver default (mpv default as well).
         fast
@@ -347,9 +789,9 @@ Available video output drivers are:
         nla
             ``non-linear anamorphic scaling``
 
-    ``--vo-vaapi-deint-mode=<mode>``
+    ``deint-mode=<mode>``
         Select deinterlacing algorithm. Note that by default deinterlacing is
-        initially always off, and needs to be enabled with the ``d`` key
+        initially always off, and needs to be enabled with the ``D`` key
         (default key binding for ``cycle deinterlace``).
 
         This option doesn't apply if libva supports video post processing (vpp).
@@ -366,7 +808,7 @@ Available video output drivers are:
         bob
             bob deinterlacing (default for older libva).
 
-    ``--vo-vaapi-scaled-osd=<yes|no>``
+    ``scaled-osd=<yes|no>``
         If enabled, then the OSD is rendered at video resolution and scaled to
         display resolution. By default, this is disabled, and the OSD is
         rendered at display resolution if the driver supports it.
@@ -374,14 +816,6 @@ Available video output drivers are:
 ``null``
     Produces no video output. Useful for benchmarking.
 
-    Usually, it's better to disable video with ``--no-video`` instead.
-
-    The following global options are supported by this video output:
-
-    ``--vo-null-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.
 
@@ -391,9 +825,7 @@ Available video output drivers are:
     Output each frame into an image file in the current directory. Each file
     takes the frame number padded with leading zeros as name.
 
-    The following global options are supported by this video output:
-
-    ``--vo-image-format=<format>``
+    ``format=<format>``
         Select the image file format.
 
         jpg
@@ -411,24 +843,24 @@ Available video output drivers are:
         tga
             Truevision TGA.
 
-    ``--vo-image-png-compression=<0-9>``
+    ``png-compression=<0-9>``
         PNG compression factor (speed vs. file size tradeoff) (default: 7)
-    ``--vo-image-png-filter=<0-5>``
+    ``png-filter=<0-5>``
         Filter applied prior to PNG compression (0 = none; 1 = sub; 2 = up;
         3 = average; 4 = Paeth; 5 = mixed) (default: 5)
-    ``--vo-image-jpeg-quality=<0-100>``
+    ``jpeg-quality=<0-100>``
         JPEG quality factor (default: 90)
-    ``--vo-image-jpeg-progressive=<yes|no>``
+    ``(no-)jpeg-progressive``
         Specify standard or progressive JPEG (default: no).
-    ``--vo-image-jpeg-baseline=<yes|no>``
+    ``(no-)jpeg-baseline``
         Specify use of JPEG baseline or not (default: yes).
-    ``--vo-image-jpeg-optimize=<0-100>``
+    ``jpeg-optimize=<0-100>``
         JPEG optimization factor (default: 100)
-    ``--vo-image-jpeg-smooth=<0-100>``
+    ``jpeg-smooth=<0-100>``
         smooth factor (default: 0)
-    ``--vo-image-jpeg-dpi=<1->``
+    ``jpeg-dpi=<1->``
         JPEG DPI (default: 72)
-    ``--vo-image-outdir=<dirname>``
+    ``outdir=<dirname>``
         Specify the directory to save the image files to (default: ``./``).
 
 ``wayland`` (Wayland only)
@@ -437,73 +869,67 @@ Available video output drivers are:
     .. note:: This driver is for compatibility with systems that don't provide
               working OpenGL drivers.
 
-    The following global options are supported by this video output:
-
-    ``--vo-wayland-alpha``
+    ``alpha``
         Use a buffer format that supports videos and images with alpha
         information
-    ``--vo-wayland-rgb565``
+    ``rgb565``
         Use RGB565 as buffer format. This format is implemented on most
         platforms, especially on embedded where it is far more efficient then
         RGB8888.
-    ``--vo-wayland-triple-buffering``
+    ``triple-buffering``
         Use 3 buffers instead of 2. This can lead to more fluid playback, but
         uses more memory.
 
 ``opengl-cb``
     For use with libmpv direct OpenGL embedding; useless in any other contexts.
     (See ``<mpv/opengl_cb.h>``.)
+    Usually, ``opengl-cb`` renders frames asynchronously by client and this
+    can cause some frame drops. In order to provide a way to handle this
+    situation, ``opengl-cb`` has its own frame queue and calls update callback
+    more frequently if the queue is not empty regardless of existence of new frame.
+    Once the queue is filled, ``opengl-cb`` drops frames automatically.
 
-    This also supports many of the options the ``opengl`` VO has.
+    With default options, ``opengl-cb`` renders only the latest frame and drops
+    all frames handed over while waiting render function after update callback.
+
+    ``frame-queue-size=<1..100>``
+        The maximum count of frames which the frame queue can hold (default: 1)
+
+    ``frame-drop-mode=<pop|clear>``
+        Select the behavior when the frame queue is full.
+
+        pop
+            Drop the oldest frame in the frame queue. (default)
+        clear
+            Drop all frames in the frame queue.
+
+    This also supports many of the suboptions the ``opengl`` VO has. Run
+    ``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.
 
-    This is deprecated. Use ``--vo=opengl`` instead, which is the default and
-    provides the same functionality. The ``rpi`` VO will be removed in
-    mpv 0.22.0. Its functionality was folded into --vo=opengl, which now uses
-    RPI hardware decoding by treating it as a hardware overlay (without applying
-    GL filtering). Also to be changed in 0.22.0: the --fs flag will be reset to
-    "no" by default (like on the other platforms).
-
-    The following deprecated global options are supported by this video output:
-
-    ``--rpi-display=<number>``
+    ``display=<number>``
         Select the display number on which the video overlay should be shown
         (default: 0).
 
-    ``--rpi-layer=<number>``
+    ``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.
 
-    ``--rpi-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.
-
-    ``--rpi-osd=<yes|no>``
-        Enabled by default. If disabled with ``no``, no OSD layer is created.
-        This also means there will be no subtitles rendered.
-
 ``drm`` (Direct Rendering Manager)
     Video output driver using Kernel Mode Setting / Direct Rendering Manager.
-    Should be used when one doesn't want to install full-blown graphical
-    environment (e.g. no X). Does not support hardware acceleration (if you
-    need this, check the ``drm-egl`` backend for ``opengl`` VO).
+    Does not support hardware acceleration. Should be used when one doesn't
+    want to install full-blown graphical environment (e.g. no X).
 
-    The following global options are supported by this video output:
-
-    ``--drm-connector=[<gpu_number>.]<name>``
-        Select the connector to use (usually this is a monitor.) If ``<name>``
-        is empty or ``auto``, mpv renders the output on the first available
-        connector. Use ``--drm-connector=help`` to get list of available
-        connectors. When using multiple graphic cards, use the ``<gpu_number>``
-        argument to disambiguate.
-        (default: empty)
-
-    ``--drm-mode=<number>``
-        Mode ID to use (resolution, bit depth and frame rate).
-        (default: 0)
+    ``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)

DOCS/mplayer-changes.rst

@@ -7,8 +7,8 @@ large part of which is incompatible or completely changes how the player
 behaves. Although there are still many similarities to its ancestors, **mpv**
 should generally be treated as a completely different program.
 
-.. admonition:: Warning
-    This document is not updated anymore, and is incomplete and outdated.
+.. note::
+    These lists are incomplete.
 
 General Changes from MPlayer to mpv
 -----------------------------------
@@ -30,7 +30,7 @@ Player
 * Slave mode compatibility broken (see below).
 * Re-enable screensaver while the player is paused.
 * Allow resuming playback at a later point with ``Shift+q``, also see the
-  ``quit-watch-later`` input command.
+  ``quit_watch_later`` input command.
 * ``--keep-open`` option to stop the player from closing the window and
   exiting after playback ends.
 * A client API, that allows embedding **mpv** into applications
@@ -52,7 +52,6 @@ Input
   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
 ~~~~~
@@ -79,8 +78,7 @@ Video
   using most of FFmpeg's filters, which improve greatly on the old MPlayer
   filters in features, performance, and correctness.
 * More correct color reproduction (color matrix generation), including support
-  for BT.2020 (Ultra HD). linear XYZ (Digital Cinema) and SMPTE ST2084 (HDR)
-  inputs.
+  for BT.2020 (Ultra HD) and linear XYZ (Digital Cinema) inputs.
 * Support for color managed displays, via ICC profiles.
 * High-quality image resamplers (see the ``opengl`` ``scale`` suboption).
 * Support for scaling in (sigmoidized) linear light.
@@ -95,8 +93,6 @@ 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
 ~~~~~~~~~~~~~~~~
@@ -262,14 +258,13 @@ Command Line Switches
     ``-ssf <sub>``              ``--sws-...``
     ``-stop-xscreensaver``      ``--stop-screensaver``
     ``-sub-fuzziness``          ``--sub-auto``
-    ``-sub-text-*``             ``--sub-*``
     ``-sub``                    ``--sub-file``
     ``-subcp``                  ``--sub-codepage``
     ``-subdelay``               ``--sub-delay``
     ``-subfile``                ``--sub-file``
-    ``-subfont-*``              ``--sub-*``, ``--osd-*``
+    ``-subfont-*``              ``--sub-text-*``, ``--osd-*``
     ``-subfont-text-scale``     ``--sub-scale``
-    ``-subfont``                ``--sub-font``
+    ``-subfont``                ``--sub-text-font``
     ``-subfps``                 ``--sub-fps``
     ``-subpos``                 ``--sub-pos``
     ``-sws``                    ``--sws-scaler``
@@ -281,7 +276,7 @@ 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)
     =========================== ========================================
 
@@ -313,11 +308,11 @@ input.conf and Slave Commands
     +--------------------------------+----------------------------------------+
     | Old                            | New                                    |
     +================================+========================================+
-    | ``pt_step 1 [0|1]``            | ``playlist-next [weak|force]``         |
+    | ``pt_step 1 [0|1]``            | ``playlist_next [weak|force]``         |
     |                                | (translation layer cannot deal with    |
     |                                | whitespace)                            |
     +--------------------------------+----------------------------------------+
-    | ``pt_step -1 [0|1]``           | ``playlist-prev [weak|force] (same)``  |
+    | ``pt_step -1 [0|1]``           | ``playlist_prev [weak|force] (same)``  |
     +--------------------------------+----------------------------------------+
     | ``switch_ratio [<ratio>]``     | ``set video-aspect <ratio>``           |
     |                                |                                        |
@@ -332,7 +327,7 @@ input.conf and Slave Commands
     | ``<step> <dir>``               | ``no-osd``: ``no-osd cycle <prop>``    |
     |                                | ``<step>``                             |
     +--------------------------------+----------------------------------------+
-    | ``osd_show_property_text``     | ``show-text <text>``                   |
+    | ``osd_show_property_text``     | ``show_text <text>``                   |
     | ``<text>``                     | The property expansion format string   |
     |                                | syntax slightly changed.               |
     +--------------------------------+----------------------------------------+
@@ -341,9 +336,9 @@ input.conf and Slave Commands
     |                                | ``raw`` prefix to disable property     |
     |                                | expansion.                             |
     +--------------------------------+----------------------------------------+
-    | ``show_tracks``                | ``show-text ${track-list}``            |
+    | ``show_tracks``                | ``show_text ${track-list}``            |
     +--------------------------------+----------------------------------------+
-    | ``show_chapters``              | ``show-text ${chapter-list}``          |
+    | ``show_chapters``              | ``show_text ${chapter-list}``          |
     +--------------------------------+----------------------------------------+
     | ``af_switch``, ``af_add``, ... | ``af set|add|...``                     |
     +--------------------------------+----------------------------------------+
@@ -401,8 +396,7 @@ 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

@@ -1,107 +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` and `DOCS/interface-changes.rst`
-  (in particular, update the last version numbers if necessary)
+- 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.
 ```
 
@@ -109,6 +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.
@@ -123,7 +124,7 @@ options/options.h, options/options.c
     parser-mpcmd.c, and uses the option table in options.c.
 
 input/input.c:
-    This translates keyboard input coming from VOs and other sources (such
+    This translates keyboard input comming from VOs and other sources (such
     as remote control devices like Apple IR or client API commands) to the
     key bindings listed in the user's (or the builtin) input.conf and turns
     them into items of type struct mp_cmd. These commands are queued, and read

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
@@ -65,7 +65,7 @@ This defines a feature called ``vdpau`` which can be enabled or disabled by
 the users with configure flags (that's the meaning of ``--``). This feature
 depends on another feature whose name is ``x11``, and the autodetection check
 consists of running ``pkg-config`` and looking for ``vdpau`` with version
-``>= 0.2``. If the check 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

@@ -26,21 +26,6 @@ variety of video file formats, audio and video codecs, and subtitle types.
 
 Releases can be found on the [release list][releases].
 
-## System requirements
-
-- A not too ancient Linux, or Windows Vista or later, or OSX 10.8 or later.
-- A somewhat capable CPU. Hardware decoding might sometimes help if the CPU
-  is too slow to decode video realtime, but must be explicitly enabled with
-  the `--hwdec` option. On Windows, a CPU with SSE4 instruction set is required
-  to get decent hardware decoding performance.
-- A not too crappy GPU. mpv is not intended to be used with bad GPUs. There are
-  many caveats with drivers or system compositors causing tearing, stutter,
-  etc. On Windows, you might want to make sure the graphics drivers are
-  current, especially OpenGL. In some cases, ancient fallback video output
-  methods can help (such as `--vo=xv` on Linux), but this use is not
-  recommended or supported.
-
-
 ## Downloads
 
 
@@ -72,6 +57,9 @@ 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
@@ -85,7 +73,7 @@ Essential dependencies (incomplete list):
 - libass (OSD, OSC, text subtitles)
 - Lua (optional, required for the OSC pseudo-GUI and youtube-dl integration)
 - libjpeg (optional, used for screenshots only)
-- uchardet (optional, for subtitle charset detection)
+- Enca (optional, for subtitle charset detection)
 - vdpau and vaapi libraries for hardware decoding on Linux (optional)
 
 Libass dependencies:
@@ -111,10 +99,13 @@ the separately available build wrapper ([mpv-build][mpv-build]) that first compi
 libraries and libass, and then compiles the player statically linked against
 those.
 
+If you are running Mac OSX and using homebrew we provide [homebrew-mpv][homebrew-mpv], an up
+to date formula that compiles mpv with sensible dependencies and defaults for
+OSX.
+
 If you want to build a Windows binary, you either have to use MSYS2 and MinGW,
 or cross-compile from Linux with MinGW. See [Windows compilation][windows_compilation].
 
-
 ## FFmpeg vs. Libav
 
 
@@ -123,42 +114,20 @@ of both FFmpeg and Libav. But FFmpeg is preferred, and some mpv features work
 with FFmpeg only (subtitle formats in particular).
 
 
-## Preferred FFmpeg version
-
-
-Using the latest FFmpeg release (or FFmpeg git master) is strongly recommended.
-Older versions are unsupported, even if the build system still happens to
-accept them. The main reason mpv still builds with older FFmpeg versions is to
-evade arguing with people (users, distros) who insist on using older FFmpeg
-versions for no rational reason.
-
-If you want to use a stable FFmpeg release, use the latest release, which has
-most likely the best maintenance out of all stable releases. Older releases
-are for distros, and at best receive basic changes like fixing critical security
-issues or build fixes, and at worst are completely abandoned.
-
-## FFmpeg ABI compatibility
-
-mpv does not support linking against FFmpeg versions it was not built with, even
-if the linked version is supposedly ABI-compatible with the version it was
-compiled against. Expect malfunctions, crashes, and security issues if you
-do it anyway.
-
-The reason for not supporting this is because it creates far too much complexity
-with little to no benefit, coupled with absurd and unusable FFmpeg API
-artifacts.
-
-Newer mpv versions will refuse to start if runtime and compile time FFmpeg
-library versions mismatch.
-
 ## Release cycle
 
-Every other month, an arbitrary git snapshot is made, and is assigned
-a 0.X.0 version number. No further maintenance is done.
 
-The goal of releases is to make Linux distributions happy. Linux distributions
-are also expected to apply their own patches in case of bugs and security
-issues.
+Every few months, a new release is cut off of the master branch and is assigned
+a 0.X.0 version number.
+
+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 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.
 
@@ -205,9 +174,11 @@ list of changes is located [here][mplayer-changes].
 Most activity happens on the IRC channel and the github issue tracker. The
 mailing lists are mostly unused.
 
- - **GitHub issue tracker**: [issue tracker][issue-tracker] (report bugs here)
+ - **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`
+ - **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.
@@ -216,12 +187,14 @@ 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
+## License 
 
 
-Mostly GPLv2 or later. See [details.](https://github.com/mpv-player/mpv/blob/master/Copyright)
+[GPLv2](https://github.com/mpv-player/mpv/blob/master/LICENSE)

TOOLS/appveyor-build.sh

@@ -1,22 +0,0 @@
-#!/usr/bin/bash
-set -e
-
-export DEST_OS=win32
-export CC=gcc
-export PKG_CONFIG=/usr/bin/pkg-config
-export PERL=/usr/bin/perl
-export PYTHON=/usr/bin/python3
-
-"$PYTHON" bootstrap.py
-"$PYTHON" waf configure \
-    --check-c-compiler=gcc \
-    --disable-cdda \
-    --enable-egl-angle \
-    --enable-jpeg \
-    --enable-lcms2 \
-    --enable-libarchive \
-    --enable-libass \
-    --enable-lua \
-    --enable-rubberband \
-    --enable-uchardet
-"$PYTHON" waf build

TOOLS/appveyor-install.sh

@@ -1,35 +0,0 @@
-#!/usr/bin/bash
-set -e
-
-case $MSYSTEM in
-MINGW32)
-    export MINGW_PACKAGE_PREFIX=mingw-w64-i686
-    ;;
-MINGW64)
-    export MINGW_PACKAGE_PREFIX=mingw-w64-x86_64
-    ;;
-esac
-
-# Write an empty fonts.conf to speed up fc-cache
-export FONTCONFIG_FILE=/dummy-fonts.conf
-cat >"$FONTCONFIG_FILE" <<EOF
-<?xml version="1.0"?>
-<!DOCTYPE fontconfig SYSTEM "fonts.dtd">
-<fontconfig></fontconfig>
-EOF
-
-# Install build dependencies for mpv
-pacman -S --noconfirm --needed \
-    $MINGW_PACKAGE_PREFIX-gcc \
-    $MINGW_PACKAGE_PREFIX-angleproject-git \
-    $MINGW_PACKAGE_PREFIX-ffmpeg \
-    $MINGW_PACKAGE_PREFIX-lcms2 \
-    $MINGW_PACKAGE_PREFIX-libarchive \
-    $MINGW_PACKAGE_PREFIX-libass \
-    $MINGW_PACKAGE_PREFIX-libjpeg-turbo \
-    $MINGW_PACKAGE_PREFIX-lua51 \
-    $MINGW_PACKAGE_PREFIX-rubberband \
-    $MINGW_PACKAGE_PREFIX-uchardet-git
-
-# Delete unused packages to reduce space used in the Appveyor cache
-pacman -Sc --noconfirm

TOOLS/file2string.pl

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

TOOLS/gen-x11-icon.sh

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

TOOLS/idet.sh

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

TOOLS/lib/Parse/Matroska/Definitions.pm

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

TOOLS/lua/README.md

@@ -11,10 +11,4 @@ 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.
 
-This is only a small selection of internally maintained scripts. Some of them
-are just for testing mpv internals, or serve as examples. An extensive
-user-edited list of 3rd party scripts is available here:
-
-    https://github.com/mpv-player/mpv/wiki/User-Scripts
-
-(Anyone can add their own scripts to that list.)
+Some of these are just for testing mpv internals.

TOOLS/lua/ao-null-reload.lua

@@ -1,20 +0,0 @@
--- Handles the edge case where previous attempts to init audio have failed, but
--- might start working due to a newly added device. This is required in
--- particular for ao=wasapi, since the internal IMMNotificationClient code that
--- normally triggers ao-reload will not be running in this case.
-
-function do_reload()
-    mp.command("ao-reload")
-    reloading = nil
-end
-
-function on_audio_device_list_change()
-    if mp.get_property("current-ao") == "null" and not reloading then
-        mp.msg.verbose("audio-device-list changed: reloading audio")
-        -- avoid calling ao-reload too often
-        reloading = mp.add_timeout(0.5, do_reload)
-    end
-end
-
-mp.set_property("options/audio-fallback-to-null", "yes")
-mp.observe_property("audio-device-list", "native", on_audio_device_list_change)

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

@@ -4,8 +4,8 @@
 -- alphabetically, and adds entries before and after the current file to
 -- the internal playlist. (It stops if the it would add an already existing
 -- playlist entry at the same position - this makes it "stable".)
--- Add at most 5000 * 2 files when starting a file (before + after).
-MAXENTRIES = 5000
+-- Add at most 5 * 2 files when starting a file (before + after).
+MAXENTRIES = 5
 
 function Set (t)
     local set = {}
@@ -25,17 +25,12 @@ function add_files_at(index, files)
     local oldcount = mp.get_property_number("playlist-count", 1)
     for i = 1, #files do
         mp.commandv("loadfile", files[i], "append")
-        mp.commandv("playlist-move", oldcount + i - 1, index + i - 1)
+        mp.commandv("playlist_move", oldcount + i - 1, index + i - 1)
     end
 end
 
 function get_extension(path)
-    match = string.match(path, "%.([^%.]+)$" )
-    if match == nil then
-        return "nomatch"
-    else
-        return match
-    end
+    return string.match(path, "%.([^%.]+)$" )
 end
 
 table.filter = function(t, iter)
@@ -52,13 +47,6 @@ function find_and_add_entries()
     if #dir == 0 then
         return
     end
-    local pl_count = mp.get_property_number("playlist-count", 1)
-    if (pl_count > 1 and autoload == nil) or
-       (pl_count == 1 and EXTENSIONS[string.lower(get_extension(filename))] == nil) then
-        return
-    else
-        autoload = true
-    end
 
     local files = mputils.readdir(dir, "files")
     if files == nil then
@@ -72,11 +60,6 @@ function find_and_add_entries()
         return EXTENSIONS[string.lower(ext)]
     end)
     table.sort(files, function (a, b)
-        local len = string.len(a) - string.len(b)
-        if len ~= 0 then -- case for ordering filename ending with such as X.Y.Z
-            local ext = string.len(get_extension(a)) + 1
-            return string.sub(a, 1, -ext) < string.sub(b, 1, -ext)
-        end
         return string.lower(a) < string.lower(b)
     end)
 

TOOLS/lua/observe-all.lua

@@ -6,17 +6,9 @@
 
 local utils = require("mp.utils")
 
-function observe(name)
+for i,name in ipairs(mp.get_property_native("property-list")) do
     mp.observe_property(name, "native", function(name, val)
         print("property '" .. name .. "' changed to '" ..
               utils.to_string(val) .. "'")
     end)
 end
-
-for i,name in ipairs(mp.get_property_native("property-list")) do
-    observe(name)
-end
-
-for i,name in ipairs(mp.get_property_native("options")) do
-    observe("options/" .. name)
-end

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/youtube-starttime.lua

@@ -0,0 +1,34 @@
+--sets the startime of a youtube video as specified in the "t=HHhMMmSSs" part of the url
+--NOTE: This might become obsolete once youtube-dl adds the functionality
+
+local msg = require 'mp.msg'
+
+function youtube_starttime()
+  url = mp.get_property("path", "")
+  start = 0
+
+  if string.find(url, "youtu%.?be") and
+    ((url:find("http://") == 1) or (url:find("https://") == 1)) then
+      time = string.match(url, "[#&%?]t=%d*h?%d*m?%d+s?m?h?")
+      --the time-string can start with #, & or ? followed by t= and the timing parameters
+      --at least one number needs to be present after t=, followed by h, m, s or nothing (>implies s)
+
+      if time then
+        for pos in string.gmatch(time,"%d+%a?") do
+          if string.match(pos,"%d+h") then            --find out multiplier for
+            multiplier = 60*60                        --hours
+          elseif string.match(pos,"%d+m") then
+            multiplier = 60                           --minutes
+          else multiplier = 1 end                     --seconds
+
+          start = start + (string.match(pos,"%d+") * multiplier)
+        end
+
+        msg.info("parsed '" .. time .. "' into '" .. start .. "' seconds")
+      end
+
+      mp.set_property("file-local-options/start",start)
+  end
+end
+
+mp.add_hook("on_load", 50, youtube_starttime)

TOOLS/osxbundle.py

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

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

@@ -1 +1,2 @@
-player-operation-mode=pseudo-gui
+screenshot-template=~/Desktop/shot%n
+profile=pseudo-gui

TOOLS/stats-conv.py

@@ -1,15 +1,11 @@
 #!/usr/bin/env python3
-from pyqtgraph.Qt import QtGui, QtCore
-import pyqtgraph as pg
+import matplotlib.pyplot as plot
 import sys
 import re
 
 filename = sys.argv[1]
 
-events = ".*"
-if len(sys.argv) > 2:
-    events = sys.argv[2]
-event_regex = re.compile(events)
+event_regex = re.compile(".*")
 
 """
 This script is meant to display stats written by mpv --dump-stats=filename.
@@ -30,8 +26,8 @@ Currently, the following event types are supported:
     'end' <name>                end of the named event
     'value' <float> <name>      a normal value (as opposed to event)
     'event-timed' <ts> <name>   singular event at the given timestamp
-    'value-timed' <ts> <float> <name>       a value for an event at the given timestamp
-    'range-timed' <ts1> <ts2> <name>        like start/end, but explicit times
+    'value-timed' <ts> <float> <name>
+                                a value for an event at the given timestamp
     <name>                      singular event (same as 'signal')
 
 """
@@ -39,8 +35,8 @@ Currently, the following event types are supported:
 class G:
     events = {}
     start = None
-    markers = ["o", "s", "t", "d"]
-    curveno = {}
+    # http://matplotlib.org/api/markers_api.html#module-matplotlib.markers
+    markers = ["o", "8", "s", "p", "*", "h", "+", "x", "D"]
 
 def find_marker():
     if len(G.markers) == 0:
@@ -66,10 +62,6 @@ def get_event(event, evtype):
         G.events[event] = e
     return G.events[event]
 
-colors = [(0.0, 0.5, 0.0), (0.0, 0.0, 1.0), (0.0, 0.0, 0.0), (1.0, 0.0, 0.0), (0.75, 0.75, 0), (0.0, 0.75, 0.75), (0.75, 0, 0.75)]
-def mkColor(t):
-    return pg.mkColor(int(t[0] * 255), int(t[1] * 255), int(t[2] * 255))
-
 SCALE = 1e6 # microseconds to seconds
 
 for line in [line.split("#")[0].strip() for line in open(filename, "r")]:
@@ -99,15 +91,6 @@ for line in [line.split("#")[0].strip() for line in open(filename, "r")]:
         val = int(val) / SCALE - G.start
         e = get_event(name, "event-signal")
         e.vals.append((val, 1))
-    elif event.startswith("range-timed "):
-        _, ts1, ts2, name = event.split(" ", 3)
-        ts1 = int(ts1) / SCALE - G.start
-        ts2 = int(ts2) / SCALE - G.start
-        e = get_event(name, "event")
-        e.vals.append((ts1, 0))
-        e.vals.append((ts1, 1))
-        e.vals.append((ts2, 1))
-        e.vals.append((ts2, 0))
     elif event.startswith("value-timed "):
         _, tsval, val, name = event.split(" ", 3)
         tsval = int(tsval) / SCALE - G.start
@@ -133,34 +116,20 @@ for e, index in zip(G.sevents, range(len(G.sevents))):
     else:
         e.vals = [(x, y * (m - index) / m) for (x, y) in e.vals]
 
-pg.setConfigOption('background', 'w')
-pg.setConfigOption('foreground', 'k')
-app = QtGui.QApplication([])
-win = pg.GraphicsWindow()
-#win.resize(1500, 900)
-
+fig = plot.figure()
+fig.hold(True)
 ax = [None, None]
 plots = 2 if hasval else 1
-ax[0] = win.addPlot()
+ax[0] = fig.add_subplot(plots, 1, 1)
 if hasval:
-    win.nextRow()
-    ax[1] = win.addPlot()
-    ax[1].setXLink(ax[0])
-for cur in ax:
-   if cur is not None:
-       cur.addLegend(offset = (-1, 1))
+    ax[1] = fig.add_subplot(plots, 1, 2, sharex=ax[0])
+legends = [[], []]
 for e in G.sevents:
     cur = ax[1 if e.type == "value" else 0]
-    if not cur in G.curveno:
-        G.curveno[cur] = 0
-    args = {'name': e.name,'antialias':True}
-    color = mkColor(colors[G.curveno[cur] % len(colors)])
+    pl, = cur.plot([x for x,y in e.vals], [y for x,y in e.vals], label=e.name)
     if e.type == "event-signal":
-        args['symbol'] = e.marker
-        args['symbolBrush'] = pg.mkBrush(color, width=0)
-    else:
-        args['pen'] = pg.mkPen(color, width=0)
-    G.curveno[cur] += 1
-    n = cur.plot([x for x,y in e.vals], [y for x,y in e.vals], **args)
-
-QtGui.QApplication.instance().exec_()
+        plot.setp(pl, marker = e.marker, linestyle = "None")
+for cur in ax:
+    if cur is not None:
+        cur.legend()
+plot.show()

TOOLS/travis-rebuild-website

@@ -1,29 +0,0 @@
-#!/bin/sh
-
-if [ "x$LIBAV" != "xffmpeg-git" ] || [ "x$TRAVIS_OS_NAME" != "xlinux" ]; then
-  # trigger build only on one of the matrix nodes
-  exit;
-fi
-
-if [ "x$TRAVIS_BRANCH" != "xmaster" ]; then
-  # only rebuild website with pushes to master
-  exit;
-fi
-
-TRAVIS_COMMIT_RANGE=${TRAVIS_COMMIT_RANGE:-HEAD^..}
-if git diff --quiet --name-only --exit-code "$TRAVIS_COMMIT_RANGE" DOCS/; then
-  exit;
-fi
-
-body='{
-"request": {
-  "branch":"master"
-}}'
-
-curl -s -X POST \
--H "Content-Type: application/json" \
--H "Accept: application/json" \
--H "Travis-API-Version: 3" \
--H "Authorization: token $WEBSITE_TRAVIS_TOKEN" \
--d "$body" \
-https://api.travis-ci.org/repo/mpv-player%2Fmpv.io/requests

TOOLS/zsh.pl

@@ -4,14 +4,11 @@
 
 use strict;
 use warnings;
-use warnings FATAL => 'uninitialized';
 
 my $mpv = $ARGV[0] || 'mpv';
 
 my @opts = parse_main_opts('--list-options', '^ (\-\-[^\s\*]*)\*?\s*(.*)');
 
-die "Couldn't find any options" unless (@opts);
-
 my @ao = parse_opts('--ao=help', '^  ([^\s\:]*)\s*: (.*)');
 my @vo = parse_opts('--vo=help', '^  ([^\s\:]*)\s*: (.*)');
 
@@ -39,8 +36,7 @@ chomp $vf_str;
 
 $protos_str = join(' ', @protos);
 
-my $runtime_completions = <<'EOS';
-  profile|show-profile)
+my $profile_comp = <<'EOS';
     local -a profiles
     local current
     for current in "${(@f)$($words[1] --profile=help)}"; do
@@ -67,40 +63,17 @@ my $runtime_completions = <<'EOS';
       profiles=($profiles 'help[list profiles]')
       _values -s , 'profile(s)' $profiles && rc=0
     fi
-  ;;
-
-  audio-device)
-    local -a audio_devices
-    local current
-    for current in "${(@f)$($words[1] --audio-device=help)}"; do
-      current=${current//\*/\\\*}
-      current=${current//\:/\\\:}
-      current=${current//\[/\\\[}
-      current=${current//\]/\\\]}
-      if [[ $current =~ '  '\'([^\']*)\'' \('(.*)'\)' ]]; then
-        audio_devices=($audio_devices "$match[1][$match[2]]")
-      fi
-    done
-    audio_devices=($audio_devices 'help[list audio devices]')
-    _values 'audio device' $audio_devices && rc=0
-  ;;
 EOS
-chomp $runtime_completions;
+chomp $profile_comp;
 
 my $tmpl = <<"EOS";
 #compdef mpv
 
-# For customization, see:
-#  https://github.com/mpv-player/mpv/wiki/Zsh-completion-customization
+# mpv zsh completion
 
 local curcontext="\$curcontext" state state_descr line
 typeset -A opt_args
 
-# By default, don't complete URLs unless no files match
-local -a tag_order
-zstyle -a ":completion:*:*:\$service:*" tag-order tag_order || \
-  zstyle  ":completion:*:*:\$service:*" tag-order '!urls'
-
 local rc=1
 
 _arguments -C -S \\
@@ -132,7 +105,9 @@ $vf_str
     && rc=0
   ;;
 
-$runtime_completions
+  profile|show-profile)
+$profile_comp
+  ;;
 
   files)
     compset -P '*,'
@@ -144,7 +119,8 @@ $runtime_completions
     local expl
     _tags files urls
     while _tags; do
-      _requested files expl 'media file' _files && rc=0
+      _requested files expl 'media file' _files -g \\
+        "*.(#i)(asf|asx|avi|flac|flv|m1v|m2p|m2v|m4v|mjpg|mka|mkv|mov|mp3|mp4|mpe|mpeg|mpg|ogg|ogm|ogv|qt|rm|ts|vob|wav|webm|wma|wmv)(-.)" && rc=0
       if _requested urls; then
         while _next_label urls expl URL; do
           _urls "\$expl[@]" && rc=0
@@ -165,7 +141,7 @@ sub parse_main_opts {
     my ($cmd, $regex) = @_;
 
     my @list;
-    my @lines = call_mpv($cmd);
+    my @lines = split /\n/, `"$mpv" --no-config $cmd`;
 
     foreach my $line (@lines) {
         my ($name, $desc) = ($line =~ /^$regex/) or next;
@@ -207,7 +183,7 @@ sub parse_main_opts {
                 }
             } elsif ($line =~ /\[file\]/) {
                 $entry .= '->files';
-            } elsif ($name =~ /^--(ao|vo|af|vf|profile|show-profile|audio-device)$/) {
+            } elsif ($name =~ /^--(ao|vo|af|vf|profile|show-profile)$/) {
                 $entry .= "->$1";
             }
             push @list, $entry;
@@ -230,7 +206,7 @@ sub parse_opts {
     my ($cmd, $regex) = @_;
 
     my @list;
-    my @lines = call_mpv($cmd);
+    my @lines = split /\n/, `"$mpv" --no-config $cmd`;
 
     foreach my $line (@lines) {
         if ($line !~ /^$regex/) {
@@ -250,14 +226,3 @@ 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 @@
-0.21.0
+0.9.2

appveyor.yml

@@ -1,29 +0,0 @@
-branches:
-  only:
-    - master
-    - appveyor
-
-environment:
-  matrix:
-    - MSYSTEM: MINGW64
-    - MSYSTEM: MINGW32
-
-shallow_clone: true
-test: off
-
-install:
-  # Update core packages
-  - C:\msys64\usr\bin\pacman -Syyuu --noconfirm
-  # Update non-core packages
-  - C:\msys64\usr\bin\pacman -Suu --noconfirm
-  # Install required MSYS2 packages
-  - C:\msys64\usr\bin\pacman -S --noconfirm --needed perl python pkg-config
-  # Now MSYS2 is up to date, do the rest of the install from a bash script
-  - C:\msys64\usr\bin\bash -lc "cd \"$APPVEYOR_BUILD_FOLDER\" && exec ./TOOLS/appveyor-install.sh"
-build_script:
-  - C:\msys64\usr\bin\bash -lc "cd \"$APPVEYOR_BUILD_FOLDER\" && exec ./TOOLS/appveyor-build.sh"
-on_failure:
-  - type "%APPVEYOR_BUILD_FOLDER%\build\config.log"
-
-cache:
-  - C:\msys64\var\cache\pacman\pkg

audio/audio.c

@@ -22,10 +22,9 @@
 
 #include <libavutil/buffer.h>
 #include <libavutil/frame.h>
-#include <libavutil/mem.h>
 #include <libavutil/version.h>
 
-#include "mpv_talloc.h"
+#include "talloc.h"
 #include "common/common.h"
 #include "fmt-conversion.h"
 #include "audio.h"
@@ -35,8 +34,8 @@ static void update_redundant_info(struct mp_audio *mpa)
     assert(mp_chmap_is_empty(&mpa->channels) ||
            mp_chmap_is_valid(&mpa->channels));
     mpa->nch = mpa->channels.num;
-    mpa->bps = af_fmt_to_bytes(mpa->format);
-    if (af_fmt_is_planar(mpa->format)) {
+    mpa->bps = af_fmt2bps(mpa->format);
+    if (AF_FORMAT_IS_PLANAR(mpa->format)) {
         mpa->spf = 1;
         mpa->num_planes = mpa->nch;
         mpa->sstride = mpa->bps;
@@ -59,6 +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;
@@ -87,19 +93,15 @@ 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;
 }
 
 void mp_audio_force_interleaved_format(struct mp_audio *mpa)
 {
-    if (af_fmt_is_planar(mpa->format))
+    if (AF_FORMAT_IS_PLANAR(mpa->format))
         mp_audio_set_format(mpa, af_fmt_from_planar(mpa->format));
 }
 
@@ -116,6 +118,7 @@ void mp_audio_set_null_data(struct mp_audio *mpa)
         mpa->allocated[n] = NULL;
     }
     mpa->samples = 0;
+    mpa->readonly = false;
 }
 
 static int get_plane_size(const struct mp_audio *mpa, int samples)
@@ -138,9 +141,6 @@ static void mp_audio_destructor(void *ptr)
  * available on every plane. The previous data is kept (for the smallest
  * common number of samples before/after resize).
  *
- * This also makes sure the resulting buffer is writable (even in the case
- * the buffer has the correct size).
- *
  * mpa->samples is not set or used.
  *
  * This function is flexible enough to handle format and channel layout
@@ -156,18 +156,12 @@ void mp_audio_realloc(struct mp_audio *mpa, int samples)
     int size = get_plane_size(mpa, samples);
     if (size < 0)
         abort(); // oom or invalid parameters
-    if (!mp_audio_is_writeable(mpa)) {
-        for (int n = 0; n < MP_NUM_CHANNELS; n++) {
-            av_buffer_unref(&mpa->allocated[n]);
-            mpa->planes[n] = NULL;
-        }
-    }
     for (int n = 0; n < mpa->num_planes; n++) {
         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]);
@@ -180,7 +174,7 @@ void mp_audio_realloc(struct mp_audio *mpa, int samples)
 // If the buffer is reallocated, also preallocate.
 void mp_audio_realloc_min(struct mp_audio *mpa, int samples)
 {
-    if (samples > mp_audio_get_allocated_size(mpa) || !mp_audio_is_writeable(mpa)) {
+    if (samples > mp_audio_get_allocated_size(mpa)) {
         size_t alloc = ta_calc_prealloc_elems(samples);
         if (alloc > INT_MAX)
             abort(); // oom
@@ -198,18 +192,8 @@ 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;
 }
@@ -261,56 +245,13 @@ void mp_audio_skip_samples(struct mp_audio *data, int samples)
         data->planes[n] = (uint8_t *)data->planes[n] + samples * data->sstride;
 
     data->samples -= samples;
-
-    if (data->pts != MP_NOPTS_VALUE)
-        data->pts += samples / (double)data->rate;
 }
 
-// Return the timestamp of the sample just after the end of this frame.
-double mp_audio_end_pts(struct mp_audio *f)
-{
-    if (f->pts == MP_NOPTS_VALUE || f->rate < 1)
-        return MP_NOPTS_VALUE;
-    return f->pts + f->samples / (double)f->rate;
-}
-
-// Clip the given frame to the given timestamp range. Adjusts the frame size
-// and timestamp.
-void mp_audio_clip_timestamps(struct mp_audio *f, double start, double end)
-{
-    double f_end = mp_audio_end_pts(f);
-    if (f_end == MP_NOPTS_VALUE)
-        return;
-    if (end != MP_NOPTS_VALUE) {
-        if (f_end >= end) {
-            if (f->pts >= end) {
-                f->samples = 0;
-            } else {
-                int new = (end - f->pts) * f->rate;
-                f->samples = MPCLAMP(new, 0, f->samples);
-            }
-        }
-    }
-    if (start != MP_NOPTS_VALUE) {
-        if (f->pts < start) {
-            if (f_end <= start) {
-                f->samples = 0;
-                f->pts = f_end;
-            } else {
-                int skip = (start - f->pts) * f->rate;
-                skip = MPCLAMP(skip, 0, f->samples);
-                mp_audio_skip_samples(f, skip);
-            }
-        }
-    }
-}
-
-
 // Return false if the frame data is shared, true otherwise.
 // Will return true for non-refcounted frames.
 bool mp_audio_is_writeable(struct mp_audio *data)
 {
-    bool ok = true;
+    bool ok = !data->readonly;
     for (int n = 0; n < MP_NUM_CHANNELS; n++) {
         if (data->allocated[n])
             ok &= av_buffer_is_writable(data->allocated[n]);
@@ -356,9 +297,9 @@ struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe)
     mp_chmap_from_lavc(&lavc_chmap, avframe->channel_layout);
 
 #if LIBAVUTIL_VERSION_MICRO >= 100
-    // FFmpeg being stupid POS again
-    if (lavc_chmap.num != av_frame_get_channels(avframe))
-        mp_chmap_from_channels(&lavc_chmap, av_frame_get_channels(avframe));
+    // FFmpeg being special again
+    if (lavc_chmap.num != avframe->channels)
+        mp_chmap_from_channels(&lavc_chmap, avframe->channels);
 #endif
 
     new->rate = avframe->sample_rate;
@@ -376,23 +317,21 @@ struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe)
     }
 
     // If we can't handle the format (e.g. too many channels), bail out.
-    if (!mp_audio_config_valid(new))
+    if (!mp_audio_config_valid(new) || avframe->nb_extended_buf)
         goto fail;
 
-    for (int n = 0; n < AV_NUM_DATA_POINTERS + avframe->nb_extended_buf; n++) {
-        AVBufferRef *buf = n < AV_NUM_DATA_POINTERS ? avframe->buf[n]
-                            : avframe->extended_buf[n - AV_NUM_DATA_POINTERS];
-        if (!buf)
+    for (int n = 0; n < AV_NUM_DATA_POINTERS; n++) {
+        if (!avframe->buf[n])
             break;
         if (n >= MP_NUM_CHANNELS)
             goto fail;
-        new->allocated[n] = av_buffer_ref(buf);
+        new->allocated[n] = av_buffer_ref(avframe->buf[n]);
         if (!new->allocated[n])
             goto fail;
     }
 
     for (int n = 0; n < new->num_planes; n++)
-        new->planes[n] = avframe->extended_data[n];
+        new->planes[n] = avframe->data[n];
     new->samples = avframe->nb_samples;
 
     return new;
@@ -403,92 +342,6 @@ fail:
     return NULL;
 }
 
-int mp_audio_to_avframe(struct mp_audio *frame, struct AVFrame *avframe)
-{
-    av_frame_unref(avframe);
-
-    avframe->nb_samples = frame->samples;
-    avframe->format = af_to_avformat(frame->format);
-    if (avframe->format == AV_SAMPLE_FMT_NONE)
-        goto fail;
-
-    avframe->channel_layout = mp_chmap_to_lavc(&frame->channels);
-    if (!avframe->channel_layout)
-        goto fail;
-#if LIBAVUTIL_VERSION_MICRO >= 100
-    // FFmpeg being a stupid POS again
-    av_frame_set_channels(avframe, frame->channels.num);
-#endif
-    avframe->sample_rate = frame->rate;
-
-    if (frame->num_planes > AV_NUM_DATA_POINTERS) {
-        avframe->extended_data =
-            av_mallocz_array(frame->num_planes, sizeof(avframe->extended_data[0]));
-        int extbufs = frame->num_planes - AV_NUM_DATA_POINTERS;
-        avframe->extended_buf =
-            av_mallocz_array(extbufs, sizeof(avframe->extended_buf[0]));
-        if (!avframe->extended_data || !avframe->extended_buf)
-            goto fail;
-        avframe->nb_extended_buf = extbufs;
-    }
-
-    for (int p = 0; p < frame->num_planes; p++)
-        avframe->extended_data[p] = frame->planes[p];
-    avframe->linesize[0] = frame->samples * frame->sstride;
-
-    for (int p = 0; p < AV_NUM_DATA_POINTERS; p++)
-        avframe->data[p] = avframe->extended_data[p];
-
-    for (int p = 0; p < frame->num_planes; p++) {
-        if (!frame->allocated[p])
-            break;
-        AVBufferRef *nref = av_buffer_ref(frame->allocated[p]);
-        if (!nref)
-            goto fail;
-        if (p < AV_NUM_DATA_POINTERS) {
-            avframe->buf[p] = nref;
-        } else {
-            avframe->extended_buf[p - AV_NUM_DATA_POINTERS] = nref;
-        }
-    }
-
-    // Force refcounted frame.
-    if (!avframe->buf[0]) {
-        AVFrame *tmp = av_frame_alloc();
-        if (!tmp)
-            goto fail;
-        if (av_frame_ref(tmp, avframe) < 0)
-            goto fail;
-        av_frame_free(&avframe);
-        avframe = tmp;
-    }
-
-    return 0;
-
-fail:
-    av_frame_unref(avframe);
-    return -1;
-}
-
-// Returns NULL on failure. The input is always unreffed.
-struct AVFrame *mp_audio_to_avframe_and_unref(struct mp_audio *frame)
-{
-    struct AVFrame *avframe = av_frame_alloc();
-    if (!avframe)
-        goto fail;
-
-    if (mp_audio_to_avframe(frame, avframe) < 0)
-        goto fail;
-
-    talloc_free(frame);
-    return avframe;
-
-fail:
-    av_frame_free(&avframe);
-    talloc_free(frame);
-    return NULL;
-}
-
 struct mp_audio_pool {
     AVBufferPool *avpool;
     int element_size;

audio/audio.h

@@ -35,19 +35,16 @@ struct mp_audio {
     int nch;            // number of channels (redundant with chmap)
     int spf;            // sub-samples per sample on each plane
     int num_planes;     // number of planes
-    int bps;            // size of sub-samples (af_fmt_to_bytes(format))
+    int bps;            // size of sub-samples (af_fmt2bps(format))
 
-    double pts;         // currently invalid within the filter chain
-
-    // --- private
-    // These do not necessarily map directly to planes[]. They can have
-    // different order or count. There shouldn't be more buffers than planes.
-    // If allocated[n] is NULL, allocated[n+1] must also be NULL.
+    // private
+    bool readonly;
     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);
@@ -72,16 +69,12 @@ void mp_audio_copy(struct mp_audio *dst, int dst_offset,
                    struct mp_audio *src, int src_offset, int length);
 void mp_audio_copy_attributes(struct mp_audio *dst, struct mp_audio *src);
 void mp_audio_skip_samples(struct mp_audio *data, int samples);
-void mp_audio_clip_timestamps(struct mp_audio *f, double start, double end);
-double mp_audio_end_pts(struct mp_audio *data);
 
 bool mp_audio_is_writeable(struct mp_audio *data);
 int mp_audio_make_writeable(struct mp_audio *data);
 
 struct AVFrame;
 struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe);
-struct AVFrame *mp_audio_to_avframe_and_unref(struct mp_audio *frame);
-int mp_audio_to_avframe(struct mp_audio *frame, struct AVFrame *avframe);
 
 struct mp_audio_pool;
 struct mp_audio_pool *mp_audio_pool_create(void *ta_parent);

audio/audio_buffer.c

@@ -1,18 +1,18 @@
 /*
  * This file is part of mpv.
  *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
+ * 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 Lesser General Public License for more details.
+ * GNU General Public License for more details.
  *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
  */
 
 #include <stddef.h>
@@ -119,15 +119,6 @@ void mp_audio_buffer_prepend_silence(struct mp_audio_buffer *ab, int samples)
     mp_audio_fill_silence(ab->buffer, 0, samples);
 }
 
-void mp_audio_buffer_duplicate(struct mp_audio_buffer *ab, int samples)
-{
-    assert(samples >= 0 && samples <= ab->buffer->samples);
-    int oldlen = ab->buffer->samples;
-    ab->buffer->samples += samples;
-    mp_audio_realloc_min(ab->buffer, ab->buffer->samples);
-    mp_audio_copy(ab->buffer, oldlen, ab->buffer, oldlen - samples, samples);
-}
-
 // Get the start of the current readable buffer.
 void mp_audio_buffer_peek(struct mp_audio_buffer *ab, struct mp_audio *out_mpa)
 {

audio/audio_buffer.h

@@ -1,18 +1,18 @@
 /*
  * This file is part of mpv.
  *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
+ * 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 Lesser General Public License for more details.
+ * GNU General Public License for more details.
  *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
  */
 
 #ifndef MP_AUDIO_BUFFER_H
@@ -35,7 +35,6 @@ void mp_audio_buffer_get_write_buffer(struct mp_audio_buffer *ab, int minsamples
 void mp_audio_buffer_finish_write(struct mp_audio_buffer *ab, int samples);
 void mp_audio_buffer_append(struct mp_audio_buffer *ab, struct mp_audio *mpa);
 void mp_audio_buffer_prepend_silence(struct mp_audio_buffer *ab, int samples);
-void mp_audio_buffer_duplicate(struct mp_audio_buffer *ab, int samples);
 void mp_audio_buffer_peek(struct mp_audio_buffer *ab, struct mp_audio *out_mpa);
 void mp_audio_buffer_skip(struct mp_audio_buffer *ab, int samples);
 void mp_audio_buffer_clear(struct mp_audio_buffer *ab);

audio/chmap.c

@@ -1,25 +1,23 @@
 /*
  * This file is part of mpv.
  *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
+ * 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 Lesser General Public License for more details.
+ * GNU General Public License for more details.
  *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ * 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 <libavutil/common.h>
-
 #include "common/common.h"
 #include "common/msg.h"
 #include "chmap.h"
@@ -70,7 +68,6 @@ static const char *const std_layout_names[][2] = {
     {"quad",            "fl-fr-bl-br"},
     {"quad(side)",      "fl-fr-sl-sr"},
     {"3.1",             "fl-fr-fc-lfe"},
-    {"3.1(back)",       "fl-fr-lfe-bc"}, // not in lavc
     {"5.0",             "fl-fr-fc-bl-br"},
     {"5.0(alsa)",       "fl-fr-bl-br-fc"}, // not in lavc
     {"5.0(side)",       "fl-fr-fc-sl-sr"},
@@ -84,23 +81,21 @@ static const char *const std_layout_names[][2] = {
     {"hexagonal",       "fl-fr-fc-bl-br-bc"},
     {"6.1",             "fl-fr-fc-lfe-bc-sl-sr"},
     {"6.1(back)",       "fl-fr-fc-lfe-bl-br-bc"}, // lavc calls this "6.1" too
-    {"6.1(top)",        "fl-fr-fc-lfe-bl-br-tc"}, // not in lavc
     {"6.1(front)",      "fl-fr-lfe-flc-frc-sl-sr"},
     {"7.0",             "fl-fr-fc-bl-br-sl-sr"},
     {"7.0(front)",      "fl-fr-fc-flc-frc-sl-sr"},
-    {"7.0(rear)",       "fl-fr-fc-bl-br-sdl-sdr"}, // not in lavc
     {"7.1",             "fl-fr-fc-lfe-bl-br-sl-sr"},
     {"7.1(alsa)",       "fl-fr-bl-br-fc-lfe-sl-sr"}, // not in lavc
     {"7.1(wide)",       "fl-fr-fc-lfe-bl-br-flc-frc"},
     {"7.1(wide-side)",  "fl-fr-fc-lfe-flc-frc-sl-sr"},
-    {"7.1(rear)",       "fl-fr-fc-lfe-bl-br-sdl-sdr"}, // not in lavc
+    {"7.1(rear)",       "fl-fr-fc-lfe-bl-br-sdl-sdr"},
     {"octagonal",       "fl-fr-fc-bl-br-bc-sl-sr"},
     {"downmix",         "dl-dr"},
     {"auto",            ""}, // not in lavc
     {0}
 };
 
-static const struct mp_chmap default_layouts[] = {
+static const struct mp_chmap default_layouts[MP_NUM_CHANNELS + 1] = {
     {0},                                        // empty
     MP_CHMAP_INIT_MONO,                         // mono
     MP_CHMAP2(FL, FR),                          // stereo
@@ -112,6 +107,19 @@ static const struct mp_chmap default_layouts[] = {
     MP_CHMAP8(FL, FR, FC, LFE, BL, BR, SL, SR), // 7.1
 };
 
+// The channel order was lavc/waveex, but differs from lavc for 5, 6 and 8
+// channels. 3 and 7 channels were likely undefined (no ALSA support).
+// I'm not sure about the 4 channel case: ALSA uses "quad", while the ffmpeg
+// default layout is "4.0".
+static const char *const mplayer_layouts[MP_NUM_CHANNELS + 1] = {
+    [1] = "mono",
+    [2] = "stereo",
+    [4] = "quad",
+    [5] = "5.0(alsa)",
+    [6] = "5.1(alsa)",
+    [8] = "7.1(alsa)",
+};
+
 // Returns true if speakers are mapped uniquely, and there's at least 1 channel.
 bool mp_chmap_is_valid(const struct mp_chmap *src)
 {
@@ -207,11 +215,24 @@ void mp_chmap_fill_na(struct mp_chmap *map, int num)
 // mp_chmap_is_valid(dst) will return false.
 void mp_chmap_from_channels(struct mp_chmap *dst, int num_channels)
 {
-    *dst = (struct mp_chmap) {0};
-    if (num_channels >= 0 && num_channels < MP_ARRAY_SIZE(default_layouts))
+    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
+        *dst = (struct mp_chmap) {0};
+    } else {
         *dst = default_layouts[num_channels];
-    if (!dst->num)
-        mp_chmap_set_unknown(dst, num_channels);
+    }
+}
+
+// Try to do what mplayer/mplayer2/mpv did before channel layouts were
+// introduced, i.e. get the old default channel order.
+void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels)
+{
+    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
+        *dst = (struct mp_chmap) {0};
+    } else {
+        mp_chmap_from_str(dst, bstr0(mplayer_layouts[num_channels]));
+        if (!dst->num)
+            mp_chmap_from_channels(dst, num_channels);
+    }
 }
 
 // Set *dst to an unknown layout for the given numbers of channels.
@@ -230,22 +251,54 @@ void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels)
     }
 }
 
+// Return channel index of the given speaker, or -1.
+static int mp_chmap_find_speaker(const struct mp_chmap *map, int speaker)
+{
+    for (int n = 0; n < map->num; n++) {
+        if (map->speaker[n] == speaker)
+            return n;
+    }
+    return -1;
+}
+
+static void mp_chmap_remove_speaker(struct mp_chmap *map, int speaker)
+{
+    int index = mp_chmap_find_speaker(map, speaker);
+    if (index >= 0) {
+        for (int n = index; n < map->num - 1; n++)
+            map->speaker[n] = map->speaker[n + 1];
+        map->num--;
+    }
+}
+
+// Some decoders output additional, redundant channels, which are usually
+// useless and will mess up proper audio output channel handling.
+// map: channel map from which the channels should be removed
+// requested: if not NULL, and if it contains any of the "useless" channels,
+//            don't remove them (this is for convenience)
+void mp_chmap_remove_useless_channels(struct mp_chmap *map,
+                                      const struct mp_chmap *requested)
+{
+    if (requested &&
+        mp_chmap_find_speaker(requested, MP_SPEAKER_ID_DL) >= 0)
+        return;
+
+    if (map->num > 2) {
+        mp_chmap_remove_speaker(map, MP_SPEAKER_ID_DL);
+        mp_chmap_remove_speaker(map, MP_SPEAKER_ID_DR);
+    }
+}
+
 // Return the ffmpeg/libav channel layout as in <libavutil/channel_layout.h>.
 // Speakers not representable by ffmpeg/libav are dropped.
 // Warning: this ignores the order of the channels, and will return a channel
 //          mask even if the order is different from libavcodec's.
-//          Also, "unknown" channel maps are translated to non-sense channel
-//          maps with the same number of channels.
 uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src)
 {
+    // lavc has no concept for unknown layouts yet, so pick a default
     struct mp_chmap t = *src;
-    if (t.num > 64)
-        return 0;
-    // lavc has no concept for unknown layouts yet, so pick something that does
-    // the job of signaling the number of channels, even if it makes no sense
-    // as a proper layout.
     if (mp_chmap_is_unknown(&t))
-        return t.num == 64 ? (uint64_t)-1 : (1ULL << t.num) - 1;
+        mp_chmap_from_channels(&t, t.num);
     uint64_t mask = 0;
     for (int n = 0; n < t.num; n++) {
         if (t.speaker[n] < 64) // ignore MP_SPEAKER_ID_NA etc.
@@ -301,8 +354,6 @@ bool mp_chmap_is_lavc(const struct mp_chmap *src)
     return true;
 }
 
-// Warning: for "unknown" channel maps, this returns something that may not
-//          make sense. Invalid channel maps are not changed.
 void mp_chmap_reorder_to_lavc(struct mp_chmap *map)
 {
     if (!mp_chmap_is_valid(map))
@@ -331,8 +382,8 @@ void mp_chmap_get_reorder(int src[MP_NUM_CHANNELS], const struct mp_chmap *from,
         return;
     }
 
-    for (int n = 0; n < to->num; n++) {
-        for (int i = 0; i < from->num; i++) {
+    for (int n = 0; n < from->num; n++) {
+        for (int i = 0; i < to->num; i++) {
             if (to->speaker[n] == from->speaker[i]) {
                 src[n] = i;
                 break;
@@ -344,12 +395,20 @@ void mp_chmap_get_reorder(int src[MP_NUM_CHANNELS], const struct mp_chmap *from,
         assert(src[n] < 0 || (to->speaker[n] == from->speaker[src[n]]));
 }
 
+static int popcount64(uint64_t bits)
+{
+    int r = 0;
+    for (int n = 0; n < 64; n++)
+        r += !!(bits & (1ULL << n));
+    return r;
+}
+
 // Return the number of channels only in a.
 int mp_chmap_diffn(const struct mp_chmap *a, const struct mp_chmap *b)
 {
     uint64_t a_mask = mp_chmap_to_lavc_unchecked(a);
     uint64_t b_mask = mp_chmap_to_lavc_unchecked(b);
-    return av_popcount64((a_mask ^ b_mask) & a_mask);
+    return popcount64((a_mask ^ b_mask) & a_mask);
 }
 
 // Returns something like "fl-fr-fc". If there's a standard layout in lavc
@@ -452,25 +511,6 @@ 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

@@ -1,18 +1,18 @@
 /*
  * This file is part of mpv.
  *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
+ * 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 Lesser General Public License for more details.
+ * GNU General Public License for more details.
  *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
  */
 
 #ifndef MP_CHMAP_H
@@ -22,7 +22,7 @@
 #include <stdbool.h>
 #include "misc/bstr.h"
 
-#define MP_NUM_CHANNELS 16
+#define MP_NUM_CHANNELS 8
 
 // Speaker a channel can be assigned to.
 // This corresponds to WAVEFORMATEXTENSIBLE channel mask bit indexes.
@@ -47,7 +47,7 @@ enum mp_speaker_id {
     MP_SPEAKER_ID_TBL,          // TOP_BACK_LEFT
     MP_SPEAKER_ID_TBC,          // TOP_BACK_CENTER
     MP_SPEAKER_ID_TBR,          // TOP_BACK_RIGHT
-     // Unofficial/libav* extensions
+     // Inofficial/libav* extensions
     MP_SPEAKER_ID_DL = 29,      // STEREO_LEFT (stereo downmix special speakers)
     MP_SPEAKER_ID_DR,           // STEREO_RIGHT
     MP_SPEAKER_ID_WL,           // WIDE_LEFT
@@ -108,6 +108,10 @@ void mp_chmap_fill_na(struct mp_chmap *map, int num);
 
 void mp_chmap_from_channels(struct mp_chmap *dst, int num_channels);
 void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels);
+void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels);
+
+void mp_chmap_remove_useless_channels(struct mp_chmap *map,
+                                      const struct mp_chmap *requested);
 
 uint64_t mp_chmap_to_lavc(const struct mp_chmap *src);
 uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src);
@@ -124,9 +128,6 @@ int mp_chmap_diffn(const struct mp_chmap *a, const struct mp_chmap *b);
 char *mp_chmap_to_str_buf(char *buf, size_t buf_size, const struct mp_chmap *src);
 #define mp_chmap_to_str(m) mp_chmap_to_str_buf((char[64]){0}, 64, (m))
 
-char *mp_chmap_to_str_hr_buf(char *buf, size_t buf_size, const struct mp_chmap *src);
-#define mp_chmap_to_str_hr(m) mp_chmap_to_str_hr_buf((char[128]){0}, 128, (m))
-
 bool mp_chmap_from_str(struct mp_chmap *dst, bstr src);
 
 struct mp_log;

audio/chmap_sel.c

@@ -1,18 +1,18 @@
 /*
  * This file is part of mpv.
  *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
+ * 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 Lesser General Public License for more details.
+ * GNU General Public License for more details.
  *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ * 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>
@@ -20,7 +20,6 @@
 #include <limits.h>
 
 #include "common/common.h"
-#include "common/msg.h"
 #include "chmap_sel.h"
 
 static const struct mp_chmap speaker_replacements[][2] = {
@@ -59,29 +58,6 @@ static bool replace_speakers(struct mp_chmap *map, struct mp_chmap list[2])
     return false;
 }
 
-// These go strictly from the first to the second entry and always use the
-// full layout (possibly reordered and/or padding channels added).
-static const struct mp_chmap preferred_remix[][2] = {
-    // mono can be perfectly played as stereo
-    { MP_CHMAP_INIT_MONO, MP_CHMAP_INIT_STEREO },
-};
-
-// Conversion from src to dst is explicitly encouraged and should be preferred
-// over "mathematical" upmixes or downmixes (which minimize lost channels).
-static bool test_preferred_remix(const struct mp_chmap *src,
-                                 const struct mp_chmap *dst)
-{
-    struct mp_chmap src_p = *src, dst_p = *dst;
-    mp_chmap_remove_na(&src_p);
-    mp_chmap_remove_na(&dst_p);
-    for (int n = 0; n < MP_ARRAY_SIZE(preferred_remix); n++) {
-        if (mp_chmap_equals_reordered(&src_p, &preferred_remix[n][0]) &&
-            mp_chmap_equals_reordered(&dst_p, &preferred_remix[n][1]))
-            return true;
-    }
-    return false;
-}
-
 // Allow all channel layouts that can be expressed with mp_chmap.
 // (By default, all layouts are rejected.)
 void mp_chmap_sel_add_any(struct mp_chmap_sel *s)
@@ -95,6 +71,17 @@ void mp_chmap_sel_add_waveext(struct mp_chmap_sel *s)
     s->allow_waveext = true;
 }
 
+// Classic ALSA-based MPlayer layouts.
+void mp_chmap_sel_add_alsa_def(struct mp_chmap_sel *s)
+{
+    for (int n = 1; n <= MP_NUM_CHANNELS; n++) {
+        struct mp_chmap t;
+        mp_chmap_from_channels_alsa(&t, n);
+        if (t.num)
+            mp_chmap_sel_add_map(s, &t);
+    }
+}
+
 // Add a channel map that should be allowed.
 void mp_chmap_sel_add_map(struct mp_chmap_sel *s, const struct mp_chmap *map)
 {
@@ -189,10 +176,6 @@ bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map)
             return true;
         }
     }
-
-    if (mp_chmap_sel_fallback(s, map))
-        return true;
-
     for (int i = 0; i < MP_ARRAY_SIZE(speaker_replacements); i++) {
         struct mp_chmap  t = *map;
         struct mp_chmap *r = (struct mp_chmap *)speaker_replacements[i];
@@ -202,6 +185,9 @@ bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map)
         }
     }
 
+    if (mp_chmap_sel_fallback(s, map))
+        return true;
+
     // Fallback to mono/stereo as last resort
     *map = (struct mp_chmap) MP_CHMAP_INIT_STEREO;
     if (test_layout(s, map))
@@ -213,100 +199,40 @@ 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;
-
-    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;
-    }
-
-    // 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;
-
-    return mindiff;
-}
-
 // Decide whether we should prefer old or new for the requested layout.
 // Return true if new should be used, false if old should be used.
 // If old is empty, always return new (initial case).
 static bool mp_chmap_is_better(struct mp_chmap *req, struct mp_chmap *old,
                                struct mp_chmap *new)
 {
+    int old_lost = mp_chmap_diffn(req, old); // num. channels only in req
+    int new_lost = mp_chmap_diffn(req, new);
+
     // Initial case
     if (!old->num)
         return true;
 
-    // Exact pick - this also ensures that the best layout is chosen if the
-    // layouts are the same, but with different order of channels.
-    if (mp_chmap_equals(req, old))
-        return false;
-    if (mp_chmap_equals(req, new))
-        return true;
-
-    // If there's no exact match, strictly do a preferred conversion.
-    bool old_pref = test_preferred_remix(req, old);
-    bool new_pref = test_preferred_remix(req, new);
-    if (old_pref && !new_pref)
-        return false;
-    if (!old_pref && new_pref)
-        return true;
-
-    int old_lost_r = mp_chmap_diffn_r(req, old); // num. channels only in req
-    int new_lost_r = mp_chmap_diffn_r(req, new);
-
     // Imperfect upmix (no real superset) - minimize lost channels
-    if (new_lost_r != old_lost_r)
-        return new_lost_r < old_lost_r;
-
-    struct mp_chmap old_p = *old, new_p = *new;
-    mp_chmap_remove_na(&old_p);
-    mp_chmap_remove_na(&new_p);
-
-    // If the situation is equal with replaced speakers, but the replacement is
-    // perfect for only one of them, let the better one win. This prefers
-    // inexact equivalents over exact supersets.
-    bool perfect_r_new = !new_lost_r && new_p.num <= old_p.num;
-    bool perfect_r_old = !old_lost_r && old_p.num <= new_p.num;
-    if (perfect_r_new != perfect_r_old)
-        return perfect_r_new;
-
-    int old_lost = mp_chmap_diffn(req, old);
-    int new_lost = mp_chmap_diffn(req, new);
-    // If the situation is equal with replaced speakers, pick the better one,
-    // even if it means an upmix.
     if (new_lost != old_lost)
         return new_lost < old_lost;
 
     // Some kind of upmix. If it's perfect, prefer the smaller one. Even if not,
     // both have equal loss, so also prefer the smaller one.
-    // Drop padding channels (NA) for the sake of this check, as the number of
-    // padding channels isn't really meaningful.
-    if (new_p.num != old_p.num)
-        return new_p.num < old_p.num;
-
-    // Again, with physical channels (minimizes number of NA channels).
     return new->num < old->num;
 }
 
 // Determine which channel map to fallback to given a source channel map.
 bool mp_chmap_sel_fallback(const struct mp_chmap_sel *s, struct mp_chmap *map)
 {
+    // special case: if possible always fallback mono to stereo (instead of
+    // looking for a multichannel upmix)
+    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;
+    }
+
     struct mp_chmap best = {0};
 
     for (int n = 0; n < s->num_chmaps; n++) {
@@ -315,8 +241,17 @@ 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;
+        // in case we didn't match any fallback retry after replacing speakers
+        for (int i = -1; i < (int)MP_ARRAY_SIZE(speaker_replacements); i++) {
+            struct mp_chmap  t = *map;
+            if (i >= 0) {
+                struct mp_chmap *r = (struct mp_chmap *)speaker_replacements[i];
+                if (!replace_speakers(&t, r))
+                    continue;
+            }
+            if (mp_chmap_is_better(&t, &best, &e))
+                best = e;
+        }
     }
 
     if (best.num) {
@@ -353,37 +288,3 @@ bool mp_chmap_sel_get_def(const struct mp_chmap_sel *s, struct mp_chmap *map,
     }
     return map->num > 0;
 }
-
-// Print the set of allowed channel layouts.
-void mp_chmal_sel_log(const struct mp_chmap_sel *s, struct mp_log *log, int lev)
-{
-    if (!mp_msg_test(log, lev))
-        return;
-
-    for (int i = 0; i < s->num_chmaps; i++)
-        mp_msg(log, lev, " - %s\n", mp_chmap_to_str(&s->chmaps[i]));
-    for (int i = 0; i < MP_SPEAKER_ID_COUNT; i++) {
-        if (!s->speakers[i])
-            continue;
-        struct mp_chmap l = {.num = 1, .speaker = { i }};
-        mp_msg(log, lev, " - #%s\n",
-                    i == MP_SPEAKER_ID_FC ? "fc" : mp_chmap_to_str_hr(&l));
-    }
-    if (s->allow_waveext)
-        mp_msg(log, lev, " - waveext\n");
-    if (s->allow_any)
-        mp_msg(log, lev, " - anything\n");
-}
-
-// Select a channel map from the given list that fits best to c. Don't change
-// *c if there's no match, or the list is empty.
-void mp_chmap_sel_list(struct mp_chmap *c, struct mp_chmap *maps, int num_maps)
-{
-    // This is a separate function to keep messing with mp_chmap_sel internals
-    // within this source file.
-    struct mp_chmap_sel sel = {
-        .chmaps = maps,
-        .num_chmaps = num_maps,
-    };
-    mp_chmap_sel_fallback(&sel, c);
-}

audio/chmap_sel.h

@@ -1,18 +1,18 @@
 /*
  * This file is part of mpv.
  *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
+ * 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 Lesser General Public License for more details.
+ * GNU General Public License for more details.
  *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
  */
 
 #ifndef MP_CHMAP_SEL_H
@@ -37,6 +37,7 @@ struct mp_chmap_sel {
 void mp_chmap_sel_add_any(struct mp_chmap_sel *s);
 void mp_chmap_sel_add_waveext(struct mp_chmap_sel *s);
 void mp_chmap_sel_add_waveext_def(struct mp_chmap_sel *s);
+void mp_chmap_sel_add_alsa_def(struct mp_chmap_sel *s);
 void mp_chmap_sel_add_map(struct mp_chmap_sel *s, const struct mp_chmap *map);
 void mp_chmap_sel_add_speaker(struct mp_chmap_sel *s, int id);
 bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map);
@@ -44,9 +45,4 @@ bool mp_chmap_sel_fallback(const struct mp_chmap_sel *s, struct mp_chmap *map);
 bool mp_chmap_sel_get_def(const struct mp_chmap_sel *s, struct mp_chmap *map,
                           int num);
 
-struct mp_log;
-void mp_chmal_sel_log(const struct mp_chmap_sel *s, struct mp_log *log, int lev);
-
-void mp_chmap_sel_list(struct mp_chmap *c, struct mp_chmap *maps, int num_maps);
-
 #endif

audio/decode/ad.h

@@ -35,8 +35,7 @@ struct ad_functions {
     int (*init)(struct dec_audio *da, const char *decoder);
     void (*uninit)(struct dec_audio *da);
     int (*control)(struct dec_audio *da, int cmd, void *arg);
-    int (*decode_packet)(struct dec_audio *da, struct demux_packet *pkt,
-                         struct mp_audio **out);
+    int (*decode_packet)(struct dec_audio *da, struct mp_audio **out);
 };
 
 enum ad_ctrl {

audio/decode/ad_lavc.c

@@ -26,7 +26,7 @@
 #include <libavutil/common.h>
 #include <libavutil/intreadwrite.h>
 
-#include "mpv_talloc.h"
+#include "talloc.h"
 
 #include "config.h"
 #include "common/av_common.h"
@@ -42,11 +42,8 @@ struct priv {
     AVFrame *avframe;
     struct mp_audio frame;
     bool force_channel_map;
-    uint32_t skip_samples, trim_samples;
-    bool preroll_done;
-    double next_pts;
-    bool needs_reset;
-    AVRational codec_timebase;
+    struct demux_packet *packet;
+    uint32_t skip_samples;
 };
 
 static void uninit(struct dec_audio *da);
@@ -63,7 +60,7 @@ const struct m_sub_options ad_lavc_conf = {
     .opts = (const m_option_t[]) {
         OPT_FLOATRANGE("ac3drc", ac3drc, 0, 0, 6),
         OPT_FLAG("downmix", downmix, 0),
-        OPT_INTRANGE("threads", threads, 0, 0, 16),
+        OPT_INTRANGE("threads", threads, 0, 1, 16),
         OPT_KEYVALUELIST("o", avopts, 0),
         {0}
     },
@@ -81,14 +78,13 @@ static int init(struct dec_audio *da, const char *decoder)
     struct ad_lavc_params *opts = mpopts->ad_lavc_params;
     AVCodecContext *lavc_context;
     AVCodec *lavc_codec;
-    struct mp_codec_params *c = da->codec;
+    struct sh_stream *sh = da->header;
+    struct sh_audio *sh_audio = sh->audio;
 
     struct priv *ctx = talloc_zero(NULL, struct priv);
     da->priv = ctx;
 
-    ctx->codec_timebase = mp_get_codec_timebase(da->codec);
-
-    ctx->force_channel_map = c->force_channels;
+    ctx->force_channel_map = sh_audio->force_channels;
 
     lavc_codec = avcodec_find_decoder_by_name(decoder);
     if (!lavc_codec) {
@@ -104,13 +100,9 @@ static int init(struct dec_audio *da, const char *decoder)
     lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
     lavc_context->codec_id = lavc_codec->id;
 
-#if LIBAVCODEC_VERSION_MICRO >= 100
-    lavc_context->pkt_timebase = ctx->codec_timebase;
-#endif
-
-    if (opts->downmix && mpopts->audio_output_channels.num_chmaps == 1) {
+    if (opts->downmix) {
         lavc_context->request_channel_layout =
-            mp_chmap_to_lavc(&mpopts->audio_output_channels.chmaps[0]);
+            mp_chmap_to_lavc(&mpopts->audio_output_channels);
     }
 
     // Always try to set - option only exists for AC3 at the moment
@@ -124,19 +116,24 @@ static int init(struct dec_audio *da, const char *decoder)
 
     mp_set_avopts(da->log, lavc_context, opts->avopts);
 
-    lavc_context->codec_tag = c->codec_tag;
-    lavc_context->sample_rate = c->samplerate;
-    lavc_context->bit_rate = c->bitrate;
-    lavc_context->block_align = c->block_align;
-    lavc_context->bits_per_coded_sample = c->bits_per_coded_sample;
-    lavc_context->channels = c->channels.num;
-    if (!mp_chmap_is_unknown(&c->channels))
-        lavc_context->channel_layout = mp_chmap_to_lavc(&c->channels);
+    lavc_context->codec_tag = sh->format;
+    lavc_context->sample_rate = sh_audio->samplerate;
+    lavc_context->bit_rate = sh_audio->bitrate;
+    lavc_context->block_align = sh_audio->block_align;
+    lavc_context->bits_per_coded_sample = sh_audio->bits_per_coded_sample;
+    lavc_context->channels = sh_audio->channels.num;
+    if (!mp_chmap_is_unknown(&sh_audio->channels))
+        lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
 
     // demux_mkv
-    mp_lavc_set_extradata(lavc_context, c->extradata, c->extradata_size);
+    if (sh_audio->codecdata_len && sh_audio->codecdata &&
+            !lavc_context->extradata) {
+        mp_lavc_set_extradata(lavc_context, sh_audio->codecdata,
+                              sh_audio->codecdata_len);
+    }
 
-    mp_set_lav_codec_headers(lavc_context, c);
+    if (sh->lav_headers)
+        mp_copy_lav_codec_headers(lavc_context, sh->lav_headers);
 
     mp_set_avcodec_threads(da->log, lavc_context, opts->threads);
 
@@ -147,8 +144,6 @@ static int init(struct dec_audio *da, const char *decoder)
         return 0;
     }
 
-    ctx->next_pts = MP_NOPTS_VALUE;
-
     return 1;
 }
 
@@ -174,53 +169,40 @@ static int control(struct dec_audio *da, int cmd, void *arg)
     switch (cmd) {
     case ADCTRL_RESET:
         avcodec_flush_buffers(ctx->avctx);
+        talloc_free(ctx->packet);
+        ctx->packet = NULL;
         ctx->skip_samples = 0;
-        ctx->trim_samples = 0;
-        ctx->preroll_done = false;
-        ctx->next_pts = MP_NOPTS_VALUE;
-        ctx->needs_reset = false;
         return CONTROL_TRUE;
     }
     return CONTROL_UNKNOWN;
 }
 
-static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt,
-                         struct mp_audio **out)
+static int decode_packet(struct dec_audio *da, struct mp_audio **out)
 {
     struct priv *priv = da->priv;
     AVCodecContext *avctx = priv->avctx;
 
-    // If the decoder discards the timestamp for some reason, we use the
-    // interpolated PTS. Initialize it so that it works for the initial
-    // packet as well.
-    if (mpkt && priv->next_pts == MP_NOPTS_VALUE)
-        priv->next_pts = mpkt->pts;
+    struct demux_packet *mpkt = priv->packet;
+    if (!mpkt) {
+        if (demux_read_packet_async(da->header, &mpkt) == 0)
+            return AD_WAIT;
+    }
+
+    priv->packet = talloc_steal(priv, mpkt);
 
     int in_len = mpkt ? mpkt->len : 0;
 
     AVPacket pkt;
-    mp_set_av_packet(&pkt, mpkt, &priv->codec_timebase);
+    mp_set_av_packet(&pkt, mpkt, NULL);
+
+    // If we don't have a PTS yet, use the first packet PTS we can get.
+    if (da->pts == MP_NOPTS_VALUE && mpkt && mpkt->pts != MP_NOPTS_VALUE) {
+        da->pts = mpkt->pts;
+        da->pts_offset = 0;
+    }
 
     int got_frame = 0;
     av_frame_unref(priv->avframe);
-
-    if (priv->needs_reset)
-        control(da, ADCTRL_RESET, NULL);
-
-#if HAVE_AVCODEC_NEW_CODEC_API
-    int ret = avcodec_send_packet(avctx, &pkt);
-    if (ret >= 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
-        if (ret >= 0 && mpkt)
-            mpkt->len = 0;
-        ret = avcodec_receive_frame(avctx, priv->avframe);
-        if (ret >= 0)
-            got_frame = 1;
-        if (ret == AVERROR_EOF)
-            priv->needs_reset = true;
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-            ret = 0;
-    }
-#else
     int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
     if (mpkt) {
         // At least "shorten" decodes sub-frames, instead of the whole packet.
@@ -231,71 +213,59 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt,
             mpkt->len    -= ret;
             mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time
         }
-        // LATM may need many packets to find mux info
-        if (ret == AVERROR(EAGAIN)) {
-            mpkt->len = 0;
-            return 0;
+        if (mpkt->len == 0 || ret < 0) {
+            talloc_free(mpkt);
+            priv->packet = NULL;
         }
+        // LATM may need many packets to find mux info
+        if (ret == AVERROR(EAGAIN))
+            return AD_OK;
     }
-#endif
     if (ret < 0) {
         MP_ERR(da, "Error decoding audio.\n");
-        return -1;
+        return AD_ERR;
     }
     if (!got_frame)
-        return 0;
+        return mpkt ? AD_OK : AD_EOF;
 
-    double out_pts = mp_pts_from_av(MP_AVFRAME_DEC_PTS(priv->avframe),
-                                    &priv->codec_timebase);
+    double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
+    if (out_pts != MP_NOPTS_VALUE) {
+        da->pts = out_pts;
+        da->pts_offset = 0;
+    }
 
     struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
     if (!mpframe)
-        return -1;
+        return AD_ERR;
 
     struct mp_chmap lavc_chmap = mpframe->channels;
     if (lavc_chmap.num != avctx->channels)
         mp_chmap_from_channels(&lavc_chmap, avctx->channels);
     if (priv->force_channel_map) {
-        if (lavc_chmap.num == da->codec->channels.num)
-            lavc_chmap = da->codec->channels;
+        struct sh_audio *sh_audio = da->header->audio;
+        if (lavc_chmap.num == sh_audio->channels.num)
+            lavc_chmap = sh_audio->channels;
     }
     mp_audio_set_channels(mpframe, &lavc_chmap);
 
-    mpframe->pts = out_pts;
-
-    if (mpframe->pts == MP_NOPTS_VALUE)
-        mpframe->pts = priv->next_pts;
-    if (mpframe->pts != MP_NOPTS_VALUE)
-        priv->next_pts = mpframe->pts + mpframe->samples / (double)mpframe->rate;
-
 #if HAVE_AVFRAME_SKIP_SAMPLES
     AVFrameSideData *sd =
         av_frame_get_side_data(priv->avframe, AV_FRAME_DATA_SKIP_SAMPLES);
     if (sd && sd->size >= 10) {
         char *d = sd->data;
         priv->skip_samples += AV_RL32(d + 0);
-        priv->trim_samples += AV_RL32(d + 4);
+        uint32_t pad = AV_RL32(d + 4);
+        uint32_t skip = MPMIN(priv->skip_samples, mpframe->samples);
+        if (skip) {
+            mp_audio_skip_samples(mpframe, skip);
+            da->pts_offset += skip;
+            priv->skip_samples -= skip;
+        }
+        if (pad <= mpframe->samples)
+            mpframe->samples -= pad;
     }
 #endif
 
-    if (!priv->preroll_done) {
-        // Skip only if this isn't already handled by AV_FRAME_DATA_SKIP_SAMPLES.
-        if (!priv->skip_samples)
-            priv->skip_samples = avctx->delay;
-        priv->preroll_done = true;
-    }
-
-    uint32_t skip = MPMIN(priv->skip_samples, mpframe->samples);
-    if (skip) {
-        mp_audio_skip_samples(mpframe, skip);
-        priv->skip_samples -= skip;
-    }
-    uint32_t trim = MPMIN(priv->trim_samples, mpframe->samples);
-    if (trim) {
-        mpframe->samples -= trim;
-        priv->trim_samples -= trim;
-    }
-
     *out = mpframe;
 
     av_frame_unref(priv->avframe);

audio/decode/ad_spdif.c

@@ -34,14 +34,11 @@
 
 struct spdifContext {
     struct mp_log   *log;
-    enum AVCodecID   codec_id;
     AVFormatContext *lavf_ctx;
     int              out_buffer_len;
     uint8_t          out_buffer[OUTBUF_SIZE];
     bool             need_close;
-    bool             use_dts_hd;
     struct mp_audio  fmt;
-    struct mp_audio_pool *pool;
 };
 
 static int write_packet(void *p, uint8_t *buf, int buf_size)
@@ -79,86 +76,17 @@ 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;
-    spdif_ctx->pool = mp_audio_pool_create(spdif_ctx);
-
-    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;
-    }
-
-#if HAVE_AVCODEC_NEW_CODEC_API
-    if (avcodec_send_packet(ctx, pkt) < 0)
-        goto done;
-    if (avcodec_receive_frame(ctx, frame) < 0)
-        goto done;
-#else
-    int got_frame = 0;
-    if (avcodec_decode_audio4(ctx, frame, &got_frame, pkt) < 1 || !got_frame)
-        goto done;
-#endif
-
-    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();
@@ -178,18 +106,14 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
     if (!stream)
         goto fail;
 
-#if HAVE_AVCODEC_HAS_CODECPAR
-    stream->codecpar->codec_id = spdif_ctx->codec_id;
-#else
-    stream->codec->codec_id = spdif_ctx->codec_id;
-#endif
+    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:
         sample_format                   = AF_FORMAT_S_AAC;
         samplerate                      = 48000;
@@ -200,22 +124,18 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
         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 ||
-                     profile == FF_PROFILE_UNKNOWN;
-        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;
+            sample_format                   = AF_FORMAT_S_DTSHD;
+            samplerate                      = 192000;
+            num_channels                    = 2*4;
         } else {
-            sample_format               = AF_FORMAT_S_DTS;
-            samplerate                  = 48000;
-            num_channels                = 2;
+            sample_format                   = AF_FORMAT_S_DTS;
+            samplerate                      = 48000;
+            num_channels                    = 2;
         }
         break;
-    }
     case AV_CODEC_ID_EAC3:
         sample_format                   = AF_FORMAT_S_EAC3;
         samplerate                      = 192000;
@@ -247,45 +167,46 @@ 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 demux_packet *mpkt,
-                         struct mp_audio **out)
+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;
 
-    if (!mpkt)
-        return 0;
+    struct demux_packet *mpkt;
+    if (demux_read_packet_async(da->header, &mpkt) == 0)
+        return AD_WAIT;
 
-    double pts = mpkt->pts;
+    if (!mpkt)
+        return AD_EOF;
 
     AVPacket pkt;
     mp_set_av_packet(&pkt, mpkt, NULL);
-    mpkt->len = 0; // will be fully consumed
     pkt.pts = pkt.dts = 0;
-    if (!spdif_ctx->lavf_ctx) {
-        if (init_filter(da, &pkt) < 0)
-            return -1;
+    if (mpkt->pts != MP_NOPTS_VALUE) {
+        da->pts        = mpkt->pts;
+        da->pts_offset = 0;
     }
-    int ret = av_write_frame(spdif_ctx->lavf_ctx, &pkt);
-    avio_flush(spdif_ctx->lavf_ctx->pb);
+    int ret = av_write_frame(lavf_ctx, &pkt);
+    talloc_free(mpkt);
+    avio_flush(lavf_ctx->pb);
     if (ret < 0)
-        return -1;
+        return AD_ERR;
 
     int samples = spdif_ctx->out_buffer_len / spdif_ctx->fmt.sstride;
-    *out = mp_audio_pool_get(spdif_ctx->pool, &spdif_ctx->fmt, samples);
+    *out = mp_audio_pool_get(da->pool, &spdif_ctx->fmt, samples);
     if (!*out)
-        return -1;
+        return AD_ERR;
 
     memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
-    (*out)->pts = pts;
 
     return 0;
 }
@@ -314,8 +235,6 @@ 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

@@ -18,7 +18,6 @@
 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
-#include <math.h>
 #include <assert.h>
 
 #include <libavutil/mem.h>
@@ -53,7 +52,6 @@ static const struct ad_functions * const ad_drivers[] = {
 
 static void uninit_decoder(struct dec_audio *d_audio)
 {
-    audio_reset_decoding(d_audio);
     if (d_audio->ad_driver) {
         MP_VERBOSE(d_audio, "Uninit audio decoder.\n");
         d_audio->ad_driver->uninit(d_audio);
@@ -83,21 +81,11 @@ 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->codec->codec;
-
     struct mp_decoder_list *list = audio_decoder_list();
-    struct mp_decoder_list *new =
-        mp_select_decoders(list, codec, opts->audio_decoders);
-    if (d_audio->try_spdif) {
-        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;
 }
@@ -111,13 +99,14 @@ 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);
 
@@ -144,7 +133,7 @@ int audio_init_best_codec(struct dec_audio *d_audio)
         MP_VERBOSE(d_audio, "Selected audio codec: %s\n", d_audio->decoder_desc);
     } else {
         MP_ERR(d_audio, "Failed to initialize an audio decoder for codec '%s'.\n",
-               d_audio->codec->codec);
+               d_audio->header->codec ? d_audio->header->codec : "<unknown>");
     }
 
     talloc_free(list);
@@ -155,139 +144,111 @@ void audio_uninit(struct dec_audio *d_audio)
 {
     if (!d_audio)
         return;
+    MP_VERBOSE(d_audio, "Uninit audio filters...\n");
+    af_destroy(d_audio->afilter);
     uninit_decoder(d_audio);
+    talloc_free(d_audio->waiting);
     talloc_free(d_audio);
 }
 
+static int decode_new_frame(struct dec_audio *da)
+{
+    while (!da->waiting) {
+        int ret = da->ad_driver->decode_packet(da, &da->waiting);
+        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;
+            mp_audio_set_null_data(&da->decode_format);
+        }
+    }
+    return mp_audio_config_valid(da->waiting) ? AD_OK : AD_ERR;
+}
+
+/* Decode packets until we know the audio format. Then reinit the buffer.
+ * Returns AD_OK on success, negative AD_* code otherwise.
+ * Also returns AD_OK if already initialized (and does nothing).
+ */
+int initial_audio_decode(struct dec_audio *da)
+{
+    return decode_new_frame(da);
+}
+
+static bool copy_output(struct af_stream *afs, struct mp_audio_buffer *outbuf,
+                        int minsamples, bool eof)
+{
+    while (mp_audio_buffer_samples(outbuf) < minsamples) {
+        if (af_output_frame(afs, eof) < 0)
+            return true; // error, stop doing stuff
+        struct mp_audio *mpa = af_read_output_frame(afs);
+        if (!mpa)
+            return false; // out of data
+        mp_audio_buffer_append(outbuf, mpa);
+        talloc_free(mpa);
+    }
+    return true;
+}
+
+/* Try to get at least minsamples decoded+filtered samples in outbuf
+ * (total length including possible existing data).
+ * Return 0 on success, or negative AD_* error code.
+ * In the former case outbuf has at least minsamples buffered on return.
+ * In case of EOF/error it might or might not be. */
+int audio_decode(struct dec_audio *da, struct mp_audio_buffer *outbuf,
+                 int minsamples)
+{
+    struct af_stream *afs = da->afilter;
+    if (afs->initialized < 1)
+        return AD_ERR;
+
+    MP_STATS(da, "start audio");
+
+    int res;
+    while (1) {
+        res = 0;
+
+        if (copy_output(afs, outbuf, minsamples, false))
+            break;
+
+        res = decode_new_frame(da);
+        if (res < 0) {
+            // drain filters first (especially for true EOF case)
+            copy_output(afs, outbuf, minsamples, true);
+            break;
+        }
+
+        // On format change, make sure to drain the filter chain.
+        if (!mp_audio_config_equals(&afs->input, da->waiting)) {
+            copy_output(afs, outbuf, minsamples, true);
+            res = AD_NEW_FMT;
+            break;
+        }
+
+        struct mp_audio *mpa = da->waiting;
+        da->waiting = NULL;
+        if (af_filter_frame(afs, mpa) < 0)
+            return AD_ERR;
+    }
+
+    MP_STATS(da, "end audio");
+
+    return res;
+}
+
 void audio_reset_decoding(struct dec_audio *d_audio)
 {
     if (d_audio->ad_driver)
         d_audio->ad_driver->control(d_audio, ADCTRL_RESET, NULL);
+    af_seek_reset(d_audio->afilter);
     d_audio->pts = MP_NOPTS_VALUE;
-    talloc_free(d_audio->current_frame);
-    d_audio->current_frame = NULL;
-    talloc_free(d_audio->packet);
-    d_audio->packet = NULL;
-    talloc_free(d_audio->new_segment);
-    d_audio->new_segment = NULL;
-    d_audio->start = d_audio->end = MP_NOPTS_VALUE;
-}
-
-static void fix_audio_pts(struct dec_audio *da)
-{
-    if (!da->current_frame)
-        return;
-
-    if (da->current_frame->pts != MP_NOPTS_VALUE) {
-        double newpts = da->current_frame->pts;
-
-        if (da->pts != MP_NOPTS_VALUE)
-            MP_STATS(da, "value %f audio-pts-err", da->pts - newpts);
-
-        // Keep the interpolated timestamp if it doesn't deviate more
-        // than 1 ms from the real one. (MKV rounded timestamps.)
-        if (da->pts == MP_NOPTS_VALUE || fabs(da->pts - newpts) > 0.001)
-            da->pts = newpts;
-    }
-
-    if (da->pts == MP_NOPTS_VALUE && da->header->missing_timestamps)
-        da->pts = 0;
-
-    da->current_frame->pts = da->pts;
-
-    if (da->pts != MP_NOPTS_VALUE)
-        da->pts += da->current_frame->samples / (double)da->current_frame->rate;
-}
-
-void audio_work(struct dec_audio *da)
-{
-    if (da->current_frame)
-        return;
-
-    if (!da->packet && !da->new_segment &&
-        demux_read_packet_async(da->header, &da->packet) == 0)
-    {
-        da->current_state = DATA_WAIT;
-        return;
-    }
-
-    if (da->packet && da->packet->new_segment) {
-        assert(!da->new_segment);
-        da->new_segment = da->packet;
-        da->packet = NULL;
-    }
-
-    bool had_input_packet = !!da->packet;
-    bool had_packet = da->packet || da->new_segment;
-
-    int ret = da->ad_driver->decode_packet(da, da->packet, &da->current_frame);
-    if (ret < 0 || (da->packet && da->packet->len == 0)) {
-        talloc_free(da->packet);
-        da->packet = NULL;
-    }
-
-    if (da->current_frame && !mp_audio_config_valid(da->current_frame)) {
-        talloc_free(da->current_frame);
-        da->current_frame = NULL;
-    }
-
-    da->current_state = DATA_OK;
-    if (!da->current_frame) {
-        da->current_state = DATA_EOF;
-        if (had_packet)
-            da->current_state = DATA_AGAIN;
-    }
-
-    fix_audio_pts(da);
-
-    bool segment_end = !da->current_frame && !had_input_packet;
-
-    if (da->current_frame) {
-        mp_audio_clip_timestamps(da->current_frame, da->start, da->end);
-        if (da->current_frame->pts != MP_NOPTS_VALUE && da->start != MP_NOPTS_VALUE)
-            segment_end = da->current_frame->pts >= da->end;
-        if (da->current_frame->samples == 0) {
-            talloc_free(da->current_frame);
-            da->current_frame = NULL;
-        }
-    }
-
-    // If there's a new segment, start it as soon as we're drained/finished.
-    if (segment_end && da->new_segment) {
-        struct demux_packet *new_segment = da->new_segment;
-        da->new_segment = NULL;
-
-        // Could avoid decoder reinit; would still need flush.
-        da->codec = new_segment->codec;
-        if (da->ad_driver)
-            da->ad_driver->uninit(da);
-        da->ad_driver = NULL;
-        audio_init_best_codec(da);
-
-        da->start = new_segment->start;
-        da->end = new_segment->end;
-
-        new_segment->new_segment = false;
-
-        da->packet = new_segment;
-        da->current_state = DATA_AGAIN;
+    d_audio->pts_offset = 0;
+    if (d_audio->waiting) {
+        talloc_free(d_audio->waiting);
+        d_audio->waiting = NULL;
     }
 }
-
-// Fetch an audio frame decoded with audio_work(). Returns one of:
-//  DATA_OK:    *out_frame is set to a new image
-//  DATA_WAIT:  waiting for demuxer; will receive a wakeup signal
-//  DATA_EOF:   end of file, no more frames to be expected
-//  DATA_AGAIN: dropped frame or something similar
-int audio_get_frame(struct dec_audio *da, struct mp_audio **out_frame)
-{
-    *out_frame = NULL;
-    if (da->current_frame) {
-        *out_frame = da->current_frame;
-        da->current_frame = NULL;
-        return DATA_OK;
-    }
-    if (da->current_state == DATA_OK)
-        return DATA_AGAIN;
-    return da->current_state;
-}

audio/decode/dec_audio.h

@@ -32,31 +32,36 @@ struct dec_audio {
     struct mpv_global *global;
     const struct ad_functions *ad_driver;
     struct sh_stream *header;
-    struct mp_codec_params *codec;
+    struct af_stream *afilter;
     char *decoder_desc;
-
-    bool try_spdif;
-
+    int init_retries;
+    struct mp_audio_pool *pool;
+    struct mp_audio decode_format;
+    struct mp_audio *waiting;   // used on format-change
+    // set by decoder
+    int bitrate;                // input bitrate, can change with VBR sources
+    // last known pts value in output from decoder
+    double pts;
+    // number of samples output by decoder after last known pts
+    int pts_offset;
     // For free use by the ad_driver
     void *priv;
+};
 
-    // Strictly internal (dec_audio.c).
-
-    double pts; // endpts of previous frame
-    double start, end;
-    struct demux_packet *packet;
-    struct demux_packet *new_segment;
-    struct mp_audio *current_frame;
-    int current_state;
+enum {
+    AD_OK = 0,
+    AD_ERR = -1,
+    AD_EOF = -2,
+    AD_NEW_FMT = -3,
+    AD_WAIT = -4,
 };
 
 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);
+void audio_reset_decoding(struct dec_audio *d_audio);
 void audio_uninit(struct dec_audio *d_audio);
 
-void audio_work(struct dec_audio *d_audio);
-int audio_get_frame(struct dec_audio *d_audio, struct mp_audio **out_frame);
-
-void audio_reset_decoding(struct dec_audio *d_audio);
-
 #endif /* MPLAYER_DEC_AUDIO_H */

audio/filter/af.c

@@ -31,33 +31,70 @@
 #include "af.h"
 
 // Static list of filters
+extern const struct af_info af_info_dummy;
+extern const struct af_info af_info_delay;
 extern const struct af_info af_info_channels;
 extern const struct af_info af_info_format;
+extern const struct af_info af_info_force;
 extern const struct af_info af_info_volume;
 extern const struct af_info af_info_equalizer;
 extern const struct af_info af_info_pan;
+extern const struct af_info af_info_surround;
+extern const struct af_info af_info_sub;
+extern const struct af_info af_info_export;
 extern const struct af_info af_info_drc;
+extern const struct af_info af_info_extrastereo;
 extern const struct af_info af_info_lavcac3enc;
 extern const struct af_info af_info_lavrresample;
+extern const struct af_info af_info_sweep;
+extern const struct af_info af_info_hrtf;
+extern const struct af_info af_info_ladspa;
+extern const struct af_info af_info_center;
+extern const struct af_info af_info_sinesuppress;
+extern const struct af_info af_info_karaoke;
 extern const struct af_info af_info_scaletempo;
 extern const struct af_info af_info_bs2b;
 extern const struct af_info af_info_lavfi;
+extern const struct af_info af_info_convert24;
+extern const struct af_info af_info_convertsignendian;
 extern const struct af_info af_info_rubberband;
 
 static const struct af_info *const filter_list[] = {
+    &af_info_dummy,
+    &af_info_delay,
     &af_info_channels,
     &af_info_format,
     &af_info_volume,
     &af_info_equalizer,
     &af_info_pan,
+    &af_info_surround,
+    &af_info_sub,
+    &af_info_export,
     &af_info_drc,
+    &af_info_extrastereo,
     &af_info_lavcac3enc,
     &af_info_lavrresample,
+    &af_info_sweep,
+    &af_info_hrtf,
+#if HAVE_LADSPA
+    &af_info_ladspa,
+#endif
 #if HAVE_RUBBERBAND
     &af_info_rubberband,
 #endif
+    &af_info_center,
+    &af_info_sinesuppress,
+    &af_info_karaoke,
     &af_info_scaletempo,
+#if HAVE_LIBBS2B
+    &af_info_bs2b,
+#endif
+#if HAVE_LIBAVFILTER
     &af_info_lavfi,
+#endif
+    // Must come last, because they're fallback format conversion filter
+    &af_info_convert24,
+    &af_info_convertsignendian,
     NULL
 };
 
@@ -155,21 +192,33 @@ 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),
-        .opts = s->opts,
         .replaygain_data = s->replaygain_data,
         .out_pool = mp_audio_pool_create(af),
     };
-    struct m_config *config =
-        m_config_from_obj_desc_and_args(af, s->log, NULL, &desc,
-                                        name, s->opts->af_defs, args);
-    if (!config)
+    struct m_config *config = m_config_from_obj_desc(af, s->log, &desc);
+    if (m_config_apply_defaults(config, name, s->opts->af_defs) < 0)
+        goto error;
+    if (m_config_set_obj_params(config, args) < 0)
         goto error;
     af->priv = config->optstruct;
 
@@ -208,6 +257,29 @@ static struct af_instance *af_prepend(struct af_stream *s,
     return new;
 }
 
+/* Create and insert a new filter of type name after the filter in the
+   argument. This function can be called during runtime, the return
+   value is the new filter */
+static struct af_instance *af_append(struct af_stream *s,
+                                     struct af_instance *af,
+                                     char *name, char **args)
+{
+    if (!af)
+        af = s->first;
+    if (af == s->last)
+        af = s->last->prev;
+    // Create the new filter and make sure it is OK
+    struct af_instance *new = af_create(s, name, args);
+    if (!new)
+        return NULL;
+    // Update pointers
+    new->prev = af;
+    new->next = af->next;
+    af->next = new;
+    new->next->prev = new;
+    return new;
+}
+
 // Uninit and remove the filter "af"
 static void af_remove(struct af_stream *s, struct af_instance *af)
 {
@@ -250,12 +322,8 @@ static void af_print_filter_chain(struct af_stream *s, struct af_instance *at,
     while (af) {
         char b[128] = {0};
         mp_snprintf_cat(b, sizeof(b), "  [%s] ", af->info->name);
-        if (af->label)
-            mp_snprintf_cat(b, sizeof(b), "\"%s\" ", af->label);
         if (af->data)
             mp_snprintf_cat(b, sizeof(b), "%s", mp_audio_config_to_str(af->data));
-        if (af->auto_inserted)
-            mp_snprintf_cat(b, sizeof(b), " [a]");
         if (af == at)
             mp_snprintf_cat(b, sizeof(b), " <-");
         MP_MSG(s, msg_level, "%s\n", b);
@@ -266,159 +334,249 @@ 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 void reset_formats(struct af_stream *s)
+static int af_count_filters(struct af_stream *s)
 {
-    struct mp_audio none = {0};
-    for (struct af_instance *af = s->first; af; af = af->next) {
-        if (af != s->first && af != s->last)
-            mp_audio_copy_config(af->data, &none);
-    }
+    int count = 0;
+    for (struct af_instance *af = s->first; af; af = af->next)
+        count++;
+    return count;
 }
 
-static int filter_reinit(struct af_instance *af)
+// Finds the first conversion filter on the way from srcfmt to dstfmt.
+// Conversions form a DAG: each node is a format/filter pair, and possible
+// conversions are edges. We search the DAG for the shortest path.
+// Some cases visit the same filter multiple times, but with different formats
+// (like u24le->s8), so one node per format or filter separate is not enough.
+// Returns the filter and dest. format for the first conversion step.
+// (So we know what conversion filter with what format to insert next.)
+static char *af_find_conversion_filter(int srcfmt, int *dstfmt)
 {
+#define MAX_NODES (64 * 32)
+    int num_fmt = 0, num_filt = 0;
+    for (int n = 0; filter_list[n]; n++)
+        num_filt = n + 1;
+    for (int n = 0; af_fmtstr_table[n].format; n++)
+        num_fmt = n + 1;
+
+    int num_nodes = num_fmt * num_filt;
+    assert(num_nodes < MAX_NODES);
+
+    bool visited[MAX_NODES] = {0};
+    unsigned char distance[MAX_NODES];
+    short previous[MAX_NODES] = {0};
+    for (int n = 0; n < num_nodes; n++) {
+        distance[n] = 255;
+        if (af_fmtstr_table[n % num_fmt].format == srcfmt)
+            distance[n] = 0;
+    }
+
+    while (1) {
+        int next = -1;
+        for (int n = 0; n < num_nodes; n++) {
+            if (!visited[n] && (next < 0 || (distance[n] < distance[next])))
+                next = n;
+        }
+        if (next < 0 || distance[next] == 255)
+            return NULL;
+        visited[next] = true;
+
+        int fmt = next % num_fmt;
+        if (af_fmtstr_table[fmt].format == *dstfmt) {
+            // Best match found
+            for (int cur = next; cur >= 0; cur = previous[cur] - 1) {
+                if (distance[cur] == 1) {
+                    *dstfmt = af_fmtstr_table[cur % num_fmt].format;
+                    return (char *)filter_list[cur / num_fmt]->name;
+                }
+            }
+            return NULL;
+        }
+
+        for (int n = 0; filter_list[n]; n++) {
+            const struct af_info *af = filter_list[n];
+            if (!af->test_conversion)
+                continue;
+            for (int i = 0; af_fmtstr_table[i].format; i++) {
+                if (i != fmt && af->test_conversion(af_fmtstr_table[fmt].format,
+                                                    af_fmtstr_table[i].format))
+                {
+                    int other = n * num_fmt + i;
+                    int ndist = distance[next] + 1;
+                    if (ndist < distance[other]) {
+                        distance[other] = ndist;
+                        previous[other] = next + 1;
+                    }
+                }
+            }
+        }
+    }
+    assert(0);
+#undef MAX_NODES
+}
+
+static bool af_is_conversion_filter(struct af_instance *af)
+{
+    return af && af->info->test_conversion != NULL;
+}
+
+// in is what af can take as input - insert a conversion filter if the actual
+// input format doesn't match what af expects.
+// Returns:
+//   AF_OK: must call af_reinit() or equivalent, format matches (or is closer)
+//   AF_FALSE: nothing was changed, format matches
+//   else: error
+static int af_fix_format_conversion(struct af_stream *s,
+                                    struct af_instance **p_af,
+                                    struct mp_audio in)
+{
+    int rv;
+    struct af_instance *af = *p_af;
     struct af_instance *prev = af->prev;
-    assert(prev);
-
-    // Check if this is the first filter
-    struct mp_audio in = *prev->data;
-    // Reset just in case...
-    mp_audio_set_null_data(&in);
-
-    if (!mp_audio_config_valid(&in))
+    struct mp_audio actual = *prev->data;
+    if (actual.format == in.format)
+        return AF_FALSE;
+    int dstfmt = in.format;
+    char *filter = af_find_conversion_filter(actual.format, &dstfmt);
+    if (!filter)
         return AF_ERROR;
-
-    af->fmt_in = in;
-    int rv = af->control(af, AF_CONTROL_REINIT, &in);
-    if (rv == AF_OK && !mp_audio_config_equals(&in, prev->data))
-        rv = AF_FALSE; // conversion filter needed
-    if (rv == AF_FALSE)
-        af->fmt_in = in;
-
-    if (rv == AF_OK) {
-        if (!mp_audio_config_valid(af->data))
-            return AF_ERROR;
-        af->fmt_out = *af->data;
-    }
-
-    return rv;
-}
-
-static int filter_reinit_with_conversion(struct af_stream *s, struct af_instance *af)
-{
-    int rv = filter_reinit(af);
-
-    // Conversion filter is needed
-    if (rv == AF_FALSE) {
-        // First try if we can change the output format of the previous
-        // filter to the input format the current filter is expecting.
-        struct mp_audio in = af->fmt_in;
-        if (af->prev != s->first && !mp_audio_config_equals(af->prev->data, &in)) {
-            // This should have been successful (because it succeeded
-            // before), even if just reverting to the old output format.
-            mp_audio_copy_config(af->prev->data, &in);
-            rv = filter_reinit(af->prev);
-            if (rv != AF_OK)
-                return rv;
+    if (strcmp(filter, prev->info->name) == 0) {
+        if (prev->control(prev, AF_CONTROL_SET_FORMAT, &dstfmt) == AF_OK) {
+            *p_af = prev;
+            return AF_OK;
         }
-        if (!mp_audio_config_equals(af->prev->data, &in)) {
-            // Retry with conversion filter added.
-            struct af_instance *new =
-                af_prepend(s, af, "lavrresample", NULL);
-            if (!new)
-                return AF_ERROR;
-            new->auto_inserted = true;
-            mp_audio_copy_config(new->data, &in);
-            rv = filter_reinit(new);
-            if (rv != AF_OK)
-                af_remove(s, new);
-        }
-        if (rv == AF_OK)
-            rv = filter_reinit(af);
     }
-
-    return rv;
-}
-
-static int af_find_output_conversion(struct af_stream *s, struct mp_audio *cfg)
-{
-    assert(mp_audio_config_valid(&s->output));
-    assert(s->initialized > 0);
-
-    if (mp_chmap_equals_reordered(&s->input.channels, &s->output.channels))
+    struct af_instance *new = af_prepend(s, af, filter, NULL);
+    if (new == NULL)
         return AF_ERROR;
-
-    // Heuristic to detect point of conversion. If it looks like something
-    // more complicated is going on, better bail out.
-    // We expect that the last filter converts channels.
-    struct af_instance *conv = s->last->prev;
-    if (!conv->auto_inserted)
-        return AF_ERROR;
-    if (!(mp_chmap_equals_reordered(&conv->fmt_in.channels, &s->input.channels) &&
-          mp_chmap_equals_reordered(&conv->fmt_out.channels, &s->output.channels)))
-        return AF_ERROR;
-    // Also, should be the only one which does auto conversion.
-    for (struct af_instance *af = s->first->next; af != s->last; af = af->next)
-    {
-        if (af != conv && af->auto_inserted &&
-            !mp_chmap_equals_reordered(&af->fmt_in.channels, &af->fmt_out.channels))
-            return AF_ERROR;
-    }
-    // And not if it's the only filter.
-    if (conv->prev == s->first && conv->next == s->last)
-        return AF_ERROR;
-
-    *cfg = s->output;
+    new->auto_inserted = true;
+    if (AF_OK != (rv = new->control(new, AF_CONTROL_SET_FORMAT, &dstfmt)))
+        return rv;
+    *p_af = new;
     return AF_OK;
 }
 
-// Return AF_OK on success or AF_ERROR on failure.
-static int af_do_reinit(struct af_stream *s, bool second_pass)
+// same as af_fix_format_conversion - only wrt. channels
+static int af_fix_channels(struct af_stream *s, struct af_instance **p_af,
+                           struct mp_audio in)
 {
-    struct mp_audio convert_early = {0};
-    if (second_pass) {
-        // If a channel conversion happens, and it is done by an auto-inserted
-        // filter, then insert a filter to convert it early. Otherwise, do
-        // nothing and return immediately.
-        if (af_find_output_conversion(s, &convert_early) != AF_OK)
-            return AF_OK;
+    int rv;
+    struct af_instance *af = *p_af;
+    struct af_instance *prev = af->prev;
+    struct mp_audio actual = *prev->data;
+    if (mp_chmap_equals(&actual.channels, &in.channels))
+        return AF_FALSE;
+    if (prev->control(prev, AF_CONTROL_SET_CHANNELS, &in.channels) == AF_OK) {
+        *p_af = prev;
+        return AF_OK;
     }
+    char *filter = "lavrresample";
+    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_CHANNELS, &in.channels)))
+        return rv;
+    *p_af = new;
+    return AF_OK;
+}
 
+static int af_fix_rate(struct af_stream *s, struct af_instance **p_af,
+                       struct mp_audio in)
+{
+    int rv;
+    struct af_instance *af = *p_af;
+    struct af_instance *prev = af->prev;
+    struct mp_audio actual = *prev->data;
+    if (actual.rate == in.rate)
+        return AF_FALSE;
+    if (prev->control(prev, AF_CONTROL_SET_RESAMPLE_RATE, &in.rate) == AF_OK) {
+        *p_af = prev;
+        return AF_OK;
+    }
+    char *filter = "lavrresample";
+    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_RESAMPLE_RATE, &in.rate)))
+        return rv;
+    *p_af = new;
+    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
+// state (for example, format filters that were tentatively inserted stay
+// inserted).
+// In that case, you should always rebuild the filter chain, or abort.
+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;
-
-    if (mp_audio_config_valid(&convert_early)) {
-        struct af_instance *new = af_prepend(s, s->first, "lavrresample", NULL);
-        if (!new)
-            return AF_ERROR;
-        new->auto_inserted = true;
-        mp_audio_copy_config(new->data, &convert_early);
-        int rv = filter_reinit(new);
-        if (rv != AF_DETACH && rv != AF_OK)
-            return AF_ERROR;
-        MP_VERBOSE(s, "Moving up output conversion.\n");
-    }
-
     // Start with the second filter, as the first filter is the special input
     // filter which needs no initialization.
     struct af_instance *af = s->first->next;
+    // Up to 7 retries per filter (channel, rate, 5x format conversions)
+    int max_retry = af_count_filters(s) * 7;
+    int retry = 0;
     while (af) {
-        int rv = filter_reinit_with_conversion(s, af);
+        if (retry >= max_retry)
+            goto negotiate_error;
 
+        // Check if this is the first filter
+        struct mp_audio in = *af->prev->data;
+        // Reset just in case...
+        mp_audio_set_null_data(&in);
+
+        if (!mp_audio_config_valid(&in))
+            goto error;
+
+        af->fmt_in = in;
+        int rv = af->control(af, AF_CONTROL_REINIT, &in);
+        if (rv == AF_OK && !mp_audio_config_equals(&in, af->prev->data))
+            rv = AF_FALSE; // conversion filter needed
         switch (rv) {
         case AF_OK:
+            if (!mp_audio_config_valid(af->data))
+                goto error;
+            af->fmt_out = *af->data;
             af = af->next;
             break;
-        case AF_FALSE: {
+        case AF_FALSE: { // Configuration filter is needed
+            if (af_fix_channels(s, &af, in) == AF_OK) {
+                retry++;
+                continue;
+            }
+            if (af_fix_rate(s, &af, in) == AF_OK) {
+                retry++;
+                continue;
+            }
+            // Do this last, to prevent "format->lavrresample" being added to
+            // the filter chain when output formats not supported by
+            // af_lavrresample are in use.
+            if (af_fix_format_conversion(s, &af, in) == AF_OK) {
+                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 = af->fmt_in.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);
+                bool spd1 = AF_FORMAT_IS_IEC61937(fmt_in1);
+                bool spd2 = AF_FORMAT_IS_IEC61937(fmt_in2);
                 if (spd1 != spd2 && af->next) {
                     MP_WARN(af, "Filter %s apparently cannot be used due to "
                                 "spdif passthrough - removing it.\n",
@@ -426,7 +584,7 @@ static int af_do_reinit(struct af_stream *s, bool second_pass)
                     struct af_instance *aft = af->prev;
                     af_remove(s, af);
                     af = aft->next;
-                    continue;
+                    break;
                 }
             }
             goto negotiate_error;
@@ -466,19 +624,6 @@ error:
     return AF_ERROR;
 }
 
-static int af_reinit(struct af_stream *s)
-{
-    int r = af_do_reinit(s, false);
-    if (r == AF_OK && mp_audio_config_valid(&s->output)) {
-        r = af_do_reinit(s, true);
-        if (r != AF_OK) {
-            MP_ERR(s, "Failed second pass filter negotiation.\n");
-            r = af_do_reinit(s, false);
-        }
-    }
-    return r;
-}
-
 // Uninit and remove all filters
 void af_uninit(struct af_stream *s)
 {
@@ -533,7 +678,7 @@ void af_destroy(struct af_stream *s)
    format of the preferred output respectively. The function is
    reentrant i.e. if called with an already initialized stream the
    stream will be reinitialized.
-   If one of the preferred output parameters is 0 the one that needs
+   If one of the prefered output parameters is 0 the one that needs
    no conversion is used (i.e. the output format in the last filter).
    The return value is 0 if success and -1 if failure */
 int af_init(struct af_stream *s)
@@ -578,12 +723,17 @@ struct af_instance *af_add(struct af_stream *s, char *name, char *label,
     if (af_find_by_label(s, label))
         return NULL;
 
-    struct af_instance *new = af_prepend(s, s->last, name, args);
+    struct af_instance *new;
+    // Insert the filter somewhere nice
+    if (af_is_conversion_filter(s->first->next))
+        new = af_append(s, s->first->next, name, args);
+    else
+        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);
         return NULL;
@@ -653,29 +803,6 @@ 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;
-    }
-}
-
-int af_send_command(struct af_stream *s, char *label, char *cmd, char *arg)
-{
-    char *args[2] = {cmd, arg};
-    if (strcmp(label, "all") == 0) {
-        af_control_all(s, AF_CONTROL_COMMAND, args);
-        return 0;
-    } else {
-        return af_control_by_label(s, AF_CONTROL_COMMAND, args, bstr0(label));
-    }
-}
-
 // 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)
@@ -705,16 +832,6 @@ 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)
@@ -754,7 +871,6 @@ 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;
@@ -779,12 +895,6 @@ struct mp_audio *af_read_output_frame(struct af_stream *s)
     return af_dequeue_output_frame(s->last);
 }
 
-void af_unread_output_frame(struct af_stream *s, struct mp_audio *frame)
-{
-    struct af_instance *af = s->last;
-    MP_TARRAY_INSERT_AT(af, af->out_queued, af->num_out_queued, 0, frame);
-}
-
 // Make sure the caller can change data referenced by the frame.
 // Return negative error code on failure (i.e. you can't write).
 int af_make_writeable(struct af_instance *af, struct mp_audio *frame)

audio/filter/af.h

@@ -27,7 +27,6 @@
 #include "audio/chmap.h"
 #include "audio/audio.h"
 #include "common/msg.h"
-#include "common/common.h"
 
 struct af_instance;
 struct mpv_global;
@@ -35,6 +34,10 @@ 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 +46,9 @@ 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;
@@ -53,7 +58,6 @@ struct af_info {
 struct af_instance {
     const struct af_info *info;
     struct mp_log *log;
-    struct MPOpts *opts;
     struct replaygain_data *replaygain_data;
     int (*control)(struct af_instance *af, int cmd, void *arg);
     void (*uninit)(struct af_instance *af);
@@ -101,17 +105,21 @@ 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 {
     AF_CONTROL_REINIT = 1,
     AF_CONTROL_RESET,
+    AF_CONTROL_SET_RESAMPLE_RATE,
+    AF_CONTROL_SET_FORMAT,
+    AF_CONTROL_SET_CHANNELS,
     AF_CONTROL_SET_VOLUME,
     AF_CONTROL_GET_VOLUME,
     AF_CONTROL_SET_PAN_LEVEL,
@@ -120,8 +128,6 @@ enum af_control {
     AF_CONTROL_GET_PAN_BALANCE,
     AF_CONTROL_SET_PLAYBACK_SPEED,
     AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE,
-    AF_CONTROL_GET_METADATA,
-    AF_CONTROL_COMMAND,
 };
 
 // Argument for AF_CONTROL_SET_PAN_LEVEL
@@ -140,21 +146,19 @@ 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);
-int af_send_command(struct af_stream *s, char *label, char *cmd, char *arg);
 
 void af_add_output_frame(struct af_instance *af, struct mp_audio *frame);
 int af_filter_frame(struct af_stream *s, struct mp_audio *frame);
 int af_output_frame(struct af_stream *s, bool eof);
 struct mp_audio *af_read_output_frame(struct af_stream *s);
-void af_unread_output_frame(struct af_stream *s, struct mp_audio *frame);
 int af_make_writeable(struct af_instance *af, struct mp_audio *frame);
 
 double af_calc_delay(struct af_stream *s);
 
 int af_test_output(struct af_instance *af, struct mp_audio *out);
 
+int af_from_dB(int n, float *in, float *out, float k, float mi, float ma);
 int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma);
 float af_softclip(float a);
 

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 mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <bs2b.h>
+#include <inttypes.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "af.h"
+#include "options/m_option.h"
+
+/// Internal specific data of the filter
+struct af_bs2b {
+    int fcut;           ///< cut frequency in Hz
+    int feed;           ///< feed level for low frequencies in 0.1*dB
+    int profile  ;      ///< profile (available crossfeed presets)
+    t_bs2bdp filter;    ///< instance of a library filter
+};
+
+#define DEF_FILTER(fmt, name) \
+static int filter_##name(struct af_instance *af, struct mp_audio *data) \
+{ \
+    if (!data) \
+        return 0; \
+    if (af_make_writeable(af, data) < 0) { \
+        talloc_free(data); \
+        return -1; \
+    } \
+    bs2b_cross_feed_##name(((struct af_bs2b*)(af->priv))->filter, \
+        (data->planes[0]), data->samples); \
+    af_add_output_frame(af, data); \
+    return 0; \
+}
+
+#define GET_FILTER(fmt, name) \
+    case AF_FORMAT_##fmt: return filter_##name;
+
+#define FILTERS \
+    FILTER(FLOAT,   f)      \
+    FILTER(S32,     s32)    \
+    FILTER(U32,     u32)    \
+    FILTER(S24,     s24)    \
+    FILTER(U24,     u24)    \
+    FILTER(S16,     s16)    \
+    FILTER(U16,     u16)    \
+    FILTER(S8,      s8)     \
+    FILTER(U8,      u8)
+
+#define FILTER DEF_FILTER
+FILTERS
+#undef FILTER
+
+typedef int (*filter)(struct af_instance *af, struct mp_audio *d);
+static filter get_filter(int fmt)
+{
+    switch (fmt) {
+#define FILTER GET_FILTER
+FILTERS
+#undef FILTER
+    default: return NULL;
+    }
+}
+
+/// Initialization and runtime control
+static int control(struct af_instance *af, int cmd, void *arg)
+{
+    struct af_bs2b *s = af->priv;
+
+    switch (cmd) {
+    case AF_CONTROL_REINIT: {
+        int format;
+        // Sanity check
+        if (!arg) return AF_ERROR;
+
+        format           = ((struct mp_audio*)arg)->format;
+        af->data->rate   = ((struct mp_audio*)arg)->rate;
+        mp_audio_set_num_channels(af->data, 2);     // bs2b is useful only for 2ch audio
+        mp_audio_set_format(af->data, format);
+
+        /* check for formats supported by libbs2b
+           and assign corresponding handlers */
+        af->filter_frame = get_filter(format);
+        if (!af->filter_frame) {
+            af->filter_frame = filter_f;
+            mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
+        }
+
+        // bs2b have srate limits, try to resample if needed
+        if (af->data->rate > BS2B_MAXSRATE || af->data->rate < BS2B_MINSRATE) {
+            af->data->rate = BS2B_DEFAULT_SRATE;
+            MP_WARN(af, "Requested sample rate %d Hz is out of bounds [%d..%d] Hz.\n"
+                   "Trying to resample to %d Hz.\n",
+                   af->data->rate, BS2B_MINSRATE, BS2B_MAXSRATE, BS2B_DEFAULT_SRATE);
+        }
+        bs2b_set_srate(s->filter, (long)af->data->rate);
+        MP_VERBOSE(af, "using format %s\n",
+               af_fmt_to_str(af->data->format));
+
+        return af_test_output(af,(struct mp_audio*)arg);
+    }
+    }
+    return AF_UNKNOWN;
+}
+
+/// Deallocate memory and close library
+static void uninit(struct af_instance *af)
+{
+    struct af_bs2b *s = af->priv;
+    if (s->filter)
+        bs2b_close(s->filter);
+}
+
+/// Allocate memory, set function pointers and init library
+static int af_open(struct af_instance *af)
+{
+    struct af_bs2b *s = af->priv;
+    af->control = control;
+    af->uninit  = uninit;
+
+    // NULL means failed initialization
+    if (!(s->filter = bs2b_open())) {
+        return AF_ERROR;
+    }
+
+    if (s->profile)
+        bs2b_set_level(s->filter, s->profile);
+    // set fcut and feed only if specified, otherwise defaults will be used
+    if (s->fcut)
+        bs2b_set_level_fcut(s->filter, s->fcut);
+    if (s->feed)
+        bs2b_set_level_feed(s->filter, s->feed);
+    return AF_OK;
+}
+
+#define OPT_BASE_STRUCT struct af_bs2b
+
+/// Description of this filter
+const struct af_info af_info_bs2b = {
+    .info = "Bauer stereophonic-to-binaural audio filter",
+    .name = "bs2b",
+    .open = af_open,
+    .priv_size = sizeof(struct af_bs2b),
+    .options = (const struct m_option[]) {
+        OPT_INTRANGE("fcut", fcut, 0, BS2B_MINFCUT, BS2B_MAXFCUT),
+        OPT_INTRANGE("feed", feed, 0, BS2B_MINFEED, BS2B_MAXFEED),
+        OPT_CHOICE("profile", profile, 0,
+                   ({"unset", 0},
+                    {"default", BS2B_DEFAULT_CLEVEL},
+                    {"cmoy", BS2B_CMOY_CLEVEL},
+                    {"jmeier", BS2B_JMEIER_CLEVEL})),
+        {0}
+    },
+};

audio/filter/af_center.c

@@ -0,0 +1,104 @@
+/*
+ * This filter adds a center channel to the audio stream by
+ * averaging the left and right channel.
+ * There are two runtime controls one for setting which channel
+ * to insert the center-audio into called AF_CONTROL_SUB_CH.
+ *
+ * FIXME: implement a high-pass filter for better results.
+ *
+ * copyright (c) 2005 Alex Beregszaszi
+ *
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "common/common.h"
+#include "af.h"
+
+// Data for specific instances of this filter
+typedef struct af_center_s
+{
+  int ch;               // Channel number which to insert the filtered data
+}af_center_t;
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+  af_center_t* s   = af->priv;
+
+  switch(cmd){
+  case AF_CONTROL_REINIT:{
+    // Sanity check
+    if(!arg) return AF_ERROR;
+
+    af->data->rate   = ((struct mp_audio*)arg)->rate;
+    mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
+
+    return af_test_output(af,(struct mp_audio*)arg);
+  }
+  }
+  return AF_UNKNOWN;
+}
+
+static int filter_frame(struct af_instance* af, struct mp_audio* data)
+{
+  if (!data)
+    return 0;
+  if (af_make_writeable(af, data) < 0) {
+    talloc_free(data);
+    return -1;
+  }
+  struct mp_audio*    c   = data;        // Current working data
+  af_center_t*  s   = af->priv; // Setup for this instance
+  float*        a   = c->planes[0];      // Audio data
+  int           nch = c->nch;    // Number of channels
+  int           len = c->samples*c->nch;         // Number of samples in current audio block
+  int           ch  = s->ch;     // Channel in which to insert the center audio
+  register int  i;
+
+  // Run filter
+  for(i=0;i<len;i+=nch){
+    // Average left and right
+    a[i+ch] = (a[i]/2) + (a[i+1]/2);
+  }
+
+  af_add_output_frame(af, data);
+  return 0;
+}
+
+// Allocate memory and set function pointers
+static int af_open(struct af_instance* af){
+  af->control=control;
+  af->filter_frame = filter_frame;
+  return AF_OK;
+}
+
+#define OPT_BASE_STRUCT af_center_t
+const struct af_info af_info_center = {
+    .info = "Audio filter for adding a center channel",
+    .name = "center",
+    .flags = AF_FLAGS_NOT_REENTRANT,
+    .open = af_open,
+    .priv_size = sizeof(af_center_t),
+    .options = (const struct m_option[]) {
+        OPT_INTRANGE("channel", ch, 0, 0, AF_NCH - 1, OPTDEF_INT(1)),
+        {0}
+    },
+};

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

@@ -0,0 +1,194 @@
+/*
+ * This audio filter delays the output signal for the different
+ * channels and can be used for simple position panning.
+ * An extension for this filter would be a reverb.
+ *
+ * Original author: Anders
+ *
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <inttypes.h>
+
+#include "common/common.h"
+#include "af.h"
+
+#define L 65536
+
+#define UPDATEQI(qi) qi=(qi+1)&(L-1)
+
+// Data for specific instances of this filter
+typedef struct af_delay_s
+{
+  void* q[AF_NCH];      // Circular queues used for delaying audio signal
+  int   wi[AF_NCH];     // Write index
+  int   ri;             // Read index
+  float d[AF_NCH];      // Delay [ms]
+  char *delaystr;
+}af_delay_t;
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+  af_delay_t* s = af->priv;
+  switch(cmd){
+  case AF_CONTROL_REINIT:{
+    int i;
+    struct mp_audio *in = arg;
+
+    if (in->bps != 1 && in->bps != 2 && in->bps != 4) {
+      MP_FATAL(af, "Sample format not supported\n");
+      return AF_ERROR;
+    }
+
+    // Free prevous delay queues
+    for(i=0;i<af->data->nch;i++)
+      free(s->q[i]);
+
+    mp_audio_force_interleaved_format(in);
+    mp_audio_copy_config(af->data, in);
+
+    // Allocate new delay queues
+    for(i=0;i<af->data->nch;i++){
+      s->q[i] = calloc(L,af->data->bps);
+      if(NULL == s->q[i])
+        MP_FATAL(af, "Out of memory\n");
+    }
+
+    if(AF_OK != af_from_ms(AF_NCH, s->d, s->wi, af->data->rate, 0.0, 1000.0))
+      return AF_ERROR;
+    s->ri = 0;
+    for(i=0;i<AF_NCH;i++){
+      MP_DBG(af, "Channel %i delayed by %0.3fms\n",
+             i,MPCLAMP(s->d[i],0.0,1000.0));
+      MP_TRACE(af, "Channel %i delayed by %i samples\n",
+             i,s->wi[i]);
+    }
+    return AF_OK;
+  }
+  }
+  return AF_UNKNOWN;
+}
+
+// Deallocate memory
+static void uninit(struct af_instance* af)
+{
+  int i;
+
+  for(i=0;i<AF_NCH;i++)
+      free(((af_delay_t*)(af->priv))->q[i]);
+}
+
+static int filter_frame(struct af_instance *af, struct mp_audio *c)
+{
+  if (!c)
+    return 0;
+  af_delay_t*   s   = af->priv; // Setup for this instance
+  int           nch = c->nch;    // Number of channels
+  int           len = mp_audio_psize(c)/c->bps; // Number of sample in data chunk
+  int           ri  = 0;
+  int           ch,i;
+  if (af_make_writeable(af, c) < 0) {
+    talloc_free(c);
+    return -1;
+  }
+  for(ch=0;ch<nch;ch++){
+    switch(c->bps){
+    case 1:{
+      int8_t* a = c->planes[0];
+      int8_t* q = s->q[ch];
+      int wi = s->wi[ch];
+      ri = s->ri;
+      for(i=ch;i<len;i+=nch){
+        q[wi] = a[i];
+        a[i]  = q[ri];
+        UPDATEQI(wi);
+        UPDATEQI(ri);
+      }
+      s->wi[ch] = wi;
+      break;
+    }
+    case 2:{
+      int16_t* a = c->planes[0];
+      int16_t* q = s->q[ch];
+      int wi = s->wi[ch];
+      ri = s->ri;
+      for(i=ch;i<len;i+=nch){
+        q[wi] = a[i];
+        a[i]  = q[ri];
+        UPDATEQI(wi);
+        UPDATEQI(ri);
+      }
+      s->wi[ch] = wi;
+      break;
+    }
+    case 4:{
+      int32_t* a = c->planes[0];
+      int32_t* q = s->q[ch];
+      int wi = s->wi[ch];
+      ri = s->ri;
+      for(i=ch;i<len;i+=nch){
+        q[wi] = a[i];
+        a[i]  = q[ri];
+        UPDATEQI(wi);
+        UPDATEQI(ri);
+      }
+      s->wi[ch] = wi;
+      break;
+    }
+    }
+  }
+  s->ri = ri;
+  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->uninit=uninit;
+    af->filter_frame = filter_frame;
+    af_delay_t *s = af->priv;
+    int n = 1;
+    int i = 0;
+    char* cl = s->delaystr;
+    while(cl && n && i < AF_NCH ){
+      sscanf(cl,"%f%n",&s->d[i],&n);
+      if(n==0 || cl[n-1] == '\0')
+        break;
+      cl=&cl[n];
+      if (*cl != ',')
+          break;
+      cl++;
+      i++;
+    }
+    return AF_OK;
+}
+
+#define OPT_BASE_STRUCT af_delay_t
+const struct af_info af_info_delay = {
+    .info = "Delay audio filter",
+    .name = "delay",
+    .open = af_open,
+    .priv_size = sizeof(af_delay_t),
+    .options = (const struct m_option[]) {
+        OPT_STRING("delays", delaystr, 0),
+        {0}
+    },
+};

audio/filter/af_drc.c

@@ -131,7 +131,7 @@ 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 +168,7 @@ 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 +216,7 @@ 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 +262,7 @@ 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 +324,7 @@ 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 +1,61 @@
+/*
+ * The name speaks for itself. This filter is a dummy and will
+ * not blow up regardless of what you do with it.
+ *
+ * Original author: Anders
+ *
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "af.h"
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+  switch(cmd){
+  case AF_CONTROL_REINIT: ;
+    *af->data = *(struct mp_audio*)arg;
+    MP_VERBOSE(af, "Was reinitialized: %iHz/%ich/%s\n",
+        af->data->rate,af->data->nch,af_fmt_to_str(af->data->format));
+    return AF_OK;
+  }
+  return AF_UNKNOWN;
+}
+
+// Filter data through filter
+static int filter(struct af_instance* af, struct mp_audio* data)
+{
+    af_add_output_frame(af, data);
+    return 0;
+}
+
+// Allocate memory and set function pointers
+static int af_open(struct af_instance* af){
+  af->control=control;
+  af->filter_frame=filter;
+  return AF_OK;
+}
+
+// Description of this filter
+const struct af_info af_info_dummy = {
+    .info = "dummy",
+    .name = "dummy",
+    .open = af_open,
+};

audio/filter/af_equalizer.c

@@ -203,6 +203,7 @@ 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 +1,237 @@
+/*
+ * This audio filter exports the incoming signal to other processes
+ * using memory mapping. The memory mapped area contains a header:
+ *   int nch,
+ *   int size,
+ *   unsigned long long counter (updated every time the  contents of
+ *                               the area changes),
+ * the rest is payload (non-interleaved).
+ *
+ * Authors: Anders; Gustavo Sverzut Barbieri <gustavo.barbieri@ic.unicamp.br>
+ *
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <inttypes.h>
+#include <unistd.h>
+#include "config.h"
+
+#include <sys/types.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
+#include "osdep/io.h"
+
+#include "talloc.h"
+#include "af.h"
+#include "options/path.h"
+
+#define DEF_SZ 512 // default buffer size (in samples)
+#define SHARED_FILE "mpv-af_export" /* default file name
+                                           (relative to ~/.mpv/ */
+
+#define SIZE_HEADER (2 * sizeof(int) + sizeof(unsigned long long))
+
+// Data for specific instances of this filter
+typedef struct af_export_s
+{
+  unsigned long long  count; // Used for sync
+  void* buf[AF_NCH];    // Buffers for storing the data before it is exported
+  int   sz;             // Size of buffer in samples
+  int   wi;             // Write index
+  int   fd;             // File descriptor to shared memory area
+  char* filename;       // File to export data
+  uint8_t *mmap_area;   // MMap shared area
+} af_export_t;
+
+
+/* Initialization and runtime control
+   af audio filter instance
+   cmd control command
+   arg argument
+*/
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+  af_export_t* s = af->priv;
+  switch (cmd){
+  case AF_CONTROL_REINIT:{
+    int i=0;
+    int mapsize;
+
+    // Free previous buffers
+    free(s->buf[0]);
+
+    // unmap previous area
+    if(s->mmap_area)
+      munmap(s->mmap_area, SIZE_HEADER + (af->data->bps*s->sz*af->data->nch));
+    // close previous file descriptor
+    if(s->fd)
+      close(s->fd);
+
+    // Accept only int16_t as input format (which sucks)
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_format(af->data, AF_FORMAT_S16);
+
+    // Allocate new buffers (as one continuous block)
+    s->buf[0] = calloc(s->sz*af->data->nch, af->data->bps);
+    if(NULL == s->buf[0]) {
+      MP_FATAL(af, "Out of memory\n");
+      return AF_ERROR;
+    }
+    for(i = 1; i < af->data->nch; i++)
+      s->buf[i] = (uint8_t *)s->buf[0] + i*s->sz*af->data->bps;
+
+    if (!s->filename) {
+        MP_FATAL(af, "No filename set.\n");
+        return AF_ERROR;
+    }
+
+    // Init memory mapping
+    s->fd = open(s->filename, O_RDWR | O_CREAT | O_TRUNC | O_CLOEXEC, 0640);
+    MP_INFO(af, "Exporting to file: %s\n", s->filename);
+    if(s->fd < 0) {
+      MP_FATAL(af, "Could not open/create file: %s\n",
+             s->filename);
+      return AF_ERROR;
+    }
+
+    // header + buffer
+    mapsize = (SIZE_HEADER + (af->data->bps * s->sz * af->data->nch));
+
+    // grow file to needed size
+    for(i = 0; i < mapsize; i++){
+      char null = 0;
+      write(s->fd, (void*) &null, 1);
+    }
+
+    // mmap size
+    s->mmap_area = mmap(0, mapsize, PROT_READ|PROT_WRITE,MAP_SHARED, s->fd, 0);
+    if(s->mmap_area == NULL)
+      MP_FATAL(af, "Could not mmap file %s\n", s->filename);
+    MP_INFO(af, "Memory mapped to file: %s (%p)\n",
+           s->filename, s->mmap_area);
+
+    // Initialize header
+    *((int*)s->mmap_area) = af->data->nch;
+    *((int*)s->mmap_area + 1) = s->sz * af->data->bps * af->data->nch;
+    msync(s->mmap_area, mapsize, MS_ASYNC);
+
+    // Use test_output to return FALSE if necessary
+    return af_test_output(af, (struct mp_audio*)arg);
+  }
+  }
+  return AF_UNKNOWN;
+}
+
+/* Free allocated memory and clean up other stuff too.
+   af audio filter instance
+*/
+static void uninit( struct af_instance* af )
+{
+    af_export_t* s = af->priv;
+
+    free(s->buf[0]);
+
+    // Free mmaped area
+    if(s->mmap_area)
+      munmap(s->mmap_area, sizeof(af_export_t));
+
+    if(s->fd > -1)
+      close(s->fd);
+}
+
+/* Filter data through filter
+   af audio filter instance
+   data audio data
+*/
+static int filter(struct af_instance *af, struct mp_audio *data)
+{
+  if (!data)
+    return 0;
+  struct mp_audio*      c   = data;          // Current working data
+  af_export_t*  s   = af->priv;     // Setup for this instance
+  int16_t*      a   = c->planes[0];          // Incomming sound
+  int           nch = c->nch;        // Number of channels
+  int           len = c->samples*c->nch; // Number of sample in data chunk
+  int           sz  = s->sz;         // buffer size (in samples)
+  int           flag = 0;            // Set to 1 if buffer is filled
+
+  int           ch, i;
+
+  // Fill all buffers
+  for(ch = 0; ch < nch; ch++){
+    int         wi = s->wi;      // Reset write index
+    int16_t*    b  = s->buf[ch]; // Current buffer
+
+    // Copy data to export buffers
+    for(i = ch; i < len; i += nch){
+      b[wi++] = a[i];
+      if(wi >= sz){ // Don't write outside the end of the buffer
+        flag = 1;
+        break;
+      }
+    }
+    s->wi = wi % s->sz;
+  }
+
+  // Export buffer to mmaped area
+  if(flag){
+    // update buffer in mapped area
+    memcpy(s->mmap_area + SIZE_HEADER, s->buf[0], sz * c->bps * nch);
+    s->count++; // increment counter (to sync)
+    memcpy(s->mmap_area + SIZE_HEADER - sizeof(s->count),
+           &(s->count), sizeof(s->count));
+  }
+
+  af_add_output_frame(af, data);
+  return 0;
+}
+
+/* Allocate memory and set function pointers
+   af audio filter instance
+   returns AF_OK or AF_ERROR
+*/
+static int af_open( struct af_instance* af )
+{
+  af->control = control;
+  af->uninit  = uninit;
+  af->filter_frame = filter;
+  af_export_t *priv = af->priv;
+
+  if (!priv->filename || !priv->filename[0]) {
+      MP_FATAL(af, "no export filename given");
+      return AF_ERROR;
+  }
+
+  return AF_OK;
+}
+
+#define OPT_BASE_STRUCT af_export_t
+const struct af_info af_info_export = {
+    .info = "Sound export filter",
+    .name = "export",
+    .open = af_open,
+    .priv_size = sizeof(af_export_t),
+    .options = (const struct m_option[]) {
+        OPT_STRING("filename", filename, 0),
+        OPT_INTRANGE("buffersamples", sz, 0, 1, 2048, OPTDEF_INT(DEF_SZ)),
+        {0}
+    },
+};

audio/filter/af_extrastereo.c

@@ -0,0 +1,132 @@
+/*
+ * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard
+ *
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <inttypes.h>
+#include <math.h>
+#include <limits.h>
+
+#include "common/common.h"
+#include "af.h"
+
+// Data for specific instances of this filter
+typedef struct af_extrastereo_s
+{
+    float mul;
+}af_extrastereo_t;
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+  switch(cmd){
+  case AF_CONTROL_REINIT:{
+    // Sanity check
+    if(!arg) return AF_ERROR;
+
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_force_interleaved_format(af->data);
+    mp_audio_set_num_channels(af->data, 2);
+    if (af->data->format != AF_FORMAT_FLOAT)
+        mp_audio_set_format(af->data, AF_FORMAT_S16);
+
+    return af_test_output(af,(struct mp_audio*)arg);
+  }
+  }
+  return AF_UNKNOWN;
+}
+
+// Filter data through filter
+static void play_s16(af_extrastereo_t *s, struct mp_audio* data)
+{
+  register int i = 0;
+  int16_t *a = (int16_t*)data->planes[0];       // Audio data
+  int len = data->samples*data->nch;            // Number of samples
+  int avg, l, r;
+
+  for (i = 0; i < len; i+=2)
+  {
+    avg = (a[i] + a[i + 1]) / 2;
+
+    l = avg + (int)(s->mul * (a[i] - avg));
+    r = avg + (int)(s->mul * (a[i + 1] - avg));
+
+    a[i] = MPCLAMP(l, SHRT_MIN, SHRT_MAX);
+    a[i + 1] = MPCLAMP(r, SHRT_MIN, SHRT_MAX);
+  }
+}
+
+static void play_float(af_extrastereo_t *s, struct mp_audio* data)
+{
+  register int i = 0;
+  float *a = (float*)data->planes[0];   // Audio data
+  int len = data->samples * data->nch;  // Number of samples
+  float avg, l, r;
+
+  for (i = 0; i < len; i+=2)
+  {
+    avg = (a[i] + a[i + 1]) / 2;
+
+    l = avg + (s->mul * (a[i] - avg));
+    r = avg + (s->mul * (a[i + 1] - avg));
+
+    a[i] = af_softclip(l);
+    a[i + 1] = af_softclip(r);
+  }
+}
+
+static int filter_frame(struct af_instance *af, struct mp_audio *data)
+{
+    if (!data)
+        return 0;
+    if (af_make_writeable(af, data) < 0) {
+        talloc_free(data);
+        return -1;
+    }
+    if (data->format == AF_FORMAT_FLOAT) {
+        play_float(af->priv, data);
+    } else {
+        play_s16(af->priv, data);
+    }
+    af_add_output_frame(af, data);
+    return 0;
+}
+
+// Allocate memory and set function pointers
+static int af_open(struct af_instance* af){
+  af->control=control;
+  af->filter_frame = filter_frame;
+
+  return AF_OK;
+}
+
+#define OPT_BASE_STRUCT af_extrastereo_t
+const struct af_info af_info_extrastereo = {
+    .info = "Increase difference between audio channels",
+    .name = "extrastereo",
+    .flags = AF_FLAGS_NOT_REENTRANT,
+    .open = af_open,
+    .priv_size = sizeof(af_extrastereo_t),
+    .options = (const struct m_option[]) {
+        OPT_FLOAT("mul", mul, 0, OPTDEF_FLOAT(2.5)),
+        {0}
+    },
+};

audio/filter/af_format.c

@@ -29,10 +29,10 @@ struct priv {
 
     int in_format;
     int in_srate;
-    struct m_channels in_channels;
+    struct mp_chmap in_channels;
     int out_format;
     int out_srate;
-    struct m_channels out_channels;
+    struct mp_chmap out_channels;
 
     int fail;
 };
@@ -44,8 +44,8 @@ static void force_in_params(struct af_instance *af, struct mp_audio *in)
     if (priv->in_format != AF_FORMAT_UNKNOWN)
         mp_audio_set_format(in, priv->in_format);
 
-    if (priv->in_channels.num_chmaps > 0)
-        mp_audio_set_channels(in, &priv->in_channels.chmaps[0]);
+    if (priv->in_channels.num)
+        mp_audio_set_channels(in, &priv->in_channels);
 
     if (priv->in_srate)
         in->rate = priv->in_srate;
@@ -58,8 +58,8 @@ static void force_out_params(struct af_instance *af, struct mp_audio *out)
     if (priv->out_format != AF_FORMAT_UNKNOWN)
         mp_audio_set_format(out, priv->out_format);
 
-    if (priv->out_channels.num_chmaps > 0)
-        mp_audio_set_channels(out, &priv->out_channels.chmaps[0]);
+    if (priv->out_channels.num)
+        mp_audio_set_channels(out, &priv->out_channels);
 
     if (priv->out_srate)
         out->rate = priv->out_srate;
@@ -124,10 +124,10 @@ const struct af_info af_info_format = {
     .options = (const struct m_option[]) {
         OPT_AUDIOFORMAT("format", in_format, 0),
         OPT_INTRANGE("srate", in_srate, 0, 1000, 8*48000),
-        OPT_CHANNELS("channels", in_channels, 0, .min = 1),
+        OPT_CHMAP("channels", in_channels, CONF_MIN, .min = 0),
         OPT_AUDIOFORMAT("out-format", out_format, 0),
         OPT_INTRANGE("out-srate", out_srate, 0, 1000, 8*48000),
-        OPT_CHANNELS("out-channels", out_channels, 0, .min = 1),
+        OPT_CHMAP("out-channels", out_channels, CONF_MIN, .min = 0),
         OPT_FLAG("fail", fail, 0),
         {0}
     },

audio/filter/af_hrtf.c

@@ -0,0 +1,670 @@
+/*
+ * Experimental audio filter that mixes 5.1 and 5.1 with matrix
+ * encoded rear channels into headphone signal using FIR filtering
+ * with HRTF.
+ *
+ * Author: ylai
+ *
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+//#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <inttypes.h>
+
+#include <math.h>
+#include <libavutil/common.h>
+
+#include "af.h"
+#include "dsp.h"
+
+/* HRTF filter coefficients and adjustable parameters */
+#include "af_hrtf.h"
+
+typedef struct af_hrtf_s {
+    /* Lengths */
+    int dlbuflen, hrflen, basslen;
+    /* L, C, R, Ls, Rs channels */
+    float *lf, *rf, *lr, *rr, *cf, *cr;
+    const float *cf_ir, *af_ir, *of_ir, *ar_ir, *or_ir, *cr_ir;
+    int cf_o, af_o, of_o, ar_o, or_o, cr_o;
+    /* Bass */
+    float *ba_l, *ba_r;
+    float *ba_ir;
+    /* Whether to matrix decode the rear center channel */
+    int matrix_mode;
+    /* How to decode the input:
+       0 = 5/5+1 channels
+       1 = 2 channels
+       2 = matrix encoded 2 channels */
+    int decode_mode;
+    /* Full wave rectified (FWR) amplitudes and gain used to steer the
+       active matrix decoding of front channels (variable names
+       lpr/lmr means Lt + Rt, Lt - Rt) */
+    float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
+    float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
+    /* Matrix input decoding require special FWR buffer, since the
+       decoding is done in place. */
+    float *fwrbuf_l, *fwrbuf_r, *fwrbuf_lr, *fwrbuf_rr;
+    /* Rear channel delay buffer for matrix decoding */
+    float *rear_dlbuf;
+    /* Full wave rectified amplitude and gain used to steer the active
+       matrix decoding of center rear channel */
+    float lr_fwr, rr_fwr, lrprr_fwr, lrmrr_fwr;
+    float adapt_lr_gain, adapt_rr_gain;
+    float adapt_lrprr_gain, adapt_lrmrr_gain;
+    /* Cyclic position on the ring buffer */
+    int cyc_pos;
+    int print_flag;
+    int mode;
+} af_hrtf_t;
+
+/* Convolution on a ring buffer
+ *    nx:       length of the ring buffer
+ *    nk:       length of the convolution kernel
+ *    sx:       ring buffer
+ *    sk:       convolution kernel
+ *    offset:   offset on the ring buffer, can be
+ */
+static float conv(const int nx, const int nk, const float *sx, const float *sk,
+                  const int offset)
+{
+    /* k = reminder of offset / nx */
+    int k = offset >= 0 ? offset % nx : nx + (offset % nx);
+
+    if(nk + k <= nx)
+        return af_filter_fir(nk, sx + k, sk);
+    else
+        return af_filter_fir(nk + k - nx, sx, sk + nx - k) +
+            af_filter_fir(nx - k, sx + k, sk);
+}
+
+/* Detect when the impulse response starts (significantly) */
+static int pulse_detect(const float *sx)
+{
+    /* nmax must be the reference impulse response length (128) minus
+       s->hrflen */
+    const int nmax = 128 - HRTFFILTLEN;
+    const float thresh = IRTHRESH;
+    int i;
+
+    for(i = 0; i < nmax; i++)
+        if(fabs(sx[i]) > thresh)
+            return i;
+    return 0;
+}
+
+/* Fuzzy matrix coefficient transfer function to "lock" the matrix on
+   a effectively passive mode if the gain is approximately 1 */
+static inline float passive_lock(float x)
+{
+   const float x1 = x - 1;
+   const float ax1s = fabs(x - 1) * (1.0 / MATAGCLOCK);
+
+   return x1 - x1 / (1 + ax1s * ax1s) + 1;
+}
+
+/* Unified active matrix decoder for 2 channel matrix encoded surround
+   sources */
+static inline void matrix_decode(short *in, const int k, const int il,
+                          const int ir, const int decode_rear,
+                          const int dlbuflen,
+                          float l_fwr, float r_fwr,
+                          float lpr_fwr, float lmr_fwr,
+                          float *adapt_l_gain, float *adapt_r_gain,
+                          float *adapt_lpr_gain, float *adapt_lmr_gain,
+                          float *lf, float *rf, float *lr,
+                          float *rr, float *cf)
+{
+   const int kr = (k + MATREARDELAY) % dlbuflen;
+   float l_gain = (l_fwr + r_fwr) /
+      (1 + l_fwr + l_fwr);
+   float r_gain = (l_fwr + r_fwr) /
+      (1 + r_fwr + r_fwr);
+   /* The 2nd axis has strong gain fluctuations, and therefore require
+      limits.  The factor corresponds to the 1 / amplification of (Lt
+      - Rt) when (Lt, Rt) is strongly correlated. (e.g. during
+      dialogues).  It should be bigger than -12 dB to prevent
+      distortion. */
+   float lmr_lim_fwr = lmr_fwr > M9_03DB * lpr_fwr ?
+      lmr_fwr : M9_03DB * lpr_fwr;
+   float lpr_gain = (lpr_fwr + lmr_lim_fwr) /
+      (1 + lpr_fwr + lpr_fwr);
+   float lmr_gain = (lpr_fwr + lmr_lim_fwr) /
+      (1 + lmr_lim_fwr + lmr_lim_fwr);
+   float lmr_unlim_gain = (lpr_fwr + lmr_fwr) /
+      (1 + lmr_fwr + lmr_fwr);
+   float lpr, lmr;
+   float l_agc, r_agc, lpr_agc, lmr_agc;
+   float f, d_gain, c_gain, c_agc_cfk;
+
+#if 0
+   static int counter = 0;
+   static FILE *fp_out;
+
+   if(counter == 0)
+      fp_out = fopen("af_hrtf.log", "w");
+   if(counter % 240 == 0)
+      fprintf(fp_out, "%g %g %g %g %g ", counter * (1.0 / 48000),
+              l_gain, r_gain, lpr_gain, lmr_gain);
+#endif
+
+   /*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/
+   /* AGC adaption */
+   d_gain = (fabs(l_gain - *adapt_l_gain) +
+             fabs(r_gain - *adapt_r_gain)) * 0.5;
+   f = d_gain * (1.0 / MATAGCTRIG);
+   f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
+   *adapt_l_gain = (1 - f) * *adapt_l_gain + f * l_gain;
+   *adapt_r_gain = (1 - f) * *adapt_r_gain + f * r_gain;
+   /* Matrix */
+   l_agc = in[il] * passive_lock(*adapt_l_gain);
+   r_agc = in[ir] * passive_lock(*adapt_r_gain);
+   cf[k] = (l_agc + r_agc) * M_SQRT1_2;
+   if(decode_rear) {
+      lr[kr] = rr[kr] = (l_agc - r_agc) * M_SQRT1_2;
+      /* Stereo rear channel is steered with the same AGC steering as
+         the decoding matrix. Note this requires a fast updating AGC
+         at the order of 20 ms (which is the case here). */
+      lr[kr] *= (l_fwr + l_fwr) /
+         (1 + l_fwr + r_fwr);
+      rr[kr] *= (r_fwr + r_fwr) /
+         (1 + l_fwr + r_fwr);
+   }
+
+   /*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
+   lpr = (in[il] + in[ir]) * M_SQRT1_2;
+   lmr = (in[il] - in[ir]) * M_SQRT1_2;
+   /* AGC adaption */
+   d_gain = fabs(lmr_unlim_gain - *adapt_lmr_gain);
+   f = d_gain * (1.0 / MATAGCTRIG);
+   f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
+   *adapt_lpr_gain = (1 - f) * *adapt_lpr_gain + f * lpr_gain;
+   *adapt_lmr_gain = (1 - f) * *adapt_lmr_gain + f * lmr_gain;
+   /* Matrix */
+   lpr_agc = lpr * passive_lock(*adapt_lpr_gain);
+   lmr_agc = lmr * passive_lock(*adapt_lmr_gain);
+   lf[k] = (lpr_agc + lmr_agc) * M_SQRT1_2;
+   rf[k] = (lpr_agc - lmr_agc) * M_SQRT1_2;
+
+   /*** CENTER FRONT CANCELLATION ***/
+   /* A heuristic approach exploits that Lt + Rt gain contains the
+      information about Lt, Rt correlation.  This effectively reshapes
+      the front and rear "cones" to concentrate Lt + Rt to C and
+      introduce Lt - Rt in L, R. */
+   /* 0.67677 is the emprical lower bound for lpr_gain. */
+   c_gain = 8 * (*adapt_lpr_gain - 0.67677);
+   c_gain = c_gain > 0 ? c_gain : 0;
+   /* c_gain should not be too high, not even reaching full
+      cancellation (~ 0.50 - 0.55 at current AGC implementation), or
+      the center will s0und too narrow. */
+   c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
+   c_agc_cfk = c_gain * cf[k];
+   lf[k] -= c_agc_cfk;
+   rf[k] -= c_agc_cfk;
+   cf[k] += c_agc_cfk + c_agc_cfk;
+#if 0
+   if(counter % 240 == 0)
+      fprintf(fp_out, "%g %g %g %g %g\n",
+              *adapt_l_gain, *adapt_r_gain,
+              *adapt_lpr_gain, *adapt_lmr_gain,
+              c_gain);
+   counter++;
+#endif
+}
+
+static inline void update_ch(af_hrtf_t *s, short *in, const int k)
+{
+    const int fwr_pos = (k + FWRDURATION) % s->dlbuflen;
+    /* Update the full wave rectified total amplitude */
+    /* Input matrix decoder */
+    if(s->decode_mode == HRTF_MIX_MATRIX2CH) {
+       s->l_fwr += abs(in[0]) - fabs(s->fwrbuf_l[fwr_pos]);
+       s->r_fwr += abs(in[1]) - fabs(s->fwrbuf_r[fwr_pos]);
+       s->lpr_fwr += abs(in[0] + in[1]) -
+          fabs(s->fwrbuf_l[fwr_pos] + s->fwrbuf_r[fwr_pos]);
+       s->lmr_fwr += abs(in[0] - in[1]) -
+          fabs(s->fwrbuf_l[fwr_pos] - s->fwrbuf_r[fwr_pos]);
+    }
+    /* Rear matrix decoder */
+    if(s->matrix_mode) {
+       s->lr_fwr += abs(in[2]) - fabs(s->fwrbuf_lr[fwr_pos]);
+       s->rr_fwr += abs(in[3]) - fabs(s->fwrbuf_rr[fwr_pos]);
+       s->lrprr_fwr += abs(in[2] + in[3]) -
+          fabs(s->fwrbuf_lr[fwr_pos] + s->fwrbuf_rr[fwr_pos]);
+       s->lrmrr_fwr += abs(in[2] - in[3]) -
+          fabs(s->fwrbuf_lr[fwr_pos] - s->fwrbuf_rr[fwr_pos]);
+    }
+
+    switch (s->decode_mode) {
+    case HRTF_MIX_51:
+       /* 5/5+1 channel sources */
+       s->lf[k] = in[0];
+       s->cf[k] = in[4];
+       s->rf[k] = in[1];
+       s->fwrbuf_lr[k] = s->lr[k] = in[2];
+       s->fwrbuf_rr[k] = s->rr[k] = in[3];
+       break;
+    case HRTF_MIX_MATRIX2CH:
+       /* Matrix encoded 2 channel sources */
+       s->fwrbuf_l[k] = in[0];
+       s->fwrbuf_r[k] = in[1];
+       matrix_decode(in, k, 0, 1, 1, s->dlbuflen,
+                     s->l_fwr, s->r_fwr,
+                     s->lpr_fwr, s->lmr_fwr,
+                     &(s->adapt_l_gain), &(s->adapt_r_gain),
+                     &(s->adapt_lpr_gain), &(s->adapt_lmr_gain),
+                     s->lf, s->rf, s->lr, s->rr, s->cf);
+       break;
+    case HRTF_MIX_STEREO:
+       /* Stereo sources */
+       s->lf[k] = in[0];
+       s->rf[k] = in[1];
+       s->cf[k] = s->lr[k] = s->rr[k] = 0;
+       break;
+    }
+
+    /* We need to update the bass compensation delay line, too. */
+    // TODO: should this use lf/cf/rf etc. instead?
+    s->ba_l[k] = in[0];
+    s->ba_r[k] = in[1];
+    if (s->decode_mode == HRTF_MIX_51) {
+        s->ba_l[k] += in[4] + in[2];
+        s->ba_r[k] += in[4] + in[3];
+    }
+}
+
+static void clear_coeff(af_hrtf_t *s, float *c)
+{
+    memset(c, 0, s->dlbuflen * sizeof(float));
+}
+
+static void reset(af_hrtf_t *s)
+{
+    clear_coeff(s, s->lf);
+    clear_coeff(s, s->rf);
+    clear_coeff(s, s->lr);
+    clear_coeff(s, s->rr);
+    clear_coeff(s, s->cf);
+    clear_coeff(s, s->cr);
+    clear_coeff(s, s->ba_l);
+    clear_coeff(s, s->ba_r);
+    clear_coeff(s, s->fwrbuf_l);
+    clear_coeff(s, s->fwrbuf_r);
+    clear_coeff(s, s->fwrbuf_lr);
+    clear_coeff(s, s->fwrbuf_rr);
+}
+
+/* Initialization and runtime control */
+static int control(struct af_instance *af, int cmd, void* arg)
+{
+    af_hrtf_t *s = af->priv;
+    int test_output_res;
+
+    switch(cmd) {
+    case AF_CONTROL_REINIT:
+        reset(s);
+        af->data->rate = 48000;
+        mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch);
+        if(af->data->nch == 2) {
+            /* 2 channel input */
+            if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
+                /* Default behavior is stereo mixing. */
+                s->decode_mode = HRTF_MIX_STEREO;
+            }
+        } else if (af->data->nch < 5) {
+            mp_audio_set_channels_old(af->data, 5);
+        }
+        mp_audio_set_format(af->data, AF_FORMAT_S16);
+        test_output_res = af_test_output(af, (struct mp_audio*)arg);
+        // after testing input set the real output format
+        mp_audio_set_num_channels(af->data, 2);
+        s->print_flag = 1;
+        return test_output_res;
+    case AF_CONTROL_RESET:
+        reset(s);
+        return AF_OK;
+    }
+
+    return AF_UNKNOWN;
+}
+
+/* Deallocate memory */
+static void uninit(struct af_instance *af)
+{
+        af_hrtf_t *s = af->priv;
+
+        free(s->lf);
+        free(s->rf);
+        free(s->lr);
+        free(s->rr);
+        free(s->cf);
+        free(s->cr);
+        free(s->ba_l);
+        free(s->ba_r);
+        free(s->ba_ir);
+        free(s->fwrbuf_l);
+        free(s->fwrbuf_r);
+        free(s->fwrbuf_lr);
+        free(s->fwrbuf_rr);
+}
+
+/* Filter data through filter
+
+Two "tricks" are used to compensate the "color" of the KEMAR data:
+
+1. The KEMAR data is refiltered to ensure that the front L, R channels
+on the same side of the ear are equalized (especially in the high
+frequencies).
+
+2. A bass compensation is introduced to ensure that 0-200 Hz are not
+damped (without any real 3D acoustical image, however).
+*/
+static int filter(struct af_instance *af, struct mp_audio *data)
+{
+    af_hrtf_t *s = af->priv;
+
+    if (!data)
+        return 0;
+    struct mp_audio *outframe =
+        mp_audio_pool_get(af->out_pool, &af->fmt_out, data->samples);
+    if (!outframe) {
+        talloc_free(data);
+        return -1;
+    }
+    mp_audio_copy_attributes(outframe, data);
+
+    short *in = data->planes[0]; // Input audio data
+    short *out = outframe->planes[0]; // Output audio data
+    short *end = in + data->samples * data->nch; // Loop end
+    float common, left, right, diff, left_b, right_b;
+    const int dblen = s->dlbuflen, hlen = s->hrflen, blen = s->basslen;
+
+    if(s->print_flag) {
+        s->print_flag = 0;
+        switch (s->decode_mode) {
+        case HRTF_MIX_51:
+          MP_INFO(af, "Using HRTF to mix %s discrete surround into "
+                 "L, R channels\n", s->matrix_mode ? "5+1" : "5");
+          break;
+        case HRTF_MIX_STEREO:
+          MP_INFO(af, "Using HRTF to mix stereo into "
+                 "L, R channels\n");
+          break;
+        case HRTF_MIX_MATRIX2CH:
+          MP_INFO(af, "Using active matrix to decode 2 channel "
+                 "input, HRTF to mix %s matrix surround into "
+                 "L, R channels\n", "3/2");
+          break;
+        default:
+          MP_WARN(af, "bogus decode_mode: %d\n", s->decode_mode);
+          break;
+        }
+
+       if(s->matrix_mode)
+          MP_INFO(af, "Using active matrix to decode rear center "
+                 "channel\n");
+    }
+
+    /* MPlayer's 5 channel layout (notation for the variable):
+     *
+     * 0: L (LF), 1: R (RF), 2: Ls (LR), 3: Rs (RR), 4: C (CF), matrix
+     * encoded: Cs (CR)
+     *
+     * or: L = left, C = center, R = right, F = front, R = rear
+     *
+     * Filter notation:
+     *
+     *      CF
+     * OF        AF
+     *      Ear->
+     * OR        AR
+     *      CR
+     *
+     * or: C = center, A = same side, O = opposite, F = front, R = rear
+     */
+
+    while(in < end) {
+        const int k = s->cyc_pos;
+
+        update_ch(s, in, k);
+
+        /* Simulate a 7.5 ms -20 dB echo of the center channel in the
+           front channels (like reflection from a room wall) - a kind of
+           psycho-acoustically "cheating" to focus the center front
+           channel, which is normally hard to be perceived as front */
+        s->lf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
+        s->rf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
+
+        switch (s->decode_mode) {
+        case HRTF_MIX_51:
+        case HRTF_MIX_MATRIX2CH:
+           /* Mixer filter matrix */
+           common = conv(dblen, hlen, s->cf, s->cf_ir, k + s->cf_o);
+           if(s->matrix_mode) {
+              /* In matrix decoding mode, the rear channel gain must be
+                 renormalized, as there is an additional channel. */
+              matrix_decode(in, k, 2, 3, 0, s->dlbuflen,
+                            s->lr_fwr, s->rr_fwr,
+                            s->lrprr_fwr, s->lrmrr_fwr,
+                            &(s->adapt_lr_gain), &(s->adapt_rr_gain),
+                            &(s->adapt_lrprr_gain), &(s->adapt_lrmrr_gain),
+                            s->lr, s->rr, NULL, NULL, s->cr);
+              common +=
+                 conv(dblen, hlen, s->cr, s->cr_ir, k + s->cr_o) *
+                 M1_76DB;
+              left    =
+                 ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
+                   conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
+                   (conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
+                    conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o)) *
+                   M1_76DB + common);
+              right   =
+                 ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
+                   conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
+                   (conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
+                    conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o)) *
+                   M1_76DB + common);
+           } else {
+              left    =
+                 ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
+                   conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
+                   conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
+                   conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o) +
+                   common);
+              right   =
+                 ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
+                   conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
+                   conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
+                   conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o) +
+                   common);
+           }
+           break;
+        case HRTF_MIX_STEREO:
+           left    =
+              ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
+                conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o));
+           right   =
+              ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
+                conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o));
+           break;
+        default:
+            /* make gcc happy */
+            left = 0.0;
+            right = 0.0;
+            break;
+        }
+
+        /* Bass compensation for the lower frequency cut of the HRTF.  A
+           cross talk of the left and right channel is introduced to
+           match the directional characteristics of higher frequencies.
+           The bass will not have any real 3D perception, but that is
+           OK (note at 180 Hz, the wavelength is about 2 m, and any
+           spatial perception is impossible). */
+        left_b  = conv(dblen, blen, s->ba_l, s->ba_ir, k);
+        right_b = conv(dblen, blen, s->ba_r, s->ba_ir, k);
+        left  += (1 - BASSCROSS) * left_b  + BASSCROSS * right_b;
+        right += (1 - BASSCROSS) * right_b + BASSCROSS * left_b;
+        /* Also mix the LFE channel (if available) */
+        if(data->nch >= 6) {
+            left  += in[5] * M3_01DB;
+            right += in[5] * M3_01DB;
+        }
+
+        /* Amplitude renormalization. */
+        left  *= AMPLNORM;
+        right *= AMPLNORM;
+
+        switch (s->decode_mode) {
+        case HRTF_MIX_51:
+        case HRTF_MIX_STEREO:
+           /* "Cheating": linear stereo expansion to amplify the 3D
+              perception.  Note: Too much will destroy the acoustic space
+              and may even result in headaches. */
+           diff = STEXPAND2 * (left - right);
+           out[0] = av_clip_int16(left  + diff);
+           out[1] = av_clip_int16(right - diff);
+           break;
+        case HRTF_MIX_MATRIX2CH:
+           /* Do attempt any stereo expansion with matrix encoded
+              sources.  The L, R channels are already stereo expanded
+              by the steering, any further stereo expansion will sound
+              very unnatural. */
+           out[0] = av_clip_int16(left);
+           out[1] = av_clip_int16(right);
+           break;
+        }
+
+        /* Next sample... */
+        in = &in[data->nch];
+        out = &out[af->data->nch];
+        (s->cyc_pos)--;
+        if(s->cyc_pos < 0)
+            s->cyc_pos += dblen;
+    }
+
+    talloc_free(data);
+    af_add_output_frame(af, outframe);
+    return 0;
+}
+
+static int allocate(af_hrtf_t *s)
+{
+    if ((s->lf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->rf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->lr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->rr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->cf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->cr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->ba_l = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->ba_r = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->fwrbuf_l =
+         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->fwrbuf_r =
+         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->fwrbuf_lr =
+         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    if ((s->fwrbuf_rr =
+         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+    return 0;
+}
+
+/* Allocate memory and set function pointers */
+static int af_open(struct af_instance* af)
+{
+    int i;
+    af_hrtf_t *s;
+    float fc;
+
+    af->control = control;
+    af->uninit = uninit;
+    af->filter_frame = filter;
+
+    s = af->priv;
+
+    s->dlbuflen = DELAYBUFLEN;
+    s->hrflen = HRTFFILTLEN;
+    s->basslen = BASSFILTLEN;
+
+    s->cyc_pos = s->dlbuflen - 1;
+    /* With a full (two axis) steering matrix decoder, s->matrix_mode
+       should not be enabled lightly (it will also steer the Ls, Rs
+       channels). */
+    s->matrix_mode = 0;
+    s->decode_mode = HRTF_MIX_51;
+
+    switch (s->mode) {
+    case 0: /* Use matrix rear decoding. */
+        s->matrix_mode = 1;
+        break;
+    case 1: /* Input needs matrix decoding. */
+        s->decode_mode = HRTF_MIX_MATRIX2CH;
+        break;
+    case 2:
+        s->matrix_mode = 0;
+        break;
+    }
+
+    s->print_flag = 1;
+
+    if (allocate(s) != 0) {
+        MP_ERR(af, "Memory allocation error.\n");
+        return AF_ERROR;
+    }
+
+    for(i = 0; i < s->dlbuflen; i++)
+        s->lf[i] = s->rf[i] = s->lr[i] = s->rr[i] = s->cf[i] =
+            s->cr[i] = 0;
+
+    s->lr_fwr =
+        s->rr_fwr = 0;
+
+    s->cf_ir = cf_filt + (s->cf_o = pulse_detect(cf_filt));
+    s->af_ir = af_filt + (s->af_o = pulse_detect(af_filt));
+    s->of_ir = of_filt + (s->of_o = pulse_detect(of_filt));
+    s->ar_ir = ar_filt + (s->ar_o = pulse_detect(ar_filt));
+    s->or_ir = or_filt + (s->or_o = pulse_detect(or_filt));
+    s->cr_ir = cr_filt + (s->cr_o = pulse_detect(cr_filt));
+
+    if((s->ba_ir = malloc(s->basslen * sizeof(float))) == NULL) {
+        MP_ERR(af, "Memory allocation error.\n");
+        return AF_ERROR;
+    }
+    fc = 2.0 * BASSFILTFREQ / (float)af->data->rate;
+    if(af_filter_design_fir(s->basslen, s->ba_ir, &fc, LP | KAISER, 4 * M_PI) ==
+       -1) {
+        MP_ERR(af, "Unable to design low-pass "
+               "filter.\n");
+        return AF_ERROR;
+    }
+    for(i = 0; i < s->basslen; i++)
+        s->ba_ir[i] *= BASSGAIN;
+
+    return AF_OK;
+}
+
+#define OPT_BASE_STRUCT af_hrtf_t
+const struct af_info af_info_hrtf = {
+    .info = "HRTF Headphone",
+    .name = "hrtf",
+    .open = af_open,
+    .priv_size = sizeof(af_hrtf_t),
+    .options = (const struct m_option[]) {
+        OPT_CHOICE("mode", mode, 0, ({"m", 0}, {"s", 1}, {"0", 2})),
+        {0}
+    },
+};

audio/filter/af_hrtf.h

@@ -0,0 +1,510 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef MPLAYER_AF_HRTF_H
+#define MPLAYER_AF_HRTF_H
+
+#define HRTF_MIX_51 0
+#define HRTF_MIX_STEREO 1
+#define HRTF_MIX_MATRIX2CH 2
+
+/* Amplitude scaling factors */
+#define M17_0DB         0.1414213562
+#define M9_03DB         0.3535533906
+#define M6_99DB         0.4472135955
+#define M4_77DB         0.5773502692
+#define M3_01DB         0.7071067812
+#define M1_76DB         0.8164965809
+
+#define DELAYBUFLEN     1024    /* Length of the delay buffer */
+#define HRTFFILTLEN     64      /* HRTF filter length */
+#define IRTHRESH        0.001   /* Impulse response pruning thresh. */
+
+#define AMPLNORM        M6_99DB /* Overall amplitude renormalization */
+
+#define BASSFILTFREQ    180     /* Bass compensation filter cut (Hz) */
+#define BASSFILTLEN     193     /* Bass compensation filter length */
+#define BASSGAIN        M_SQRT2 /* Bass compensation gain */
+#define BASSCROSS       0.35    /* Bass cross talk */
+
+#define FWRDURATION     240     /* FWR average duration (samples) */
+#define MATREARDELAY    720     /* Matrix mode rear delay (samples) */
+
+#define MATAGCTRIG      8.0     /* (Fuzzy) AGC trigger */
+#define MATAGCDECAY     1.0     /* AGC baseline decay rate (1/samp.) */
+#define MATAGCLOCK      0.2     /* AGC range (around 1) where the
+                                   matrix behaves passively */
+#define MATCOMPGAIN     0.37    /* Cross talk compensation gain,
+                                   0.50 - 0.55 is full cancellation. */
+
+#define CFECHODELAY     360     /* Center front echo delay (samples) */
+#define CFECHOAMPL      M17_0DB /* Center front echo amplitude */
+
+#define STEXPAND2       0.07    /* Stereo expansion / 2 */
+
+/* Head related impulse response (HRIR) derived from KEMAR measurement
+   data by Bill Gardner <billg@media.mit.edu> and Keith Martin
+   <kdm@media.mit.edu>
+
+   URL: http://sound.media.mit.edu/KEMAR.html
+
+   Distributed under GPL with authors' permission
+*/
+
+/* EQUALIZED KEMAR HRIR
+
+How to generate these data:
+
+1. You need the MIT Media Lab's KEMAR data, read it into an software
+   capable of signal/time series analysis (like Mathematica, Matlab,
+   ...)
+
+2. Construct an equalizing FIR inverse filter by calculating the
+   transfer function of the front, same side impulse response, then
+   take 1 over the absolute magnitude.
+
+3. Cut the poles in the inverse filter's transfer function
+   specification by limiting the amplification to 2.5 (note, this
+   number assumes that you have correct signal processing
+   normalization of the Fourier transform).
+
+4. Design the FIR inverse filter by calculating the inverse Fourier
+   transform, then chopping the coefficients down to a reasonable
+   number (N = 15 is used here), apply the Kaiser-Bessel window (alpha
+   = 2 is used here).  Note the objective is remove the color bias
+   only (as if you are using an equalizer), _not_ to do a full inverse
+   filtering. (Note: beta = pi*alpha in other notation.)
+
+   For N = 15, alpha = 2, you should get the following impulse
+   response:
+
+    0.001001558668605168,  0.00698457265741865,  0.040453643039829436,
+    0.012230541722147855, -0.11939760844854072,  0.16468099899755967,
+   -0.30297563073747436,   1.3140211791355982,  -0.30297563073747436,
+    0.16468099899755967,  -0.11939760844854072,  0.012230541722147855,
+    0.040453643039829436,  0.00698457265741865,  0.001001558668605168
+
+5. Linearly convolve all KEMAR HRIR with this inverse filter.
+
+6. Resample from 44.1 kHz sampling frequency to 48 kHz.
+*/
+
+/* Center front (-5 degree) - not 0 degree in order to create a clear
+   front image from a finite distance */
+static const float cf_filt[128] = {
+   -0.00008638082319075036, 0.0003198059946385229,
+   -0.0005010631339162132, 0.0011424741331126876,
+   -0.001584220794688753, 0.001742715363246275,
+   -0.0011080796626780694, 0.0001651829990860167,
+   0.005235028337314985, 0.0035223828473357776,
+   0.010057681388431303, -0.033469432129545514,
+   0.013391253316233523, 0.004858462839827063,
+   0.08172161220103627, 0.26158596134500023,
+   -0.12420314583323326, -0.14298458356097565,
+   0.14421897280453896, -0.1635792507629016,
+   -0.02187136722480014, 0.2426863044711817,
+   0.07229814207917194, 0.0942742651913879,
+   0.29856830878076834, 0.2944146162057754,
+   -0.12122157003421209, -0.19640092165631157,
+   0.11623836502034968, -0.05794027397995521,
+   -0.34313138341973776, -0.19487516249168105,
+   0.010118993953802401, -0.09460218797710966,
+   -0.16761521117359582, 0.004154461610153861,
+   0.052768641758969316, -0.00041823982226147407,
+   0.021634960445143514, 0.07562793486871108,
+   0.08895407129506479, 0.039857755093416214,
+   0.044257936180422945, 0.061557584906101664,
+   0.015547268541895703, -0.023908191934932484,
+   0.009498030443468223, 0.03816269003221337,
+   -0.009820500607303615, -0.042003975527908084,
+   -0.03335447117311547, -0.029294510859746596,
+   -0.05212623136198511, -0.073427547153205,
+   -0.061190797824958836, -0.04620925059966413,
+   -0.04204619420702159, -0.02331915902615157,
+   0.00211481411477094, 0.00852563995740107,
+   0.008766809731743399, 0.008666632180812078,
+   0.018279202191625352, 0.02924332751289675,
+   0.022293148257836494, 0.012362146008584188,
+   0.008572582458807008, 0.006491370763597344,
+   -0.0019366944997535774, -0.006318669309634434,
+   -0.006457921690218669, -0.015050265524669974,
+   -0.02110660282616213, -0.017027809096377904,
+   -0.01651052305334348, -0.022770064150046673,
+   -0.01999875754219472, -0.012294792027337775,
+   -0.011506057031057188, -0.011448970577312903,
+   -0.004823572302580925, 0.0022451134042777883,
+   0.004145473526859826, 0.005629030064546135,
+   0.008588029213398785, 0.010092048834844231,
+   0.007182013245552008, 0.0014600979508720656,
+   -0.0038314646272511756, -0.003443901997881347,
+   -0.0029483418254804047, -0.007609357112679647,
+   -0.006518368948030822, -0.004495803701497202,
+   -0.007109113004849672, -0.008346237278084265,
+   -0.005560847336252453, -0.002993453167553188,
+   -0.005122897816824269, -0.004389782626604215,
+   -0.0010912633695218108, -0.0019712029474458835,
+   -0.005870162265802235, -0.005626159534954128,
+   -0.00027254977910844407, 0.0013794425431633785,
+   -0.0005919083190430062, -0.0007861203545416682,
+   -0.0007049560240893946, -0.0032720188494468868,
+   -0.004460645567968504, -0.0032018528193571696,
+   -0.0030579229375062105, -0.003593998902656612,
+   -0.0038032977997776445, -0.004613776010454773,
+   -0.0059796549143736845, -0.00420126194319768,
+   -0.0012374419948287222, 0.0008572699213050608,
+   0.0021490971020081094, 0.00047295283198381995,
+   -0.0009670277915884887, -0.001354440866080231,
+   -0.002962902746547851, -0.00533935813338141,
+   -0.005469203016468759, -0.004355784273189485
+};
+/* Front,   same side (30 degree) */
+static const float af_filt[128] = {
+   -0.004140580614755493, 0.005790934614385445,
+   0.003318916682081112, 0.014257145544366063,
+   0.007328442487127339, -0.06550381777876194,
+   0.03502225818161845, -0.013090579770708259,
+   0.2517776798694195, 0.420770489950659,
+   -0.3418854608834852, -0.24371032493696737,
+   0.04901356150030018, -0.1778083521632833,
+   0.26448004245714163, 0.23245199964546834,
+   0.033053145803936305, 0.46811222821062415,
+   0.5359265986255611, -0.011912195468533787,
+   -0.39763432601411647, -0.034482864386898314,
+   0.029445398240649626, -0.3850940407863262,
+   -0.3272328478175581, -0.14701421403616477,
+   -0.08522137400169517, -0.14936851633336035,
+   -0.09432605283433723, 0.0991200405937827,
+   0.011075012089917331, -0.0051036489980274685,
+   0.0933903289749412, 0.1344189369609565,
+   0.10507466913017807, 0.04240159324684365,
+   0.06559270110638656, 0.026984119875617524,
+   -0.03359846103238096, -0.018000197099174275,
+   0.042031818548178244, 0.03849039666888434,
+   -0.02450829674011345, -0.03407882403088576,
+   -0.029230189282961977, -0.046964865291761734,
+   -0.09458258700116245, -0.1527349330901158,
+   -0.15411577687826097, -0.08761679790956928,
+   -0.033623549089171874, -0.007204768531481949,
+   0.008273425020444852, 0.021368717994908505,
+   0.04366608267875025, 0.05660907333076205,
+   0.06775726495503939, 0.05621881735271431,
+   0.03576231950669927, 0.02500825834889175,
+   0.015423811076054601, 0.007903258334503761,
+   -0.0053873014137761945, -0.006987955469434698,
+   -0.012027972007598602, -0.025228281243816594,
+   -0.026225091797257318, -0.023809293997344882,
+   -0.03250172017712295, -0.03195696301067249,
+   -0.01784813952189948, -0.01663267233760342,
+   -0.016184530450468065, -0.011659883749357463,
+   -0.0035378511240219163, -0.0005485800790443406,
+   0.0018432660108168625, 0.011634844139907534,
+   0.018333603402051105, 0.020447379185133056,
+   0.00850783664147828, 0.0004694148911037838,
+   -0.0017047130409786676, -0.0022409152834483997,
+   -0.000860472174892845, -0.004111075059198666,
+   -0.003527843382056666, -0.009640160874903018,
+   -0.01750044574231376, -0.015613389403672443,
+   -0.010790028120953001, -0.0095313499955768,
+   -0.007469721416726809, -0.0019186578145117315,
+   -0.00014977322572890802, -0.0029803838028179728,
+   -0.006520567233727221, 0.000035015132033882596,
+   0.009245098100543752, 0.009896930052308656,
+   0.008316744929565786, 0.004575207140193997,
+   -0.0000647420103997081, -0.004502916832871627,
+   -0.004225962213251224, -0.002886014126381486,
+   -0.006416834142585976, -0.007156609995423569,
+   -0.008840274447579472, -0.01441763751386817,
+   -0.015435817484659574, -0.00924487254924743,
+   -0.0021571721940235205, 0.0028540722992305453,
+   0.00273577475088536, -0.000788412365513805,
+   -0.0032650029728365907, -0.003880217646231338,
+   -0.0035302087299613778, -0.0038531436176586246,
+   -0.0011921632190514074, -0.0020722967099011938,
+   -0.004985351145629344, -0.0042375588844648735,
+   -0.003030360463006021, -0.0014161075428041471,
+   -0.0005083025643192044, 0.00035096963769606926
+};
+/* Front,   opposite (-30 degree) */
+static const float of_filt[128] = {
+   -0.000013472538374193126, -0.00008048061877079751,
+   0.000043927265781258155, -0.000017931700794858892,
+   -0.000034774602476112886, -0.00009576223008735474,
+   0.0001557797638630691, -0.00018742885883751094,
+   0.00026512448626705716, -0.0001451040203319678,
+   -0.00008263233117758043, 0.0006486245853639179,
+   -0.0010631408451846698, 0.0026571994100746143,
+   0.0014179177997092787, 0.0062326502956616256,
+   -0.008194149324545333, -0.006568029415878379,
+   0.009538759710818582, 0.012309193558632693,
+   0.12336638055838955, 0.046164307101829005,
+   -0.10228706407884815, 0.04047687260345798,
+   -0.00296595313977046, -0.07057949208414134,
+   0.08172114840714612, 0.08736490764127891,
+   0.05105250431333021, 0.11627179512747428,
+   0.20717888490340705, 0.09375052213570291,
+   -0.09784374168330194, -0.010493571845901443,
+   0.053131894303891716, -0.10157443971694806,
+   -0.16264032634244974, -0.05402369511361273,
+   -0.0274403608654217, -0.09860277022495063,
+   -0.06841875821090282, -0.004434574400066223,
+   -0.0005222661652743502, -0.006231881259827263,
+   0.014410397820340159, 0.04885649512730243,
+   0.04361962569042684, 0.03399214029009391,
+   0.04961073933475931, 0.04067325604132289,
+   0.007850647519227257, 0.004564440466905299,
+   0.02257107958021618, 0.008183791928884486,
+   -0.014913479343180557, -0.018685938460856224,
+   -0.01745737397226911, -0.02327177054233603,
+   -0.03723048632685227, -0.044739390162299685,
+   -0.042651220125613766, -0.03730017561004743,
+   -0.029039465434276192, -0.01885087458914294,
+   -0.01207127752277769, -0.006779800724164512,
+   -0.001930416967444157, 0.000029454577995528385,
+   0.0013822760965755472, 0.0014799128583230202,
+   0.0002068200609199832, 0.0022254295286201083,
+   0.005143858159434566, 0.0018580542060917013,
+   -0.0019426046325146259, -0.0014464042108543495,
+   -0.0034430083560735582, -0.009692758426099499,
+   -0.011840035292593485, -0.010716508855893968,
+   -0.012939889036853034, -0.0121846427935653,
+   -0.006198503315630782, -0.0023186723099380305,
+   -0.002679872498314837, -0.003086020446226295,
+   -0.0015709623347698936, -0.0008147490468332398,
+   -0.0012384575726770983, -0.0005212877089109362,
+   0.0017707578744906142, 0.001324932723905786,
+   -0.0017023653780617696, -0.0045108927752919425,
+   -0.005422155613096912, -0.0039489323837623835,
+   -0.005295995750506547, -0.00629706566356189,
+   -0.004685732198036754, -0.0048076735568143,
+   -0.005978864279217503, -0.005928999306332966,
+   -0.004187703549017582, -0.003213999896976475,
+   -0.0028068699816073414, -0.0010889703907593833,
+   0.0003276714243495386, -0.0013015007040186994,
+   -0.003208050402434782, -0.0025115319088208545,
+   -0.0013787553006401076, -0.0018279087370218635,
+   -0.0025904836507747754, -0.002071221947222004,
+   -0.0026424212922485594, -0.0039837031817577,
+   -0.0041635566057380965, -0.004355223489150822,
+   -0.004350395332709937, -0.0036693292471930263,
+   -0.003386384394185026, -0.003972568655001128,
+   -0.004332336840023821, -0.002648767912111827,
+   -0.001384410080218114, -0.0011353792711849466,
+   -0.0013726264946164232, -0.0020075119315034313
+};
+/* Rear,   same side (135 degree) */
+static const float ar_filt[128] = {
+   0.004573315040810066, 0.0013592578059426913,
+   0.01553271930902704, -0.0002356117224941317,
+   -0.05746098219774702, 0.03430688963370445,
+   0.00808371687447385, 0.21893535841158596,
+   0.2984357591724814, -0.3302799746504719,
+   -0.3194029149806245, 0.21633225051331056,
+   0.24371260938097083, -0.08843705549751085,
+   0.03939684701343366, 0.45386926431114494,
+   0.07599118140753386, -0.18114706160474578,
+   0.285640624686038, 0.4049515236666218,
+   -0.05347890222071792, -0.31464359045319074,
+   -0.1033502246468194, -0.04553593949283157,
+   -0.1880747731157464, -0.13629090230626037,
+   -0.10435789106123239, -0.19818232801888755,
+   -0.16701805476330397, -0.022793111199284,
+   0.058049696762683685, 0.007048321372693906,
+   -0.002966419183225961, 0.10140569697797763,
+   0.11648999956673124, 0.05218347182779882,
+   0.028427001212735392, 0.04151900310166159,
+   -0.0006960604221423734, -0.05898623212226975,
+   -0.03801934531173312, -0.029306970535287986,
+   -0.04549125782835908, -0.0599222718506552,
+   -0.058299618975430116, -0.03765579129720727,
+   -0.03559302657499581, -0.020647901025903054,
+   -0.005720957338744348, -0.0041915732688915545,
+   -0.0011470880098346143, 0.008737404798553,
+   0.023444168098121512, 0.024204226042172663,
+   0.01894897166475026, 0.020807655257479588,
+   0.021570431128040954, 0.006800556178576289,
+   -0.009000089216921362, -0.010969824547067934,
+   -0.0033653428332822374, -0.012676936164668659,
+   -0.026739938673413587, -0.023427869194287573,
+   -0.023302007105117244, -0.023647155590533712,
+   -0.021289317515613106, -0.009120487305069884,
+   0.0009251551667728967, 0.00004285344125653763,
+   -0.00009042365479456271, 0.00022573242339446494,
+   0.00720168491586098, 0.007111875505402431,
+   0.003186514817683482, 0.00810087718334745,
+   0.012619557025922575, 0.007854726400013397,
+   -0.0024013592881066267, -0.001452457473161119,
+   -0.0025535188366093945, -0.012428911627809337,
+   -0.013729251536694145, -0.0070099675146427344,
+   -0.007165284278706593, -0.01639289295622301,
+   -0.015831795079778305, -0.007305768485523729,
+   -0.003608863157004021, -0.0032640528878698084,
+   0.0030901263998481944, 0.00749497566124848,
+   0.002515185532327241, 0.00004840875738621367,
+   0.0017596043486043966, 0.0046229941553338144,
+   0.0034259167322926096, 0.003707347634186093,
+   0.0035584806528586328, -0.0019078936035275198,
+   -0.006057343815214898, -0.0069262470468817,
+   -0.004345020728618624, -0.004177623355574794,
+   -0.005373506556122508, -0.006624933928893836,
+   -0.008679541408588839, -0.010718719681595322,
+   -0.011392246979594496, -0.007893917064389902,
+   -0.0027572935365832536, -0.00006064707149834412,
+   -0.0012512537319656323, -0.0024501501002409786,
+   -0.0022106788572895998, -0.00234124933370301,
+   -0.0008953445167066823, 0.0005393670625637734,
+   -0.00033175600142209297, -0.004023994309351289,
+   -0.008655472335784443, -0.009899957354849682,
+   -0.008664952919996412, -0.00553483124503576,
+   -0.003735336089277662, -0.002754824348643885,
+   -0.0026884314856593368, -0.004084181815125924
+};
+/* Rear,   opposite (-135 degree) */
+static const float or_filt[128] = {
+   0.0001220944028243897, -0.000021785381808441314,
+   5.823057988603169e-6, -0.00001217768176447613,
+   -0.00006123604397345513, 5.574117262531134e-6,
+   -0.00004935331914366778, 1.1771577934768211e-6,
+   -0.000059236211621095756, 9.503536190497286e-6,
+   -0.0001494445696103564, 0.00012248858284145305,
+   -0.0000963975321456313, 6.017905197665205e-6,
+   0.00003353395360402643, -0.0001931511015359506,
+   0.0005113536523931485, -0.0005676652619386114,
+   0.0012057159755477467, 0.0009370492250339692,
+   0.004596472288877981, -0.0018831773384237068,
+   -0.008208535225621212, 0.0038178646400751056,
+   0.008726517739105965, 0.06664363898418262,
+   0.06788684221502142, -0.04492315162807267,
+   -0.04019906311255255, 0.026203059677375153,
+   0.013678129114847544, -0.014334962223993527,
+   0.010141709596167392, 0.11559131576945537,
+   0.1586402064538425, 0.059975334707967023,
+   0.004671725963777715, 0.031498678282775874,
+   0.014338626006524587, -0.014749719448472231,
+   -0.02055508237941379, -0.05407690143992048,
+   -0.07767559836886143, -0.05029091786216801,
+   -0.030808335706574427, -0.03401958135442541,
+   -0.030520368430288967, -0.014168302104259355,
+   0.011907621259989802, 0.014286081013069,
+   0.006377467879613449, 0.018546823568277478,
+   0.028297012771618273, 0.025222339408338186,
+   0.021931611353415138, 0.019708894333646355,
+   0.01729258494072014, 0.017468204169564034,
+   0.009729094845051928, -0.002976992018531901,
+   -0.00956986166277019, -0.016125733548332074,
+   -0.02934094241442545, -0.04133767871051455,
+   -0.043536981145416466, -0.0385966307108608,
+   -0.02784453599342459, -0.018995135307247116,
+   -0.012849534096536747, -0.004437491064613308,
+   0.00028385411598204655, 0.003776874988516643,
+   0.008069432041547833, 0.008764754183751848,
+   0.008020908861878062, 0.006830351461360802,
+   0.002218330884267235, -0.0020478725582339444,
+   -0.003997428121462543, -0.007066287373515421,
+   -0.00940847412544698, -0.010938998446237963,
+   -0.011775483016151306, -0.011391103919484287,
+   -0.010586061195163017, -0.009842793078929053,
+   -0.007753202010139829, -0.00569213732353025,
+   -0.006506783349722073, -0.005346134281903736,
+   -0.003913089814898934, -0.0036091443854759727,
+   -0.0020328564301266287, 0.00017932870773467683,
+   0.0032779786679056357, 0.003969695813293966,
+   0.0020339334412434987, -0.00011345940675415259,
+   -0.0018344103399567666, -0.003556764701666365,
+   -0.004263523639408391, -0.002940568582022133,
+   -0.0034341188272627556, -0.006023399920020824,
+   -0.0077456903203677865, -0.007912219312377842,
+   -0.00625202770436523, -0.00530785086116117,
+   -0.005569722659634311, -0.004664448462594344,
+   -0.0037747773914077747, -0.004175649656985592,
+   -0.004659601521384289, -0.005008602967819641,
+   -0.004730625079902729, -0.0034039554356604146,
+   -0.0017110333873406587, -0.0006091938771510242,
+   -0.0016051679050678297, -0.003312864664007262,
+   -0.004505512715977288, -0.004152222189861692,
+   -0.003218596419678823, -0.0027277806209877343,
+   -0.001715005444317267, -0.0012589960071233749,
+   -0.001852908777923165, -0.002540339553144362
+};
+/* Center rear (180 degree) */
+static const float cr_filt[128] = {
+   -0.00005989110716536726, -0.00022790291829128702,
+   0.0002659166098971966, -0.0003774772716776257,
+   0.0004540309551867803, -0.000420238187386368,
+   0.00025518536450885686, 0.00028285526288953955,
+   -0.001016391007574093, 0.0028634984299063795,
+   0.0021574799687976045, 0.01035121276682072,
+   -0.010481720917298163, -0.013197198495899292,
+   0.0031928225328717195, 0.02903137843618603,
+   0.1632429772511569, 0.1047487989875262,
+   -0.10464685060623742, -0.09260196288035998,
+   -0.007514241993554443, 0.013596249226741712,
+   -0.019876166508450258, 0.1255626123599804,
+   0.3648170359521724, 0.19458249753223478,
+   -0.04434070930031298, 0.046582528121935265,
+   0.09484427347230277, -0.03137795311969644,
+   -0.10297437925363695, -0.09351091015917065,
+   -0.1129521092162064, -0.14925322995658827,
+   -0.1231466295584665, -0.06356719756705227,
+   -0.05442277895126282, -0.07568433015661316,
+   -0.023314932828602003, 0.04260950721318558,
+   0.02249026315598923, 0.02048195669571197,
+   0.05651342117268278, 0.05885038917623213,
+   0.03797102097397795, 0.011767394419953451,
+   0.00560502503429481, 0.005051125343961189,
+   -0.012925933188033823, -0.023918884651306566,
+   -0.013251659441678816, -0.010694772488866284,
+   -0.03080486448617846, -0.03661278237783158,
+   -0.0379227303416262, -0.042189005718490775,
+   -0.026595666344596286, -0.009759025956801257,
+   -0.002064986663513004, -0.002420117028098389,
+   -0.006629991977552491, 0.004619970897631026,
+   0.019450642967537877, 0.0173521119057514,
+   0.017641425439988062, 0.02270029598048491,
+   0.018976431925275348, 0.009299852902290885,
+   -0.001695039371619912, -0.00675162574265618,
+   -0.009380968871003034, -0.011208396125485165,
+   -0.01308640049201482, -0.0165636375633249,
+   -0.022004099870933345, -0.025173458684139286,
+   -0.016918759559175375, -0.00865150653575917,
+   -0.006999929082792643, -0.005454830010518988,
+   -0.0021129521131095317, 0.00018717090054046307,
+   -0.0002864344788569993, 0.0017615225381095569,
+   0.006985907557802283, 0.010577308310476465,
+   0.006466104789306027, -0.0014988738575948326,
+   -0.0039669755229277195, -0.0065156971200080235,
+   -0.009343206924192169, -0.0076430644693577495,
+   -0.004395214976600924, -0.003052735340422483,
+   -0.007019103043066595, -0.00974109267696527,
+   -0.007968015032797376, -0.007801513845528344,
+   -0.007535748903681969, -0.003543341967287925,
+   0.0015083125553729722, 0.0023345972556147025,
+   -0.0010043623069557037, -0.0025295765105203746,
+   -0.0023701840891643634, -0.0005908186035024362,
+   0.0029826252289082847, 0.004829048542117764,
+   0.004488360022902081, 0.00002643748103005408,
+   -0.0042100779212597295, -0.006170600558114495,
+   -0.007267149164680168, -0.006825522903494639,
+   -0.006899834372739123, -0.0073493916110062675,
+   -0.009554351265163382, -0.011790297433830197,
+   -0.010645796603734424, -0.0064661575394022106,
+   -0.002026743466524137, -0.0004337034584909932,
+   -0.0011172647031654614, -0.0017947816283674731,
+   -0.00255615052036616, -0.0017721562881944813,
+   -0.0002379619297227554, 0.0007130120121089036
+};
+
+#endif /* MPLAYER_AF_HRTF_H */

audio/filter/af_karaoke.c

@@ -0,0 +1,86 @@
+/*
+ * simple voice removal filter
+ *
+ * copyright (c) 2006 Reynaldo H. Verdejo Pinochet
+ * Based on code by Alex Beregszaszi for his 'center' filter.
+ *
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "af.h"
+
+// Data for specific instances of this filter
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+        switch(cmd){
+                case AF_CONTROL_REINIT:
+                mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+                mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
+                return af_test_output(af,(struct mp_audio*)arg);
+        }
+        return AF_UNKNOWN;
+}
+
+static int filter_frame(struct af_instance *af, struct mp_audio *c)
+{
+        if (!c)
+                return 0;
+        if (af_make_writeable(af, c) < 0) {
+                talloc_free(c);
+                return 0;
+        }
+
+        float*          a       = c->planes[0];  // Audio data
+        int                     nch     = c->nch;        // Number of channels
+        int                     len     = c->samples*nch;        // Number of samples in current audio block
+        register int  i;
+
+        /*
+                FIXME1 add a low band pass filter to avoid suppressing
+                centered bass/drums
+                FIXME2 better calculated* attenuation factor
+        */
+
+        for(i=0;i<len;i+=nch)
+        {
+                a[i] = (a[i] - a[i+1]) * 0.7;
+                a[i+1]=a[i];
+        }
+
+        af_add_output_frame(af, c);
+        return 0;
+}
+
+// Allocate memory and set function pointers
+static int af_open(struct af_instance* af){
+        af->control     = control;
+        af->filter_frame = filter_frame;
+        return AF_OK;
+}
+
+// Description of this filter
+const struct af_info af_info_karaoke = {
+        .info = "Simple karaoke/voice-removal audio filter",
+        .name = "karaoke",
+        .flags = AF_FLAGS_NOT_REENTRANT,
+        .open = af_open,
+};

audio/filter/af_ladspa.c

@@ -0,0 +1,851 @@
+/*
+ * LADSPA plugin loader
+ *
+ * Written by Ivo van Poorten <ivop@euronet.nl>
+ * Copyright (C) 2004, 2005
+ *
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+/* ------------------------------------------------------------------------- */
+
+/* Global Includes */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <inttypes.h>
+#include <math.h>
+#include <limits.h>
+
+#include <dlfcn.h>
+#include <ladspa.h>
+
+/* ------------------------------------------------------------------------- */
+
+/* Local Includes */
+
+#include "af.h"
+
+#define _(x) (x)
+
+/* ------------------------------------------------------------------------- */
+
+/* Filter specific data */
+
+typedef struct af_ladspa_s
+{
+    int status;     /**< Status of the filter.
+                     *   Either AF_OK or AF_ERROR
+                     *   Because MPlayer re-inits audio filters that
+                     *   _clearly_ returned AF_ERROR anyway, I use this
+                     *   in play() to skip the processing and return
+                     *   the data unchanged.
+                     */
+
+    int activated;  /**< 0 or 1. Activate LADSPA filters only once, even
+                     *   if the buffers get resized, to avoid a stuttering
+                     *   filter.
+                     */
+
+    char *file;
+    char *label;
+    char *controls;
+
+    char *myname;   /**< It's easy to have a concatenation of file and label */
+
+    void *libhandle;
+    const LADSPA_Descriptor *plugin_descriptor;
+
+    int nports;
+
+    int ninputs;
+    int *inputs;
+
+    int noutputs;
+    int *outputs;
+
+    int ninputcontrols;
+    int *inputcontrolsmap;  /**< Map input port number [0-] to actual port */
+    float *inputcontrols;
+
+    int noutputcontrols;
+    int *outputcontrolsmap;
+    float *outputcontrols;
+
+    int nch;                /**< number of channels */
+    int bufsize;
+    float **inbufs;
+    float **outbufs;
+    LADSPA_Handle *chhandles;
+
+} af_ladspa_t;
+
+/* ------------------------------------------------------------------------- */
+
+static int af_open(struct af_instance *af);
+static int af_ladspa_malloc_failed(char*);
+
+/* ------------------------------------------------------------------------- */
+
+#define OPT_BASE_STRUCT af_ladspa_t
+const struct af_info af_info_ladspa = {
+    .info = "LADSPA plugin loader",
+    .name = "ladspa",
+    .open = af_open,
+    .priv_size = sizeof(af_ladspa_t),
+    .options = (const struct m_option[]) {
+        OPT_STRING("file", file, 0),
+        OPT_STRING("label", label, 0),
+        OPT_STRING("controls", controls, 0),
+        {0}
+    },
+};
+
+/* ------------------------------------------------------------------------- */
+
+/* By lack of a better word (in my vocabulary) this is called 'parse'.
+ * Feel free to suggest an alternative.
+ */
+
+/** \brief Check for inputs, outputs and controls of a given filter.
+ *
+ * This function counts and checks all input, output and control ports
+ * of the filter that was loaded. If it turns out to be a valid
+ * filter for MPlayer use, it prints out a list of all controls and
+ * the corresponding range of its value at message level  MSGL_V.
+ *
+ * \param setup     Current setup of the filter. Must have its
+ *                  plugin_descriptor set!
+ *
+ * \return  Returns AF_OK if it has a valid input/output/controls
+ *          configuration. Else, it returns AF_ERROR.
+ */
+
+static int af_ladspa_parse_plugin(struct af_instance *af) {
+    af_ladspa_t *setup = af->priv;
+    int p, i;
+    const LADSPA_Descriptor *pdes = setup->plugin_descriptor;
+    LADSPA_PortDescriptor d;
+    LADSPA_PortRangeHint hint;
+
+    if (!setup->libhandle)
+        return AF_ERROR; /* only call parse after a succesful load */
+    if (!setup->plugin_descriptor)
+        return AF_ERROR; /* same as above */
+
+    /* let's do it */
+
+    setup->nports = pdes->PortCount;
+
+    /* allocate memory for all inputs/outputs/controls */
+
+    setup->inputs = calloc(setup->nports, sizeof(int));
+    if (!setup->inputs) return af_ladspa_malloc_failed(setup->myname);
+
+    setup->outputs = calloc(setup->nports, sizeof(int));
+    if (!setup->outputs) return af_ladspa_malloc_failed(setup->myname);
+
+    setup->inputcontrolsmap = calloc(setup->nports, sizeof(int));
+    if (!setup->inputcontrolsmap) return af_ladspa_malloc_failed(setup->myname);
+
+    setup->inputcontrols = calloc(setup->nports, sizeof(float));
+    if (!setup->inputcontrols) return af_ladspa_malloc_failed(setup->myname);
+
+    setup->outputcontrolsmap = calloc(setup->nports, sizeof(int));
+    if (!setup->outputcontrolsmap) return af_ladspa_malloc_failed(setup->myname);
+
+    setup->outputcontrols = calloc(setup->nports, sizeof(float));
+    if (!setup->outputcontrols) return af_ladspa_malloc_failed(setup->myname);
+
+    /* set counts to zero */
+
+    setup->ninputs = 0;
+    setup->noutputs = 0;
+    setup->ninputcontrols = 0;
+    setup->noutputcontrols = 0;
+
+    /* check all ports, see what type it is and set variables according to
+     * what we have found
+     */
+
+    for (p=0; p<setup->nports; p++) {
+        d = pdes->PortDescriptors[p];
+
+        if (LADSPA_IS_PORT_AUDIO(d)) {
+            if (LADSPA_IS_PORT_INPUT(d)) {
+                setup->inputs[setup->ninputs] = p;
+                setup->ninputs++;
+            } else if (LADSPA_IS_PORT_OUTPUT(d)) {
+                setup->outputs[setup->noutputs] = p;
+                setup->noutputs++;
+            }
+        }
+
+        if (LADSPA_IS_PORT_CONTROL(d)) {
+            if (LADSPA_IS_PORT_INPUT(d)) {
+                setup->inputcontrolsmap[setup->ninputcontrols] = p;
+                setup->ninputcontrols++;
+                /* set control to zero. set values after reading the rest
+                 * of the suboptions and check LADSPA_?_HINT's later.
+                 */
+                setup->inputcontrols[p] = 0.0f;
+            } else if (LADSPA_IS_PORT_OUTPUT(d)) {
+                /* read and handle these too, otherwise filters that have them
+                 * will sig11
+                 */
+                setup->outputcontrolsmap[setup->noutputcontrols]=p;
+                setup->noutputcontrols++;
+                setup->outputcontrols[p] = 0.0f;
+            }
+        }
+
+    }
+
+    if (setup->ninputs == 0) {
+        MP_WARN(af, "%s: %s\n", setup->myname,
+                                                _("WARNING! This LADSPA plugin has no audio inputs.\n  The incoming audio signal will be lost."));
+    } else if (setup->ninputs == 1) {
+        MP_VERBOSE(af, "%s: this is a mono effect\n", setup->myname);
+    } else if (setup->ninputs == 2) {
+        MP_VERBOSE(af, "%s: this is a stereo effect\n", setup->myname);
+    } else {
+        MP_VERBOSE(af, "%s: this is a %i-channel effect, "
+               "support is experimental\n", setup->myname, setup->ninputs);
+    }
+
+    if (setup->noutputs == 0) {
+        MP_ERR(af, "%s: %s\n", setup->myname,
+                                                _("This LADSPA plugin has no audio outputs."));
+        return AF_ERROR;
+    }
+
+    if (setup->noutputs != setup->ninputs ) {
+        MP_ERR(af, "%s: %s\n", setup->myname,
+                                                _("The number of audio inputs and audio outputs of the LADSPA plugin differ."));
+        return AF_ERROR;
+    }
+
+    MP_VERBOSE(af, "%s: this plugin has %d input control(s)\n",
+                                        setup->myname, setup->ninputcontrols);
+
+    /* Print list of controls and its range of values it accepts */
+
+    for (i=0; i<setup->ninputcontrols; i++) {
+        p = setup->inputcontrolsmap[i];
+        hint = pdes->PortRangeHints[p];
+        MP_VERBOSE(af, "  --- %d %s [", i, pdes->PortNames[p]);
+
+        if (LADSPA_IS_HINT_BOUNDED_BELOW(hint.HintDescriptor)) {
+            MP_VERBOSE(af, "%0.2f , ", hint.LowerBound);
+        } else {
+            MP_VERBOSE(af, "... , ");
+        }
+
+        if (LADSPA_IS_HINT_BOUNDED_ABOVE(hint.HintDescriptor)) {
+            MP_VERBOSE(af, "%0.2f]\n", hint.UpperBound);
+        } else {
+            MP_VERBOSE(af, "...]\n");
+        }
+
+    }
+
+    return AF_OK;
+}
+
+/* ------------------------------------------------------------------------- */
+
+/* This function might "slightly" look like dlopenLADSPA in the LADSPA SDK :-)
+ * But, I changed a few things, because imho it was broken. It did not support
+ * relative paths, only absolute paths that start with a /
+ * I think ../../some/dir/foobar.so is just as valid. And if one wants to call
+ * his library '...somename...so' he's crazy, but it should be allowed.
+ * So, search the path first, try plain *filename later.
+ * Also, try adding .so first! I like the recursion the SDK did, but it's
+ * better the other way around. -af ladspa=cmt:amp_stereo:0.5 is easier to type
+ * than -af ladspa=cmt.so:amp_stereo:0.5 :-))
+ */
+
+/** \brief dlopen() wrapper
+ *
+ * This is a wrapper around dlopen(). It tries various variations of the
+ * filename (with or without the addition of the .so extension) in various
+ * directories specified by the LADSPA_PATH environment variable. If all fails
+ * it tries the filename directly as an absolute path to the library.
+ *
+ * \param filename  filename of the library to load.
+ * \param flag      see dlopen(3) for a description of the flags.
+ *
+ * \return          returns a pointer to the loaded library on success, or
+ *                  NULL if it fails to load.
+ */
+
+static void* mydlopen(const char *filename, int flag) {
+    char *buf;
+    const char *end, *start, *ladspapath;
+    int endsinso;
+    size_t filenamelen;
+    void *result = NULL;
+
+#if defined(__MINGW32__) || defined(__CYGWIN__)
+    /* For Windows there's only absolute path support.
+     * If you have a Windows machine, feel free to fix this.
+     * (path separator, shared objects extension, et cetera). */
+        MP_VERBOSE(af, "\ton windows, only absolute pathnames "
+                "are supported\n");
+        MP_VERBOSE(af, "\ttrying %s\n", filename);
+        return dlopen(filename, flag);
+#endif
+
+    filenamelen = strlen(filename);
+
+    endsinso = 0;
+    if (filenamelen > 3)
+        endsinso = (strcmp(filename+filenamelen-3, ".so") == 0);
+    if (!endsinso) {
+        buf=malloc(filenamelen+4);
+        strcpy(buf, filename);
+        strcat(buf, ".so");
+        result=mydlopen(buf, flag);
+        free(buf);
+    }
+
+    if (result)
+        return result;
+
+    ladspapath=getenv("LADSPA_PATH");
+
+    if (ladspapath) {
+        start=ladspapath;
+        while (*start != '\0') {
+            int needslash;
+            end=start;
+            while ( (*end != ':') && (*end != '\0') )
+                end++;
+
+            buf=malloc(filenamelen + 2 + (end-start) );
+            if (end > start)
+                strncpy(buf, start, end-start);
+            needslash=0;
+            if (end > start)
+                if (*(end-1) != '/') {
+                    needslash = 1;
+                    buf[end-start] = '/';
+                }
+            strcpy(buf+needslash+(end-start), filename);
+
+            result=dlopen(buf, flag);
+
+            free(buf);
+            if (result)
+                return result;
+
+            start = end;
+            if (*start == ':')
+                start++;
+        } /* end while there's still more in the path */
+    } /* end if there's a ladspapath */
+
+    /* last resort, just open it again, so the dlerror() message is correct */
+    return dlopen(filename,flag);
+}
+
+/* ------------------------------------------------------------------------- */
+
+/** \brief Load a LADSPA Plugin
+ *
+ * This function loads the LADSPA plugin specified by the file and label
+ * that are present in the setup variable. First, it loads the library.
+ * If it fails, it returns AF_ERROR. If not, it continues to look for the
+ * specified label. If it finds it, it sets the plugin_descriptor inside
+ * setup and returns AF_OK. If it doesn't, it returns AF_ERROR. Special case
+ * is a label called 'help'. In that case, it prints a list of all available
+ * labels (filters) in the library specified by file.
+ *
+ * \param setup     Current setup of the filter. Contains filename and label.
+ *
+ * \return  Either AF_ERROR or AF_OK, depending on the success of the operation.
+ */
+
+static int af_ladspa_load_plugin(struct af_instance *af) {
+    af_ladspa_t *setup = af->priv;
+    const LADSPA_Descriptor *ladspa_descriptor;
+    LADSPA_Descriptor_Function descriptor_function;
+    int i;
+
+    /* load library */
+    MP_VERBOSE(af, "%s: loading ladspa plugin library %s\n",
+                                                setup->myname, setup->file);
+
+    setup->libhandle = mydlopen(setup->file, RTLD_NOW);
+
+    if (!setup->libhandle) {
+        MP_ERR(af, "%s: %s %s\n\t%s\n", setup->myname,
+                    _("failed to load"), setup->file, dlerror() );
+        return AF_ERROR;
+    }
+
+    MP_VERBOSE(af, "%s: library found.\n", setup->myname);
+
+    /* find descriptor function */
+    dlerror();
+    descriptor_function = (LADSPA_Descriptor_Function) dlsym (setup->libhandle,
+                                                        "ladspa_descriptor");
+
+    if (!descriptor_function) {
+        MP_ERR(af, "%s: %s\n\t%s\n", setup->myname,
+                                _("Couldn't find ladspa_descriptor() function in the specified library file."), dlerror());
+        return AF_ERROR;
+    }
+
+    /* if label == help, list all labels in library and exit */
+
+    if (strcmp(setup->label, "help") == 0) {
+        MP_INFO(af, "%s: %s %s:\n", setup->myname,
+                _("available labels in"), setup->file);
+        for (i=0; ; i++) {
+            ladspa_descriptor = descriptor_function(i);
+            if (ladspa_descriptor == NULL) {
+                return AF_ERROR;
+            }
+            MP_INFO(af, "  %-16s - %s (%lu)\n",
+                    ladspa_descriptor->Label,
+                    ladspa_descriptor->Name,
+                    ladspa_descriptor->UniqueID);
+        }
+    }
+
+    MP_VERBOSE(af, "%s: looking for label\n", setup->myname);
+
+    /* find label in library */
+    for (i=0; ; i++) {
+        ladspa_descriptor = descriptor_function(i);
+        if (ladspa_descriptor == NULL) {
+            MP_ERR(af, "%s: %s\n", setup->myname,
+                                            _("Couldn't find label in plugin library."));
+            return AF_ERROR;
+        }
+        if (strcmp(ladspa_descriptor->Label, setup->label) == 0) {
+            setup->plugin_descriptor = ladspa_descriptor;
+            MP_VERBOSE(af, "%s: %s found\n", setup->myname,
+                                                                setup->label);
+            return AF_OK;
+        }
+    }
+
+    return AF_OK;
+}
+
+/* ------------------------------------------------------------------------- */
+
+/** \brief Print a malloc() failed error message.
+ *
+ * Generic function which can be called if a call to malloc(), calloc(),
+ * strdup(), et cetera, failed. It prints a message to the console and
+ * returns AF_ERROR.
+ *
+ * \return  AF_ERROR
+ */
+
+static int af_ladspa_malloc_failed(char *myname) {
+    return AF_ERROR;
+}
+
+/* ------------------------------------------------------------------------- */
+
+/** \brief Controls the filter.
+ *
+ * Control the behaviour of the filter.
+ *
+ * Commands:
+ * CONTROL_REINIT   Sets the af structure with proper values for number
+ *                  of channels, rate, format, et cetera.
+ * \param af    Audio filter instance
+ * \param cmd   The command to execute
+ * \param arg   Arguments to the command
+ *
+ * \return      Either AF_ERROR or AF_OK, depending on the succes of the
+ *              operation.
+ */
+
+static int control(struct af_instance *af, int cmd, void *arg) {
+    af_ladspa_t *setup = (af_ladspa_t*) af->priv;
+
+    switch(cmd) {
+    case AF_CONTROL_REINIT:
+        MP_VERBOSE(af, "%s: (re)init\n", setup->myname);
+
+        if (!arg) return AF_ERROR;
+
+        /* accept FLOAT, let af_format do conversion */
+
+        mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+        mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
+
+        return af_test_output(af, (struct mp_audio*)arg);
+    }
+
+    return AF_UNKNOWN;
+}
+
+/* ------------------------------------------------------------------------- */
+
+/** \brief Uninitialise the LADSPA Plugin Loader filter.
+ *
+ * This function deactivates the plugin(s), cleans up, frees all allocated
+ * memory and exits.
+ *
+ * \return  No return value.
+ */
+
+static void uninit(struct af_instance *af) {
+    if (af->priv) {
+        af_ladspa_t *setup = (af_ladspa_t*) af->priv;
+        const LADSPA_Descriptor *pdes = setup->plugin_descriptor;
+
+        if (setup->myname) {
+            MP_VERBOSE(af, "%s: cleaning up\n", setup->myname);
+            free(setup->myname);
+        }
+
+        if (setup->chhandles) {
+            for (int i = 0; i < setup->nch; i+=setup->ninputs) {
+                if (pdes->deactivate) pdes->deactivate(setup->chhandles[i]);
+                if (pdes->cleanup) pdes->cleanup(setup->chhandles[i]);
+            }
+            free(setup->chhandles);
+        }
+
+        free(setup->inputcontrolsmap);
+        free(setup->inputcontrols);
+        free(setup->outputcontrolsmap);
+        free(setup->outputcontrols);
+        free(setup->inputs);
+        free(setup->outputs);
+
+        if (setup->inbufs) {
+            for(int i = 0; i < setup->nch; i++)
+                free(setup->inbufs[i]);
+            free(setup->inbufs);
+        }
+
+        if (setup->outbufs) {
+            for (int i = 0; i < setup->nch; i++)
+                free(setup->outbufs[i]);
+            free(setup->outbufs);
+        }
+
+        if (setup->libhandle)
+            dlclose(setup->libhandle);
+    }
+}
+
+/* ------------------------------------------------------------------------- */
+
+/** \brief Process chunk of audio data through the selected LADSPA Plugin.
+ *
+ * \param af    Pointer to audio filter instance
+ * \param data  Pointer to chunk of audio data
+ *
+ * \return      Either AF_ERROR or AF_OK
+ */
+
+static int filter_frame(struct af_instance *af, struct mp_audio *data)
+{
+    if (!data)
+        return 0;
+    af_ladspa_t *setup = af->priv;
+    const LADSPA_Descriptor *pdes = setup->plugin_descriptor;
+    float *audio = (float*)data->planes[0];
+    int nsamples = data->samples*data->nch;
+    int nch = data->nch;
+    int rate = data->rate;
+    int i, p;
+
+    if (setup->status !=AF_OK) {
+        talloc_free(data);
+        return -1;
+    }
+    if (af_make_writeable(af, data) < 0) {
+        talloc_free(data);
+        return -1;
+    }
+
+    /* See if it's the first call. If so, setup inbufs/outbufs, instantiate
+     * plugin, connect ports and activate plugin
+     */
+
+    if ( (setup->bufsize != nsamples/nch) || (setup->nch != nch) ) {
+
+        /* if setup->nch==0, it's the first call, if not, something has
+         * changed and all previous mallocs have to be freed
+         */
+
+        if (setup->nch != 0) {
+            MP_TRACE(af, "%s: bufsize change; free old buffer\n",
+                                                                setup->myname);
+
+            if(setup->inbufs) {
+                for(i=0; i<setup->nch; i++)
+                    free(setup->inbufs[i]);
+                free(setup->inbufs);
+            }
+            if(setup->outbufs) {
+                for(i=0; i<setup->nch; i++)
+                    free(setup->outbufs[i]);
+                free(setup->outbufs);
+            }
+        } /* everything is freed */
+
+        setup->bufsize = nsamples/nch;
+        setup->nch = nch;
+
+        setup->inbufs = calloc(nch, sizeof(float*));
+        setup->outbufs = calloc(nch, sizeof(float*));
+
+        MP_TRACE(af, "%s: bufsize = %d\n",
+                                        setup->myname, setup->bufsize);
+
+        for(i=0; i<nch; i++) {
+            setup->inbufs[i] = calloc(setup->bufsize, sizeof(float));
+            setup->outbufs[i] = calloc(setup->bufsize, sizeof(float));
+        }
+
+        /* only on the first call, there are no handles. */
+
+        if (!setup->chhandles) {
+            setup->chhandles = calloc(nch, sizeof(LADSPA_Handle));
+
+            /* create handles
+             * for stereo effects, create one handle for two channels
+             */
+
+            for(i=0; i<nch; i++) {
+
+                if (i % setup->ninputs) { /* stereo effect */
+                    /* copy the handle from previous channel */
+                    setup->chhandles[i] = setup->chhandles[i-1];
+                    continue;
+                }
+
+                setup->chhandles[i] = pdes->instantiate(pdes, rate);
+            }
+        }
+
+        /* connect input/output ports for each channel/filter instance
+         *
+         * always (re)connect ports
+         */
+
+        for(i=0; i<nch; i++) {
+            pdes->connect_port(setup->chhandles[i],
+                               setup->inputs[i % setup->ninputs],
+                               setup->inbufs[i]);
+            pdes->connect_port(setup->chhandles[i],
+                               setup->outputs[i % setup->ninputs],
+                               setup->outbufs[i]);
+
+            /* connect (input) controls */
+
+            for (p=0; p<setup->nports; p++) {
+                LADSPA_PortDescriptor d = pdes->PortDescriptors[p];
+                if (LADSPA_IS_PORT_CONTROL(d)) {
+                    if (LADSPA_IS_PORT_INPUT(d)) {
+                        pdes->connect_port(setup->chhandles[i], p,
+                                                &(setup->inputcontrols[p]) );
+                    } else {
+                        pdes->connect_port(setup->chhandles[i], p,
+                                                &(setup->outputcontrols[p]) );
+                    }
+                }
+            }
+
+            /* Activate filter (if it isn't already :) ) */
+
+            if (pdes->activate && !setup->activated && i % setup->ninputs == 0)
+                pdes->activate(setup->chhandles[i]);
+
+        } /* All channels/filters done! except for... */
+        setup->activated = 1;
+
+        /* Stereo effect with one channel left. Use same buffer for left
+         * and right. connect it to the second port.
+         */
+
+        for (p = i; p % setup->ninputs; p++) {
+            pdes->connect_port(setup->chhandles[i-1],
+                               setup->inputs[p % setup->ninputs],
+                               setup->inbufs[i-1]);
+            pdes->connect_port(setup->chhandles[i-1],
+                               setup->outputs[p % setup->ninputs],
+                               setup->outbufs[i-1]);
+        } /* done! */
+
+    } /* setup for first call/change of bufsize is done.
+       * normal playing routine follows...
+       */
+
+    /* Right now, I use a separate input and output buffer.
+     * I could change this to in-place processing (inbuf==outbuf), but some
+     * ladspa filters are broken and are not able to handle that. This seems
+     * fast enough, so unless somebody complains, it stays this way :)
+     */
+
+    /* Fill inbufs */
+
+    for (p=0; p<setup->bufsize; p++) {
+        for (i=0; i<nch; i++) {
+            setup->inbufs[i][p] = audio[p*nch + i];
+        }
+    }
+
+    /* Run filter(s) */
+
+    for (i=0; i<nch; i+=setup->ninputs) {
+        pdes->run(setup->chhandles[i], setup->bufsize);
+    }
+
+    /* Extract outbufs */
+
+    for (p=0; p<setup->bufsize; p++) {
+        for (i=0; i<nch; i++) {
+            audio[p*nch + i] = setup->outbufs[i][p];
+        }
+    }
+
+    /* done */
+
+    af_add_output_frame(af, data);
+    return 0;
+}
+
+/* ------------------------------------------------------------------------- */
+
+/** \brief Open LADSPA Plugin Loader Filter
+ *
+ * \param af    Audio Filter instance
+ *
+ * \return      Either AF_ERROR or AF_OK
+ */
+
+static int af_open(struct af_instance *af) {
+
+    af->control=control;
+    af->uninit=uninit;
+    af->filter_frame = filter_frame;
+
+    af_ladspa_t *setup = af->priv;
+
+    setup->status = AF_ERROR; /* will be set to AF_OK if
+                                                   * all went OK and filter()
+                                                   * should proceed.
+                                                   */
+
+    setup->myname = strdup(af_info_ladspa.name);
+    if (!setup->myname)
+        return af_ladspa_malloc_failed((char*)af_info_ladspa.name);
+
+    if (!setup->file || !setup->file[0]) {
+        MP_ERR(af, "%s: %s\n", setup->myname,
+                                            _("No library file specified."));
+        uninit(af);
+        return AF_ERROR;
+    }
+    if (!setup->label || !setup->label[0]) {
+        MP_ERR(af, "%s: %s\n", setup->myname,
+                                            _("No filter label specified."));
+        uninit(af);
+        return AF_ERROR;
+    }
+
+    free(setup->myname);
+    setup->myname = calloc(strlen(af_info_ladspa.name)+strlen(setup->file)+
+                                                strlen(setup->label)+6, 1);
+    snprintf(setup->myname, strlen(af_info_ladspa.name)+
+            strlen(setup->file)+strlen(setup->label)+6, "%s: (%s:%s)",
+                        af_info_ladspa.name, setup->file, setup->label);
+
+    /* load plugin :) */
+
+    if ( af_ladspa_load_plugin(af) != AF_OK )
+        return AF_ERROR;
+
+    /* see what inputs, outputs and controls this plugin has */
+    if ( af_ladspa_parse_plugin(af) != AF_OK )
+        return AF_ERROR;
+
+    /* ninputcontrols is set by now, read control values from arg */
+
+    float val;
+    char *line = setup->controls;
+    for (int i = 0; i < setup->ninputcontrols; i++) {
+        if (!line || (i != 0 && *line != ',')) {
+            MP_ERR(af, "%s: %s\n", setup->myname,
+                                    _("Not enough controls specified on the command line."));
+            return AF_ERROR;
+        }
+        if (i != 0)
+            line++;
+        if (sscanf(line, "%f", &val) != 1) {
+            MP_ERR(af, "%s: %s\n", setup->myname,
+                                    _("Not enough controls specified on the command line."));
+            return AF_ERROR;
+        }
+        setup->inputcontrols[setup->inputcontrolsmap[i]] = val;
+        line = strchr(line, ',');
+    }
+
+    MP_VERBOSE(af, "%s: input controls: ", setup->myname);
+    for (int i = 0; i < setup->ninputcontrols; i++) {
+        MP_VERBOSE(af, "%0.4f ",
+                        setup->inputcontrols[setup->inputcontrolsmap[i]]);
+    }
+    MP_VERBOSE(af, "\n");
+
+    /* check boundaries of inputcontrols */
+
+    MP_VERBOSE(af, "%s: checking boundaries of input controls\n",
+                                                            setup->myname);
+    for (int i = 0; i < setup->ninputcontrols; i++) {
+        int p = setup->inputcontrolsmap[i];
+        LADSPA_PortRangeHint hint =
+                            setup->plugin_descriptor->PortRangeHints[p];
+        val = setup->inputcontrols[p];
+
+        if (LADSPA_IS_HINT_BOUNDED_BELOW(hint.HintDescriptor) &&
+                val < hint.LowerBound) {
+            MP_ERR(af, "%s: Input control #%d is below lower boundary of %0.4f.\n",
+                                        setup->myname, i, hint.LowerBound);
+            return AF_ERROR;
+        }
+        if (LADSPA_IS_HINT_BOUNDED_ABOVE(hint.HintDescriptor) &&
+                val > hint.UpperBound) {
+            MP_ERR(af, "%s: Input control #%d is above upper boundary of %0.4f.\n",
+                                        setup->myname, i, hint.UpperBound);
+            return AF_ERROR;
+        }
+    }
+    MP_VERBOSE(af, "%s: all controls have sane values\n",
+                                                            setup->myname);
+
+    /* All is well! */
+    setup->status = AF_OK;
+
+    return AF_OK;
+}
+
+/* ------------------------------------------------------------------------- */

audio/filter/af_lavcac3enc.c

@@ -26,18 +26,15 @@
 #include <assert.h>
 
 #include <libavcodec/avcodec.h>
+#include <libavutil/audioconvert.h>
 #include <libavutil/intreadwrite.h>
 #include <libavutil/common.h>
 #include <libavutil/bswap.h>
 #include <libavutil/mem.h>
 
-#include "config.h"
-
-#include "common/av_common.h"
 #include "common/common.h"
 #include "af.h"
 #include "audio/audio_buffer.h"
-#include "audio/chmap_sel.h"
 #include "audio/fmt-conversion.h"
 
 
@@ -53,57 +50,19 @@ const uint16_t ac3_bitrate_tab[19] = {
 typedef struct af_ac3enc_s {
     struct AVCodec        *lavc_acodec;
     struct AVCodecContext *lavc_actx;
+    AVPacket pkt;
     int bit_rate;
     struct mp_audio *input;     // frame passed to libavcodec
     struct mp_audio *pending;   // unconsumed input data
     int in_samples;     // samples of input per AC3 frame
     int out_samples;    // upper bound on encoded output per AC3 frame
-    int64_t encoder_buffered;
+    int in_sampleformat;
 
     int cfg_add_iec61937_header;
     int cfg_bit_rate;
     int cfg_min_channel_num;
-    char *cfg_encoder;
-    char **cfg_avopts;
 } af_ac3enc_t;
 
-// fmt carries the input format. Change it to the best next-possible format
-// the encoder likely accepts.
-static void select_encode_format(AVCodecContext *c, struct mp_audio *fmt)
-{
-    int formats[AF_FORMAT_COUNT];
-    af_get_best_sample_formats(fmt->format, formats);
-
-    for (int n = 0; formats[n]; n++) {
-        const enum AVSampleFormat *lf = c->codec->sample_fmts;
-        for (int i = 0; lf && lf[i] != AV_SAMPLE_FMT_NONE; i++) {
-            int mpfmt = af_from_avformat(lf[i]);
-            if (mpfmt && mpfmt == formats[n]) {
-                mp_audio_set_format(fmt, mpfmt);
-                goto done_fmt;
-            }
-        }
-    }
-done_fmt: ;
-
-    int rate =
-        af_select_best_samplerate(fmt->rate, c->codec->supported_samplerates);
-    if (rate > 0)
-        fmt->rate = rate;
-
-    struct mp_chmap_sel sel = {0};
-    const uint64_t *lch = c->codec->channel_layouts;
-    for (int n = 0; lch && lch[n]; n++) {
-        struct mp_chmap chmap = {0};
-        mp_chmap_from_lavc(&chmap, lch[n]);
-        mp_chmap_sel_add_map(&sel, &chmap);
-    }
-    struct mp_chmap res = fmt->channels;
-    mp_chmap_sel_adjust(&sel, &res);
-    if (!mp_chmap_is_empty(&res))
-        mp_audio_set_channels(fmt, &res);
-}
-
 // Initialization and runtime control
 static int control(struct af_instance *af, int cmd, void *arg)
 {
@@ -116,20 +75,26 @@ 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;
 
-        // At least currently, the AC3 encoder doesn't export sample rates.
-        in->rate = 48000;
-        select_encode_format(s->lavc_actx, in);
+        mp_audio_set_format(in, s->in_sampleformat);
 
+        if (in->rate != 48000 && in->rate != 44100 && in->rate != 32000)
+            in->rate = 48000;
         af->data->rate = in->rate;
+
+        mp_chmap_reorder_to_lavc(&in->channels);
+        if (in->nch > AC3_MAX_CHANNELS)
+            mp_audio_set_num_channels(in, AC3_MAX_CHANNELS);
+
         mp_audio_set_format(af->data, AF_FORMAT_S_AC3);
         mp_audio_set_num_channels(af->data, 2);
 
         if (!mp_audio_config_equals(in, &orig_in))
             return AF_FALSE;
 
+        s->in_samples = AC3_FRAME_SIZE;
         if (s->cfg_add_iec61937_header) {
             s->out_samples = AC3_FRAME_SIZE;
         } else {
@@ -137,11 +102,14 @@ static int control(struct af_instance *af, int cmd, void *arg)
         }
 
         mp_audio_copy_config(s->input, in);
+        mp_audio_realloc(s->input, s->in_samples);
+        s->input->samples = 0;
 
         talloc_free(s->pending);
         s->pending = NULL;
 
-        MP_DBG(af, "reinit: %d, %d, %d.\n", in->nch, in->rate, s->in_samples);
+        MP_DBG(af, "af_lavcac3enc reinit: %d, %d, %d.\n",
+               in->nch, in->rate, s->in_samples);
 
         int bit_rate = s->bit_rate ? s->bit_rate : default_bit_rate[in->nch];
 
@@ -152,7 +120,6 @@ static int control(struct af_instance *af, int cmd, void *arg)
             avcodec_close(s->lavc_actx);
 
             // Put sample parameters
-            s->lavc_actx->sample_fmt = af_to_avformat(in->format);
             s->lavc_actx->channels = in->nch;
             s->lavc_actx->channel_layout = mp_chmap_to_lavc(&in->channels);
             s->lavc_actx->sample_rate = in->rate;
@@ -162,26 +129,14 @@ static int control(struct af_instance *af, int cmd, void *arg)
                 MP_ERR(af, "Couldn't open codec %s, br=%d.\n", "ac3", bit_rate);
                 return AF_ERROR;
             }
-
-            if (s->lavc_actx->frame_size < 1) {
-                MP_ERR(af, "encoder didn't specify input frame size\n");
-                return AF_ERROR;
-            }
         }
-        s->in_samples = s->lavc_actx->frame_size;
-        mp_audio_realloc(s->input, s->in_samples);
-        s->input->samples = 0;
-        s->encoder_buffered = 0;
+        if (s->lavc_actx->frame_size != AC3_FRAME_SIZE) {
+            MP_ERR(af, "lavcac3enc: unexpected ac3 "
+                   "encoder frame size %d\n", s->lavc_actx->frame_size);
+            return AF_ERROR;
+        }
         return AF_OK;
     }
-    case AF_CONTROL_RESET:
-        if (avcodec_is_open(s->lavc_actx))
-            avcodec_flush_buffers(s->lavc_actx);
-        talloc_free(s->pending);
-        s->pending = NULL;
-        s->input->samples = 0;
-        s->encoder_buffered = 0;
-        return AF_OK;
     }
     return AF_UNKNOWN;
 }
@@ -192,6 +147,7 @@ static void uninit(struct af_instance* af)
     af_ac3enc_t *s = af->priv;
 
     if (s) {
+        av_free_packet(&s->pkt);
         if(s->lavc_actx) {
             avcodec_close(s->lavc_actx);
             av_free(s->lavc_actx);
@@ -200,13 +156,6 @@ 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 +
-                  s->encoder_buffered) / (double)s->input->rate;
-}
-
 static int filter_frame(struct af_instance *af, struct mp_audio *audio)
 {
     af_ac3enc_t *s = af->priv;
@@ -217,7 +166,6 @@ static int filter_frame(struct af_instance *af, struct mp_audio *audio)
 
     talloc_free(s->pending);
     s->pending = audio;
-    update_delay(af);
     return 0;
 }
 
@@ -229,120 +177,62 @@ 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) {
-        if (!mp_audio_is_writeable(s->input))
-            assert(s->input->samples == 0); // we can't have sent a partial frame
-        mp_audio_realloc_min(s->input, s->in_samples);
         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;
 }
 
-// Return <0 on error, 0 on need more input, 1 on success (and *frame set).
-// To actually advance the read pointer, set s->input->samples=0 afterwards.
-static int read_input_frame(struct af_instance *af, AVFrame *frame)
-{
-    af_ac3enc_t *s = af->priv;
-    if (!fill_buffer(af))
-        return 0; // need more input
-
-    if (mp_audio_to_avframe(s->input, frame) < 0)
-        return -1;
-
-    return 1;
-}
-
 static int filter_out(struct af_instance *af)
 {
     af_ac3enc_t *s = af->priv;
-
-    if (!s->pending)
-        return 0;
+    if (!fill_buffer(s))
+        return 0; // need more input
 
     AVFrame *frame = av_frame_alloc();
     if (!frame) {
         MP_FATAL(af, "Could not allocate memory \n");
         return -1;
     }
-    int err = -1;
+    frame->nb_samples = s->in_samples;
+    frame->format = s->lavc_actx->sample_fmt;
+    frame->channel_layout = s->lavc_actx->channel_layout;
+    assert(s->input->num_planes <= AV_NUM_DATA_POINTERS);
+    frame->extended_data = frame->data;
+    for (int n = 0; n < s->input->num_planes; n++)
+        frame->data[n] = s->input->planes[n];
+    frame->linesize[0] = s->input->samples * s->input->sstride;
 
-    AVPacket pkt = {0};
-    av_init_packet(&pkt);
-
-#if HAVE_AVCODEC_NEW_CODEC_API
-    // Send input as long as it wants.
-    while (1) {
-        err = read_input_frame(af, frame);
-        if (err < 0)
-            goto done;
-        if (err == 0)
-            break;
-        err = -1;
-        int lavc_ret = avcodec_send_frame(s->lavc_actx, frame);
-        // On EAGAIN, we're supposed to read remaining output.
-        if (lavc_ret == AVERROR(EAGAIN))
-            break;
-        if (lavc_ret < 0) {
-            MP_FATAL(af, "Encode failed.\n");
-            goto done;
-        }
-        s->encoder_buffered += s->input->samples;
-        s->input->samples = 0;
-    }
-    int lavc_ret = avcodec_receive_packet(s->lavc_actx, &pkt);
-    if (lavc_ret == AVERROR(EAGAIN)) {
-        // Need to buffer more input.
-        err = 0;
-        goto done;
-    }
-    if (lavc_ret < 0) {
-        MP_FATAL(af, "Encode failed.\n");
-        goto done;
-    }
-#else
-    err = read_input_frame(af, frame);
-    if (err < 0)
-        goto done;
-    if (err == 0)
-        goto done;
-    err = -1;
     int ok;
-    int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &pkt, frame, &ok);
+    int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
+    av_frame_free(&frame);
     s->input->samples = 0;
     if (lavc_ret < 0 || !ok) {
         MP_FATAL(af, "Encode failed.\n");
-        goto done;
+        return -1;
     }
-#endif
 
     MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
-           pkt.size, s->pending->samples + s->input->samples);
-
-    s->encoder_buffered -= AC3_FRAME_SIZE;
+            s->pkt.size, s->pending->samples);
 
     struct mp_audio *out =
         mp_audio_pool_get(af->out_pool, af->data, s->out_samples);
     if (!out)
-        goto done;
+        return -1;
     mp_audio_copy_attributes(out, s->pending);
 
-    int frame_size = pkt.size;
+    int frame_size = s->pkt.size;
     int header_len = 0;
     char hdr[8];
 
-    if (s->cfg_add_iec61937_header && pkt.size > 5) {
-        int bsmod = pkt.data[5] & 0x7;
+    if (s->cfg_add_iec61937_header && s->pkt.size > 5) {
+        int bsmod = s->pkt.data[5] & 0x7;
         int len = frame_size;
 
         frame_size = AC3_FRAME_SIZE * 2 * 2;
@@ -360,19 +250,13 @@ static int filter_out(struct af_instance *af)
 
     char *buf = (char *)out->planes[0];
     memcpy(buf, hdr, header_len);
-    memcpy(buf + header_len, pkt.data, pkt.size);
-    memset(buf + header_len + pkt.size, 0,
-           frame_size - (header_len + pkt.size));
-    swap_16((uint16_t *)(buf + header_len), pkt.size / 2);
+    memcpy(buf + header_len, s->pkt.data, s->pkt.size);
+    memset(buf + header_len + s->pkt.size, 0,
+            frame_size - (header_len + s->pkt.size));
+    swap_16((uint16_t *)(buf + header_len), s->pkt.size / 2);
     out->samples = frame_size / out->sstride;
     af_add_output_frame(af, out);
-
-    err = 0;
-done:
-    av_packet_unref(&pkt);
-    av_frame_free(&frame);
-    update_delay(af);
-    return err;
+    return 0;
 }
 
 static int af_open(struct af_instance* af){
@@ -383,9 +267,9 @@ static int af_open(struct af_instance* af){
     af->filter_frame = filter_frame;
     af->filter_out = filter_out;
 
-    s->lavc_acodec = avcodec_find_encoder_by_name(s->cfg_encoder);
+    s->lavc_acodec = avcodec_find_encoder_by_name("ac3");
     if (!s->lavc_acodec) {
-        MP_ERR(af, "Couldn't find encoder %s.\n", s->cfg_encoder);
+        MP_ERR(af, "Audio LAVC, couldn't find encoder for codec %s.\n", "ac3");
         return AF_ERROR;
     }
 
@@ -394,34 +278,37 @@ static int af_open(struct af_instance* af){
         MP_ERR(af, "Audio LAVC, couldn't allocate context!\n");
         return AF_ERROR;
     }
-
-    if (mp_set_avopts(af->log, s->lavc_actx, s->cfg_avopts) < 0)
-        return AF_ERROR;
-
-    // For this one, we require the decoder to expert lists of all supported
-    // parameters. (Not all decoders do that, but the ones we're interested
-    // in do.)
-    if (!s->lavc_acodec->sample_fmts ||
-        !s->lavc_acodec->channel_layouts)
-    {
-        MP_ERR(af, "Audio encoder doesn't list supported parameters.\n");
+    const enum AVSampleFormat *fmts = s->lavc_acodec->sample_fmts;
+    for (int i = 0; fmts[i] != AV_SAMPLE_FMT_NONE; i++) {
+        s->in_sampleformat = af_from_avformat(fmts[i]);
+        if (s->in_sampleformat) {
+            s->lavc_actx->sample_fmt = fmts[i];
+            break;
+        }
+    }
+    if (!s->in_sampleformat) {
+        MP_ERR(af, "Audio LAVC, encoder doesn't "
+               "support expected sample formats!\n");
         return AF_ERROR;
     }
+    MP_VERBOSE(af, "[af_lavcac3enc]: in sample format: %s\n",
+           af_fmt_to_str(s->in_sampleformat));
+
+    av_init_packet(&s->pkt);
 
     s->input = talloc_zero(s, struct mp_audio);
 
     if (s->cfg_bit_rate) {
         int i;
         for (i = 0; i < 19; i++) {
-            if (ac3_bitrate_tab[i] == s->cfg_bit_rate) {
-                s->bit_rate = ac3_bitrate_tab[i] * 1000;
+            if (ac3_bitrate_tab[i] == s->cfg_bit_rate)
                 break;
-            }
         }
         if (i >= 19) {
-            MP_WARN(af, "unable set unsupported bitrate %d, use default "
-                    "bitrate (check manpage to see supported bitrates).\n",
-                    s->cfg_bit_rate);
+            MP_WARN(af, "af_lavcac3enc unable set unsupported "
+                    "bitrate %d, use default bitrate (check manpage to see "
+                    "supported bitrates).\n", s->cfg_bit_rate);
+            s->cfg_bit_rate = 0;
         }
     }
 
@@ -439,15 +326,12 @@ const struct af_info af_info_lavcac3enc = {
         .cfg_add_iec61937_header = 1,
         .cfg_bit_rate = 640,
         .cfg_min_channel_num = 3,
-        .cfg_encoder = "ac3",
     },
     .options = (const struct m_option[]) {
         OPT_FLAG("tospdif", cfg_add_iec61937_header, 0),
         OPT_CHOICE_OR_INT("bitrate", cfg_bit_rate, 0, 32, 640,
                           ({"auto", 0}, {"default", 0})),
         OPT_INTRANGE("minch", cfg_min_channel_num, 0, 2, 6),
-        OPT_STRING("encoder", cfg_encoder, 0),
-        OPT_KEYVALUELIST("o", cfg_avopts, 0),
         {0}
     },
 };

audio/filter/af_lavfi.c

@@ -3,18 +3,18 @@
  *
  * Filter graph creation code taken from FFmpeg ffplay.c (LGPL 2.1 or later)
  *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
+ * 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 Lesser General Public License for more details.
+ * GNU General Public License for more details.
  *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ * 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>
@@ -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"
 
@@ -50,7 +47,6 @@
 #if LIBAVFILTER_VERSION_MICRO < 100
 #define graph_parse(graph, filters, inputs, outputs, log_ctx) \
     avfilter_graph_parse(graph, filters, inputs, outputs, log_ctx)
-#define avfilter_graph_send_command(a, b, c, d, e, f, g) -1
 #else
 #define graph_parse(graph, filters, inputs, outputs, log_ctx) \
     avfilter_graph_parse_ptr(graph, filters, &(inputs), &(outputs), log_ctx)
@@ -67,8 +63,6 @@ struct priv {
 
     bool eof;
 
-    struct mp_tags *metadata;
-
     // options
     char *cfg_graph;
     char **cfg_avopts;
@@ -196,10 +190,6 @@ static int control(struct af_instance *af, int cmd, void *arg)
         if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
             mp_audio_set_format(in, AF_FORMAT_FLOAT);
 
-        // Removing this requires fixing AVFrame.data vs. AVFrame.extended_data
-        if (in->channels.num > AV_NUM_DATA_POINTERS)
-            return AF_ERROR;
-
         if (!mp_chmap_is_lavc(&in->channels))
             mp_chmap_reorder_to_lavc(&in->channels); // will always work
 
@@ -216,27 +206,13 @@ static int control(struct af_instance *af, int cmd, void *arg)
         mp_chmap_from_lavc(&out_cm, l_out->channel_layout);
         mp_audio_set_channels(out, &out_cm);
 
-        if (!mp_audio_config_valid(out) || out->channels.num > AV_NUM_DATA_POINTERS)
+        if (!mp_audio_config_valid(out))
             return AF_ERROR;
 
         p->timebase_out = l_out->time_base;
 
         return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
     }
-    case AF_CONTROL_COMMAND: {
-        if (!p->graph)
-            break;
-        char **args = arg;
-        return avfilter_graph_send_command(p->graph, "all",
-                                           args[0], args[1], &(char){0}, 0, 0)
-                >= 0 ? CONTROL_OK : CONTROL_ERROR;
-    }
-    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;
@@ -244,17 +220,6 @@ static int control(struct af_instance *af, int cmd, void *arg)
     return AF_UNKNOWN;
 }
 
-static void get_metadata_from_av_frame(struct af_instance *af, AVFrame *frame)
-{
-#if HAVE_AVFRAME_METADATA
-    struct priv *p = af->priv;
-    if (!p->metadata)
-        p->metadata = talloc_zero(p, struct mp_tags);
-
-    mp_tags_copy_from_av_dictionary(p->metadata, av_frame_get_metadata(frame));
-#endif
-}
-
 static int filter_frame(struct af_instance *af, struct mp_audio *data)
 {
     struct priv *p = af->priv;
@@ -266,15 +231,32 @@ static int filter_frame(struct af_instance *af, struct mp_audio *data)
     if (!p->graph)
         goto error;
 
+    AVFilterLink *l_in = p->in->outputs[0];
+
     if (data) {
-        frame = mp_audio_to_avframe_and_unref(data);
-        data = NULL;
+        frame = av_frame_alloc();
         if (!frame)
             goto error;
 
+        frame->nb_samples = data->samples;
+        frame->format = l_in->format;
+
         // Timebase is 1/sample_rate
         frame->pts = p->samples_in;
-        p->samples_in += frame->nb_samples;
+
+        frame->channel_layout = l_in->channel_layout;
+        frame->sample_rate = l_in->sample_rate;
+#if LIBAVFILTER_VERSION_MICRO >= 100
+        // FFmpeg being a stupid POS
+        frame->channels = l_in->channels;
+#endif
+
+        frame->extended_data = frame->data;
+        for (int n = 0; n < data->num_planes; n++)
+            frame->data[n] = data->planes[n];
+        frame->linesize[0] = frame->nb_samples * data->sstride;
+
+        p->samples_in += data->samples;
     }
 
     if (av_buffersrc_add_frame(p->in, frame) < 0)
@@ -325,7 +307,6 @@ 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;
@@ -345,6 +326,8 @@ static int af_open(struct af_instance *af)
     af->uninit = uninit;
     af->filter_frame = filter_frame;
     af->filter_out = filter_out;
+    // Removing this requires fixing AVFrame.data vs. AVFrame.extended_data
+    assert(MP_NUM_CHANNELS <= AV_NUM_DATA_POINTERS);
     return AF_OK;
 }
 

audio/filter/af_lavrresample.c

@@ -28,9 +28,9 @@
 #include <assert.h>
 
 #include <libavutil/opt.h>
+#include <libavutil/audioconvert.h>
 #include <libavutil/common.h>
 #include <libavutil/samplefmt.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/mathematics.h>
 
 #include "common/common.h"
@@ -49,7 +49,6 @@
 #define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \
     swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples)
 #define avresample_set_channel_mapping swr_set_channel_mapping
-#define avresample_set_compensation swr_set_compensation
 #else
 #error "config.h broken or no resampler found"
 #endif
@@ -59,30 +58,12 @@
 #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
@@ -93,45 +74,51 @@ struct af_resample {
     struct mp_chmap out_channels;
 };
 
+struct af_resample {
+    int allow_detach;
+    char **avopts;
+    double playback_speed;
+    struct mp_audio *pending;
+    bool avrctx_ok;
+    struct AVAudioResampleContext *avrctx;
+    struct mp_audio avrctx_fmt; // output format of 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 int get_out_samples(struct af_resample *s, int in_samples)
+static void drop_all_output(struct af_resample *s)
 {
-    return avresample_get_out_samples(s->avrctx, in_samples);
+    while (avresample_read(s->avrctx, NULL, 1000) > 0) {}
+}
+static int get_drain_samples(struct af_resample *s)
+{
+    return avresample_get_out_samples(s->avrctx, 0);
 }
 #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 int get_out_samples(struct af_resample *s, int in_samples)
+static void drop_all_output(struct af_resample *s)
 {
-#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
+    while (swr_drop_output(s->avrctx, 1000) > 0) {}
+}
+static int get_drain_samples(struct af_resample *s)
+{
+    return 4096; // libswscale does not have this
 }
 #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)
 {
@@ -154,46 +141,27 @@ static int rate_from_speed(int rate, double speed)
     return lrint(rate * speed);
 }
 
-// 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)
+static bool needs_lavrctx_reconfigure(struct af_resample *s,
+                                      struct mp_audio *in,
+                                      struct mp_audio *out)
 {
-    if (mp_format == AF_FORMAT_S24)
-        return AV_SAMPLE_FMT_S32;
-    return af_to_avformat(mp_format);
+    return s->ctx.in_rate_af  != in->rate ||
+           s->ctx.in_format   != in->format ||
+           !mp_chmap_equals(&s->ctx.in_channels, &in->channels) ||
+           s->ctx.out_rate    != out->rate ||
+           s->ctx.out_format  != out->format ||
+           !mp_chmap_equals(&s->ctx.out_channels, &out->channels) ||
+           s->ctx.filter_size != s->opts.filter_size ||
+           s->ctx.phase_shift != s->opts.phase_shift ||
+           s->ctx.linear      != s->opts.linear ||
+           s->ctx.cutoff      != s->opts.cutoff;
+
 }
 
-static struct mp_chmap fudge_pairs[][2] = {
-    {MP_CHMAP2(BL,  BR),  MP_CHMAP2(SL,  SR)},
-    {MP_CHMAP2(SL,  SR),  MP_CHMAP2(BL,  BR)},
-    {MP_CHMAP2(SDL, SDR), MP_CHMAP2(SL,  SR)},
-    {MP_CHMAP2(SL,  SR),  MP_CHMAP2(SDL, SDR)},
-};
-
-// Modify out_layout and return the new value. The intention is reducing the
-// loss libswresample's rematrixing will cause by exchanging similar, but
-// strictly speaking incompatible channel pairs. For example, 7.1 should be
-// changed to 7.1(wide) without dropping the SL/SR channels. (We still leave
-// it to libswresample to create the remix matrix.)
-static uint64_t fudge_layout_conversion(struct af_instance *af,
-                                        uint64_t in, uint64_t out)
+static bool test_conversion(int src_format, int dst_format)
 {
-    for (int n = 0; n < MP_ARRAY_SIZE(fudge_pairs); n++) {
-        uint64_t a = mp_chmap_to_lavc(&fudge_pairs[n][0]);
-        uint64_t b = mp_chmap_to_lavc(&fudge_pairs[n][1]);
-        if ((in & a) == a && (in & b) == 0 &&
-            (out & a) == 0 && (out & b) == b)
-        {
-            out = (out & ~b) | a;
-
-            MP_VERBOSE(af, "Fudge: %s -> %s\n",
-                       mp_chmap_to_str(&fudge_pairs[n][0]),
-                       mp_chmap_to_str(&fudge_pairs[n][1]));
-        }
-    }
-    return out;
+    return af_to_avformat(src_format) != AV_SAMPLE_FMT_NONE &&
+           af_to_avformat(dst_format) != AV_SAMPLE_FMT_NONE;
 }
 
 // mp_chmap_get_reorder() performs:
@@ -213,51 +181,52 @@ static void transpose_order(int *map, int num)
 }
 
 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;
+    s->avrctx_ok = false;
 
     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);
+    enum AVSampleFormat out_samplefmtp =
+        af_to_avformat(af_fmt_to_planar(out->format));
 
     if (in_samplefmt == AV_SAMPLE_FMT_NONE ||
         out_samplefmt == AV_SAMPLE_FMT_NONE ||
         out_samplefmtp == AV_SAMPLE_FMT_NONE)
-        goto error;
+        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);
+    talloc_free(s->pending);
+    s->pending = NULL;
 
-    av_opt_set_double(s->avrctx, "cutoff",          s->opts.cutoff, 0);
+    s->ctx.out_rate    = out->rate;
+    s->ctx.in_rate_af  = in->rate;
+    s->ctx.in_rate     = rate_from_speed(in->rate, s->playback_speed);
+    s->ctx.out_format  = out->format;
+    s->ctx.in_format   = in->format;
+    s->ctx.out_channels= out->channels;
+    s->ctx.in_channels = in->channels;
+    s->ctx.filter_size = s->opts.filter_size;
+    s->ctx.phase_shift = s->opts.phase_shift;
+    s->ctx.linear      = s->opts.linear;
+    s->ctx.cutoff      = s->opts.cutoff;
+
+    av_opt_set_int(s->avrctx, "filter_size",        s->ctx.filter_size, 0);
+    av_opt_set_int(s->avrctx, "phase_shift",        s->ctx.phase_shift, 0);
+    av_opt_set_int(s->avrctx, "linear_interp",      s->ctx.linear, 0);
+
+    av_opt_set_double(s->avrctx, "cutoff",          s->ctx.cutoff, 0);
 
-    int normalize = s->opts.normalize;
-    if (normalize < 0)
-        normalize = af->opts->audio_normalize;
 #if HAVE_LIBSWRESAMPLE
-    av_opt_set_double(s->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0);
-#else
-    av_opt_set_int(s->avrctx, "normalize_mix_level", !!normalize, 0);
+    av_opt_set_double(s->avrctx, "rematrix_maxval", 1.0, 0);
 #endif
 
     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;
@@ -272,24 +241,19 @@ 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;
+    struct mp_chmap in_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;
+        return AF_ERROR;
     }
 
     mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
     transpose_order(s->reorder_in, map_in.num);
 
+    struct mp_chmap out_lavc;
+    mp_chmap_from_lavc(&out_lavc, out_ch_layout);
     if (mp_chmap_equals(&out_lavc, &map_out)) {
         // No intermediate step required - output new format directly.
         out_samplefmtp = out_samplefmt;
@@ -298,63 +262,48 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
         struct mp_chmap withna = out_lavc;
         mp_chmap_fill_na(&withna, map_out.num);
         if (withna.num != map_out.num)
-            goto error;
+            return AF_ERROR;
+        mp_chmap_get_reorder(s->reorder_out, &out_lavc, &map_out);
     }
-    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);
-
-    out_ch_layout = fudge_layout_conversion(af, in_ch_layout, out_ch_layout);
-
     // 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);
 
-    // Just needs the correct number of channels for deplanarization.
-    struct mp_chmap fake_chmap;
-    mp_chmap_set_unknown(&fake_chmap, map_out.num);
-    uint64_t fake_out_ch_layout = mp_chmap_to_lavc_unchecked(&fake_chmap);
+    // 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;
+        return AF_ERROR;
+
+    // Deplanarize if needed.
     av_opt_set_int(s->avrctx_out, "in_channel_layout",  fake_out_ch_layout, 0);
     av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
-
     av_opt_set_int(s->avrctx_out, "in_sample_fmt",      out_samplefmtp, 0);
     av_opt_set_int(s->avrctx_out, "out_sample_fmt",     out_samplefmt, 0);
-    av_opt_set_int(s->avrctx_out, "in_sample_rate",     s->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);
 
-    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;
     }
+    s->avrctx_ok = true;
     return AF_OK;
-
-error:
-    close_lavrr(af);
-    return AF_ERROR;
 }
 
 
@@ -382,133 +331,129 @@ 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 && af_to_avformat(format) == AV_SAMPLE_FMT_NONE)
+            return AF_FALSE;
+
+        mp_audio_set_format(af->data, format);
+        return AF_OK;
+    }
+    case AF_CONTROL_SET_CHANNELS: {
+        mp_audio_set_channels(af->data, (struct mp_chmap *)arg);
+        return AF_OK;
+    }
+    case AF_CONTROL_SET_RESAMPLE_RATE:
+        af->data->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->ctx.in_rate_af, s->playback_speed);
+        if (new_rate != s->ctx.in_rate && s->avrctx_ok && af->fmt_out.format) {
+            // Before reconfiguring, drain the audio that is still buffered
+            // in the resampler.
+            struct mp_audio *pending = talloc_zero(NULL, struct mp_audio);
+            mp_audio_copy_config(pending, &af->fmt_out);
+            pending->samples = get_drain_samples(s);
+            if (pending->samples > 0) {
+                mp_audio_realloc_min(pending, pending->samples);
+                int r = resample_frame(s->avrctx, pending, NULL);
+                pending->samples = MPMAX(r, 0);
+            }
+            // Reinitialize resampler.
+            configure_lavrr(af, &af->fmt_in, &af->fmt_out);
+            s->pending = pending;
+        }
         return AF_OK;
     }
     case AF_CONTROL_RESET:
-        if (s->avrctx) {
-#if HAVE_LIBSWRESAMPLE
-            swr_close(s->avrctx);
-            if (swr_init(s->avrctx) < 0) {
-                close_lavrr(af);
-                return AF_ERROR;
-            }
-#else
-            while (avresample_read(s->avrctx, NULL, 1000) > 0) {}
-#endif
-        }
+        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);
+    talloc_free(s->pending);
 }
 
-// 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)
-{
-    if (mpa->format == AF_FORMAT_S32 && af->data->format == AF_FORMAT_S24) {
-        size_t len = mp_audio_psize(mpa) / mpa->bps;
-        for (int s = 0; s < len; s++) {
-            uint32_t val = *((uint32_t *)mpa->planes[0] + s);
-            uint8_t *ptr = (uint8_t *)mpa->planes[0] + s * 3;
-            ptr[0] = val >> SHIFT24(0);
-            ptr[1] = val >> SHIFT24(1);
-            ptr[2] = val >> SHIFT24(2);
-        }
-        mp_audio_set_format(mpa, AF_FORMAT_S24);
-    }
-
-    for (int p = 0; p < mpa->num_planes; p++) {
-        void *ptr = mpa->planes[p];
-        int total = mpa->samples * mpa->spf;
-        if (af_fmt_from_planar(mpa->format) == AF_FORMAT_FLOAT) {
-            for (int s = 0; s < total; s++)
-                ((float *)ptr)[s] = av_clipf(((float *)ptr)[s], -1.0f, 1.0f);
-        } else if (af_fmt_from_planar(mpa->format) == AF_FORMAT_DOUBLE) {
-            for (int s = 0; s < total; s++)
-                ((double *)ptr)[s] = MPCLAMP(((double *)ptr)[s], -1.0, 1.0);
-        }
-    }
-}
-
-// This relies on the tricky way mpa was allocated.
 static void reorder_planes(struct mp_audio *mpa, int *reorder,
                            struct mp_chmap *newmap)
 {
     struct mp_audio prev = *mpa;
-    mp_audio_set_channels(mpa, newmap);
 
-    // The trailing planes were never written by avrctx, they're the NA channels.
-    int next_na = prev.num_planes;
-
-    for (int n = 0; n < mpa->num_planes; n++) {
+    for (int n = 0; n < newmap->num; 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);
+        if (src < 0) {
+            src = 0;    // Use the first plane for padding channels.
+            mpa->readonly = true;
         }
+        assert(src < mpa->num_planes);
+        mpa->planes[n] = prev.planes[src];
     }
+
+    mp_audio_set_channels(mpa, newmap);
 }
 
-static int filter_resample(struct af_instance *af, struct mp_audio *in)
+static int filter(struct af_instance *af, struct mp_audio *in)
 {
     struct af_resample *s = af->priv;
-    struct mp_audio *out = NULL;
 
-    if (!s->avrctx)
-        goto error;
+    if (s->pending) {
+        if (s->pending->samples) {
+            af_add_output_frame(af, s->pending);
+        } else {
+            talloc_free(s->pending);
+        }
+        s->pending = NULL;
+    }
 
-    int samples = get_out_samples(s, in ? in->samples : 0);
+    int samples = avresample_available(s->avrctx) +
+        av_rescale_rnd(get_delay(s) + (in ? in->samples : 0),
+                       s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP);
 
-    struct mp_audio out_format = s->pool_fmt;
-    out = mp_audio_pool_get(af->out_pool, &out_format, samples);
+    struct mp_audio out_format = s->avrctx_fmt;
+    struct mp_audio *out = mp_audio_pool_get(af->out_pool, &out_format, samples);
     if (!out)
         goto error;
     if (in)
         mp_audio_copy_attributes(out, in);
 
+    af->delay = get_delay(s) / (double)s->ctx.in_rate;
+
     if (out->samples) {
         out->samples = resample_frame(s->avrctx, out, in);
         if (out->samples < 0)
             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 (out->samples && !mp_audio_config_equals(out, af->data)) {
+        assert(AF_FORMAT_IS_PLANAR(out->format));
+        reorder_planes(out, s->reorder_out, &af->data->channels);
+        if (!mp_audio_config_equals(out, af->data)) {
+            struct mp_audio *new = mp_audio_pool_get(s->reorder_buffer, af->data,
                                                      out->samples);
             if (!new)
                 goto error;
@@ -521,17 +466,12 @@ static int filter_resample(struct af_instance *af, struct mp_audio *in)
         }
     }
 
-    extra_output_conversion(af, out);
-
     talloc_free(in);
     if (out->samples) {
         af_add_output_frame(af, out);
     } else {
         talloc_free(out);
     }
-
-    af->delay = get_delay(s);
-
     return 0;
 error:
     talloc_free(in);
@@ -539,39 +479,6 @@ error:
     return -1;
 }
 
-static int filter(struct af_instance *af, struct mp_audio *in)
-{
-    struct af_resample *s = af->priv;
-
-    int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
-    bool need_reinit = fabs(new_rate / (double)s->in_rate - 1) > 0.01;
-
-    if (s->avrctx) {
-        AVRational r = av_d2q(s->playback_speed * s->in_rate_af / s->in_rate,
-                              INT_MAX / 2);
-        // Essentially, swr/avresample_set_compensation() does 2 things:
-        // - adjust output sample rate by sample_delta/compensation_distance
-        // - reset the adjustment after compensation_distance output samples
-        // Increase the compensation_distance to avoid undesired reset
-        // semantics - we want to keep the ratio for the whole frame we're
-        // feeding it, until the next filter() call.
-        int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1);
-        r = (AVRational){ r.num * mult, r.den * mult };
-        if (avresample_set_compensation(s->avrctx, r.den - r.num, r.den) < 0)
-            need_reinit = true;
-    }
-
-    if (need_reinit && new_rate != s->in_rate) {
-        // Before reconfiguring, drain the audio that is still buffered
-        // in the resampler.
-        filter_resample(af, NULL);
-        // Reinitialize resampler.
-        configure_lavrr(af, &af->fmt_in, &af->fmt_out, false);
-    }
-
-    return filter_resample(af, in);
-}
-
 static int af_open(struct af_instance *af)
 {
     struct af_resample *s = af->priv;
@@ -583,9 +490,17 @@ 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
@@ -594,13 +509,13 @@ const struct af_info af_info_lavrresample = {
     .info = "Sample frequency conversion using libavresample",
     .name = "lavrresample",
     .open = af_open,
+    .test_conversion = test_conversion,
     .priv_size = sizeof(struct af_resample),
     .priv_defaults = &(const struct af_resample) {
         .opts = {
             .filter_size = 16,
             .cutoff      = 0.0,
             .phase_shift = 10,
-            .normalize   = -1,
         },
         .playback_speed = 1.0,
         .allow_detach = 1,
@@ -611,8 +526,6 @@ 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_CHOICE("normalize", opts.normalize, 0,
-                   ({"no", 0}, {"yes", 1}, {"auto", -1})),
         OPT_KEYVALUELIST("o", avopts, 0),
         {0}
     },

audio/filter/af_pan.c

@@ -28,11 +28,12 @@
 #include "af.h"
 
 // Data for specific instances of this filter
-typedef struct af_pan_s {
-    int nch; // Number of output channels; zero means same as input
-    float level[AF_NCH][AF_NCH];  // Gain level for each channel
-    char *matrixstr;
-} af_pan_t;
+typedef struct af_pan_s
+{
+  int nch; // Number of output channels; zero means same as input
+  float level[AF_NCH][AF_NCH];  // Gain level for each channel
+  char *matrixstr;
+}af_pan_t;
 
 static void set_channels(struct mp_audio *mpa, int num)
 {
@@ -43,17 +44,129 @@ static void set_channels(struct mp_audio *mpa, int num)
     mp_audio_set_channels(mpa, &map);
 }
 
-static void parse_matrix(struct af_instance *af, const char *cp)
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
 {
+  af_pan_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_format(af->data, AF_FORMAT_FLOAT);
+    set_channels(af->data, s->nch ? s->nch: ((struct mp_audio*)arg)->nch);
+
+    if((af->data->format != ((struct mp_audio*)arg)->format) ||
+       (af->data->bps != ((struct mp_audio*)arg)->bps)){
+      mp_audio_set_format((struct mp_audio*)arg, af->data->format);
+      return AF_FALSE;
+    }
+    return AF_OK;
+  case AF_CONTROL_SET_PAN_LEVEL:{
+    int    i;
+    int    ch = ((af_control_ext_t*)arg)->ch;
+    float* level = ((af_control_ext_t*)arg)->arg;
+    if (ch >= AF_NCH)
+      return AF_FALSE;
+    for(i=0;i<AF_NCH;i++)
+      s->level[ch][i] = level[i];
+    return AF_OK;
+  }
+  case AF_CONTROL_SET_PAN_NOUT:
+    // Reinit must be called after this function has been called
+
+    // Sanity check
+    if(((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH){
+      MP_ERR(af, "The number of output channels must be"
+            " between 1 and %i. Current value is %i\n",AF_NCH,((int*)arg)[0]);
+      return AF_ERROR;
+    }
+    s->nch=((int*)arg)[0];
+    return AF_OK;
+  case AF_CONTROL_SET_PAN_BALANCE:{
+    float val = *(float*)arg;
+    if (s->nch)
+      return AF_ERROR;
+    if (af->data->nch >= 2) {
+      s->level[0][0] = MPMIN(1.f, 1.f - val);
+      s->level[0][1] = MPMAX(0.f, val);
+      s->level[1][0] = MPMAX(0.f, -val);
+      s->level[1][1] = MPMIN(1.f, 1.f + val);
+    }
+    return AF_OK;
+  }
+  case AF_CONTROL_GET_PAN_BALANCE:
+    if (s->nch)
+      return AF_ERROR;
+    *(float*)arg = s->level[0][1] - s->level[1][0];
+    return AF_OK;
+  }
+  return AF_UNKNOWN;
+}
+
+static int filter_frame(struct af_instance *af, struct mp_audio *c)
+{
+  if (!c)
+    return 0;
+  struct mp_audio *l = mp_audio_pool_get(af->out_pool, &af->fmt_out, c->samples);
+  if (!l) {
+    talloc_free(c);
+    return -1;
+  }
+  mp_audio_copy_attributes(l, c);
+
+  af_pan_t*     s    = af->priv;        // Setup for this instance
+  float*        in   = c->planes[0];    // Input audio data
+  float*        out  = NULL;            // Output audio data
+  float*        end  = in+c->samples*c->nch;    // End of loop
+  int           nchi = c->nch;          // Number of input channels
+  int           ncho = l->nch;          // Number of output channels
+  register int  j,k;
+
+  out = l->planes[0];
+  // Execute panning
+  // FIXME: Too slow
+  while(in < end){
+    for(j=0;j<ncho;j++){
+      register float  x   = 0.0;
+      register float* tin = in;
+      for(k=0;k<nchi;k++)
+        x += tin[k] * s->level[j][k];
+      out[j] = x;
+    }
+    out+= ncho;
+    in+= nchi;
+  }
+
+  talloc_free(c);
+  af_add_output_frame(af, l);
+  return 0;
+}
+
+// Allocate memory and set function pointers
+static int af_open(struct af_instance* af){
+    af->control=control;
+    af->filter_frame = filter_frame;
     af_pan_t *s = af->priv;
-    int j = 0, k = 0, n;
-    while (*cp && k < AF_NCH) {
+    int   n = 0;
+    int   j,k;
+
+    int nch = s->nch;
+    if(nch && AF_OK != control(af,AF_CONTROL_SET_PAN_NOUT, &nch))
+        return AF_ERROR;
+
+    // Read pan values
+    char *cp = s->matrixstr;
+    j = 0; k = 0;
+    while(cp && k < AF_NCH){
         sscanf(cp, "%f%n" , &s->level[j][k], &n);
         MP_VERBOSE(af, "Pan level from channel %i to"
-                   " channel %i = %f\n", k, j, s->level[j][k]);
-        cp = &cp[n];
+                " channel %i = %f\n",k,j,s->level[j][k]);
+        cp =&cp[n];
         j++;
-        if (j >= s->nch) {
+        if(j>=nch){
             j = 0;
             k++;
         }
@@ -61,133 +174,6 @@ static void parse_matrix(struct af_instance *af, const char *cp)
             break;
         cp++;
     }
-
-}
-
-// Initialization and runtime control
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    af_pan_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_format(af->data, AF_FORMAT_FLOAT);
-        set_channels(af->data, s->nch ? s->nch : ((struct mp_audio*)arg)->nch);
-
-        if ((af->data->format != ((struct mp_audio*)arg)->format) ||
-            (af->data->bps != ((struct mp_audio*)arg)->bps)) {
-            mp_audio_set_format((struct mp_audio*)arg, af->data->format);
-            return AF_FALSE;
-        }
-        return AF_OK;
-    case AF_CONTROL_SET_PAN_LEVEL: {
-        int    i;
-        int    ch = ((af_control_ext_t*)arg)->ch;
-        float  *level = ((af_control_ext_t*)arg)->arg;
-        if (ch >= AF_NCH)
-            return AF_FALSE;
-        for (i = 0; i < AF_NCH; i++)
-            s->level[ch][i] = level[i];
-        return AF_OK;
-    }
-    case AF_CONTROL_SET_PAN_NOUT:
-        // Reinit must be called after this function has been called
-        // Sanity check
-        if (((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH) {
-            MP_ERR(af, "The number of output channels must be"
-                       " between 1 and %i. Current value is %i\n",
-                       AF_NCH, ((int*)arg)[0]);
-            return AF_ERROR;
-        }
-        s->nch = ((int*)arg)[0];
-        return AF_OK;
-    case AF_CONTROL_SET_PAN_BALANCE: {
-        float val = *(float*)arg;
-        if (s->nch)
-            return AF_ERROR;
-        if (af->data->nch >= 2) {
-            s->level[0][0] = MPMIN(1.f, 1.f - val);
-            s->level[0][1] = MPMAX(0.f, val);
-            s->level[1][0] = MPMAX(0.f, -val);
-            s->level[1][1] = MPMIN(1.f, 1.f + val);
-        }
-        return AF_OK;
-    }
-    case AF_CONTROL_GET_PAN_BALANCE:
-        if (s->nch)
-            return AF_ERROR;
-        *(float*)arg = s->level[0][1] - s->level[1][0];
-        return AF_OK;
-    case AF_CONTROL_COMMAND: {
-        char **args = arg;
-        if (!strcmp(args[0], "set-matrix")) {
-            parse_matrix(af, args[1]);
-            return CONTROL_OK;
-        } else {
-            return CONTROL_ERROR;
-        }
-    }
-    }
-    return AF_UNKNOWN;
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *c)
-{
-    if (!c)
-        return 0;
-    struct mp_audio *l = mp_audio_pool_get(af->out_pool, &af->fmt_out, c->samples);
-    if (!l) {
-        talloc_free(c);
-        return -1;
-    }
-    mp_audio_copy_attributes(l, c);
-
-    af_pan_t*     s    = af->priv;        // Setup for this instance
-    float         *in  = c->planes[0];    // Input audio data
-    float         *out = NULL;            // Output audio data
-    float         *end = in+c->samples * c->nch;    // End of loop
-    int           nchi = c->nch;          // Number of input channels
-    int           ncho = l->nch;          // Number of output channels
-    register int  j, k;
-
-    out = l->planes[0];
-    // Execute panning
-    // FIXME: Too slow
-    while (in < end) {
-        for (j = 0; j < ncho; j++) {
-            register float  x    = 0.0;
-            register float  *tin = in;
-            for (k = 0; k < nchi; k++)
-                x += tin[k] * s->level[j][k];
-            out[j] = x;
-        }
-        out += ncho;
-        in += nchi;
-    }
-
-    talloc_free(c);
-    af_add_output_frame(af, l);
-    return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance *af)
-{
-    af->control = control;
-    af->filter_frame = filter_frame;
-    af_pan_t *s = af->priv;
-    int nch = s->nch;
-    if (nch && AF_OK != control(af, AF_CONTROL_SET_PAN_NOUT, &nch))
-        return AF_ERROR;
-
-    // Read pan values
-    if (s->matrixstr)
-        parse_matrix(af, s->matrixstr);
     return AF_OK;
 }