Release 0.6.1

Oops.

Fixes #1172.

CC: @mpv-player/stable

.travis.yml

@@ -4,23 +4,17 @@ os:
   - linux
   - osx
 env:
-  matrix:
-    - LIBAV=libav-stable
-    - LIBAV=libav-git
-    - LIBAV=ffmpeg-stable
-    - LIBAV=ffmpeg-git
-  global:
-    # Coverity token
-    - secure: "H21mSRlMhk4BKS0xHZvCFGJxteCP0hRVUxTuNfM2Z9HBsyutuLEYMtViLO86VtM+Tqla3xXPzUdS4ozLwI72Ax/5ZUDXACROj73yW6QhFB5D6rLut12+FjqC7M33Qv2hl0xwgNBmR5dsm1ToP37+Wn+ecJQNvN8fkTXF+HVzOEw="
+  - LIBAV=libav-stable
+  - LIBAV=libav-git
+  - LIBAV=ffmpeg-stable
+  - LIBAV=ffmpeg-git
 compiler:
   - clang
-  - gcc
 
 branches:
   only:
     - master
     - ci
-    - coverity_scan
 
 matrix:
   exclude:
@@ -30,12 +24,6 @@ matrix:
       env: LIBAV=libav-git
     - os: osx
       env: LIBAV=ffmpeg-git
-    - os: osx
-      compiler: gcc
-    - os: linux
-      env: LIBAV=ffmpeg-stable
-    - os: linux
-      compiler: clang
 
 before_install: ./travis-deps libass-stable $LIBAV
 script:
@@ -50,13 +38,3 @@ notifications:
       - "irc.freenode.org#mpv-player-dev"
     on_success: change
     on_failure: always
-
-addons:
-  coverity_scan:
-    project:
-      name: "mpv-player/mpv"
-      description: "Build submitted via Travis CI"
-    notification_email: mpv-team@googlegroups.com
-    build_command_prepend: "./bootstrap.py && ./waf configure"
-    build_command:   "./waf build"
-    branch_pattern: coverity_scan

DOCS/client-api-changes.rst

@@ -25,36 +25,11 @@ API changes
 
 ::
 
- --- 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
-        - add MPV_END_FILE_REASON_ERROR and the mpv_event_end_file.error field
-          for slightly better error reporting on playback failure
-        - add --stop-playback-on-init-failure option, and make it the default
-          behavior for libmpv only
-        - add qthelper.hpp set_option_variant()
-        - mark the following events as deprecated:
-            MPV_EVENT_TRACKS_CHANGED
-            MPV_EVENT_TRACK_SWITCHED
-            MPV_EVENT_PAUSE
-            MPV_EVENT_UNPAUSE
-            MPV_EVENT_METADATA_UPDATE
-            MPV_EVENT_CHAPTER_CHANGE
-          They are handled better with mpv_observe_property() as mentioned in
-          the documentation comments. They are not removed and still work.
- 1.8    - add qthelper.hpp
- 1.7    - add mpv_command_node(), mpv_command_node_async()
- 1.6    - modify "core-idle" property behavior
-        - MPV_EVENT_LOG_MESSAGE now always sends complete lines
-        - introduce numeric log levels (mpv_log_level)
  --- mpv 0.6.0 is released ---
  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
           measure until XEmbed is implemented and confirmed working.
-          Note: in 1.6, "input-x11-keyboard" was renamed to "input-vo-keyboard",
-          although the old option name still works.
  1.4    - subtle change in X11 and "--wid" behavior
           (this change was added to 0.5.2, and broke some things, see #1090)
  --- mpv 0.5.0 is released ---

DOCS/client_api_examples/cocoabasic.m

@@ -8,19 +8,10 @@
 
 #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;
 }
 @end
 
@@ -28,39 +19,7 @@ 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];
-
-    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: @"Shutdown mpv" action:@selector(shutdown) keyEquivalent:@"s"];
-    [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) {
@@ -69,8 +28,6 @@ static void wakeup(void *);
     }
     NSString *filename = args[1];
 
-    [self createWindow];
-
     // Deal with MPV in the background.
     queue = dispatch_queue_create("mpv", DISPATCH_QUEUE_SERIAL);
     dispatch_async(queue, ^{
@@ -81,21 +38,10 @@ static void wakeup(void *);
             exit(1);
         }
 
-        int64_t wid = (intptr_t) [self->w contentView];
-        check_error(mpv_set_option(mpv, "wid", MPV_FORMAT_INT64, &wid));
-
         // Maybe set some options here, like default key bindings.
         // 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.
@@ -110,27 +56,17 @@ static void wakeup(void *);
 - (void) handleEvent:(mpv_event *)event
 {
     switch (event->event_id) {
-    case MPV_EVENT_SHUTDOWN:
-        // Clean up and shut down.
-        mpv_terminate_destroy(mpv);
-        mpv = NULL;
-        dispatch_async(dispatch_get_main_queue(), ^{
-            [[NSApplication sharedApplication] terminate:nil];
-        });
-        break;
+        case MPV_EVENT_SHUTDOWN:
+            // Clean up and shut down.
+            mpv_terminate_destroy(mpv);
+            mpv = NULL;
+            dispatch_async(dispatch_get_main_queue(), ^{
+                [[NSApplication sharedApplication] terminate:nil];
+            });
+            break;
 
-    case MPV_EVENT_LOG_MESSAGE: {
-        struct mpv_event_log_message *msg = (struct mpv_event_log_message *)event->data;
-        printf("[%s] %s: %s", msg->prefix, msg->level, msg->text);
-    }
-
-    case MPV_EVENT_VIDEO_RECONFIG:
-        dispatch_async(dispatch_get_main_queue(), ^{
-            [self->w selectNextKeyView:nil];
-        });
-
-    default:
-        printf("event: %s\n", mpv_event_name(event->event_id));
+        default:
+            printf("event: %s\n", mpv_event_name(event->event_id));
     }
 }
 
@@ -155,8 +91,6 @@ static void wakeup(void *context) {
 - (BOOL) windowShouldClose:(id)sender
 {
     [self shutdown];
-    if (self->w)
-        [self->w release];
     return YES;
 }
 
@@ -169,10 +103,12 @@ static void wakeup(void *context) {
 }
 @end
 
+
+
 // Delete this if you already have a main.m.
 int main(int argc, const char * argv[]) {
     @autoreleasepool {
-        NSApplication *app = [NSApplication sharedApplication];
+        NSApplication *app = [NSApplication new];
         AppDelegate *delegate = [AppDelegate new];
         app.delegate = delegate;
         [app run];

DOCS/client_api_examples/qtexample.cpp

@@ -2,23 +2,14 @@
 
 // This example can be built with: qmake && make
 
-#include <clocale>
 #include <sstream>
 
-#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"
 
@@ -35,9 +26,6 @@ static void wakeup(void *ctx)
 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);
@@ -47,23 +35,15 @@ MainWindow::MainWindow(QWidget *parent) :
 
     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 "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
+    // pass the window ID to the mpv wid option. This doesn't work on OSX,
+    // because Cocoa doesn't support this form of embedding.
     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();
@@ -74,20 +54,13 @@ MainWindow::MainWindow(QWidget *parent) :
     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");
+    // --input-x11-keyboard on the manpage.
+    mpv_set_option_string(mpv, "input-x11-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 Qt QEvent mechanism to relay
     // the wakeup in a thread-safe way.
@@ -113,19 +86,6 @@ void MainWindow::handle_mpv_event(mpv_event *event)
                 // 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;
     }
@@ -147,13 +107,6 @@ void MainWindow::handle_mpv_event(mpv_event *event)
         }
         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;
@@ -190,14 +143,6 @@ void MainWindow::on_file_open()
     }
 }
 
-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)
@@ -207,11 +152,6 @@ MainWindow::~MainWindow()
 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();
 

DOCS/client_api_examples/qtexample.h

@@ -5,8 +5,6 @@
 
 #include <mpv/client.h>
 
-class QTextEdit;
-
 class MainWindow : public QMainWindow
 {
     Q_OBJECT
@@ -24,9 +22,6 @@ private slots:
 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);

DOCS/client_api_examples/qtexample.pro

@@ -5,7 +5,7 @@ greaterThan(QT_MAJOR_VERSION, 4): QT += widgets
 TARGET = qtexample
 TEMPLATE = app
 
-CONFIG += link_pkgconfig debug
+CONFIG += link_pkgconfig
 PKGCONFIG = mpv
 
 SOURCES += qtexample.cpp

DOCS/edl-mpv.rst

@@ -27,13 +27,13 @@ The general simplified syntax is::
     <filename>,<start in seconds>,<length in seconds>
 
 If the start time is omitted, 0 is used. If the length is omitted, the
-estimated remaining duration of the source file is used.
+estimated duration of the source file is used.
 
 Note::
 
-    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.
+    mpv can't use ordered chapter files or libquvi-resolved URLs 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/man/ao.rst

@@ -114,18 +114,25 @@ Available audio output drivers are:
         with fixed setups.
 
 ``coreaudio`` (Mac OS X only)
-    Native Mac OS X audio output driver using AudioUnits and the CoreAudio
+    Native Mac OS X audio output driver using the AudioUnits and the CoreAudio
     sound server.
 
-    Automatically redirects to ``coreaudio_exclusive`` when playing compressed
-    formats.
+    ``device_id=<id>``
+        ID of output device to use (0 = default device)
+    ``help``
+        List all available output devices with their IDs.
 
-``coreaudio_exclusive`` (Mac OS X only)
+``coreaudio_device`` (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).
 
+    ``device_id=<id>``
+        ID of output device to use (0 = default device)
+    ``help``
+        List all available output devices with their IDs.
+
 ``openal``
     Experimental OpenAL audio output driver
 

DOCS/man/changes.rst

@@ -10,159 +10,104 @@ should generally be treated as a completely different program.
 .. note::
     These lists are incomplete.
 
-General Changes from MPlayer to mpv
------------------------------------
+General Changes for MPlayer to mpv
+----------------------------------
 
-This listing is about changes introduced by mplayer2 and mpv relatively to
-MPlayer.
-
-Player
-~~~~~~
-
-* New name for the binary (``mpv``). New location for config files (either
-  ``~/.config/mpv/mpv.conf``, or if you want, ``~/.mpv/config``).
-* Encoding functionality (replacement for MEncoder, see the `ENCODING`_ section).
-* Support for Lua scripting (see the `LUA SCRIPTING`_ section).
-* Better pause handling (e.g. do not unpause on a command).
-* Precise seeking support.
-* Improvements in audio/video sync handling.
-* Do not lose settings when playing a new file in the same player instance.
-* 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.
+* Switch to GPLv2+. (Technically speaking, MPlayer as a whole seems to be
+  GPLv2-only, while mplayer2 is GPLv3+ - see ``Copyright`` file for details.)
+* Removal of the internal GUI, MEncoder, OSD menu, video kernel drivers for
+  Linux 2.4 (including VIDIX)
+* Large internal cleanups
+* Removal of support for dead platforms
+* New build system
+* No embedded copy of FFmpeg and other libraries
+* Better pause handling (do not unpause on a command)
+* Better MKV support (such as ordered chapters)
+* vo_vdpau improvements
+* Precise seeking support
+* Native OpenGL backend for OS X
+* General OS X improvements
+* Improvements in audio/video sync handling
+* Cleaned up terminal output
+* Gapless audio support (``--gapless-audio``)
+* Improved responsiveness on user input
+* Support for modifier keys (alt, shift, ctrl) in input.conf
+* OSS4 volume control
+* More correct color reproduction (color matrix generation)
+* Use libass for subtitle rendering by default (better quality)
+* Generally preferring FFmpeg/Libav over internal demuxers and decoders
+* Improvements when playing multiple files (``--fixed-vo``)
+* Screenshot improvements (instant screenshots without 1-frame delay, allow
+  taking screenshots even with hardware decoding)
+* Improved support for PulseAudio
+* Generally improved MS Windows support (dealing with Unicode file names,
+  improved ``--vo=direct3d``, improved window handling)
+* Better OSD rendering (using libass). This has full Unicode support, and
+  languages like Arabic should be better supported.
+* Cleaned up terminal output (nicer status line, less useless noise)
+* Support for playing URLs of popular streaming sites directly
+  (e.g. ``mpv https://www.youtube.com/watch?v=...``)
+* Improved OpenGL output (``--vo=opengl-hq``)
+* Make ``--softvol`` default (**mpv** is not a mixer control panel)
+* Improved support for .cue files
+* Screenshot improvements (can save screenshots as JPEG or PNG, configurable
+  file names, support for taking screenshots with or without subtitles - the
+  ``screenshot`` video filter is not needed anymore, and should not be put
+  into the mpv config file)
+* Removal of teletext support
+* Removal of most built-in demuxers, using libavformat instead
+* Removal of built-in network support, using libavformat instead (also,
+  support https via libavformat)
+* Replace image VOs (``--vo=jpeg`` etc.) with ``--vo=image``
+* Do not lose settings when playing a new file in the same player instance
+* New location for config files, new name for the binary.
+* Slave mode compatibility broken (see below)
+* Encoding functionality (replacement for MEncoder, see ``DOCS/encoding.rst``)
+* Remove ``--vo=gif89a``, ``--vo=md5sum``, ``--vo=yuv4mpeg``, as encoding can
+  handle these use cases. For yuv4mpeg, for example, use:
+  ``mpv input.mkv -o output.y4m --no-audio --oautofps --oneverdrop``.
+* Image subtitles (DVDs etc.) are rendered in color and use more correct
+  positioning (color can be disabled with ``--sub-gray``)
+* Wayland support
+* Support for precise scrolling which scales the parameter of commands. If the
+  input doesn't support precise scrolling the scale factor stays 1.
+* OS X: Cocoa event loop is independent from MPlayer's event loop, so user
+  actions like accessing menus and live resizing do not block the playback.
+* OS X: Apple Remote support.
+* OS X: Media Keys support.
+* Windows: Added WASAPI audio output.
+* New OSD bar with chapter marks and not positioned in the middle of the video
+  (though this can be customized with the ``--osd-bar-align-y`` option).
+* Allow customizing whether a key binding for seeking shows the video time, the
+  OSD bar, or nothing (see section `Input Command Prefixes`_).
+* Display list of chapters and audio/subtitle tracks on OSD (see section
+  `Properties`_).
 * ``--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
-  (see ``libmpv/client.h`` in the sources).
-
-Input
-~~~~~
-
-* Improved default keybindings. MPlayer bindings are also available (see
-  ``etc/mplayer-input.conf`` in the source tree).
-* Improved responsiveness on user input.
-* Support for modifier keys (alt, shift, ctrl) in input.conf.
-* Allow customizing whether a key binding for seeking shows the video time, the
-  OSD bar, or nothing (see the `Input Command Prefixes`_ section).
-* Support mapping multiple commands to one key.
-
-Audio
-~~~~~
-
-* Support for gapless audio (see the ``--gapless-audio`` option).
-* Support for OSS4 volume control.
-* Improved support for PulseAudio.
-* Make ``--softvol`` default (**mpv** is not a mixer control panel).
-* By default, do pitch correction if playback speed is increased.
-* Improved downmixing and output of surround audio:
-
-  - Instead of using hardcoded pan filters to do remixing, libavresample is used
-  - Channel maps are used to identify the channel layout, so e.g. ``3.0`` and
-    ``2.1`` audio can be distinguished.
-
-Video
-~~~~~
-
-* Wayland support.
-* Native support for VAAPI and VDA. Improved VDPAU video output.
-* Improved OpenGL output (see the ``opengl-hq`` video output).
-* Make hardware decoding work with the ``opengl`` video output.
+* Re-enable screensaver while the player is paused.
+* Matroska edition switching at runtime.
 * Support for libavfilter (for video->video and audio->audio). This allows
   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).
-* Better subtitle rendering using libass by default.
-* Improvements when playing multiple files (``--fixed-vo`` is default, do not
-  reset settings by default when playing a new file).
-* Replace image video outputs (``--vo=jpeg`` etc.) with ``--vo=image``.
-* Removal of ``--vo=gif89a``, ``--vo=md5sum``, ``--vo=yuv4mpeg``, as encoding
-  can handle these use cases. For yuv4mpeg, for example, use::
-
-    mpv input.mkv -o output.y4m --no-audio --oautofps --oneverdrop
-
-* Image subtitles (DVDs etc.) are rendered in color and use more correct
-  positioning (color for image subs can be disabled with ``--sub-gray``).
-
-OSD and terminal
-~~~~~~~~~~~~~~~~
-
-* Cleaned up terminal output: nicer status line, less useless noise.
-* Improved OSD rendering using libass, with full Unicode support.
-* New OSD bar with chapter marks. Not positioned in the middle of the video
-  (this can be customized with the ``--osd-bar-align-y`` option).
-* Display list of chapters and audio/subtitle tracks on OSD (see the
-  `Properties`_ section).
-
-Screenshots
-~~~~~~~~~~~
-
-* Instant screenshots without 1-frame delay.
-* Support for taking screenshots even with hardware decoding.
-* Support for saving screenshots as JPEG or PNG.
-* Support for configurable file names.
-* Support for taking screenshots with or without subtitles.
-
-Note that the ``screenshot`` video filter is not needed anymore, and should not
-be put into the mpv config file.
-
-Miscellaneous
-~~~~~~~~~~~~~
-
-* Better MKV support (e.g. ordered chapters, 3D metadata).
-* Matroska edition switching at runtime.
-* Support for playing URLs of popular streaming sites directly.
-  (e.g. ``mpv https://www.youtube.com/watch?v=...``).
-  Requires a recent version of ``youtube-dl`` to be installed. Can be
-  disabled with ``ytdl=no`` in the mpv config file.
-* Support for precise scrolling which scales the parameter of commands. If the
-  input doesn't support precise scrolling the scale factor stays 1.
+* Improved downmixing and output of surround audio. Instead of using hardcoded
+  pan filters to do remixing, use libavresample. Channel maps are used to
+  identify the channel layout, so e.g. ``3.0`` and ``2.1`` audio can be
+  distinguished.
+* Allow resuming playback at a later point with ``Shift+q``, also see
+  ``quit_watch_later`` input command.
+* Support mapping multiple commands to one key.
 * Allow changing/adjusting video filters at runtime. (This is also used to make
-  the ``D`` key insert vf_yadif if deinterlacing is not supported otherwise).
-* Improved support for .cue files.
-
-Mac OS X
-~~~~~~~~
-
-* Native OpenGL backend.
-* Cocoa event loop is independent from MPlayer's event loop, so user
-  actions like accessing menus and live resizing do not block the playback.
-* Apple Remote support.
-* Media Keys support.
-* VDA support using libavcodec hwaccel API instead of FFmpeg's decoder with up
+  the ``D`` key insert vf_yadif if deinterlacing is not supported otherwise.)
+* Native VAAPI support
+* OS X: VDA support using libavcodec hwaccel API instead of FFmpeg's decoder. Up
   to 2-2.5x reduction in CPU usage.
-
-Windows
-~~~~~~~
-
-* Improved support for Unicode file names.
-* Improved window handling.
-* Do not block playback when moving the window.
-* Improved Direct3D video output.
-* Added WASAPI audio output.
-
-Internal changes
-~~~~~~~~~~~~~~~~
-
-* Switch to GPLv2+ (see ``Copyright`` file for details).
-* Removal of lots of cruft:
-
-  - Internal GUI (replaced by the OSC, see the `ON SCREEN CONTROLLER`_ section).
-  - MEncoder (replaced by native encoding, see the `ENCODING`_ section).
-  - OSD menu.
-  - Kernel video drivers for Linux 2.4 (including VIDIX).
-  - Teletext support.
-  - Support for dead platforms.
-  - Most built-in demuxers have been replaced by their libavformat counterparts.
-  - Built-in network support has been replaced by libavformat's (which also
-    supports https URLs).
-  - Embedded copies of libraries (such as FFmpeg).
-
-* General code cleanups (including refactoring or rewrites of many parts).
-* New build system.
-* Many bug fixes and removal of long-standing issues.
-* Generally preferring FFmpeg/Libav over internal demuxers, decoders, and
-  filters.
+* Make hardware decoding in general work with the ``opengl`` video output.
+* Lua scripting (see `LUA SCRIPTING`_)
+* A client API, that allows embedding **mpv** into applications
+  (see ``libmpv/client.h`` in the sources)
+* General bug fixes and removal of long-standing issues
+* General code cleanups (including refactoring or rewrites of many parts)
+* Many more changes
 
 Detailed Listing of User-visible Changes
 ----------------------------------------
@@ -389,8 +334,8 @@ Slave mode
 
   (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.
+* A JSON RPC protocol giving access to the client API is planned, but nothing
+  has emerged yet.
 
 * **mpv** also provides a client API, which can be used to embed the player
   by loading it as shared library. (See ``libmpv/client.h`` in the sources.)

DOCS/man/input.rst

@@ -193,7 +193,7 @@ List of Input Commands
     force
         Terminate playback if the first file is being played.
 
-``loadfile "<file>" [replace|append|append-play [options]]``
+``loadfile "<file>" [replace|append [options]]``
     Load the given file and play it.
 
     Second argument:
@@ -204,8 +204,6 @@ List of Input Commands
         Append the file to the playlist.
     <append-play>
         Append the file, and if nothing is currently playing, start playback.
-        (Always starts with the added file, even if the playlist was not empty
-        before running this command.)
 
     The third argument is a list of options and values which should be set
     while the file is playing. It is of the form ``opt1=value1,opt2=value2,..``.
@@ -262,33 +260,10 @@ List of Input Commands
     will seek to the previous position on start. The (optional) argument is
     exactly as in the ``quit`` command.
 
-``sub_add "<file>" [<flags> [<title> [<lang>]]]``
+``sub_add "<file>"``
     Load the given subtitle file. It is selected as current subtitle after
     loading.
 
-    The ``flags`` args is one of the following values:
-
-    <select>
-
-        Select the subtitle immediately.
-
-    <auto>
-
-        Don't select the subtitle. (Or in some special situations, let the
-        default stream selection mechanism decide.)
-
-    <cached>
-
-        Select the subtitle. If a subtitle with the same filename was already
-        added, that one is selected, instead of loading a duplicate entry.
-        (In this case, title/language are ignored, and if the was changed since
-        it was loaded, these changes won't be reflected.)
-
-    The ``title`` argument sets the track title in the UI.
-
-    The ``lang`` argument sets the track language, and can also influence
-    stream selection with ``flags`` set to ``auto``.
-
 ``sub_remove [<id>]``
     Remove the given subtitle track. If the ``id`` argument is missing, remove
     the current track. (Works on external subtitle files only.)
@@ -428,7 +403,7 @@ Input Commands that are Possibly Subject to Change
     The special argument ``!reverse`` can be used to cycle the value list in
     reverse. Compared with a command that just lists the value in reverse, this
     command will actually share the internal counter with the forward-cycling
-    key binding (as long as the rest of the arguments are the same).
+    key binding.
 
     Note that there is a static limit of (as of this writing) 10 arguments
     (this limit could be raised on demand).
@@ -482,16 +457,6 @@ Input Commands that are Possibly Subject to Change
     random points. Instead, call ``overlay_add`` again (preferably with a
     different memory region to prevent tearing).
 
-    It is also possible to pass a raw memory address for use as bitmap memory
-    by passing a memory address as integer prefixed with a ``&`` character.
-    Passing the wrong thing here will crash the player. The ``offset`` parameter
-    is not used and must be 0. This mode might be useful for use with libmpv.
-
-    On Windows, currently only raw memory addresses work. File mapping is not
-    implemented because a ``mmap`` compatibility layer is missing, and because
-    this kind of shared memory method would perhaps not be overly useful on
-    Windows.
-
     ``offset`` is the offset of the first pixel in the source file. It is
     passed directly to ``mmap`` and is subject to certain restrictions
     (see ``man mmap`` for details). In particular, this value has to be a
@@ -541,46 +506,12 @@ Input Commands that are Possibly Subject to Change
     ``<target>``. Each client (scripts etc.) has a unique name. For example,
     Lua scripts can get their name via ``mp.get_script_name()``.
 
-``script_binding "<name>"``
-    Invoke a script-provided key binding. This can be used to remap key
-    bindings provided by external Lua scripts.
+    (Scripts use this internally to dispatch key bindings, and this can also
+    be used in input.conf to reassign such bindings.)
 
-    The argument is the name of the binding.
-
-    It can optionally be prefixed with the name of the script, using ``/`` as
-    separator, e.g. ``script_binding scriptname/bindingname``.
-
-    For completeness, here is how this command works internally. The details
-    could change any time. On any matching key event, ``script_message_to``
-    or ``script_message`` is called (depending on whether the script name is
-    included), where the first argument is the string ``key-binding``, the
-    second argument is the name of the binding, and the third argument is the
-    key state as string. The key state consists of a number of letters. The
-    first letter is one of ``d`` (key was pressed down), ``u`` (was released),
-    ``r`` (key is still down, and was repeated; only if key repeat is enabled
-    for this binding), ``p`` (key was pressed; happens if up/down can't be
-    tracked). The second letter whether the event originates from the mouse,
-    either ``m`` (mouse button) or ``-`` (something else).
-
-``ab_loop``
-    Cycle through A-B loop states. The first command will set the ``A`` point
-    (the ``ab-loop-a`` property); the second the ``B`` point, and the third
-    will clear both points.
-
-``vo_cmdline "<args>"``
-    Reset the sub-option of the current VO. Currently works with ``opengl``
-    (including ``opengl-hq``). The argument is the sub-option string usually
-    passed to the VO on the command line. Not all sub-options can be set, but
-    those which can will be reset even if they don't appear in the argument.
-    This command might be changed or removed in the future.
-
-``drop_buffers``
-    Drop audio/video/demuxer buffers, and restart from fresh. Might help with
-    unseekable streams that are going out of sync.
-    This command might be changed or removed in the future.
 
 Undocumented commands: ``tv_last_channel`` (TV/DVB only), ``get_property`` (?),
-``ao_reload`` (experimental/internal).
+``vo_cmdline`` (experimental).
 
 Input Command Prefixes
 ----------------------
@@ -605,9 +536,7 @@ prefixes can be specified. They are separated by whitespace.
     Do not expand properties in string arguments. (Like ``"${property-name}"``.)
 ``expand-properties`` (default)
     All string arguments are expanded as described in `Property Expansion`_.
-``repeatable``
-    For some commands, keeping a key pressed doesn't run the command repeatedly.
-    This prefix forces enabling key repeat in any case.
+
 
 All of the osd prefixes are still overridden by the global ``--osd-level``
 settings.
@@ -690,7 +619,10 @@ Property list
     Full path of the currently played file.
 
 ``media-title``
-    If the currently played file has a ``title`` tag, use that.
+    If libquvi is used and libquvi returns a page title for the currently
+    played URL, return the page title.
+
+    Otherwise, if the currently played file has a ``title`` tag, use that.
 
     Otherwise, if the media type is DVD, return the volume ID of DVD.
 
@@ -724,10 +656,9 @@ Property list
     disabled.
 
 ``drop-frame-count``
-    Video frames dropped by decoder, because video is too far behind audio (when
-    using ``--framedrop=decoder``). Sometimes, this may be incremented in other
-    situations, e.g. when video packets are damaged, or the decoder doesn't
-    follow the usual rules. Unavailable if video is disabled.
+    Frames dropped because they arrived too late. Doesn't necessarily indicate
+    actual frame-drops, just the number of times the decoder was asked to drop.
+    Unavailable if video is disabled
 
 ``vo-drop-frame-count``
     Frames dropped by VO (when using ``--framedrop=vo``).
@@ -814,10 +745,6 @@ Property list
                 "title"             MPV_FORMAT_STRING
                 "default"           MPV_FORMAT_FLAG
 
-``ab-loop-a``, ``ab-loop-b`` (TW)
-    Set/get A-B loop points. See corresponding options and ``ab_loop`` command.
-    The special value ``no`` on either of these properties disables looping.
-
 ``angle`` (RW)
     Current DVD angle.
 
@@ -891,10 +818,6 @@ Property list
     be different ``pause`` in special situations, such as when the player
     pauses itself due to low network cache.
 
-    This also returns ``yes`` if playback is restarting or if nothing is
-    playing at all. In other words, it's only ``no`` if there's actually
-    video playing. (Behavior since mpv 0.7.0.)
-
 ``cache``
     Network cache fill state (0-100.0).
 
@@ -932,10 +855,6 @@ Property list
 ``paused-for-cache``
     Returns ``yes`` when playback is paused because of waiting for the cache.
 
-``cache-buffering-state``
-    Return the percentage (0-100) of the cache fill status until the player
-    will unpause (related to ``paused-for-cache``).
-
 ``eof-reached``
     Returns ``yes`` if end of playback was reached, ``no`` otherwise. Note
     that this is usually interesting only if ``--keep-open`` is enabled,
@@ -997,9 +916,6 @@ Property list
 ``deinterlace`` (RW)
     See ``--deinterlace``.
 
-``field-dominance`` (RW)
-    See ``--field-dominance``
-
 ``colormatrix`` (RW)
     See ``--colormatrix``.
 
@@ -1155,13 +1071,6 @@ Property list
     (or to be exact, the size the video filters output). ``2`` will set the
     double size, ``0.5`` halves the size.
 
-``window-minimized``
-    Return whether the video window is minimized or not.
-
-``display-names``
-    Names of the displays that the mpv window covers. On X11, these
-    are the xrandr names (LVDS1, HDMI1, DP1, VGA1, etc.).
-
 ``video-aspect`` (RW)
     Video aspect, see ``--video-aspect``.
 
@@ -1228,8 +1137,8 @@ Property list
     See ``--ass-style-override``.
 
 ``stream-capture`` (RW)
-    A filename, see ``--stream-capture``. Setting this will start capture using
-    the given filename. Setting it to an empty string will stop it.
+    A filename, see ``--capture``. Setting this will start capture using the
+    given filename. Setting it to an empty string will stop it.
 
 ``tv-brightness``, ``tv-contrast``, ``tv-saturation``, ``tv-hue`` (RW)
     TV stuff.
@@ -1254,13 +1163,6 @@ Property list
     ``playlist/N/filename``
         Filename of the Nth entry.
 
-    ``playlist/N/current``, ``playlist/N/playing``
-        ``yes`` if this entry is currently playing (or being loaded).
-        Unavailable or ``no`` otherwise. When changing files, ``current`` and
-        ``playing`` can be different, because the currently playing file hasn't
-        been unloaded yet; in this case, ``current`` refers to the new
-        selection. (Since mpv 0.7.0.)
-
     When querying the property with the client API using ``MPV_FORMAT_NODE``,
     or with Lua ``mp.get_property_native``, this will return a mpv_node with
     the following contents:
@@ -1270,8 +1172,6 @@ Property list
         MPV_FORMAT_NODE_ARRAY
             MPV_FORMAT_NODE_MAP (for each playlist entry)
                 "filename"  MPV_FORMAT_STRING
-                "current"   MPV_FORMAT_FLAG (might be missing; since mpv 0.7.0)
-                "playing"   MPV_FORMAT_FLAG (same)
 
 ``track-list``
     List of audio/video/sub tracks, current entry marked. Currently, the raw
@@ -1391,21 +1291,9 @@ Property list
 
     It's also possible to write the property using this format.
 
-``video-rotate`` (RW)
-    See ``--video-rotate`` option.
-
 ``seekable``
     Return whether it's generally possible to seek in the current file.
 
-``playback-abort``
-    Return whether playback is stopped or is to be stopped. (Useful in obscure
-    situations like during ``on_load`` hook processing, when the user can
-    stop playback, but the script has to explicitly end processing.)
-
-``cursor-autohide`` (RW)
-    See ``--cursor-autohide``. Setting this to a new value will always update
-    the cursor, and reset the internal timer.
-
 ``osd-sym-cc``
     Inserts the current OSD symbol as opaque OSD control code (cc). This makes
     sense only with the ``show_text`` command or options which set OSD messages.
@@ -1428,104 +1316,12 @@ Property list
     Note that you need to escape the ``\`` character, because the string is
     processed for C escape sequences before passing it to the OSD code.
 
-``vo-configured``
-    Return whether the VO is configured right now. Usually this corresponds to
-    whether the video window is visible. If the ``--force-window`` option is
-    used, this is usually always returns ``yes``.
-
-``audio-device-list``
-    Return the list of discovered audio devices. This is mostly for use with
-    the client API, and reflects what ``--audio-device=help`` with the command
-    line player returns.
-
-    When querying the property with the client API using ``MPV_FORMAT_NODE``,
-    or with Lua ``mp.get_property_native``, this will return a mpv_node with
-    the following contents:
-
-    ::
-
-        MPV_FORMAT_NODE_ARRAY
-            MPV_FORMAT_NODE_MAP (for each device entry)
-                "name"          MPV_FORMAT_STRING
-                "description"   MPV_FORMAT_STRING
-
-    The ``name`` is what is to be passed to the ``--audio-device`` option (and
-    often a rather cryptic audio API-specific ID), while ``description`` is
-    human readable free form text. The description is an empty string if none
-    was received.
-
-``audio-device`` (RW)
-    Set the audio device. This directly reads/writes the ``--audio-device``
-    option, but on write accesses, the audio output will be scheduled for
-    reloading.
-
-    Writing this property while no audio output is active will not automatically
-    enable audio. (This is also true in the case when audio was disabled due to
-    reinitialization failure after a previous write access to ``audio-device``.)
-
-    This property also doesn't tell you which audio device is actually in use.
-
-    How these details are handled may change in the future.
-
-``mpv-version``
-    Return the mpv version/copyright string. Depending on how the binary was
-    built, it might contain either a release version, or just a git hash.
-
-``mpv-configuration``
-    Return the configuration arguments which were passed to the build system
-    (typically the way ``./waf configure ...`` was invoked).
-
 ``options/<name>`` (RW)
     Read-only access to value of option ``--<name>``. Most options can be
     changed at runtime by writing to this property. Note that many options
     require reloading the file for changes to take effect. If there is an
     equivalent property, prefer setting the property instead.
 
-``file-local-options/<name>``
-    Similar to ``options/<name>``, but when setting an option through this
-    property, the option is reset to its old value once the current file has
-    stopped playing. Trying to write an option while no file is playing (or
-    is being loaded) results in an error.
-
-    (Note that if an option is marked as file-local, even ``options/`` will
-    access the local value, and the ``old`` value, which will be restored on
-    end of playback, can not be read or written until end of playback.)
-
-``option-info/<name>``
-    Additional per-option information.
-
-    This has a number of sub-properties. Replace ``<name>`` with the name of
-    a top-level option. No guarantee of stability is given to any of these
-    sub-properties - they may change radically in the feature.
-
-    ``option-info/<name>/name``
-        Returns the name of the option.
-
-    ``option-info/<name>/type``
-        Return the name of the option type, like ``String`` or ``Integer``.
-        For many complex types, this isn't very accurate.
-
-    ``option-info/<name>/set-from-commandline``
-
-        Return ``yes`` if the option was set from the mpv command line,
-        ``no`` otherwise. What this is set to if the option is e.g. changed
-        at runtime is left undefined (meaning it could change in the future).
-
-    ``option-info/<name>/default-value``
-        The default value of the option. May not always be available.
-
-    ``option-info/<name>/min``, ``option-info/<name>/max``
-        Integer minimum and maximum values allowed for the option. Only
-        available if the options are numeric, and the minimum/maximum has been
-        set internally. It's also possible that only one of these is set.
-
-    ``option-info/<name>/choices``
-        If the option is a choice option, the possible choices. Choices that
-        are integers may or may not be included (they can be implied by ``min``
-        and ``max``). Note that options which behave like choice options, but
-        are not actual choice options internally, may not have this info
-        available.
-
 ``property-list``
     Return the list of top-level properties.
 

DOCS/man/ipc.rst

@@ -1,183 +0,0 @@
-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 using the option
-``--input-unix-socket``. Clients can connect to this socket and send commands to
-the player or receive events from it. You can use the ``socat`` tool to send
-commands (and receive reply) from the shell.
-
-.. warning::
-
-    This is not intended to be a secure network protocol. It is explicitly
-    insecure: there is no authentication, no encryption, and the commands
-    themselves are insecure too. For example, the ``run`` command is exposed,
-    which can run arbitrary system commands. The use-case is controlling the
-    player locally. This is not different from the MPlayer slave protocol.
-
-Protocol
---------
-
-Clients can execute commands on the player by sending JSON messages of the
-following form:
-
-::
-
-    { "command": ["command_name", "param1", "param2", ...] }
-
-where ``command_name`` is the name of the command to be executed, followed by a
-list of parameters. Parameters must be formatted as native JSON values
-(integers, strings, booleans, ...). Every message **must** be terminated with
-``\n``. Additionally, ``\n`` must not appear anywhere inside the message. In
-practice this means that messages should be minified before being sent to mpv.
-
-mpv will then send back a reply indicating whether the command was run
-correctly, and an additional field holding the command-specific return data (it
-can also be null).
-
-::
-
-    { "error": "success", "data": null }
-
-mpv will also send events to clients with JSON messages of the following form:
-
-::
-
-    { "event": "event_name" }
-
-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.
-
-If the first character (after skipping whitespace) is not ``{``, the command
-will be interpreted as non-JSON text command, as they are used in input.conf
-(or ``mpv_command_string()`` in the client API). Additionally, line starting
-with ``#`` and empty lines are ignored.
-
-Currently, embedded 0 bytes terminate the current line, but you should not
-rely on this.
-
-Commands
---------
-
-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``
-    Return the name of the client as string. This is the string ``ipc-N`` with
-    N being an integer number.
-
-``get_time_us``
-    Return the current mpv internal time in microseconds as a number. This is
-    basically the system time, with an arbitrary offset.
-
-``get_property``
-    Return the value of the given property. The value will be sent in the data
-    field of the replay message.
-
-    Example:
-
-    ::
-
-        { "command": ["get_property", "volume"] }
-        { "data": 50.0, "error": "success" }
-
-``get_property_string``
-    Like ``get_property``, but the resulting data will always be a string.
-
-    Example:
-
-    ::
-
-        { "command": ["get_property_string", "volume"] }
-        { "data": "50.000000", "error": "success" }
-
-``set_property``
-    Set the given property to the given value. See `Properties`_ for more
-    information about properties.
-
-    Example:
-
-    ::
-
-        { "command": ["set_property", "pause", true] }
-        { "error": "success" }
-
-``set_property_string``
-    Like ``set_property``, but the argument value must be passed as string.
-
-    Example:
-
-    ::
-
-        { "command": ["set_property_string", "pause", "yes"] }
-        { "error": "success" }
-
-``observe_property``
-    Watch a property for changes. If the given property is changed, then an
-    event of type ``property-change`` will be generated
-
-    Example:
-
-    ::
-
-        { "command": ["observe_property", 1, "volume"] }
-        { "error": "success" }
-        { "event": "property-change", "id": 1, "data": 52.0, "name": "volume" }
-
-``observe_property_string``
-    Like ``observe_property``, but the resulting data will always be a string.
-
-    Example:
-
-    ::
-
-        { "command": ["observe_property_string", 1, "volume"] }
-        { "error": "success" }
-        { "event": "property-change", "id": 1, "data": "52.000000", "name": "volume" }
-
-``unobserve_property``
-    Undo ``observe_property`` or ``observe_property_string``. This requires the
-    numeric id passed to the observe command as argument.
-
-    Example:
-
-    ::
-
-        { "command": ["unobserve_property", 1] }
-        { "error": "success" }
-
-``request_log_messages``
-    Enable output of mpv log messages. They will be received as events. The
-    parameter to this command is the log-level (see ``mpv_request_log_messages``
-    C API function).
-
-    Log message output is meant for humans only (mostly for debugging).
-    Attempting to retrieve information by parsing these messages will just
-    lead to breakages with future mpv releases. Instead, make a feature request,
-    and ask for a proper event that returns the information you need.
-
-``suspend``
-    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``
-    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. (Also see ``DOCS/client-api-changes.rst``.)
-
-UTF-8
------
-
-Normally, all strings are in UTF-8. Sometimes it can happen that strings are
-in some broken encoding (often happens with file tags and such, and filenames
-on many Unixes are not required to be in UTF-8 either). This means that mpv
-sometimes sends invalid JSON. If that is a problem for the client application's
-parser, it should filter the raw data for invalid UTF-8 sequences and perform
-the desired replacement, before feeding the data to its JSON parser.
-
-mpv will not attempt to construct invalid UTF-8 with broken escape sequences.

DOCS/man/lua.rst

@@ -94,15 +94,6 @@ The ``mp`` module is preloaded, although it can be loaded manually with
     the ``expand-properties`` prefix, or the ``mp.get_property`` family of
     functions.
 
-``mp.command_native(table [,def])``
-    Similar to ``mp.commandv``, but pass the argument list as table. This has
-    the advantage that in at least some cases, arguments can be passed as
-    native types.
-
-    Returns a result table on success (usually empty), or ``def, error`` on
-    error. ``def`` is the second parameter provided to the function, and is
-    nil if it's missing.
-
 ``mp.get_property(name [,def])``
     Return the value of the given property as string. These are the same
     properties as used in input.conf. See `Properties`_ for a list of
@@ -181,7 +172,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
     Return the current mpv internal time in seconds as a number. This is
     basically the system time, with an arbitrary offset.
 
-``mp.add_key_binding(key, name|fn [,fn [,flags]])``
+``mp.add_key_binding(key, name|fn [,fn])``
     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
@@ -198,23 +189,8 @@ The ``mp`` module is preloaded, although it can be loaded manually with
     overwritten. You can omit the name, in which case a random name is generated
     internally.
 
-    The last argument is used for optional flags. This is a table, which can
-    have the following entries:
-
-        ``repeatable``
-            If set to ``true``, enables key repeat for this specific binding.
-
-        ``complex``
-            If set to ``true``, then ``fn`` is called on both key up and down
-            events (as well as key repeat, if enabled), with the first
-            argument being a table. This table has an ``event`` entry, which
-            is set to one of the strings ``down``, ``repeat``, ``up`` or
-            ``press`` (the latter if key up/down can't be tracked). It further
-            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`` input
+    command 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
@@ -237,7 +213,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 
     ::
 
-        y script_binding something
+        y script_message something
 
 
     This will print the message when the key ``y`` is pressed. (``x`` will
@@ -248,7 +224,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 
     ::
 
-        y script_binding fooscript.something
+        y script_message_to fooscript something
 
 ``mp.add_forced_key_binding(...)``
     This works almost the same as ``mp.add_key_binding``, but registers the
@@ -526,9 +502,6 @@ 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
 provided for convenience. Most compensate for Lua's scarce standard library.
 
-Be warned that any of these functions might disappear any time. They are not
-strictly part of the guaranteed API.
-
 ``utils.getcwd()``
     Returns the directory that mpv was launched from. On error, ``nil, error``
     is returned.
@@ -569,60 +542,6 @@ strictly part of the guaranteed API.
     Return the concatenation of the 2 paths. Tries to be clever. For example,
     if ```p2`` is an absolute path, p2 is returned without change.
 
-``utils.subprocess(t)``
-    Runs an external process and waits until it exits. Returns process status
-    and the captured output.
-
-    The parameter ``t`` is a table. The function reads the following entries:
-
-        ``args``
-            Array of strings. The first array entry is the executable. This
-            can be either an absolute path, or a filename with no path
-            components, in which case the ``PATH`` environment variable is
-            used to resolve the executable. The other array elements are
-            passed as command line arguments.
-
-        ``cancellable``
-            Optional. If set to ``true`` (default), then if the user stops
-            playback or goes to the next file while the process is running,
-            the process will be killed.
-
-        ``max_size``
-            Optional. The maximum size in bytes of the data that can be captured
-            from stdout. (Default: 16 MB.)
-
-    The function returns a table as result with the following entries:
-
-        ``status``
-            The raw exit status of the process. It will be negative on error.
-
-        ``stdout``
-            Captured output stream as string, limited to ``max_size``.
-
-        ``error``
-            ``nil`` on success. The string ``killed`` if the process was
-            terminated in an unusual way. The string ``init`` if the process
-            could not be started.
-
-            On Windows, ``killed`` is only returned when the process has been
-            killed by mpv as a result of ``cancellable`` being set to ``true``.
-
-    In all cases, ``mp.resume_all()`` is implicitly called.
-
-``utils.parse_json(str [, trail])``
-    Parses the given string argument as JSON, and returns it as a Lua table. On
-    error, returns ``nil, error``. (Currently, ``error`` is just a string
-    reading ``error``, because there is no fine-grained error reporting of any
-    kind.)
-
-    The returned value uses similar conventions as ``mp.get_property_native()``
-    to distinguish empty objects and arrays.
-
-    If the ``trail`` parameter is ``true`` (or any value equal to ``true``),
-    then trailing non-whitespace text is tolerated by the function, and the
-    trailing text is returned as 3rd return value. (The 3rd return value is
-    always there, but with ``trail`` set, no error is raised.)
-
 Events
 ------
 
@@ -664,11 +583,28 @@ List of events
 ``playback-restart``
     Start of playback after seek or after file was loaded.
 
+``tracks-changed``
+    The list of video/audio/sub tracks was updated. (This happens on playback
+    start, and very rarely during playback.)
+
+``track-switched``
+    A video/audio/subtitle track was switched on or off. This usually happens
+    when the user (or a script) changes the subtitle track and so on.
+
 ``idle``
     Idle mode is entered. This happens when playback ended, and the player was
     started with ``--idle`` or ``--force-window``. This mode is implicitly ended
     when the ``start-file`` or ``shutdown`` events happen.
 
+``pause``
+    Playback was paused. This also happens when for example the player is
+    paused on low network cache. Then the event type indicates the pause state
+    (like the property "pause" as opposed to the "core-idle" property), and you
+    might receive multiple ``pause`` events in a row.
+
+``unpause``
+    Playback was unpaused. See above for details.
+
 ``tick``
     Called after a video frame was displayed. This is a hack, and you should
     avoid using it. Use timers instead and maybe watch pausing/unpausing events
@@ -728,32 +664,8 @@ List of events
 ``audio-reconfig``
     Happens on audio output or filter reconfig.
 
-The following events also happen, but are deprecated: ``tracks-changed``,
-``track-switched``, ``pause``, ``unpause``, ``metadata-update``,
-``chapter-change``. Use ``mp.observe_property()`` instead.
+``metadata-update``
+    Metadata (like file tags) was updated.
 
-Extras
-------
-
-This documents experimental features, or features that are "too special" and
-we don't guarantee a stable interface to it.
-
-``mp.add_hook(type, priority, fn)``
-    Add a hook callback for ``type`` (a string identifying a certain kind of
-    hook). These hooks allow the player to call script functions and wait for
-    their result (normally, the Lua scripting interface is asynchronous from
-    the point of view of the player core). ``priority`` is an arbitrary integer
-    that allows ordering among hooks of the same kind. Using the value 50 is
-    recommended as neutral default value. ``fn`` is the function that will be
-    called during execution of the hook.
-
-    Currently existing hooks:
-
-    ``on_load``
-        Called when a file is to be opened, before anything is actually done.
-        For example, you could read and write the ``stream-open-filename``
-        property to redirect an URL to something else (consider support for
-        streaming sites which rarely give the user a direct media URL), or
-        you could set per-file options with by setting the property
-        ``file-local-options/<option name>``. The player will wait until all
-        hooks are run.
+``chapter-change``
+    The current chapter possibly changed.

DOCS/man/mpv.rst

@@ -12,7 +12,8 @@ a media player
 SYNOPSIS
 ========
 
-| **mpv** [options] [file|URL|PLAYLIST|-]
+| **mpv** [options] [file|URL|-]
+| **mpv** [options] --playlist=PLAYLIST
 | **mpv** [options] files
 
 DESCRIPTION
@@ -46,10 +47,6 @@ UP and DOWN
     Seek forward/backward 1 minute. Shift+arrow does a 5 second exact seek (see
     ``--hr-seek``).
 
-Ctrl+LEFT and Ctrl+RIGHT
-    Seek to the previous/next subtitle. Subject to some restrictions and
-    might not work always; see ``sub_seek`` command.
-
 [ and ]
     Decrease/increase current playback speed by 10%.
 
@@ -63,7 +60,7 @@ BACKSPACE
     Go backward/forward in the playlist.
 
 ENTER
-    Go forward in the playlist.
+    Go forward in the playlist, even over the end.
 
 p / SPACE
     Pause (pressing again unpauses).
@@ -129,9 +126,6 @@ j and J
 x and z
     Adjust subtitle delay by +/- 0.1 seconds.
 
-l
-    Set/clear A-B loop points. See ``ab_loop`` command for details.
-
 Ctrl + and Ctrl -
     Adjust audio delay by +/- 0.1 seconds.
 
@@ -362,7 +356,7 @@ You can also write file-specific configuration files. If you wish to have a
 configuration file for a file called 'video.avi', create a file named
 'video.avi.conf' with the file-specific options in it and put it in
 ``~/.config/mpv/``. You can also put the configuration file in the same directory
-as the file to be played. Both require you to set the ``--use-filedir-conf``
+as the file to be played, as long as you give the ``--use-filedir-conf``
 option (either on the command line or in your global config file). If a
 file-specific configuration file is found in the same directory, no
 file-specific configuration is loaded from ``~/.config/mpv``. In addition, the
@@ -451,15 +445,14 @@ 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.
-- Dropped frames, e.g. ``Dropped: 4``. Shows up only if the count is not 0. Can
-  grow if the video framerate is higher than that of the display, or if video
-  rendering is too slow. Also can be incremented on "hiccups" and when the video
-  frame couldn't be displayed on time. (``vo-drop-frame-count`` property.)
-  If the decoder drops frames, the number of decoder-dropped frames is appended
-  to the display as well, e.g.: ``Dropped: 4/34``. This should almost never
-  happen, unless decoder-framedropping is enabled with one of the
-  ``--framedrop`` options, the stream contains errors, or a weird codec is in
-  use. (``drop-frame-count`` property.)
+- Decoder-dropped video frames, e.g. ``SD: 2``. Shows up only if the count is
+  not 0. Normally should never show up, unless ``--framedrop`` is set to enable
+  this mode, and your CPU is too slow. (``drop-frame-count`` property.)
+- VO-dropped video frames, e.g. ``D: 4``. Shows up only if the count is not 0.
+  Can grow if the video framerate is higher than that of the display, or if
+  video rendering is too slow. Also can be incremented on "hiccups" and when
+  the video frame couldn't be displayed on time. (``vo-drop-frame-count``
+  property.)
 - Cache state, e.g. ``Cache:  2s+134KB``. Visible if the stream cache is enabled.
   The first value shows the amount of video buffered in the demuxer in seconds,
   the second value shows *additional* data buffered in the stream cache in
@@ -474,10 +467,6 @@ PROTOCOLS
     be specified. mpv will never attempt to guess whether a filename is
     actually a network address. A protocol prefix is always required.
 
-    Note that not all prefixes are documented here. Undocumented prefixes are
-    either aliases to documented protocols, or are just redirections to
-    protocols implemented and documented in FFmpeg.
-
 ``-``
     Play data from stdin.
 
@@ -567,8 +556,6 @@ PROTOCOLS
 
 .. include:: lua.rst
 
-.. include:: ipc.rst
-
 .. include:: changes.rst
 
 ENVIRONMENT VARIABLES
@@ -724,39 +711,6 @@ override the standard directory ``~/.config/mpv/``.
 Also, the old config location at ``~/.mpv/`` is still read, and if the XDG
 variant does not exist, will still be preferred.
 
-FILES ON WINDOWS
-================
-
-On win32 (if compiled with MinGW, but not Cygwin), the default config file
-locations are different:
-
-``%APPDATA%/mpv/mpv.conf``
-    Preferred mpv config file. This maps to a system and user-specific path,
-    for example ``C:\users\USERNAME\Application Data\mpv\mpv.conf``. You can
-    find the exact path by running ``echo %APPDATA%\mpv\mpv.conf`` in cmd.exe.
-
-``%APPDATA%/mpv/input.conf``
-    key bindings (see `INPUT.CONF`_ section)
-
-``%APPDATA%/mpv/lua/``
-    equivalent of ``~/.config/mpv/lua/`` on Unix.
-
-The environment variable ``$MPV_HOME`` completely overrides these, like on
-UNIX.
-
-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 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
 =====================
 

DOCS/man/options.rst

@@ -45,15 +45,6 @@ Track Selection
 ``--vid=<ID|auto|no>``
     Select video channel. ``auto`` selects the default, ``no`` disables video.
 
-``--ff-aid=<ID|auto|no>``, ``--ff-sid=<ID|auto|no>``, ``--ff-vid=<ID|auto|no>``
-    Select audio/subtitle/video streams by the FFmpeg stream index. The FFmpeg
-    stream index is relatively arbitrary, but useful when interacting with
-    other software using FFmpeg (consider ``ffprobe``).
-
-    Note that with external tracks (added with ``--sub-file`` and similar
-    options) will have duplicate IDs. In that case, the first stream in order
-    is selected.
-
 ``--edition=<ID|auto>``
     (Matroska files only)
     Specify the edition (set of chapters) to use, where 0 is the first. If set
@@ -104,9 +95,6 @@ Playback Control
 ``--speed=<0.01-100>``
     Slow down or speed up playback by the factor given as parameter.
 
-    If ``--audio-pitch-correction`` is used, playing with a speed higher than
-    normal automatically inserts the ``scaletempo`` audio filter.
-
 ``--loop=<N|inf|no>``
     Loops playback ``N`` times. A value of ``1`` plays it one time (default),
     ``2`` two times, etc. ``inf`` means forever. ``no`` is the same as ``1`` and
@@ -128,23 +116,21 @@ Playback Control
     no format is detected, t will be treated as list of files, separated by
     newline characters. Note that XML playlist formats are not supported.)
 
-    You can play playlists directly and without this option, however, this
-    option disables any security mechanisms that might be in place. You may
-    also need this option to load plaintext files as playlist.
-
     .. warning::
 
-        The way mpv uses playlist files via ``--playlist`` is not safe against
-        maliciously constructed files. Such files may trigger harmful actions.
+        The way mpv uses playlist files is not safe against maliciously
+        constructed files. Such files may trigger harmful actions.
         This has been the case for all mpv and MPlayer versions, but
         unfortunately this fact was not well documented earlier, and some people
         have even misguidedly recommended use of ``--playlist`` with untrusted
         sources. Do NOT use ``--playlist`` with random internet sources or files
         you do not trust!
 
-        Playlist can contain entries using other protocols, such as local files,
-        or (most severely), special protocols like ``avdevice://``, which are
-        inherently unsafe.
+        The main problem is that playlists can point to arbitrary network
+        addresses (including local addresses inside of your LAN), and thus
+        can't be considered secure. Playlists also can contain entries using
+        other protocols, such as local files, or (most severely), special
+        protocols like ``avdevice://``, which are inherently unsafe.
 
 ``--chapter-merge-threshold=<number>``
     Threshold for merging almost consecutive ordered chapter parts in
@@ -229,12 +215,6 @@ Playback Control
     between the two option is that this option performs a seek on loop, instead
     of reloading the file.
 
-``--ab-loop-a=<time>``, ``--ab-loop-b=<time>``
-    Set loop points. If playback passes the ``b`` timestamp, it will seek to
-    the ``a`` timestamp. Seeking past the ``b`` point doesn't loop (this is
-    intentional). The loop-points can be adjusted at runtime with the
-    corresponding properties. See also ``ab_loop`` command.
-
 ``--ordered-chapters``, ``--no-ordered-chapters``
     Enabled by default.
     Disable support for Matroska ordered chapters. mpv will not load or
@@ -253,10 +233,6 @@ Playback Control
     Note: a playlist can be as simple as a text file containing filenames
     separated by newlines.
 
-``--chapters-file=<filename>``
-    Load chapters from this file, instead of using the chapter metadata found
-    in the main file.
-
 ``--sstep=<sec>``
     Skip <sec> seconds after every frame.
 
@@ -264,12 +240,6 @@ Playback Control
 
         Without ``--hr-seek``, skipping will snap to keyframes.
 
-``--stop-playback-on-init-failure=<yes|no>``
-    Stop playback if either audio or video fails to initialize. Currently,
-    the default behavior is ``no`` for the command line player, but ``yes``
-    for libmpv. With ``no``, playback will continue in video-only or audio-only
-    mode if one of them fails. This doesn't affect playback of audio-only or
-    video-only files.
 
 Program Behavior
 ----------------
@@ -358,7 +328,7 @@ Program Behavior
 ``--merge-files``
     Pretend that all files passed to mpv are concatenated into a single, big
     file. This uses timeline/EDL support internally. Note that this won't work
-    for ordered chapter files.
+    for ordered chapter files or quvi-resolved URLs (such as YouTube links).
 
     This option is interpreted at program start, and doesn't affect for
     example files or playlists loaded with the ``loadfile`` or ``loadlist``
@@ -422,24 +392,6 @@ Program Behavior
 
         May be dangerous if playing from untrusted media.
 
-``--ytdl``, ``--no-ytdl``
-    Enable the youtube-dl hook-script. It will look at the input URL, and will
-    play the video located on the website. This works with many streaming sites,
-    not just the one the scripts are named after. This requires a recent version
-    of youtube-dl to be installed on the system. (Enabled by default.)
-
-    If the script can't do anything with an URL, it will do nothing.
-
-    (Note: this is the replacement for the now removed libquvi support.)
-
-``--ytdl-format=<best|worst|mp4|webm|...>``
-    Video format/quality that is directly passed to youtube-dl. The possible
-    values are specific to the website and the video, for a given url the
-    available formats can be found with the command
-    ``youtube-dl --list-formats URL``. See youtube-dl's documentation for
-    available aliases. To use experimental DASH support for youtube, use
-    ``bestvideo+bestaudio``.
-    (Default: ``best``)
 
 Video
 -----
@@ -528,18 +480,15 @@ Video
     :vaapi:     requires ``--vo=opengl`` or ``--vo=vaapi`` (Linux with Intel GPUs only)
     :vaapi-copy: copies video back into system RAM (Linux with Intel GPUs only)
     :vda:       requires ``--vo=opengl`` (OS X only)
-    :dxva2-copy: copies video back to system RAM (Windows only) (experimental)
 
     ``auto`` tries to automatically enable hardware decoding using the first
     available method. This still depends what VO you are using. For example,
-    if you are not using ``--vo=vdpau`` or ``--vo=opengl``, vdpau decoding will
-    never be enabled. Also note that if the first found method doesn't actually
-    work, it will always fall back to software decoding, instead of trying the
-    next method (might matter on some Linux systems).
+    if you are not using ``--vo=vdpau``, vdpau decoding will never be enabled.
+    Also note that if the first found method doesn't actually work, it will
+    always fall back to software decoding, instead of trying the next method.
 
-    The ``vaapi-copy`` mode allows you to use vaapi with any VO. Because
-    this copies the decoded video back to system RAM, it's likely less efficient
-    than the ``vaapi`` mode.
+    The ``vaapi-copy`` function allows you to use vaapi with any VO. Because
+    this copies the decoded video back to system RAM, it's quite inefficient.
 
     .. note::
 
@@ -577,7 +526,7 @@ Video
     Disable scaling of the video. If the window is larger than the video,
     black bars are added. Otherwise, the video is cropped. The video still
     can be influenced by the other ``--video-...`` options. (If the
-    ``--video-zoom`` option is set to a value other than ``0``, scaling is
+    ``--video-zoom`` option is set to a value other than ``1``, scaling is
     enabled, but the video isn't automatically scaled to the window size.)
 
     The video and monitor aspects aspect will be ignored. Aspect correction
@@ -714,6 +663,26 @@ Video
         ``mpv --hwdec=vdpau --vo=vdpau --hwdec-codecs=h264,mpeg2video``
             Enable vdpau decoding for h264 and mpeg2 only.
 
+``--quvi-format=<best|default|...>``
+    Video format/quality that is directly passed to libquvi (default: ``best``).
+    This is used when opening links to streaming sites like YouTube. The
+    interpretation of this value is highly specific to the streaming site and
+    the video.
+
+    libquvi 0.4.x:
+
+        The only well-defined values that work on all sites are ``best``
+        (best quality/highest bandwidth, default), and ``default`` (lowest
+        quality).
+
+        The quvi command line tool can be used to find out which formats are
+        supported for a given URL: ``quvi --query-formats URL``.
+
+    libquvi 0.9.x:
+
+        The following explanations are relevant:
+        `<http://quvi.sourceforge.net/r/api/0.9/glossary_termino.html#m_stream_id>`_
+
 ``--vd-lavc-check-hw-profile=<yes|no>``
     Check hardware decoder profile (default: yes). If ``no`` is set, the
     highest profile of the hardware decoder is unconditionally selected, and
@@ -787,30 +756,6 @@ Video
 Audio
 -----
 
-``--audio-pitch-correction=<yes|no>``
-    If this is enabled (default), playing with a speed higher than normal
-    automatically inserts the ``scaletempo`` audio filter. For details, see
-    audio filter section.
-
-``--audio-device=<name>``
-    Use the given audio device. This consists of the audio output name, e.g.
-    ``alsa``, followed by ``/``, followed by the audio output specific device
-    name.
-
-    You can list audio devices with ``--audio-device=help``. This outputs the
-    device name in quotes, followed by a description. The device name is what
-    you have to pass to the ``--audio-device`` option.
-
-    The default value for this option is ``auto``, which tries every audio
-    output in preference order with the default device.
-
-    Note that many AOs have a ``device`` sub-option, which overrides the
-    device selection of this option (but not the audio output selection).
-    Likewise, forcing an AO with ``--ao`` will override the audio output
-    selection of ``--audio-device`` (but not the device selecton).
-
-    Currently not implemented for most AOs.
-
 ``--ao=<driver1[:suboption1[=value]:...],driver2,...[,]>``
     Specify a priority list of audio output drivers to be used. For
     interactive use one would normally specify a single one to use, but in
@@ -939,9 +884,8 @@ Audio
     lists speaker names, which can be used to express arbitrary channel
     layouts (e.g. ``fl-fr-lfe`` is 2.1).
 
-    You can use ``--audio-channels=auto`` to disable this. In this case, the AO
-    use the channel layout as the audio filter chain indicates. (``empty`` is
-    an accepted obsolete for ``auto``.)
+    You can use ``--audio-channels=empty`` to disable this. In this case, the AO
+    use the channel layout as the audio filter chain indicates.
 
     This will also request the channel layout from the decoder. If the decoder
     does not support the layout, it will fall back to its native channel layout.
@@ -978,7 +922,7 @@ Audio
     of the current media, the lavrresample audio filter will be inserted into
     the audio filter layer to compensate for the difference.
 
-``--gapless-audio=<no|yes|weak>``
+``--gapless-audio=<no|yes|weak``
     Try to play consecutive audio files with no silence or disruption at the
     point of file change. Default: ``weak``.
 
@@ -1033,11 +977,6 @@ Audio
         maximum amplification, i.e. amplify by 200%. The default volume (no
         change in volume) will be ``50`` in this case.
 
-``--audio-client-name=<name>``
-    The application name the player reports to the audio API. Can be useful
-    if you want to force a different audio profile (e.g. with PulseAudio),
-    or to set your own application name when using libmpv.
-
 ``--volume-restore-data=<string>``
     Used internally for use by playback resume (e.g. with ``quit_watch_later``).
     Restoring value has to be done carefully, because different AOs as well as
@@ -1122,7 +1061,7 @@ Subtitles
         This affects ASS subtitles as well, and may lead to incorrect subtitle
         rendering. Use with care, or use ``--sub-text-font-size`` instead.
 
-``--sub-scale-with-window=<yes|no>``
+``--sub-scale-with-window=yes|no``
     Make the subtitle font size relative to the window, instead of the video.
     This is useful if you always want the same font size, even if the video
     doesn't covert the window fully, e.g. because screen aspect and window
@@ -1165,7 +1104,7 @@ Subtitles
 
         Using this option may lead to incorrect subtitle rendering.
 
-``--ass-hinting=<none|light|normal|native>``
+``--ass-hinting=none|light|normal|native``
     Set font hinting type. <type> can be:
 
     :none:       no hinting (default)
@@ -1183,7 +1122,7 @@ Subtitles
 ``--ass-line-spacing=<value>``
     Set line spacing value for SSA/ASS renderer.
 
-``--ass-shaper=<simple|complex>``
+``--ass-shaper=simple|complex``
     Set the text layout engine used by libass.
 
     :simple:   uses Fribidi only, fast, doesn't render some languages correctly
@@ -1207,7 +1146,6 @@ Subtitles
     :yes:   Apply all the ``--ass-*`` style override options. Changing the default
             for any of these options can lead to incorrect subtitle rendering
             (default).
-    :signfs: like ``yes``, but apply ``--sub-scale`` only to signs
     :no:    Render subtitles as forced by subtitle scripts.
     :force: Try to force the font style as defined by the ``--sub-text-*``
             options. Requires a modified libass, can break rendering easily.
@@ -1268,6 +1206,15 @@ Subtitles
     a subtitle script with another video file. The ``--ass-style-override``
     option doesn't affect how this option is interpreted.
 
+``--quvi-fetch-subtitles=<yes|no>``
+    Toggles fetching of subtitles from streaming sites with libquvi. Disabled
+    by default, because it's unreliable and slow. Note that when enabled,
+    subtitles will always be fetched, even if subtitles are explicitly
+    disabled with ``--no-sub`` (because you might want to enable subtitles
+    at runtime).
+
+    Supported when using libquvi 0.9.x.
+
 ``--stretch-dvd-subs=<yes|no>``
     Stretch DVD subtitles when playing anamorphic videos for better looking
     fonts on badly mastered DVDs. This switch has no effect when the
@@ -1663,6 +1610,13 @@ Window
     there is a change in video parameters, video stream or file. This used to
     be the default behavior. Currently only affects X11 VOs.
 
+``--fs-missioncontrol``
+    (OS X only)
+    Use OS X Mission Control's fullscreen feature instead of the custom one
+    provided by mpv. This can potentially break a lot of stuff like
+    ``--geometry`` and is disabled by default. On the other hand it provides
+    a more 'OS X-like' user experience.
+
 ``--heartbeat-cmd=<command>``
     Command that is executed every 30 seconds during playback via *system()* -
     i.e. using the shell. The time between the commands can be customized with
@@ -1704,13 +1658,6 @@ Window
     disable the window manager hints that force the window aspect ratio.
     (Ignored in fullscreen mode.)
 
-``--no-keepaspect-window``, ``--keepaspect-window``
-    ``--keepaspect-window`` (the default) will lock the window size to the
-    video aspect. ``--no-keepaspect-window`` disables this behavior, and will
-    instead add black bars if window aspect and video aspect mismatch. Whether
-    this actually works depends on the VO backend.
-    (Ignored in fullscreen mode.)
-
 ``--monitoraspect=<ratio>``
     Set the aspect ratio of your monitor or TV screen. A value of 0 disables a
     previous setting (e.g. in the config file). Overrides the
@@ -1738,25 +1685,14 @@ Window
     to to work this around using ``--heartbeat-cmd`` instead.
 
 ``--wid=<ID>``
-    This tells mpv to attach to an existing window. If a VO is selected that
-    supports this option, it will use that window for video output. mpv will
-    scale the video to the size of this window, and will add black bars to
-    compensate if the aspect ratio of the video is different.
+    (X11 and Windows only)
+    This tells mpv to attach to an existing window. The ID is interpreted as
+    "Window" on X11, and as HWND on Windows. If a VO is selected that supports
+    this option, a new window will be created and the given window will be set
+    as parent. The window will always be resized to cover the parent window
+    fully, and will add black bars to compensate for the video aspect ratio.
 
-    On X11, the ID is interpreted as a ``Window`` on X11. Unlike
-    MPlayer/mplayer2, mpv always creates its own window, and sets the wid
-    window as parent. The window will always be resized to cover the parent
-    window fully. The value ``0`` is interpreted specially, and mpv will
-    draw directly on the root window.
-
-    On win32, the ID is interpreted as ``HWND``. Pass it as value cast to
-    ``intptr_t``. mpv will create its own window, and set the wid window as
-    parent, like with X11.
-
-    On OSX/Cocoa. the ID is interpreted as ``NSView*``. Pass it as value cast
-    to ``intptr_t``. mpv will creates its own sub-view. Because OSX does not
-    support window embedding of foreign processes, this works only with libmpv,
-    and will crash when used from the command line.
+    See also ``--slave-broken``.
 
 ``--no-window-dragging``
     Don't move the window when clicking on it and moving the mouse pointer.
@@ -1764,26 +1700,16 @@ Window
 ``--x11-name``
     Set the window class name for X11-based video output methods.
 
-``--x11-netwm=<yes|no|auto>``
+``--x11-netwm=no``
     (X11 only)
-    Control the use of NetWM protocol features.
-
+    Disable use of the NetWM protocol when switching to or from fullscreen.
     This may or may not help with broken window managers. This provides some
     functionality that was implemented by the now removed ``--fstype`` option.
     Actually, it is not known to the developers to which degree this option
     was needed, so feedback is welcome.
 
-    Specifically, ``yes`` will force use of NetWM fullscreen support, even if
-    not advertised by the WM. This can be useful for WMs that are broken on
-    purpose, like XMonad. (XMonad supposedly doesn't advertise fullscreen
-    support, because Flash uses it. Apparently, applications which want to
-    use fullscreen anyway are supposed to either ignore the NetWM support hints,
-    or provide a workaround. Shame on XMonad for deliberately breaking X
-    protocols (as if X isn't bad enough already).
-
-    By default, NetWM support is autodetected (``auto``).
-
-    This option might be removed in the future.
+    By default, NetWM support is autodetected, and using this option forces
+    autodetection to fail.
 
 
 Disc Devices
@@ -1843,7 +1769,7 @@ Disc Devices
     Add ``<value>`` sectors to the values reported when addressing tracks.
     May be negative.
 
-``--cdda-skip=<yes|no>``
+``--cdda-skip=<es|no``
     (Never) accept imperfect data reconstruction.
 
 ``--dvd-speed=<speed>``
@@ -2062,22 +1988,6 @@ Demuxer
     and hr-seeks, and this option changes behavior with relative or imprecise
     seeks only.
 
-    You can use the ``--demuxer-mkv-subtitle-preroll-secs`` option to specify
-    how mach data the demuxer should pre-read at most in order to find subtitle
-    packets that may overlap. Setting this to 0 will effectively disable this
-    preroll mechanism. Setting a very large value can make seeking very slow,
-    and an extremely large value would completely reread the entire file from
-    start to seek target on every seek - seeking can become slower towards the
-    end of the file. The details are messy, and the value is actually rounded
-    down to the cluster with the previous video keyframe.
-
-    Some files, especially files muxed with newer mkvmerge versions, have
-    information embedded that can be used to determine what subtitle packets
-    overlap with a seek target. In these cases, mpv will reduce the amount
-    of data read to a minimum. (Although it will still read *all* data between
-    the cluster that contains the first wanted subtitle packet, and the seek
-    target.)
-
     See also ``--hr-seek-demuxer-offset`` option. This option can achieve a
     similar effect, but only if hr-seek is active. It works with any demuxer,
     but makes seeking much slower, as it has to decode audio and video data
@@ -2085,16 +1995,6 @@ Demuxer
 
     ``--mkv-subtitle-preroll`` is a deprecated alias.
 
-``--demuxer-mkv-subtitle-preroll-secs=<value>``
-    See ``--demuxer-mkv-subtitle-preroll``.
-
-``--demuxer-mkv-probe-video-duration``
-    When opening the file, seek to the end of it, and check what timestamp the
-    last video packet has, and report that as file duration. This is strictly
-    for compatibility with Haali only. In this mode, it's possible that opening
-    will be slower (especially when playing over http), or that behavior with
-    broken files is much worse. So don't use this option.
-
 ``--demuxer-rawaudio-channels=<value>``
     Number of channels (or channel layout) if ``--demuxer=rawaudio`` is used
     (default: stereo).
@@ -2141,7 +2041,7 @@ Demuxer
     playback, but on the other hand can add delays to seeking or track
     switching.
 
-``--demuxer-readahead-secs=<seconds>``
+``--demuxer-readahead-secs=N``
     If ``--demuxer-thread`` is enabled, this controls how much the demuxer
     should buffer ahead in seconds (default: 0.2). As long as no packet has
     a timestamp difference higher than the readahead amount relative to the
@@ -2150,7 +2050,7 @@ Demuxer
     (This value tends to be fuzzy, because many file formats don't store linear
     timestamps.)
 
-``--demuxer-readahead-packets=<packets>``
+``--demuxer-readahead-packets=N``
     If ``--demuxer-thread`` is enabled, this controls how much the demuxer
     should buffer ahead. As long as the number of packets in the packet queue
     doesn't exceed ``--demuxer-readahead-packets``, and the total number of
@@ -2162,7 +2062,7 @@ Demuxer
 
     See ``--list-options`` for defaults and value range.
 
-``--demuxer-readahead-bytes=<bytes>``
+``--demuxer-readahead-bytes=N``
     See ``--demuxer-readahead-packets``.
 
 
@@ -2213,11 +2113,7 @@ Input
     like any other binding). See `INPUT.CONF`_.
 
 ``--input-file=<filename>``
-    Read commands from the given file. Mostly useful with a FIFO. Since
-    mpv 0.7.0 also understands JSON commands (see `JSON IPC`_), but you can't
-    get replies or events. Use ``--input-unix-socket`` for something
-    bi-directional. On MS Windows, JSON commands are not available.
-
+    Read commands from the given file. Mostly useful with a FIFO.
     See also ``--slave-broken``.
 
     .. note::
@@ -2234,16 +2130,8 @@ Input
     or intend to read from stdin later on via the loadfile or loadlist slave
     commands.
 
-``--input-unix-socket=<filename>``
-    Enable the IPC support and create the listening socket at the given path.
-
-    See `JSON IPC`_ for details.
-
-    Not available on MS Windows.
-
-``--input-appleremote=<yes|no>``
-    (OS X only)
-    Enable/disable Apple Remote support. Enabled by default (except for libmpv).
+``--input-appleremote``, ``--no-input-appleremote``
+    Enable/disable AppleIR remote support. Enabled by default.
 
 ``--input-cursor``, ``--no-input-cursor``
     Permit mpv to receive pointer events reported by the video output
@@ -2263,18 +2151,16 @@ Input
     (LIRC only)
     Specifies a configuration file for LIRC (default: ``~/.lircrc``).
 
-``--input-media-keys=<yes|no>``
-    (OS X only)
-    Enable/disable media keys support. Enabled by default (except for libmpv).
+``--input-media-keys``, ``--no-input-media-keys``
+    OS X only: Enabled by default. Enables/disable media keys support.
 
 ``--input-right-alt-gr``, ``--no-input-right-alt-gr``
     (Cocoa and Windows only)
     Use the right Alt key as Alt Gr to produce special characters. If disabled,
     count the right Alt as an Alt modifier key. Enabled by default.
 
-``--input-vo-keyboard=<yes|no>``
-    Disable all keyboard input on for VOs which can't participate in proper
-    keyboard input dispatching. This currently affects X11. Generally useful for
+``--input-x11-keyboard=<yes|no>``
+    Disable all keyboard input on the X11 VO window. Generally useful for
     embedding only.
 
     On X11, a sub-window with input enabled grabs all keyboard input as long
@@ -2292,13 +2178,6 @@ Input
     As a workaround, this option is disabled by default in libmpv. (Note that
     ``input-default-bindings`` is disabled by default in libmpv as well.)
 
-    (This option was renamed from ``--input-x11-keyboard``.)
-
-``--input-app-events=<yes|no>``
-    (OS X only)
-    Enable/disable application wide keyboard events so that keyboard shortcuts
-    can be processed without a window. Enabled by default (except for libmpv).
-
 OSD
 ---
 
@@ -2337,7 +2216,7 @@ OSD
     height: if the window height is larger or smaller than 720, the actual size
     of the text increases or decreases as well.
 
-    Default: 55.
+    Default: 45.
 
 ``--osd-msg1=<string>``
     Show this string as message on OSD with OSD level 1 (visible by default).
@@ -2411,7 +2290,7 @@ OSD
     Size of the OSD/sub font border in scaled pixels (see ``--osd-font-size``
     for details). A value of 0 disables borders.
 
-    Default: 3.
+    Default: 2.5.
 
 ``--osd-color=<color>``, ``--sub-text-color=<color>``
     Specify the color used for OSD/unstyled text subtitles.
@@ -2469,12 +2348,12 @@ OSD
     for unstyled text subtitles. If you just want to raise the vertical
     subtitle position, use ``--sub-pos``.
 
-    Default: 22.
+    Default: 10.
 
 ``--osd-scale=<factor>``
     OSD font size multiplier, multiplied with ``--osd-font-size`` value.
 
-``--osd-scale-by-window=<yes|no>``
+``--osd-scale-by-window=yes|no``
     Whether to scale the OSD with the window size (default: yes). If this is
     disabled, ``--osd-font-size`` and other OSD options that use scaled pixels
     are always in actual pixels. The effect is that changing the window size
@@ -2496,16 +2375,6 @@ OSD
 
     Default: 0.
 
-``--use-text-osd=<yes|no>``
-    Disable text OSD rendering completely. (This includes the complete OSC as
-    well.) This is mostly useful for avoiding loading fontconfig in situations
-    where fontconfig does not behave well, and OSD is unused - this could for
-    example allow GUI programs using libmpv to workaround fontconfig issues.
-
-    Note that selecting subtitles of any kind still initializes fontconfig.
-
-    Default: ``no``.
-
 
 Screenshot
 ----------

DOCS/man/osc.rst

@@ -229,10 +229,6 @@ Configurable Options
     | Default: yes
     | Display a tooltip over the seekbar indicating time at mouse position.
 
-``seekbarstyle``
-    | Default: slider
-    | Sets the style of the seekbar, slider (diamond marker) or bar (fill)
-
 Script Commands
 ~~~~~~~~~~~~~~~
 

DOCS/man/vf.rst

@@ -910,31 +910,6 @@ Available filters are:
         making it higher than the number of cores can actually make it
         slower.
 
-``vapoursynth-lazy``
-    The same as ``vapoursynth``, but doesn't load Python scripts. Instead, a
-    custom backend using Lua and the raw VapourSynth API is used. The syntax
-    is completely different, and absolutely no conveniencve features are
-    provided. There's no type checking either, and you can trigger crashes.
-
-    .. admonition:: Example:
-
-        ::
-
-            video_out = invoke("morpho", "Open", {clip = video_in})
-
-    The special variable ``video_in`` is the mpv video source, while the
-    special variable ``video_out`` is used to read video from. The 1st argument
-    is the plugin (queried with ``getPluginByNs``), the 2nd is the filter name,
-    and the 3rd argument is a table with the arguments. Positional arguments
-    are not supported. The types must match exactly. Since Lua is terrible and
-    can't distinguish integers and floats, integer arguments must be prefixed
-    with ``i_``, in which case the prefix is removed and the argument is cast
-    to an integer. Should the argument's name start with ``i_``, you're out of
-    luck.
-
-    Clips (VSNodeRef) are passed as light userdata, so trying to pass any
-    other userdata type will result in hard crashes.
-
 ``vavpp``
     VA-AP-API video post processing. Works with ``--vo=vaapi`` and ``--vo=opengl``
     only. Currently deinterlaces. This filter is automatically inserted if

DOCS/man/vo.rst

@@ -233,29 +233,18 @@ Available video output drivers are:
         Always force textures to power of 2, even if the device reports
         non-power-of-2 texture sizes as supported.
 
-    ``texture-memory=<mode>``
+    ``texture-memory=N``
         Only affects operation with shaders/texturing enabled, and (E)OSD.
-        Possible values:
+        Values for N:
 
-        ``default`` (default)
-            Use ``D3DPOOL_DEFAULT``, with a ``D3DPOOL_SYSTEMMEM`` texture for
-            locking. If the driver supports ``D3DDEVCAPS_TEXTURESYSTEMMEMORY``,
-            ``D3DPOOL_SYSTEMMEM`` is used directly.
-
-        ``default-pool``
-            Use ``D3DPOOL_DEFAULT``. (Like ``default``, but never use a
-            shadow-texture.)
-
-        ``default-pool-shadow``
-            Use ``D3DPOOL_DEFAULT``, with a ``D3DPOOL_SYSTEMMEM`` texture for
-            locking. (Like ``default``, but always force the shadow-texture.)
-
-        ``managed``
-            Use ``D3DPOOL_MANAGED``.
-
-        ``scratch``
-            Use ``D3DPOOL_SCRATCH``, with a ``D3DPOOL_SYSTEMMEM`` texture for
-            locking.
+            0
+                default, will often use an additional shadow texture + copy
+            1
+                use ``D3DPOOL_MANAGED``
+            2
+                use ``D3DPOOL_DEFAULT``
+            3
+                use ``D3DPOOL_SYSTEMMEM``, but without shadow texture
 
     ``swap-discard``
         Use ``D3DSWAPEFFECT_DISCARD``, which might be faster.
@@ -449,17 +438,12 @@ Available video output drivers are:
         this options will make rendering a single operation.
         Note that chroma scalers are always done as 1-pass filters.
 
-    ``cscale=<filter>``
+    ``cscale=<n>``
         As ``lscale``, but for chroma (2x slower with little visible effect).
         Note that with some scaling filters, upscaling is always done in
         RGB. If chroma is not subsampled, this option is ignored, and the
         luma scaler is used instead. Setting this option is often useless.
 
-    ``lscale-down=<filter>``, ``cscale-down=<filter>``
-        Like ``lscale`` and ``cscale``, but apply these filters on downscaling
-        instead. If these options are unset, the filter implied by ``lscale``
-        (and ``cscale``, respectively) will be applied.
-
     ``cparam1``, ``cparam2``, ``cradius``
         Set filter parameters and radius for ``cscale``.
 

README.md

@@ -3,8 +3,6 @@ mpv
 
 [![Build Status](https://api.travis-ci.org/mpv-player/mpv.png)](https://travis-ci.org/mpv-player/mpv)
 
-[![Coverity Scan Build Status](https://scan.coverity.com/projects/1232/badge.svg)](https://scan.coverity.com/projects/1232)
-
 Overview
 --------
 
@@ -50,7 +48,7 @@ Essential dependencies (incomplete list):
 - FFmpeg libraries (libavutil libavcodec libavformat libswscale libavresample)
   At least FFmpeg 2.1.4 or Libav 10 is required.
 - libjpeg (for screenshots)
-- Lua (strictly speaking optional, but required for the OSC pseudo-GUI)
+- libquvi if you want to play Youtube videos directly
 
 Libass dependencies:
 
@@ -119,17 +117,17 @@ make code review easier for both parties later on.
 Contact
 -------
 
-Most activity happens on the IRC channel and the github issue tracker. The
-mailing lists are mostly unused.
+These forms of contact are meant to ask questions about mpv usage, give
+feedback on mpv and discuss it's development.
 
- - **Github issue tracker**: [issue tracker][issue-tracker] (report bugs here)
- - **User IRC Channel**: `#mpv-player` on `irc.freenode.net`
- - **Developer IRC Channel**: `#mpv-player-dev` on `irc.freenode.net`
+If possible, please avoid posting bugs here and use the [issue tracker][issue-tracker]
+instead.
+
+ - **Users IRC Channel**: `#mpv-player` 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.
+To contact the `mpv` team in private write to `mpv-team@googlegroups.com`.
 
 [changes]: https://github.com/mpv-player/mpv/blob/master/DOCS/man/changes.rst
 [mpv-build]: https://github.com/mpv-player/mpv-build

RELEASE_NOTES

@@ -1,135 +1,102 @@
-Release 0.7.2
+Release 0.6.1
 =============
 
 Changes
 -------
 
-- Give precedence to the DVD menu navigation keyboard bindings so that user
-  defined LEFT/RIGHT/... bindings don't break DVD menu navigation.
-- Try to fallback to the "default" device if the selected device is busy in the
-  alsa AO.
-- Don't create Dock icon for audio only files on OS X.
-- Save screenshots to desktop when using the app bundle on OS X.
-- Restore ab-loop settings with playback resume.
-- Bump required youtube-dl version to 2014.11.26 and enable the ytdl_hook Lua
-  script by default (now playing videos from YouTube and the like will work
-  out of the box without any configuration change needed).
+- Don't show message on "osd" command by default (this essentially reverts a
+  change introduced in the last release).
+- Add an alternative input configuration that reverts the key binding changes
+  introduced by previous releases (see etc/restore-old-bindings.conf in the
+  sources).
+- Automatically install example input configuration files.
 
 Bug fixes
 ---------
 
-- Don't signal an error if --stream-dump is used.
-- Fix removing key bindings from Lua scripts.
-- Reject channel descriptions with too many channels in the coreaudio AO.
-- Don't async redraw when waiting for VO redraw on OS X (this fixes the very
-  annoying glitch where the black bars disappear for a single frame when going
-  fullscreen).
-- Fix mono playback with the also AO.
-- Don't crash if framebuffers are not available in the opengl VO.
-
-New features
-------------
-
-- Try to handle multi-arc videos in the ytdl_hook Lua script.
+- Make stream language work with Bluray as well (the previous release
+  introduced this for DVDs, but it didn't work for Blurays).
+- Don't try to create surfaces of size 0 in the VDPAU vo.
+- Don't print audio/video init failure message twice.
+- Fix playback on the root window on X11 (i.e. when using --wid=0).
+- Fix the autodetection of the input format in the vf_stereo3d filter when mpv
+  is built against libav (this makes --video-stereo-mode work with libav as
+  well).
+- Fix compilation with PulseAudio 1.0.
+- Fix compilation with clang.
+- Try harder to decode cover art picture only once.
+- Fix cursor autohide (--cursor-autohide) with audio-only files on file change
+  (e.g. when skipping to the next playlist file).
 
 This listing is not complete. There are many more bug fixes and changes. The
-complete change log can be viewed by running ``git log 47ec404..c7d6b21`` in
+complete change log can be viewed by running ``git log 061f509..2ffdc0d`` in
 the git repository.
 
-Release 0.7.1
+Release 0.6.0
 =============
 
 Changes
 -------
 
-- Don't show the volume neutral marker on the OSD if softvol is disabled.
-- Don't select a subtitle track when executing the sub_add input command in
-  "auto" mode.
-
-Bug fixes
----------
-
-- Fix busy loop when seeking while paused (this fixes a problem with pulseaudio
-  that caused mpv and the pulseaudio daemon to use 100% CPU).
-- Fix Lua function utils.subprocess() in Windows versions older than Vista.
-- Avoid creating a window bigger than the screen on Windows.
-- Don't ignore the last line in m3u playlists.
-- Don't crash if a codec could not be opened.
-- Dynamically allocate audio channel map entries (this should fix a crash in
-  the alsa and coreaudio AOs with audio devices that support more than 20
-  channel maps).
-- Ignore the "srgb" option in the opengl VO if hardware decoding is enabled.
-- Linearize non-RGB sRGB files correctly (eg. JPEG).
-- Fix opening reference URLs (.file/id=) on OS X.
-
-This listing is not complete. There are many more bug fixes and changes. The
-complete change log can be viewed by running ``git log 8d8b36d..6583ad6`` in
-the git repository.
-
-Release 0.7.0
-=============
-
-Changes
--------
-
-- Buffer partial log messages in the client API (the client API will now only
-  pass full log messages to clients).
-- Remove ncurses/terminfo/termcap support (it was disabled by default and
-  replaced by new code since v0.6.0).
-- Enable cdda:// support by default again (it was disabled since v0.6.0).
-- Cascade-load input.conf (if there are several input.confs in the set of valid
-  config paths, load them all).
-- Draw the OSD twice in 3D mode (this fixes subtitles display in 3D mode).
-- Make wasapi the default AO on Windows again since many of its problems have
-  been solved.
-- Use "site-functions" subdir to install the zsh completion script instead of
-  the Debian-specific "vendor-completions" (also provide the --zshdir waf
-  configure option for changing this value).
-- Improve synchronization between the Cocoa GUI and the player (this fixes some
-  long standing deadlock issues on Mac OS X).
-- Remove --fs-missioncontrol option (only relevant to Mac OS X).
+- Map ESC to exiting fullscreen.
+- Run the vo inside its own thread.
+- Enable the demuxer thread by default.
+- Prefer the opengl vo over the vdpau one.
+- Complete profile names in the zsh completion script.
+- Raise dsound ao buffer size to 200ms and make it configurable.
+- Bind ctrl+c to quit, so that it works both in the terminal and the player
+  window.
+- Automatically load playlists without needing --load-unsafe-playlists or
+  --playlist. Unsafe URLs are still rejected (e.g. remote playlists can't load
+  local files).
+- Remove quvi-format property (which never really worked).
+- Wait until all Lua scripts are loaded on start.
+- Use separate threads for input sources (terminal, lirc, joystick) instead of
+  using a central I/O loop.
+- Complete values for options that take file names in the zsh completion script.
+- Use libwaio for pipe input on Windows (if you want --input-file to work on
+  Windows you need libwaio as an additional dependency).
+- Remove the --volstep option (users should change the argument to the "add"
+  function in input.conf instead).
+- Always show the OSD message when changing volume, gamma, panscan, etc along
+  with the OSD bar.
+- When changing a property always show it on the OSD (except for "pause" and
+  "fullscreen").
+- Prefer the libavcodec decoder over libmpg123 for mp3 decoding (ffmpeg/libav
+  support for gapless audio works better for mpv's use-case).
+- Allow passing number of loops to --loop-file.
+- Change --keep-open option semantics to not terminate the player when playing
+  or seeking past the end of the file and there's no next file to be played.
+- Disable the sndio ao by default and prefer other outputs even if it's enabled.
 
 New features
 ------------
 
-- Enable pitch correction by default when playing at higher speeds (this can be
-  controlled with the --audio-pitch-correction option).
-- Open stream and demuxer asynchronously (this should avoid having the player
-  get blocked on network streams).
-- Add cache-buffering-state property for querying the cache fill status until
-  the player unpauses.
-- Add support for listing and selecting the audio device (note that it is not
-  implemented for all AOs, see the --audio-device option for more information).
-- Add support for a JSON-based IPC mechanism (note that this is not currently
-  supported on Windows, see the JSON IPC section in the manpage for more
-  information).
-- Add Lua utility function for starting processes (see utils.subprocess() in the
-  manpage).
-- Add Lua utility function for parsing JSON (see utils.parse_json() in the
-  manpage).
-- Add field-dominance property (see --field-dominance option).
-- Add video-rotate property (see --video-rotate option).
-- Add playback-abort property for querying whether playback is stopped or is to
-  be stopped.
-- Add cursor-autohide property (see --cursor-autohide option).
-- Add vo-configured property for querying whether a window is created.
-- Add support for dxva2 hardware acceleration on Windows.
-- Drop libquvi support (this has been replaced by a built-in Lua script that
-  invokes the youtube-dl tool, which needs to be installed, see the --ytdl
-  option).
-- Add support for loading chapters from an external file (see the
-  --chapters-file option).
-- Add window-minimized property for querying whether the window is minimized
-  (works for X11 only).
-- Make it possible to configure the OSC seekbar style (see the "seekbarstyle"
-  OSC option).
-- Add support for libmpv on Mac OS X (it used to be broken, now it's fully
-  functional and also provides support for embedding the mpv window inside a
-  Cocoa/Qt application).
-- Try to use the audio channel map reported by ALSA in the alsa AO.
-- Add option to disable text OSD rendering completely (useful for working around
-  certain fontconfig issues, see the --use-text-osd option).
+- Add new framedrop mode that limits the frequency of new video frames displayed
+  based on the display framerate (this is enabled by default, the old
+  decoder-based mode can be enabled with --framedrop=decoder).
+- Add --demuxer-readahead-secs option to control the readahead buffer using a
+  duration value.
+- Add estimated-frame-count and estimated-frame-number properties.
+- Add --cache-secs option which overrides the value of --demuxer-readahead-secs
+  if the stream cache is active.
+- Add demuxer-cache-duration and demuxer-cache-idle properties.
+- Make --cache-file create a temporary file if the string "TMP" is passed
+  instead of a file name.
+- Add support for Matroska 3D (if a file is marked as 3D using the StereoMode
+  element, an automatic conversion filter is added).
+- Add --osd-playing-msg option to show a message on the OSD when the playback
+  starts.
+- Add --media-title option to force the content of the media-title property.
+- Add --osd-msg1, --osd-msg2 and --osd-msg3 options to override the OSD message
+  for the corresponding OSD level (before only level 3 could be overridden with
+  --osd-status-message).
+- Add support for digital passthrough in the pulse ao.
+- Add support for opening DVD and Bluray directories directly.
+- Add --input-x11-keyboard option to control whether keyboard input should be
+  enabled on the X11 window ot not (this is disabled by default when using the
+  libmpv library).
 
 This listing is not complete. There are many more bug fixes and changes. The
-complete change log can be viewed by running ``git log 7759c18..9479daa`` in
+complete change log can be viewed by running ``git log a1be3cf..61ebe61`` in
 the git repository.

TOOLS/idet.sh

@@ -35,7 +35,11 @@ judge()
     bff=0
     progressive=0
     undetermined=0
-    while read -r _ _ _ _ _ _ tff1 _ bff1 _ progressive1 _ undetermined1 _; do
+    while IFS= read -r out; do
+        tff1=${out##* TFF:}; tff1=${tff1%% *}
+        bff1=${out##* BFF:}; bff1=${bff1%% *}
+        progressive1=${out##* Progressive:}; progressive1=${progressive1%% *}
+        undetermined1=${out##* Undetermined:}; undetermined1=${undetermined1%% *}
         case "$tff1$bff1$progressive1$undetermined1" in
             *[!0-9]*)
                 printf >&2 'ERROR: Unrecognized idet output: %s\n' "$out"

TOOLS/lib/Parse/Matroska/Definitions.pm

@@ -232,7 +232,6 @@ sub define_matroska {
                     elem('Block',           'a1', 'binary'),
                     elem('BlockDuration',   '9b', 'uint'),
                     elem('ReferenceBlock*', 'fb', 'sint'),
-                    elem('DiscardPadding',  '75A2', 'sint'),
                 }),
                 elem('SimpleBlock*', 'a3', 'binary'),
             }),
@@ -257,8 +256,6 @@ sub define_matroska {
                     elem('CodecPrivate',         '63a2', 'binary'),
                     elem('CodecName',          '258688', 'str'),
                     elem('CodecDecodeAll',         'aa', 'uint'),
-                    elem('CodecDelay',           '56AA', 'uint'),
-                    elem('SeekPreRoll',          '56BB', 'uint'),
                     elem('Video', 'e0', {
                         elem('FlagInterlaced',  '9a', 'uint'),
                         elem('PixelWidth',      'b0', 'uint'),
@@ -296,8 +293,6 @@ sub define_matroska {
                     elem('CueTrackPositions*', 'b7', {
                         elem('CueTrack',           'f7', 'uint'),
                         elem('CueClusterPosition', 'f1', 'uint'),
-                        elem('CueRelativePosition','f0', 'uint'),
-                        elem('CueDuration',        'b2', 'uint'),
                     }),
                 }),
             }),

TOOLS/lua/README.md

@@ -8,5 +8,12 @@ adding the option
 
 to mpv's command line.
 
-Where appropriate, they may also be placed in ~/.config/mpv/lua from where
-they will be automatically loaded when mpv starts.
+Unless otherwise specified, they are also suitable for inclusion in
+the `~/.mpv/lua` directory where they will be loaded every time mpv
+starts, obviating the need to load them with the above `--lua=...`
+argument. This is acceptable as they do only basic argument parsing
+and key-binding registration, until those bound keys are actually
+pressed.  They should therefore not interfere with normal playback
+(unless you have a conflicting user-defined key-binding, in which
+case, you may want to modify either the `mp.add_key_binding()` calls
+in the scripts, or your keybinding).

TOOLS/lua/autocrop.lua

@@ -42,11 +42,11 @@ function del_filter_if_present(label)
     -- error if the filter doesn't exist
     local vfs = mp.get_property_native("vf")
     for i,vf in pairs(vfs) do
-        if vf["label"] == label then
-            table.remove(vfs, i)
-            mp.set_property_native("vf", vfs)
-            return true
-        end
+	if vf["label"] == label then
+	    table.remove(vfs, i)
+	    mp.set_property_native("vf", vfs)
+	    return true
+	end
     end
     return false
 end
@@ -65,10 +65,10 @@ function autocrop_start()
 
     -- insert the cropdetect filter
     ret=mp.command(
-        string.format(
-            'vf add @%s:lavfi=graph="cropdetect=limit=24:round=2:reset=0"',
-            cropdetect_label
-        )
+	string.format(
+	    'vf add @%s:lavfi=graph="cropdetect=limit=24:round=2:reset=0"',
+	    cropdetect_label
+	)
     )
     -- wait to gather data
     timer=mp.add_timeout(detect_seconds, do_crop)
@@ -77,15 +77,15 @@ end
 function do_crop()
     -- get the metadata
     local cropdetect_metadata = mp.get_property_native(
-        string.format("vf-metadata/%s", cropdetect_label)
+	string.format("vf-metadata/%s", cropdetect_label)
     )
     -- use it to crop if its valid
     if cropdetect_metadata then
-        if cropdetect_metadata["lavfi.cropdetect.w"]
-            and cropdetect_metadata["lavfi.cropdetect.h"]
-            and cropdetect_metadata["lavfi.cropdetect.x"]
-            and cropdetect_metadata["lavfi.cropdetect.y"]
-        then
+	if cropdetect_metadata["lavfi.cropdetect.w"]
+	    and cropdetect_metadata["lavfi.cropdetect.h"]
+	    and cropdetect_metadata["lavfi.cropdetect.x"]
+	    and cropdetect_metadata["lavfi.cropdetect.y"]
+	then
             mp.command(string.format("vf add @%s:crop=%s:%s:%s:%s",
                                      crop_label,
                                      cropdetect_metadata["lavfi.cropdetect.w"],
@@ -94,9 +94,9 @@ function do_crop()
                                      cropdetect_metadata["lavfi.cropdetect.y"]))
         else
             mp.msg.error(
-                "Got empty crop data. You might need to increase detect_seconds."
-            )
-        end
+		"Got empty crop data. You might need to increase detect_seconds."
+	    )
+	end
     else
         mp.msg.error(
             "No crop data. Was the cropdetect filter successfully inserted?"

TOOLS/lua/autodeint.lua

@@ -1,164 +0,0 @@
--- This script uses the lavfi idet filter to automatically insert the
--- appropriate deinterlacing filter based on a short section of the
--- currently playing video.
---
--- It registers the key-binding ctrl+d, which when pressed, inserts
--- the filter vf=lavfi=idet. After 4 second, it examines the results
--- to determine whether the content progressive or interlaced and the
--- interlacing field dominance. It immediately sets the latter within
--- mpv.
---
--- 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.
---
--- The default detection time may be overridden by adding
---
--- --lua-opts=autodeint.detect_seconds=<number of seconds>
---
--- to mpv's arguments. This may be desirable to allow idet more
--- time to collect data.
---
--- To see counts of the various types of frames for each detection phase,
--- the verbosity can be increased with
---
--- --msg-level autodeint=v
---
--- This script requires a recent version of ffmpeg for which the idet
--- filter adds the required metadata.
-
-require "mp.msg"
-
-script_name = mp.get_script_name()
-detect_label = string.format("%s-detect", script_name)
-pullup_label = string.format("%s", script_name)
-
--- number of seconds to gather cropdetect data
-detect_seconds = tonumber(mp.get_opt(string.format("%s.detect_seconds", script_name)))
-if not detect_seconds then
-    detect_seconds = 4
-end
-
-function del_filter_if_present(label)
-    -- necessary because mp.command('vf del @label:filter') raises an
-    -- error if the filter doesn't exist
-    local vfs = mp.get_property_native("vf")
-
-    for i,vf in pairs(vfs) do
-        if vf["label"] == label then
-            table.remove(vfs, i)
-            mp.set_property_native("vf", vfs)
-            return true
-        end
-    end
-    return false
-end
-
-function start_detect()
-    -- exit if detection is already in progress
-    if timer then
-        mp.msg.warn("already detecting!")
-        return
-    end
-
-    mp.set_property("deinterlace","no")
-    del_filter_if_present(pullup_label)
-
-    percent_pos = mp.get_property("percent-pos")
-
-    -- insert the detection filter
-    local cmd = string.format('vf add @%s:lavfi=graph="idet"', detect_label)
-    if not mp.command(cmd) then
-        mp.msg.error("failed to insert detection filter")
-        return
-    end
-
-    -- wait to gather data
-    timer = mp.add_timeout(detect_seconds, judge_field_dominance)
-end
-
-function stop_detect()
-    del_filter_if_present(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()
-    -- get the metadata
-    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"])
-    num_undetermined = tonumber(result["lavfi.idet.multiple.undetermined"])
-    num_interlaced   = num_tff + num_bff
-    num_determined   = num_interlaced + num_progressive
-
-    mp.msg.verbose("progressive    = "..num_progressive)
-    mp.msg.verbose("interlaced-tff = "..num_tff)
-    mp.msg.verbose("interlaced-bff = "..num_bff) 
-    mp.msg.verbose("undetermined   = "..num_undetermined)
-
-    if num_determined < num_undetermined then
-        mp.msg.warn("majority undetermined frames")
-    end
-    if num_progressive > 20*num_interlaced then
-        return progressive
-    elseif num_tff > 10*num_bff then
-        return interlaced_tff
-    elseif num_bff > 10*num_tff then
-        return interlaced_bff
-    else
-        return interlaced
-    end
-end
-
-function judge_field_dominance()
-    verdict = judge()
-    if verdict == progressive then
-        stop_detect()
-        mp.msg.info("progressive: doing nothing")
-        return
-    elseif verdict == interlaced_tff then
-        mp.set_property("field-dominance", "top")
-    elseif verdict == interlaced_bff then
-        mp.set_property("field-dominance", "bottom")
-    elseif verdict == interlaced then
-        mp.set_property("field-dominance", "auto")
-    end
-
-    local cmd = string.format("vf pre @%s:pullup", pullup_label)
-    if not mp.command(cmd) then
-        mp.msg.error("failed to insert pullup filter")
-        stop_detect()
-        return
-    end
-
-    -- redo the detection with the pullup filter to see if it's telecined
-    mp.set_property("percent-pos", percent_pos)
-    timer = mp.add_timeout(detect_seconds, judge_ivtc)
-end
-
-function judge_ivtc()
-    verdict = judge()
-
-    if verdict == progressive then
-        mp.msg.info(string.format("telecinied with %s field dominance: using pullup", mp.get_property("field-dominance")))
-        stop_detect()
-    else
-        mp.msg.info(string.format("interlaced with %s field dominance: setting deinterlace property", mp.get_property("field-dominance")))
-        del_filter_if_present(pullup_label)
-        mp.set_property("deinterlace","yes")
-        stop_detect()
-    end
-
-    return
-end
-
-mp.add_key_binding("ctrl+d", script_name, start_detect)

TOOLS/lua/autoload.lua

@@ -26,9 +26,6 @@ function find_and_add_entries()
     end
     local files = mputils.readdir(dir, "files")
     table.sort(files)
-    if dir == "." then
-        dir = ""
-    end
     local pl = mp.get_property_native("playlist", {})
     local pl_current = mp.get_property_number("playlist-pos", 0) + 1
     -- Find the current pl entry (dir+"/"+filename) in the sorted dir list
@@ -43,28 +40,27 @@ function find_and_add_entries()
         return
     end
     local append = {[-1] = {}, [1] = {}}
-    for direction = -1, 1, 2 do -- 2 iterations, with direction = -1 and +1
+    for dir = -1, 1, 2 do -- 2 iterations, with dir = -1 and +1
         for i = 1, MAXENTRIES do
-            local file = files[current + i * direction]
-            local pl_e = pl[pl_current + i * direction]
+            local file = files[current + i * dir]
+            local pl_e = pl[pl_current + i * dir]
             if file == nil or file[1] == "." then
                 break
             end
-            local filepath = dir .. file
             if pl_e then
                 -- If there's a playlist entry, and it's the same file, stop.
-                if pl_e.filename == filepath then
+                if pl_e.filename == file then
                     break
                 end
             end
-            if direction == -1 then
+            if dir == -1 then
                 if pl_current == 1 then -- never add additional entries in the middle
                     mp.msg.info("Prepending " .. file)
-                    table.insert(append[-1], 1, filepath)
+                    table.insert(append[-1], 1, file)
                 end
             else
                 mp.msg.info("Adding " .. file)
-                table.insert(append[1], filepath)
+                table.insert(append[1], file)
             end
         end
     end

TOOLS/lua/cycle-deinterlace-pullup.lua

@@ -20,9 +20,9 @@ pullup_label = string.format("%s-pullup", script_name)
 
 function pullup_on()
     for i,vf in pairs(mp.get_property_native('vf')) do
-        if vf['label'] == pullup_label then
-            return "yes"
-        end
+	if vf['label'] == pullup_label then
+	    return "yes"
+	end
     end
     return "no"
 end

TOOLS/lua/drc-control.lua

@@ -21,30 +21,30 @@ script_name = mp.get_script_name()
 
 function print_state(params)
     if params then
-        mp.osd_message(script_name..":\n"
-                           .."method = "..params["method"].."\n"
-                           .."target = "..params["target"])
+	mp.osd_message(script_name..":\n"
+			   .."method = "..params["method"].."\n"
+			   .."target = "..params["target"])
     else
-        mp.osd_message(script_name..":\noff")
+	mp.osd_message(script_name..":\noff")
     end
 end
 
 function get_index_of_drc(afs)
     for i,af in pairs(afs) do
-        if af["label"] == script_name then
-            return i
-        end
+	if af["label"] == script_name then
+	    return i
+	end
     end
 end
 
 function append_drc(afs)
     afs[#afs+1] = {
-        name   = "drc",
-        label  = script_name,
-        params = {
-            method = "1",
-            target = "0.25"
-        }
+	name   = "drc",
+	label  = script_name,
+	params = {
+	    method = "1",
+	    target = "0.25"
+	}
     }
     print_state(afs[#afs]["params"])
 end
@@ -53,34 +53,34 @@ function modify_or_create_af(fun)
     afs = mp.get_property_native("af")
     i = get_index_of_drc(afs)
     if not i then
-        append_drc(afs)
+	append_drc(afs)
     else
-        fun(afs, i)
+	fun(afs, i)
     end
     mp.set_property_native("af", afs)
 end
 
 function drc_toggle_method_handler()
     modify_or_create_af(
-        function (afs, i)
-            new_method=(afs[i]["params"]["method"]+1)%3
-            if new_method == 0 then
-                table.remove(afs, i)
-                print_state(nil)
-            else
-                afs[i]["params"]["method"] = tostring((afs[i]["params"]["method"])%2+1)
-                print_state(afs[i]["params"])
-            end
-        end
+	function (afs, i)
+	    new_method=(afs[i]["params"]["method"]+1)%3
+	    if new_method == 0 then
+		table.remove(afs, i)
+		print_state(nil)
+	    else
+		afs[i]["params"]["method"] = tostring((afs[i]["params"]["method"])%2+1)
+		print_state(afs[i]["params"])
+	    end
+	end
     )
 end
 
 function drc_scale_target(factor)
     modify_or_create_af(
-        function (afs)
-            afs[i]["params"]["target"] = tostring(afs[i]["params"]["target"]*factor)
-            print_state(afs[i]["params"])
-        end
+	function (afs)
+	    afs[i]["params"]["target"] = tostring(afs[i]["params"]["target"]*factor)
+	    print_state(afs[i]["params"])
+	end
     )
 end
 

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

@@ -1 +0,0 @@
-screenshot-template=~/Desktop/shot%n

TOOLS/umpv

@@ -21,13 +21,21 @@ script after that will start a new mpv instance.
 
 Note that you can control the mpv instance by writing to the command fifo:
 
-    echo "cycle fullscreen" > ~/.umpv-fifo
+    echo "cycle fullscreen" > /tmp/umpv-fifo-*
 
 Note: you can supply custom mpv path and options with the MPV environment
       variable. The environment variable will be split on whitespace, and the
       first item is used as path to mpv binary and the rest is passed as options
       _if_ the script starts mpv. If mpv is not started by the script (i.e. mpv
       is already running), this will be ignored.
+
+Warning:
+
+The script attempts to make sure the FIFO is safely created (i.e. not world-
+writable), and checks that it's really a FIFO. This is important for security,
+because the FIFO allows anyone with write access to run arbitrary commands
+in mpv's context using the "run" input command. If you are worried about
+security, you should verify that the code handles these concerns correctly.
 """
 
 import sys
@@ -57,7 +65,7 @@ def make_abs(filename):
     return filename
 files = [make_abs(f) for f in files]
 
-FIFO = os.path.join(os.getenv("HOME"), ".umpv_fifo")
+FIFO = "/tmp/umpv-fifo-" + os.getenv("USER")
 
 fifo_fd = -1
 try:
@@ -70,6 +78,15 @@ except OSError as e:
     else:
         raise e
 
+if fifo_fd >= 0:
+    st = os.fstat(fifo_fd)
+    if (((st.st_mode & (stat.S_IRWXG | stat.S_IRWXO)) != 0) or
+        (not stat.S_ISFIFO(st.st_mode)) or
+        (st.st_uid != os.getuid())):
+        sys.stderr.write("error: command FIFO is not a FIFO or has bogus "
+                         "permissions\n")
+        sys.exit(1)
+
 if fifo_fd >= 0:
     # Unhandled race condition: what if mpv is terminating right now?
     fcntl.fcntl(fifo_fd, fcntl.F_SETFL, 0) # set blocking mode
@@ -86,7 +103,10 @@ else:
     try:
         os.unlink(FIFO)
     except OSError as e:
-        pass
+        if e.errno == errno.ENOENT:
+            pass
+        else:
+            raise e
     os.mkfifo(FIFO, 0o600)
 
     opts = (os.getenv("MPV") or "mpv").split()

TOOLS/vf_dlopen/ildetect.sh

@@ -11,6 +11,7 @@ esac
 
 : ${MPV:=mpv}
 : ${ILDETECT_MPV:=$MPV}
+: ${ILDETECT_MPV:=$MPV}
 : ${ILDETECT_MPVFLAGS:=--start=35% --length=35}
 : ${ILDETECT_DRY_RUN:=}
 : ${ILDETECT_QUIET:=}

VERSION

@@ -1 +1 @@
-0.7.2
+0.6.1

audio/audio.c

@@ -20,13 +20,10 @@
 #include <stdlib.h>
 #include <assert.h>
 
-#include <libavutil/buffer.h>
-#include <libavutil/frame.h>
-#include <libavutil/version.h>
+#include <libavutil/mem.h>
 
 #include "talloc.h"
 #include "common/common.h"
-#include "fmt-conversion.h"
 #include "audio.h"
 
 static void update_redundant_info(struct mp_audio *mpa)
@@ -91,12 +88,18 @@ bool mp_audio_config_valid(const struct mp_audio *mpa)
         && mpa->rate >= 1 && mpa->rate < 10000000;
 }
 
-char *mp_audio_config_to_str_buf(char *buf, size_t buf_sz, struct mp_audio *mpa)
+char *mp_audio_fmt_to_str(int srate, const struct mp_chmap *chmap, int format)
 {
-    snprintf(buf, buf_sz, "%dHz %s %dch %s", mpa->rate,
-             mp_chmap_to_str(&mpa->channels), mpa->channels.num,
-             af_fmt_to_str(mpa->format));
-    return buf;
+    char *chstr = mp_chmap_to_str(chmap);
+    char *res = talloc_asprintf(NULL, "%dHz %s %dch %s", srate, chstr,
+                                chmap->num, af_fmt_to_str(format));
+    talloc_free(chstr);
+    return res;
+}
+
+char *mp_audio_config_to_str(struct mp_audio *mpa)
+{
+    return mp_audio_fmt_to_str(mpa->rate, &mpa->channels, mpa->format);
 }
 
 void mp_audio_force_interleaved_format(struct mp_audio *mpa)
@@ -113,27 +116,19 @@ int mp_audio_psize(struct mp_audio *mpa)
 
 void mp_audio_set_null_data(struct mp_audio *mpa)
 {
-    for (int n = 0; n < MP_NUM_CHANNELS; n++) {
+    for (int n = 0; n < MP_NUM_CHANNELS; n++)
         mpa->planes[n] = NULL;
-        mpa->allocated[n] = NULL;
-    }
     mpa->samples = 0;
 }
 
-static int get_plane_size(const struct mp_audio *mpa, int samples)
-{
-    if (samples < 0 || !mp_audio_config_valid(mpa))
-        return -1;
-    if (samples >= INT_MAX / mpa->sstride)
-        return -1;
-    return MPMAX(samples * mpa->sstride, 1);
-}
-
 static void mp_audio_destructor(void *ptr)
 {
     struct mp_audio *mpa = ptr;
-    for (int n = 0; n < MP_NUM_CHANNELS; n++)
-        av_buffer_unref(&mpa->allocated[n]);
+    for (int n = 0; n < MP_NUM_CHANNELS; n++) {
+        // Note: don't free if not allocated by mp_audio_realloc
+        if (mpa->allocated[n])
+            av_free(mpa->planes[n]);
+    }
 }
 
 /* Reallocate the data stored in mpa->planes[n] so that enough samples are
@@ -152,19 +147,27 @@ static void mp_audio_destructor(void *ptr)
  */
 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
+    assert(samples >= 0);
+    if (samples >= INT_MAX / mpa->sstride)
+        abort(); // oom
+    int size = MPMAX(samples * mpa->sstride, 1);
     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;
+        if (size != mpa->allocated[n]) {
+            // Note: av_realloc() can't be used (see libavutil doxygen)
+            void *new = av_malloc(size);
+            if (!new)
+                abort();
+            if (mpa->allocated[n])
+                memcpy(new, mpa->planes[n], MPMIN(mpa->allocated[n], size));
+            av_free(mpa->planes[n]);
+            mpa->planes[n] = new;
+            mpa->allocated[n] = size;
         }
     }
     for (int n = mpa->num_planes; n < MP_NUM_CHANNELS; n++) {
-        av_buffer_unref(&mpa->allocated[n]);
+        av_free(mpa->planes[n]);
         mpa->planes[n] = NULL;
+        mpa->allocated[n] = 0;
     }
     talloc_set_destructor(mpa, mp_audio_destructor);
 }
@@ -191,7 +194,7 @@ int mp_audio_get_allocated_size(struct mp_audio *mpa)
 {
     int size = 0;
     for (int n = 0; n < mpa->num_planes; n++) {
-        int s = mpa->allocated[n] ? mpa->allocated[n]->size / mpa->sstride : 0;
+        int s = mpa->allocated[n] / mpa->sstride;
         size = n == 0 ? s : MPMIN(size, s);
     }
     return size;
@@ -237,136 +240,3 @@ void mp_audio_skip_samples(struct mp_audio *data, int samples)
 
     data->samples -= samples;
 }
-
-// Make sure the frame owns the audio data, and if not, copy the data.
-// Return negative value on failure (which means it can't be made writeable).
-// Non-refcounted frames are always considered writeable.
-int mp_audio_make_writeable(struct mp_audio *data)
-{
-    bool ok = true;
-    for (int n = 0; n < MP_NUM_CHANNELS; n++) {
-        if (data->allocated[n])
-            ok &= av_buffer_is_writable(data->allocated[n]);
-    }
-    if (!ok) {
-        struct mp_audio *new = talloc(NULL, struct mp_audio);
-        *new = *data;
-        mp_audio_set_null_data(new); // use format only
-        mp_audio_realloc(new, data->samples);
-        new->samples = data->samples;
-        mp_audio_copy(new, 0, data, 0, data->samples);
-        // "Transfer" the reference.
-        mp_audio_destructor(data);
-        *data = *new;
-        talloc_set_destructor(new, NULL);
-        talloc_free(new);
-    }
-    return 0;
-}
-
-struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe)
-{
-    AVFrame *tmp = NULL;
-    struct mp_audio *new = talloc_zero(NULL, struct mp_audio);
-    talloc_set_destructor(new, mp_audio_destructor);
-
-    mp_audio_set_format(new, af_from_avformat(avframe->format));
-
-    struct mp_chmap lavc_chmap;
-    mp_chmap_from_lavc(&lavc_chmap, avframe->channel_layout);
-
-#if LIBAVUTIL_VERSION_MICRO >= 100
-    // FFmpeg being special again
-    if (lavc_chmap.num != avframe->channels)
-        mp_chmap_from_channels(&lavc_chmap, avframe->channels);
-#endif
-
-    new->rate = avframe->sample_rate;
-
-    mp_audio_set_channels(new, &lavc_chmap);
-
-    // Force refcounted frame.
-    if (!avframe->buf[0]) {
-        tmp = av_frame_alloc();
-        if (!tmp)
-            goto fail;
-        if (av_frame_ref(tmp, avframe) < 0)
-            goto fail;
-        avframe = tmp;
-    }
-
-    // If we can't handle the format (e.g. too many channels), bail out.
-    if (!mp_audio_config_valid(new) || avframe->nb_extended_buf)
-        goto fail;
-
-    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(avframe->buf[n]);
-        if (!new->allocated[n])
-            goto fail;
-    }
-
-    for (int n = 0; n < new->num_planes; n++)
-        new->planes[n] = avframe->data[n];
-    new->samples = avframe->nb_samples;
-
-    return new;
-
-fail:
-    talloc_free(new);
-    av_frame_free(&tmp);
-    return NULL;
-}
-
-struct mp_audio_pool {
-    AVBufferPool *avpool;
-    int element_size;
-};
-
-struct mp_audio_pool *mp_audio_pool_create(void *ta_parent)
-{
-    return talloc_zero(ta_parent, struct mp_audio_pool);
-}
-
-static void mp_audio_pool_destructor(void *p)
-{
-    struct mp_audio_pool *pool = p;
-    av_buffer_pool_uninit(&pool->avpool);
-}
-
-// Allocate data using the given format and number of samples.
-// Returns NULL on error.
-struct mp_audio *mp_audio_pool_get(struct mp_audio_pool *pool,
-                                   const struct mp_audio *fmt, int samples)
-{
-    int size = get_plane_size(fmt, samples);
-    if (size < 0)
-        return NULL;
-    if (!pool->avpool || size > pool->element_size) {
-        size_t alloc = ta_calc_prealloc_elems(size);
-        if (alloc >= INT_MAX)
-            return NULL;
-        av_buffer_pool_uninit(&pool->avpool);
-        pool->avpool = av_buffer_pool_init(alloc, NULL);
-        if (!pool->avpool)
-            return NULL;
-        talloc_set_destructor(pool, mp_audio_pool_destructor);
-    }
-    struct mp_audio *new = talloc_ptrtype(NULL, new);
-    talloc_set_destructor(new, mp_audio_destructor);
-    *new = *fmt;
-    mp_audio_set_null_data(new);
-    new->samples = samples;
-    for (int n = 0; n < new->num_planes; n++) {
-        new->allocated[n] = av_buffer_pool_get(pool->avpool);
-        if (!new->allocated[n]) {
-            talloc_free(new);
-            return NULL;
-        }
-        new->planes[n] = new->allocated[n]->data;
-    }
-    return new;
-}

audio/audio.h

@@ -38,7 +38,7 @@ struct mp_audio {
     int bps;            // size of sub-samples (af_fmt2bps(format))
 
     // private
-    struct AVBufferRef *allocated[MP_NUM_CHANNELS];
+    int allocated[MP_NUM_CHANNELS]; // use mp_audio_get_allocated_size()
 };
 
 void mp_audio_set_format(struct mp_audio *mpa, int format);
@@ -49,8 +49,8 @@ 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);
 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);
-#define mp_audio_config_to_str(m) mp_audio_config_to_str_buf((char[64]){0}, 64, (m))
+char *mp_audio_fmt_to_str(int srate, const struct mp_chmap *chmap, int format);
+char *mp_audio_config_to_str(struct mp_audio *mpa);
 
 void mp_audio_force_interleaved_format(struct mp_audio *mpa);
 
@@ -68,13 +68,4 @@ void mp_audio_copy(struct mp_audio *dst, int dst_offset,
                    struct mp_audio *src, int src_offset, int length);
 void mp_audio_skip_samples(struct mp_audio *data, int samples);
 
-int mp_audio_make_writeable(struct mp_audio *data);
-
-struct AVFrame;
-struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe);
-
-struct mp_audio_pool *mp_audio_pool_create(void *ta_parent);
-struct mp_audio *mp_audio_pool_get(struct mp_audio_pool *pool,
-                                   const struct mp_audio *fmt, int samples);
-
 #endif

audio/chmap.c

@@ -18,7 +18,6 @@
 #include <stdlib.h>
 #include <assert.h>
 
-#include "common/common.h"
 #include "common/msg.h"
 #include "chmap.h"
 
@@ -87,7 +86,6 @@ static const char *const std_layout_names[][2] = {
     {"7.1(wide-side)",  "fl-fr-fc-lfe-flc-frc-sl-sr"},
     {"octagonal",       "fl-fr-fc-bl-br-bc-sl-sr"},
     {"downmix",         "dl-dr"},
-    {"auto",            ""}, // not in lavc
     {0}
 };
 
@@ -377,35 +375,34 @@ void mp_chmap_get_reorder(int dst[MP_NUM_CHANNELS], const struct mp_chmap *from,
 // Returns something like "fl-fr-fc". If there's a standard layout in lavc
 // order, return that, e.g. "3.0" instead of "fl-fr-fc".
 // Unassigned but valid speakers get names like "sp28".
-char *mp_chmap_to_str_buf(char *buf, size_t buf_size, const struct mp_chmap *src)
+char *mp_chmap_to_str(const struct mp_chmap *src)
 {
-    buf[0] = '\0';
+    char *res = talloc_strdup(NULL, "");
 
-    if (mp_chmap_is_unknown(src)) {
-        snprintf(buf, buf_size, "unknown%d", src->num);
-        return buf;
-    }
+    if (mp_chmap_is_unknown(src))
+        return talloc_asprintf_append_buffer(res, "unknown%d", src->num);
 
     for (int n = 0; n < src->num; n++) {
         int sp = src->speaker[n];
         const char *s = sp < MP_SPEAKER_ID_COUNT ? speaker_names[sp][0] : NULL;
-        char sp_buf[10];
+        char buf[10];
         if (!s) {
-            snprintf(sp_buf, sizeof(sp_buf), "sp%d", sp);
-            s = sp_buf;
+            snprintf(buf, sizeof(buf), "sp%d", sp);
+            s = buf;
         }
-        mp_snprintf_cat(buf, buf_size, "%s%s", n > 0 ? "-" : "", s);
+        res = talloc_asprintf_append_buffer(res, "%s%s", n > 0 ? "-" : "", s);
     }
 
     // To standard layout name
     for (int n = 0; std_layout_names[n][0]; n++) {
-        if (strcmp(buf, std_layout_names[n][1]) == 0) {
-            snprintf(buf, buf_size, "%s", std_layout_names[n][0]);
+        if (res && strcmp(res, std_layout_names[n][1]) == 0) {
+            talloc_free(res);
+            res = talloc_strdup(NULL, std_layout_names[n][0]);
             break;
         }
     }
 
-    return buf;
+    return res;
 }
 
 // If src can be parsed as channel map (as produced by mp_chmap_to_str()),

audio/chmap.h

@@ -61,8 +61,7 @@ enum mp_speaker_id {
     MP_SPEAKER_ID_UNKNOWN0 = 64,
     MP_SPEAKER_ID_UNKNOWN_LAST = MP_SPEAKER_ID_UNKNOWN0 + MP_NUM_CHANNELS - 1,
 
-    // Including the unassigned IDs in between. This is not a valid ID anymore,
-    // but is still within uint8_t.
+    // Including the unassigned IDs in between. This is not a valid ID anymore.
     MP_SPEAKER_ID_COUNT,
 };
 
@@ -120,9 +119,7 @@ void mp_chmap_reorder_to_lavc(struct mp_chmap *map);
 void mp_chmap_get_reorder(int dst[MP_NUM_CHANNELS], const struct mp_chmap *from,
                           const struct mp_chmap *to);
 
-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(const struct mp_chmap *src);
 bool mp_chmap_from_str(struct mp_chmap *dst, bstr src);
 
 struct mp_log;

audio/chmap_sel.c

@@ -81,21 +81,14 @@ void mp_chmap_sel_add_alsa_def(struct mp_chmap_sel *s)
     }
 }
 
+#define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0]))
+
 // Add a channel map that should be allowed.
 void mp_chmap_sel_add_map(struct mp_chmap_sel *s, const struct mp_chmap *map)
 {
-    if (!mp_chmap_is_valid(map))
-        return;
-    if (!s->chmaps)
-        s->chmaps = s->chmaps_storage;
-    if (s->num_chmaps == MP_ARRAY_SIZE(s->chmaps_storage)) {
-        if (!s->tmp)
-            return;
-        s->chmaps = talloc_memdup(s->tmp, s->chmaps, sizeof(s->chmaps_storage));
-    }
-    if (s->chmaps != s->chmaps_storage)
-        MP_TARRAY_GROW(s->tmp, s->chmaps, s->num_chmaps);
-    s->chmaps[s->num_chmaps++] = *map;
+    assert(s->num_chmaps < ARRAY_LEN(s->chmaps));
+    if (mp_chmap_is_valid(map))
+        s->chmaps[s->num_chmaps++] = *map;
 }
 
 // Allow all waveext formats in default order.

audio/chmap_sel.h

@@ -26,12 +26,8 @@ struct mp_chmap_sel {
     // should be considered opaque
     bool allow_any, allow_waveext;
     bool speakers[MP_SPEAKER_ID_COUNT];
-    struct mp_chmap *chmaps;
+    struct mp_chmap chmaps[20];
     int num_chmaps;
-
-    struct mp_chmap chmaps_storage[20];
-
-    void *tmp; // set to any talloc context to allow more chmaps entries
 };
 
 void mp_chmap_sel_add_any(struct mp_chmap_sel *s);

audio/decode/ad.h

@@ -36,7 +36,7 @@ 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 mp_audio **out);
+    int (*decode_packet)(struct dec_audio *da);
 };
 
 enum ad_ctrl {

audio/decode/ad_lavc.c

@@ -25,7 +25,6 @@
 #include <libavcodec/avcodec.h>
 #include <libavutil/opt.h>
 #include <libavutil/common.h>
-#include <libavutil/intreadwrite.h>
 
 #include "talloc.h"
 
@@ -44,7 +43,6 @@ struct priv {
     struct mp_audio frame;
     bool force_channel_map;
     struct demux_packet *packet;
-    uint32_t skip_samples;
 };
 
 static void uninit(struct dec_audio *da);
@@ -73,6 +71,37 @@ const struct m_sub_options ad_lavc_conf = {
     },
 };
 
+static void set_data_from_avframe(struct dec_audio *da)
+{
+    struct priv *priv = da->priv;
+    AVCodecContext *lavc_context = priv->avctx;
+
+    // Note: invalid parameters are rejected by dec_audio.c
+
+    int fmt = lavc_context->sample_fmt;
+    mp_audio_set_format(&da->decoded, af_from_avformat(fmt));
+    if (!da->decoded.format)
+        MP_FATAL(da, "unsupported lavc format %s", av_get_sample_fmt_name(fmt));
+
+    da->decoded.rate = lavc_context->sample_rate;
+
+    struct mp_chmap lavc_chmap;
+    mp_chmap_from_lavc(&lavc_chmap, lavc_context->channel_layout);
+    // No channel layout or layout disagrees with channel count
+    if (lavc_chmap.num != lavc_context->channels)
+        mp_chmap_from_channels(&lavc_chmap, lavc_context->channels);
+    if (priv->force_channel_map) {
+        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(&da->decoded, &lavc_chmap);
+
+    da->decoded.samples = priv->avframe->nb_samples;
+    for (int n = 0; n < da->decoded.num_planes; n++)
+        da->decoded.planes[n] = priv->avframe->data[n];
+}
+
 static int init(struct dec_audio *da, const char *decoder)
 {
     struct MPOpts *mpopts = da->opts;
@@ -110,11 +139,6 @@ static int init(struct dec_audio *da, const char *decoder)
     av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
                       AV_OPT_SEARCH_CHILDREN);
 
-#if HAVE_AVFRAME_SKIP_SAMPLES
-    // Let decoder add AV_FRAME_DATA_SKIP_SAMPLES.
-    av_opt_set(lavc_context, "flags2", "+skip_manual", AV_OPT_SEARCH_CHILDREN);
-#endif
-
     mp_set_avopts(da->log, lavc_context, opts->avopts);
 
     lavc_context->codec_tag = sh->format;
@@ -173,19 +197,21 @@ static int control(struct dec_audio *da, int cmd, void *arg)
     switch (cmd) {
     case ADCTRL_RESET:
         avcodec_flush_buffers(ctx->avctx);
+        mp_audio_set_null_data(&da->decoded);
         talloc_free(ctx->packet);
         ctx->packet = NULL;
-        ctx->skip_samples = 0;
         return CONTROL_TRUE;
     }
     return CONTROL_UNKNOWN;
 }
 
-static int decode_packet(struct dec_audio *da, struct mp_audio **out)
+static int decode_packet(struct dec_audio *da)
 {
     struct priv *priv = da->priv;
     AVCodecContext *avctx = priv->avctx;
 
+    mp_audio_set_null_data(&da->decoded);
+
     struct demux_packet *mpkt = priv->packet;
     if (!mpkt) {
         if (demux_read_packet_async(da->header, &mpkt) == 0)
@@ -232,49 +258,15 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
     if (!got_frame)
         return mpkt ? AD_OK : AD_EOF;
 
+    set_data_from_avframe(da);
+
     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 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) {
-        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);
-
-#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);
-        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
-
-    *out = mpframe;
-
-    av_frame_unref(priv->avframe);
-
-    MP_DBG(da, "Decoded %d -> %d samples\n", in_len, mpframe->samples);
+    MP_DBG(da, "Decoded %d -> %d samples\n", in_len, da->decoded.samples);
     return 0;
 }
 

audio/decode/ad_mpg123.c

@@ -43,7 +43,6 @@ struct ad_mpg123_context {
     short delay;
     /* If the stream is actually VBR. */
     char vbr;
-    struct mp_audio frame;
 };
 
 static void uninit(struct dec_audio *da)
@@ -198,25 +197,27 @@ static int set_format(struct dec_audio *da)
     int encoding;
     ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
     if (ret == MPG123_OK) {
-        mp_audio_set_num_channels(&con->frame, channels);
-        con->frame.rate = rate;
+        mp_audio_set_num_channels(&da->decoded, channels);
+        da->decoded.rate = rate;
         int af = mpg123_format_to_af(encoding);
         if (!af) {
             /* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
             MP_ERR(da, "Bad encoding from mpg123: %i.\n", encoding);
             return MPG123_ERR;
         }
-        mp_audio_set_format(&con->frame, af);
+        mp_audio_set_format(&da->decoded, af);
         con->sample_size = channels * af_fmt2bps(af);
     }
     return ret;
 }
 
-static int decode_packet(struct dec_audio *da, struct mp_audio **out)
+static int decode_packet(struct dec_audio *da)
 {
     struct ad_mpg123_context *con = da->priv;
     int ret;
 
+    mp_audio_set_null_data(&da->decoded);
+
     struct demux_packet *pkt;
     if (demux_read_packet_async(da->header, &pkt) == 0)
         return AD_WAIT;
@@ -256,11 +257,8 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
     }
 
     int got_samples = bytes / con->sample_size;
-    *out = mp_audio_pool_get(da->pool, &con->frame, got_samples);
-    if (!*out)
-        return AD_ERR;
-
-    memcpy((*out)->planes[0], audio, bytes);
+    da->decoded.planes[0] = audio;
+    da->decoded.samples = got_samples;
 
     update_info(da);
     return 0;
@@ -276,6 +274,7 @@ static int control(struct dec_audio *da, int cmd, void *arg)
 
     switch (cmd) {
     case ADCTRL_RESET:
+        mp_audio_set_null_data(&da->decoded);
         mpg123_close(con->handle);
 
         if (mpg123_open_feed(con->handle) != MPG123_OK) {

audio/decode/ad_spdif.c

@@ -40,7 +40,6 @@ struct spdifContext {
     int              out_buffer_len;
     uint8_t          out_buffer[OUTBUF_SIZE];
     bool             need_close;
-    struct mp_audio  fmt;
 };
 
 static int write_packet(void *p, uint8_t *buf, int buf_size)
@@ -163,9 +162,9 @@ static int init(struct dec_audio *da, const char *decoder)
     default:
         abort();
     }
-    mp_audio_set_num_channels(&spdif_ctx->fmt, num_channels);
-    mp_audio_set_format(&spdif_ctx->fmt, sample_format);
-    spdif_ctx->fmt.rate = samplerate;
+    mp_audio_set_num_channels(&da->decoded, num_channels);
+    mp_audio_set_format(&da->decoded, sample_format);
+    da->decoded.rate = samplerate;
 
     if (avformat_write_header(lavf_ctx, &format_opts) < 0) {
         MP_FATAL(da, "libavformat spdif initialization failed.\n");
@@ -183,11 +182,13 @@ fail:
     return 0;
 }
 
-static int decode_packet(struct dec_audio *da, struct mp_audio **out)
+static int decode_packet(struct dec_audio *da)
 {
     struct spdifContext *spdif_ctx = da->priv;
     AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;
 
+    mp_audio_set_null_data(&da->decoded);
+
     spdif_ctx->out_buffer_len  = 0;
 
     struct demux_packet *mpkt;
@@ -211,12 +212,8 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
     if (ret < 0)
         return AD_ERR;
 
-    int samples = spdif_ctx->out_buffer_len / spdif_ctx->fmt.sstride;
-    *out = mp_audio_pool_get(da->pool, &spdif_ctx->fmt, samples);
-    if (!*out)
-        return AD_ERR;
-
-    memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
+    da->decoded.planes[0] = spdif_ctx->out_buffer;
+    da->decoded.samples = spdif_ctx->out_buffer_len / da->decoded.sstride;
 
     return 0;
 }

audio/decode/dec_audio.c

@@ -56,6 +56,19 @@ static const struct ad_functions * const ad_drivers[] = {
     NULL
 };
 
+// Drop audio buffer and reinit it (after format change)
+// Returns whether the format was valid at all.
+static bool reinit_audio_buffer(struct dec_audio *da)
+{
+    if (!mp_audio_config_valid(&da->decoded)) {
+        MP_ERR(da, "Audio decoder did not specify audio "
+               "format, or requested an unsupported configuration!\n");
+        return false;
+    }
+    mp_audio_buffer_reinit(da->decode_buffer, &da->decoded);
+    return true;
+}
+
 static void uninit_decoder(struct dec_audio *d_audio)
 {
     if (d_audio->ad_driver) {
@@ -76,6 +89,7 @@ static int init_audio_codec(struct dec_audio *d_audio, const char *decoder)
         return 0;
     }
 
+    d_audio->decode_buffer = mp_audio_buffer_create(NULL);
     return 1;
 }
 
@@ -150,26 +164,136 @@ 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);
+    if (d_audio->afilter) {
+        MP_VERBOSE(d_audio, "Uninit audio filters...\n");
+        af_destroy(d_audio->afilter);
+        d_audio->afilter = NULL;
+    }
     uninit_decoder(d_audio);
+    talloc_free(d_audio->decode_buffer);
     talloc_free(d_audio);
 }
 
+
+int audio_init_filters(struct dec_audio *d_audio, int in_samplerate,
+                       int *out_samplerate, struct mp_chmap *out_channels,
+                       int *out_format)
+{
+    if (!d_audio->afilter)
+        d_audio->afilter = af_new(d_audio->global);
+    struct af_stream *afs = d_audio->afilter;
+
+    // input format: same as codec's output format:
+    mp_audio_buffer_get_format(d_audio->decode_buffer, &afs->input);
+    // Sample rate can be different when adjusting playback speed
+    afs->input.rate = in_samplerate;
+
+    // output format: same as ao driver's input format (if missing, fallback to input)
+    afs->output.rate = *out_samplerate;
+    mp_audio_set_channels(&afs->output, out_channels);
+    mp_audio_set_format(&afs->output, *out_format);
+
+    afs->replaygain_data = d_audio->replaygain_data;
+
+    char *s_from = mp_audio_config_to_str(&afs->input);
+    char *s_to = mp_audio_config_to_str(&afs->output);
+    MP_VERBOSE(d_audio, "Building audio filter chain for %s -> %s...\n", s_from, s_to);
+    talloc_free(s_from);
+    talloc_free(s_to);
+
+    // let's autoprobe it!
+    if (af_init(afs) != 0) {
+        af_destroy(afs);
+        d_audio->afilter = NULL;
+        return 0;   // failed :(
+    }
+
+    *out_samplerate = afs->output.rate;
+    *out_channels = afs->output.channels;
+    *out_format = afs->output.format;
+
+    return 1;
+}
+
 /* 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)
 {
-    while (!da->waiting) {
-        int ret = da->ad_driver->decode_packet(da, &da->waiting);
+    while (!mp_audio_config_valid(&da->decoded)) {
+        if (da->decoded.samples > 0)
+            return AD_ERR; // invalid format, rather than uninitialized
+        int ret = da->ad_driver->decode_packet(da);
         if (ret < 0)
             return ret;
     }
-    talloc_steal(da, da->waiting);
-    da->decode_format = *da->waiting;
-    return mp_audio_config_valid(da->waiting) ? AD_OK : AD_ERR;
+    if (mp_audio_buffer_samples(da->decode_buffer) > 0) // avoid accidental flush
+        return AD_OK;
+    return reinit_audio_buffer(da) ? AD_OK : AD_ERR;
+}
+
+// Filter len bytes of input, put result into outbuf.
+static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
+                          int len)
+{
+    bool format_change = false;
+    int error = 0;
+
+    assert(len > 0); // would break EOF logic below
+
+    while (mp_audio_buffer_samples(da->decode_buffer) < len) {
+        // Check for a format change
+        struct mp_audio config;
+        mp_audio_buffer_get_format(da->decode_buffer, &config);
+        format_change = !mp_audio_config_equals(&da->decoded, &config);
+        if (format_change) {
+            error = AD_EOF; // drain remaining data left in the current buffer
+            break;
+        }
+        if (da->decoded.samples > 0) {
+            int copy = MPMIN(da->decoded.samples, len);
+            struct mp_audio append = da->decoded;
+            append.samples = copy;
+            mp_audio_buffer_append(da->decode_buffer, &append);
+            mp_audio_skip_samples(&da->decoded, copy);
+            da->pts_offset += copy;
+            continue;
+        }
+        error = da->ad_driver->decode_packet(da);
+        if (error < 0)
+            break;
+    }
+
+    if (error == AD_WAIT)
+        return error;
+
+    // Filter
+    struct mp_audio filter_data;
+    mp_audio_buffer_peek(da->decode_buffer, &filter_data);
+    filter_data.rate = da->afilter->input.rate; // due to playback speed change
+    len = MPMIN(filter_data.samples, len);
+    filter_data.samples = len;
+    bool eof = error == AD_EOF && filter_data.samples == 0;
+
+    if (af_filter(da->afilter, &filter_data, eof ? AF_FILTER_FLAG_EOF : 0) < 0)
+        return AD_ERR;
+
+    mp_audio_buffer_append(outbuf, &filter_data);
+    if (error == AD_EOF && filter_data.samples > 0)
+        error = 0; // don't end playback yet
+
+    // remove processed data from decoder buffer:
+    mp_audio_buffer_skip(da->decode_buffer, len);
+
+    // if format was changed, and all data was drained, execute the format change
+    if (format_change && eof) {
+        error = AD_NEW_FMT;
+        if (!reinit_audio_buffer(da))
+            error = AD_ERR; // switch to invalid format
+    }
+
+    return error;
 }
 
 /* Try to get at least minsamples decoded+filtered samples in outbuf
@@ -177,55 +301,52 @@ int initial_audio_decode(struct dec_audio *da)
  * 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 audio_decode(struct dec_audio *d_audio, struct mp_audio_buffer *outbuf,
                  int minsamples)
 {
-    struct af_stream *afs = da->afilter;
-    if (afs->initialized < 1)
+    if (!d_audio->afilter)
         return AD_ERR;
 
-    MP_STATS(da, "start audio");
+    // Indicates that a filter seems to be buffering large amounts of data
+    int huge_filter_buffer = 0;
 
+    /* Filter output size will be about filter_multiplier times input size.
+     * If some filter buffers audio in big blocks this might only hold
+     * as average over time. */
+    double filter_multiplier = af_calc_filter_multiplier(d_audio->afilter);
+
+    int prev_buffered = -1;
     int res = 0;
+    MP_STATS(d_audio, "start audio");
     while (res >= 0 && minsamples >= 0) {
         int buffered = mp_audio_buffer_samples(outbuf);
-        if (minsamples < buffered)
+        if (minsamples < buffered || buffered == prev_buffered)
             break;
+        prev_buffered = buffered;
 
-        res = 0;
+        int decsamples = (minsamples - buffered) / filter_multiplier;
+        // + some extra for possible filter buffering, and avoid 0
+        decsamples += 512;
 
-        struct mp_audio *mpa = da->waiting;
-        if (!mpa)
-            res = da->ad_driver->decode_packet(da, &mpa);
-
-        if (res != AD_EOF) {
-            if (res < 0)
-                break;
-            if (!mpa )
-                continue;
+        if (huge_filter_buffer) {
+            /* Some filter must be doing significant buffering if the estimated
+             * input length didn't produce enough output from filters.
+             * Feed the filters 250 samples at a time until we have enough
+             * output. Very small amounts could make filtering inefficient while
+             * large amounts can make mpv demux the file unnecessarily far ahead
+             * to get audio data and buffer video frames in memory while doing
+             * so. However the performance impact of either is probably not too
+             * significant as long as the value is not completely insane. */
+            decsamples = 250;
         }
 
-        if (mpa) {
-            da->pts_offset += mpa->samples;
-            da->decode_format = *mpa;
-            mp_audio_set_null_data(&da->decode_format);
-            // On format change, make sure to drain the filter chain.
-            if (!mp_audio_config_equals(&afs->input, mpa)) {
-                res = AD_NEW_FMT;
-                da->waiting = talloc_steal(da, mpa);
-                mpa = NULL;
-            }
-        }
+        /* if this iteration does not fill buffer, we must have lots
+         * of buffering in filters */
+        huge_filter_buffer = 1;
 
-        if (mpa)
-            da->waiting = NULL;
-
-        if (af_filter(afs, mpa, outbuf) < 0)
-            return AD_ERR;
+        res = filter_n_bytes(d_audio, outbuf, decsamples);
     }
-
-    MP_STATS(da, "end audio");
-
+    MP_STATS(d_audio, "end audio");
     return res;
 }
 
@@ -233,11 +354,10 @@ 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_control_all(d_audio->afilter, AF_CONTROL_RESET, NULL);
+    if (d_audio->afilter)
+        af_control_all(d_audio->afilter, AF_CONTROL_RESET, NULL);
     d_audio->pts = MP_NOPTS_VALUE;
     d_audio->pts_offset = 0;
-    if (d_audio->waiting) {
-        talloc_free(d_audio->waiting);
-        d_audio->waiting = NULL;
-    }
+    if (d_audio->decode_buffer)
+        mp_audio_buffer_clear(d_audio->decode_buffer);
 }

audio/decode/dec_audio.h

@@ -33,13 +33,13 @@ struct dec_audio {
     struct mpv_global *global;
     const struct ad_functions *ad_driver;
     struct sh_stream *header;
+    struct mp_audio_buffer *decode_buffer;
     struct af_stream *afilter;
     char *decoder_desc;
+    struct replaygain_data *replaygain_data;
     int init_retries;
-    struct mp_audio_pool *pool;
-    struct mp_audio decode_format;
-    struct mp_audio *waiting;   // used on format-change
     // set by decoder
+    struct mp_audio decoded;    // decoded audio set by last decode_packet() call
     int bitrate;                // input bitrate, can change with VBR sources
     // last known pts value in output from decoder
     double pts;
@@ -65,4 +65,8 @@ 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);
 
+int audio_init_filters(struct dec_audio *d_audio, int in_samplerate,
+                       int *out_samplerate, struct mp_chmap *out_channels,
+                       int *out_format);
+
 #endif /* MPLAYER_DEC_AUDIO_H */

audio/filter/af.c

@@ -28,7 +28,6 @@
 #include "options/m_option.h"
 #include "options/m_config.h"
 
-#include "audio/audio_buffer.h"
 #include "af.h"
 
 // Static list of filters
@@ -54,7 +53,6 @@ extern const struct af_info af_info_center;
 extern const struct af_info af_info_sinesuppress;
 extern const struct af_info af_info_karaoke;
 extern const struct af_info af_info_scaletempo;
-extern const struct af_info af_info_forcespeed;
 extern const struct af_info af_info_bs2b;
 extern const struct af_info af_info_lavfi;
 extern const struct af_info af_info_convert24;
@@ -85,7 +83,6 @@ static const struct af_info *const filter_list[] = {
     &af_info_center,
     &af_info_sinesuppress,
     &af_info_karaoke,
-    &af_info_forcespeed,
     &af_info_scaletempo,
 #if HAVE_LIBBS2B
     &af_info_bs2b,
@@ -124,6 +121,13 @@ const struct m_obj_list af_obj_list = {
     },
 };
 
+static bool af_config_equals(struct mp_audio *a, struct mp_audio *b)
+{
+    return a->format == b->format
+        && mp_chmap_equals(&a->channels, &b->channels)
+        && a->rate   == b->rate;
+}
+
 static void af_copy_unset_fields(struct mp_audio *dst, struct mp_audio *src)
 {
     if (dst->format == AF_FORMAT_UNKNOWN)
@@ -158,7 +162,7 @@ static int output_control(struct af_instance* af, int cmd, void* arg)
         *filter_output = *output;
         af_copy_unset_fields(filter_output, in);
         *in = *filter_output;
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
+        return af_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
     }
     }
     return AF_UNKNOWN;
@@ -308,8 +312,11 @@ 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->data)
-            mp_snprintf_cat(b, sizeof(b), "%s", mp_audio_config_to_str(af->data));
+        if (af->data) {
+            char *info = mp_audio_config_to_str(af->data);
+            mp_snprintf_cat(b, sizeof(b), "%s", info);
+            talloc_free(info);
+        }
         if (af == at)
             mp_snprintf_cat(b, sizeof(b), " <-");
         MP_MSG(s, msg_level, "%s\n", b);
@@ -317,7 +324,9 @@ static void af_print_filter_chain(struct af_stream *s, struct af_instance *at,
         af = af->next;
     }
 
-    MP_MSG(s, msg_level, "  [ao] %s\n", mp_audio_config_to_str(&s->output));
+    char *info = mp_audio_config_to_str(&s->output);
+    MP_MSG(s, msg_level, "  [ao] %s\n", info);
+    talloc_free(info);
 }
 
 static int af_count_filters(struct af_stream *s)
@@ -514,16 +523,11 @@ static int af_reinit(struct af_stream *s)
         // Reset just in case...
         mp_audio_set_null_data(&in);
 
-        if (!mp_audio_config_valid(&in))
-            goto error;
-
         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 = af->next;
             break;
         case AF_FALSE: { // Configuration filter is needed
@@ -554,27 +558,22 @@ static int af_reinit(struct af_stream *s)
             MP_ERR(s, "Reinitialization did not work, "
                    "audio filter '%s' returned error code %i\n",
                    af->info->name, rv);
-            goto error;
+            af_print_filter_chain(s, af, MSGL_ERR);
+            return AF_ERROR;
         }
     }
 
+    af_print_filter_chain(s, NULL, MSGL_V);
+
     /* Set previously unset fields in s->output to those of the filter chain
      * output. This is used to make the output format fixed, and even if you
      * insert new filters or change the input format, the output format won't
      * change. (Audio outputs generally can't change format at runtime.) */
     af_copy_unset_fields(&s->output, &s->filter_output);
-    if (mp_audio_config_equals(&s->output, &s->filter_output)) {
-        s->initialized = 1;
-        af_print_filter_chain(s, NULL, MSGL_V);
-        return AF_OK;
-    }
-
-    goto error;
+    return af_config_equals(&s->output, &s->filter_output) ? AF_OK : AF_ERROR;
 
 negotiate_error:
     MP_ERR(s, "Unable to convert audio input format to output format.\n");
-error:
-    s->initialized = -1;
     af_print_filter_chain(s, af, MSGL_ERR);
     return AF_ERROR;
 }
@@ -584,7 +583,6 @@ void af_uninit(struct af_stream *s)
 {
     while (s->first->next && s->first->next != s->last)
         af_remove(s, s->first->next);
-    s->initialized = 0;
 }
 
 struct af_stream *af_new(struct mpv_global *global)
@@ -634,6 +632,10 @@ void af_destroy(struct af_stream *s)
    The return value is 0 if success and -1 if failure */
 int af_init(struct af_stream *s)
 {
+    // Sanity check
+    if (!s)
+        return -1;
+
     // Precaution in case caller is misbehaving
     mp_audio_set_null_data(&s->input);
     mp_audio_set_null_data(&s->output);
@@ -643,20 +645,17 @@ int af_init(struct af_stream *s)
         // Add all filters in the list (if there are any)
         struct m_obj_settings *list = s->opts->af_settings;
         for (int i = 0; list && list[i].name; i++) {
-            struct af_instance *af =
-                af_prepend(s, s->last, list[i].name, list[i].attribs);
-            if (!af) {
+            if (!af_prepend(s, s->last, list[i].name, list[i].attribs)) {
                 af_uninit(s);
-                s->initialized = -1;
                 return -1;
             }
-            af->label = talloc_strdup(af, list[i].label);
         }
     }
 
     if (af_reinit(s) != AF_OK) {
         // Something is stuffed audio out will not work
         MP_ERR(s, "Could not create audio filter chain.\n");
+        af_uninit(s);
         return -1;
     }
     return 0;
@@ -669,6 +668,9 @@ int af_init(struct af_stream *s)
 struct af_instance *af_add(struct af_stream *s, char *name, char **args)
 {
     struct af_instance *new;
+    // Sanity check
+    if (!s || !s->first || !name)
+        return NULL;
     // Insert the filter somewhere nice
     if (af_is_conversion_filter(s->first->next))
         new = af_append(s, s->first->next, name, args);
@@ -679,85 +681,30 @@ struct af_instance *af_add(struct af_stream *s, char *name, char **args)
 
     // Reinitalize the filter list
     if (af_reinit(s) != AF_OK) {
-        af_remove(s, new);
-        if (af_reinit(s) != AF_OK) {
-            af_uninit(s);
-            af_init(s);
-        }
+        af_uninit(s);
+        af_init(s);
         return NULL;
     }
     return new;
 }
 
-struct af_instance *af_find_by_label(struct af_stream *s, char *label)
-{
-    for (struct af_instance *af = s->first; af; af = af->next) {
-        if (af->label && strcmp(af->label, label) == 0)
-            return af;
-    }
-    return NULL;
-}
-
-/* Remove the first filter that matches this name. Return number of filters
- * removed (0, 1), or a negative error code if reinit after removing failed.
+/* Filter data chunk through the filters in the list.
+ * On success, *data is set to the filtered data/format.
+ * Warning: input audio data will be overwritten.
  */
-int af_remove_by_label(struct af_stream *s, char *label)
-{
-    struct af_instance *af = af_find_by_label(s, label);
-    if (!af)
-        return 0;
-    af_remove(s, af);
-    if (af_reinit(s) != AF_OK) {
-        af_uninit(s);
-        af_init(s);
-        return -1;
-    }
-    return 1;
-}
-
-/* Feed "data" to the chain, and write results to output. "data" needs to be
- * a refcounted frame, although refcounting is not used yet.
- * data==NULL means EOF.
- */
-int af_filter(struct af_stream *s, struct mp_audio *data,
-              struct mp_audio_buffer *output)
+int af_filter(struct af_stream *s, struct mp_audio *data, int flags)
 {
     struct af_instance *af = s->first;
-    assert(s->initialized > 0);
-    int flags = 0;
-    int r = 0;
-    struct mp_audio tmp;
-    char dummy[MP_NUM_CHANNELS];
-    if (data) {
-        assert(mp_audio_config_equals(af->data, data));
-        r = mp_audio_make_writeable(data);
-    } else {
-        data = &tmp;
-        *data = *(af->data);
-        mp_audio_set_null_data(data);
-        flags = AF_FILTER_FLAG_EOF;
-        for (int n = 0; n < MP_NUM_CHANNELS; n++)
-            data->planes[n] = &dummy[n];
-    }
-    if (r < 0)
-        goto done;
-    struct mp_audio frame = *data;
-    for (int n = 0; n < MP_NUM_CHANNELS; n++)
-        frame.allocated[n] = NULL;
+    assert(mp_audio_config_equals(af->data, data));
     // Iterate through all filters
     while (af) {
-        r = af->filter(af, &frame, flags);
+        int r = af->filter(af, data, flags);
         if (r < 0)
-            goto done;
-        assert(mp_audio_config_equals(af->data, &frame));
+            return r;
+        assert(mp_audio_config_equals(af->data, data));
         af = af->next;
     }
-    mp_audio_buffer_append(output, &frame);
-
-done:
-    if (data != &tmp)
-        talloc_free(data);
-    return r;
+    return 0;
 }
 
 // Calculate average ratio of filter output samples to input samples.

audio/filter/af.h

@@ -76,13 +76,10 @@ struct af_instance {
                  * the number of samples passed though. (Ratio of input
                  * and output, e.g. mul=4 => 1 sample becomes 4 samples) .*/
     bool auto_inserted; // inserted by af.c, such as conversion filters
-    char *label;
 };
 
 // Current audio stream
 struct af_stream {
-    int initialized; // 0: no, 1: yes, -1: attempted to, but failed
-
     // The first and last filter in the list
     struct af_instance *first;
     struct af_instance *last;
@@ -120,7 +117,6 @@ enum af_control {
     AF_CONTROL_SET_PAN_BALANCE,
     AF_CONTROL_GET_PAN_BALANCE,
     AF_CONTROL_SET_PLAYBACK_SPEED,
-    AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE,
 };
 
 // Argument for AF_CONTROL_SET_PAN_LEVEL
@@ -134,11 +130,7 @@ void af_destroy(struct af_stream *s);
 int af_init(struct af_stream *s);
 void af_uninit(struct af_stream *s);
 struct af_instance *af_add(struct af_stream *s, char *name, char **args);
-int af_remove_by_label(struct af_stream *s, char *label);
-struct af_instance *af_find_by_label(struct af_stream *s, char *label);
-struct mp_audio_buffer;
-int af_filter(struct af_stream *s, struct mp_audio *data,
-              struct mp_audio_buffer *output);
+int af_filter(struct af_stream *s, struct mp_audio *data, int flags);
 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);
 

audio/filter/af_export.c

@@ -78,7 +78,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     int mapsize;
 
     // Free previous buffers
-    free(s->buf[0]);
+    if (s->buf)
+      free(s->buf[0]);
 
     // unmap previous area
     if(s->mmap_area)
@@ -93,10 +94,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
     // Allocate new buffers (as one continuous block)
     s->buf[0] = calloc(s->sz*af->data->nch, af->data->bps);
-    if(NULL == s->buf[0]) {
+    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;
 
@@ -148,8 +147,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
 static void uninit( struct af_instance* af )
 {
     af_export_t* s = af->priv;
-
-    free(s->buf[0]);
+    if (s->buf)
+      free(s->buf[0]);
 
     // Free mmaped area
     if(s->mmap_area)

audio/filter/af_forcespeed.c

@@ -1,69 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include "af.h"
-
-struct priv {
-    double speed;
-};
-
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    struct priv *priv = af->priv;
-
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-        struct mp_audio *out = af->data;
-
-        mp_audio_copy_config(out, in);
-        out->rate = in->rate * priv->speed;
-
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
-    }
-    case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
-        priv->speed = *(double *)arg;
-        return AF_OK;
-    }
-    }
-    return AF_UNKNOWN;
-}
-
-static int filter(struct af_instance *af, struct mp_audio *data, int flags)
-{
-    mp_audio_copy_config(data, af->data);
-    return 0;
-}
-
-static int af_open(struct af_instance *af)
-{
-    struct priv *priv = af->priv;
-    af->control = control;
-    af->filter = filter;
-    priv->speed = 1.0;
-    return AF_OK;
-}
-
-#define OPT_BASE_STRUCT struct priv
-
-const struct af_info af_info_forcespeed = {
-    .info = "Force audio speed",
-    .name = "forcespeed",
-    .open = af_open,
-    .priv_size = sizeof(struct priv),
-};

audio/filter/af_format.c

@@ -80,12 +80,12 @@ static int control(struct af_instance *af, int cmd, void *arg)
         force_out_params(af, out);
 
         if (in->nch != out->nch || in->bps != out->bps) {
-            MP_ERR(af, "Forced input/output formats are incompatible.\n");
+            MP_ERR(af, "[af_format] Forced input/output formats are incompatible.\n");
             return AF_ERROR;
         }
 
         if (priv->fail) {
-            MP_ERR(af, "Failing on purpose.\n");
+            MP_ERR(af, "[af_format] Failing on purpose.\n");
             return AF_ERROR;
         }
 

audio/filter/af_hrtf.c

@@ -280,13 +280,8 @@ static inline void update_ch(af_hrtf_t *s, short *in, const int k)
     }
 
     /* 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];
-    }
+    s->ba_l[k] = in[0] + in[4] + in[2];
+    s->ba_r[k] = in[4] + in[1] + in[3];
 }
 
 static void clear_coeff(af_hrtf_t *s, float *c)

audio/filter/af_scaletempo.c

@@ -271,7 +271,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
     switch (cmd) {
     case AF_CONTROL_REINIT: {
         struct mp_audio *data = (struct mp_audio *)arg;
-        float srate  = data->rate / 1000.0;
+        float srate = data->rate / 1000;
         int nch = data->nch;
         int use_int = 0;
 

audio/mixer.c

@@ -36,7 +36,7 @@ struct mixer {
     struct ao *ao;
     struct af_stream *af;
     // Static, dependent on ao/softvol settings
-    bool softvol;                       // use AO (false) or af_volume (true)
+    bool softvol;                       // use AO (true) or af_volume (false)
     bool ao_softvol;                    // AO has private or per-app volume
     bool emulate_mute;                  // if true, emulate mute with volume=0
     // Last known values (possibly out of sync with reality)
@@ -72,7 +72,7 @@ bool mixer_audio_initialized(struct mixer *mixer)
 float mixer_getneutralvolume(struct mixer *mixer)
 {
     // gain == 1
-    return mixer->softvol ? 1.0 / mixer->opts->softvol_max * 100.0 * 100.0 : 100;
+    return 1.0 / mixer->opts->softvol_max * 100.0 * 100.0;
 }
 
 static void checkvolume(struct mixer *mixer)

audio/out/ao.c

@@ -29,7 +29,6 @@
 #include "audio/format.h"
 #include "audio/audio.h"
 
-#include "input/input.h"
 #include "options/options.h"
 #include "options/m_config.h"
 #include "common/msg.h"
@@ -64,12 +63,12 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #if HAVE_ALSA
     &audio_out_alsa,
 #endif
-#if HAVE_WASAPI
-    &audio_out_wasapi,
-#endif
 #if HAVE_DSOUND
     &audio_out_dsound,
 #endif
+#if HAVE_WASAPI
+    &audio_out_wasapi,
+#endif
 #if HAVE_OSS_AUDIO
     &audio_out_oss,
 #endif
@@ -88,9 +87,6 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #endif
 #if HAVE_SNDIO
     &audio_out_sndio,
-#endif
-#if HAVE_COREAUDIO
-    &audio_out_coreaudio_exclusive,
 #endif
     &audio_out_null,
     // should not be auto-selected:
@@ -98,6 +94,9 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #if HAVE_ENCODING
     &audio_out_lavc,
 #endif
+#if HAVE_COREAUDIO
+    &audio_out_coreaudio_exclusive,
+#endif
 #if HAVE_RSOUND
     &audio_out_rsound,
 #endif
@@ -129,34 +128,12 @@ const struct m_obj_list ao_obj_list = {
     .allow_trailer = true,
 };
 
-// Return true if the ao with the given name matches the opt (--audio-device
-// contents). If opt is empty, matching always succeeds. If out_device is not
-// NULL, it will be set to the implied audio device (or NULL).
-static bool match_ao_driver(const char *ao_name, char *opt, char **out_device)
+static struct ao *ao_create(bool probing, struct mpv_global *global,
+                            struct input_ctx *input_ctx,
+                            struct encode_lavc_context *encode_lavc_ctx,
+                            int samplerate, int format, struct mp_chmap channels,
+                            char *name, char **args)
 {
-    bstr ao_nameb = bstr0(ao_name);
-    bstr optb = bstr0(opt);
-
-    char *dummy;
-    if (!out_device)
-        out_device = &dummy;
-    *out_device = NULL;
-
-    if (!optb.len || bstr_equals0(optb, "auto"))
-        return true;
-    if (!bstr_startswith(optb, ao_nameb))
-        return false;
-    if (optb.len > ao_nameb.len && optb.start[ao_nameb.len] != '/')
-        return false;
-    char *split = strchr(opt, '/');
-    *out_device = split ? split + 1 : NULL;
-    return true;
-}
-
-static struct ao *ao_alloc(bool probing, struct mpv_global *global,
-                           struct input_ctx *input_ctx, char *name, char **args)
-{
-    struct MPOpts *opts = global->opts;
     struct mp_log *log = mp_log_new(NULL, global->log, "ao");
     struct m_obj_desc desc;
     if (!m_obj_list_find(&desc, &ao_obj_list, bstr0(name))) {
@@ -169,62 +146,34 @@ static struct ao *ao_alloc(bool probing, struct mpv_global *global,
     *ao = (struct ao) {
         .driver = desc.p,
         .probing = probing,
+        .encode_lavc_ctx = encode_lavc_ctx,
         .input_ctx = input_ctx,
+        .samplerate = samplerate,
+        .channels = channels,
+        .format = format,
         .log = mp_log_new(ao, log, name),
-        .def_buffer = opts->audio_buffer,
+        .def_buffer = global->opts->audio_buffer,
     };
+    if (ao->driver->encode != !!ao->encode_lavc_ctx)
+        goto error;
     struct m_config *config = m_config_from_obj_desc(ao, ao->log, &desc);
-    if (m_config_apply_defaults(config, name, opts->ao_defs) < 0)
+    if (m_config_apply_defaults(config, name, global->opts->ao_defs) < 0)
         goto error;
     if (m_config_set_obj_params(config, args) < 0)
         goto error;
     ao->priv = config->optstruct;
-    return ao;
-error:
-    talloc_free(ao);
-    return NULL;
-}
 
-static struct ao *ao_alloc_pb(bool probing, struct mpv_global *global,
-                              struct input_ctx *input_ctx,
-                              struct encode_lavc_context *encode_lavc_ctx,
-                              int samplerate, int format, struct mp_chmap channels,
-                              char *name, char **args)
-{
-    struct MPOpts *opts = global->opts;
-    struct ao *ao = ao_alloc(probing, global, input_ctx, name, args);
-    if (!ao)
-        return NULL;
-    ao->samplerate = samplerate;
-    ao->channels = channels;
-    ao->format = format;
-    ao->encode_lavc_ctx = encode_lavc_ctx;
-    if (ao->driver->encode != !!ao->encode_lavc_ctx) {
-        talloc_free(ao);
-        return NULL;
-    }
-
-    match_ao_driver(ao->driver->name, opts->audio_device, &ao->device);
-    ao->device = talloc_strdup(ao, ao->device);
-
-    ao->client_name = talloc_strdup(ao, opts->audio_client_name);
-
-    return ao;
-}
-
-static int ao_init(struct ao *ao)
-{
+    char *chmap = mp_chmap_to_str(&ao->channels);
     MP_VERBOSE(ao, "requested format: %d Hz, %s channels, %s\n",
-               ao->samplerate, mp_chmap_to_str(&ao->channels),
-               af_fmt_to_str(ao->format));
+               ao->samplerate, chmap, af_fmt_to_str(ao->format));
+    talloc_free(chmap);
 
     ao->api = ao->driver->play ? &ao_api_push : &ao_api_pull;
     ao->api_priv = talloc_zero_size(ao, ao->api->priv_size);
     assert(!ao->api->priv_defaults && !ao->api->options);
 
-    int r = ao->driver->init(ao);
-    if (r < 0)
-        return r;
+    if (ao->driver->init(ao) < 0)
+        goto error;
 
     ao->sstride = af_fmt2bps(ao->format);
     ao->num_planes = 1;
@@ -235,40 +184,18 @@ static int ao_init(struct ao *ao)
     }
     ao->bps = ao->samplerate * ao->sstride;
 
-    if (!ao->device_buffer && ao->driver->get_space)
+    if (!ao->device_buffer && ao->driver->get_space) {
         ao->device_buffer = ao->driver->get_space(ao);
-    if (ao->device_buffer)
         MP_VERBOSE(ao, "device buffer: %d samples.\n", ao->device_buffer);
+    }
     ao->buffer = MPMAX(ao->device_buffer, ao->def_buffer * ao->samplerate);
     MP_VERBOSE(ao, "using soft-buffer of %d samples.\n", ao->buffer);
 
-    return ao->api->init(ao);
-}
+    if (ao->api->init(ao) < 0)
+        goto error;
 
-static struct ao *ao_create(bool probing, struct mpv_global *global,
-                            struct input_ctx *input_ctx,
-                            struct encode_lavc_context *encode_lavc_ctx,
-                            int samplerate, int format, struct mp_chmap channels,
-                            char *name, char **args)
-{
-    struct ao *ao = ao_alloc_pb(probing, global, input_ctx, encode_lavc_ctx,
-                                samplerate, format, channels, name, args);
-    if (ao && ao_init(ao) >= 0)
-        return ao;
-    // Silly exception for coreaudio spdif redirection
-    if (ao && ao->redirect) {
-        char redirect[80], device[80];
-        snprintf(redirect, sizeof(redirect), "%s", ao->redirect);
-        snprintf(device, sizeof(device), "%s", ao->device ? ao->device : "");
-        talloc_free(ao);
-        ao = ao_alloc_pb(probing, global, input_ctx, encode_lavc_ctx,
-                         samplerate, format, channels, redirect, args);
-        if (ao) {
-            ao->device = talloc_strdup(ao, device);
-            if (ao_init(ao) >= 0)
-                return ao;
-        }
-    }
+    return ao;
+error:
     talloc_free(ao);
     return NULL;
 }
@@ -278,10 +205,9 @@ struct ao *ao_init_best(struct mpv_global *global,
                         struct encode_lavc_context *encode_lavc_ctx,
                         int samplerate, int format, struct mp_chmap channels)
 {
-    struct MPOpts *opts = global->opts;
     struct mp_log *log = mp_log_new(NULL, global->log, "ao");
     struct ao *ao = NULL;
-    struct m_obj_settings *ao_list = opts->audio_driver_list;
+    struct m_obj_settings *ao_list = global->opts->audio_driver_list;
     if (ao_list && ao_list[0].name) {
         for (int n = 0; ao_list[n].name; n++) {
             if (strlen(ao_list[n].name) == 0)
@@ -298,21 +224,15 @@ struct ao *ao_init_best(struct mpv_global *global,
         }
         goto done;
     }
-autoprobe: ;
+autoprobe:
     // now try the rest...
-    bool matched_dev = false;
     for (int i = 0; audio_out_drivers[i]; i++) {
-        char *name = (char *)audio_out_drivers[i]->name;
-        if (!match_ao_driver(name, opts->audio_device, NULL))
-            continue;
-        matched_dev = true;
         ao = ao_create(true, global, input_ctx, encode_lavc_ctx,
-                       samplerate, format, channels, name, NULL);
+                       samplerate, format, channels,
+                       (char *)audio_out_drivers[i]->name, NULL);
         if (ao)
             goto done;
     }
-    if (!matched_dev)
-        mp_err(log, "--audio-device option refers to missing output driver.\n");
 done:
     talloc_free(log);
     return ao;
@@ -396,22 +316,6 @@ bool ao_eof_reached(struct ao *ao)
     return ao->api->get_eof ? ao->api->get_eof(ao) : true;
 }
 
-// Query the AO_EVENT_*s as requested by the events parameter, and return them.
-int ao_query_and_reset_events(struct ao *ao, int events)
-{
-    int actual_events = 0;
-    if (atomic_load(&ao->request_reload)) // don't need to reset it
-        actual_events |= AO_EVENT_RELOAD;
-    return actual_events & events;
-}
-
-// Request that the player core destroys and recreates the AO.
-void ao_request_reload(struct ao *ao)
-{
-    atomic_store(&ao->request_reload, true);
-    mp_input_wakeup(ao->input_ctx);
-}
-
 bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
                          struct mp_chmap *map)
 {
@@ -448,48 +352,3 @@ bool ao_untimed(struct ao *ao)
 {
     return ao->untimed;
 }
-
-struct ao_device_list *ao_get_device_list(struct mpv_global *global)
-{
-    struct ao_device_list *list = talloc_zero(NULL, struct ao_device_list);
-    MP_TARRAY_APPEND(list, list->devices, list->num_devices,
-        (struct ao_device_desc){"auto", "Autoselect device"});
-    for (int n = 0; audio_out_drivers[n]; n++) {
-        const struct ao_driver *d = audio_out_drivers[n];
-        if (d == &audio_out_null)
-            break; // don't add unsafe/special entries
-        struct ao *ao = ao_alloc(true, global, NULL, (char *)d->name, NULL);
-        if (!ao)
-            continue;
-        int num = list->num_devices;
-        if (d->list_devs)
-            d->list_devs(ao, list);
-        // Add at least a default entry
-        if (list->num_devices == num)
-            ao_device_list_add(list, ao, &(struct ao_device_desc){"", "Default"});
-        talloc_free(ao);
-    }
-    return list;
-}
-
-void ao_device_list_add(struct ao_device_list *list, struct ao *ao,
-                        struct ao_device_desc *e)
-{
-    struct ao_device_desc c = *e;
-    const char *dname = ao->driver->name;
-    c.name = c.name[0] ? talloc_asprintf(list, "%s/%s", dname, c.name)
-                       : talloc_strdup(list, dname);
-    c.desc = talloc_strdup(list, c.desc);
-    MP_TARRAY_APPEND(list, list->devices, list->num_devices, c);
-}
-
-void ao_print_devices(struct mpv_global *global, struct mp_log *log)
-{
-    struct ao_device_list *list = ao_get_device_list(global);
-    mp_info(log, "List of detected audio devices:\n");
-    for (int n = 0; n < list->num_devices; n++) {
-        struct ao_device_desc *desc = &list->devices[n];
-        mp_info(log, "  '%s' (%s)\n", desc->name, desc->desc);
-    }
-    talloc_free(list);
-}

audio/out/ao.h

@@ -46,25 +46,11 @@ enum aocontrol {
 // data might be written again, instead of closing the AO.
 #define AOPLAY_FINAL_CHUNK 1
 
-enum {
-    AO_EVENT_RELOAD = 1,
-};
-
 typedef struct ao_control_vol {
     float left;
     float right;
 } ao_control_vol_t;
 
-struct ao_device_desc {
-    const char *name;   // symbolic name; will be set on ao->device
-    const char *desc;   // verbose human readable name
-};
-
-struct ao_device_list {
-    struct ao_device_desc *devices;
-    int num_devices;
-};
-
 struct ao;
 struct mpv_global;
 struct input_ctx;
@@ -89,10 +75,5 @@ void ao_pause(struct ao *ao);
 void ao_resume(struct ao *ao);
 void ao_drain(struct ao *ao);
 bool ao_eof_reached(struct ao *ao);
-int ao_query_and_reset_events(struct ao *ao, int events);
-void ao_request_reload(struct ao *ao);
-
-struct ao_device_list *ao_get_device_list(struct mpv_global *global);
-void ao_print_devices(struct mpv_global *global, struct mp_log *log);
 
 #endif /* MPLAYER_AUDIO_OUT_H */

audio/out/ao_alsa.c

@@ -40,10 +40,10 @@
 #include "common/msg.h"
 #include "osdep/endian.h"
 
-#include <alsa/asoundlib.h>
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
 
-#define HAVE_CHMAP_API \
-    (defined(SND_CHMAP_API_VERSION) && SND_CHMAP_API_VERSION >= (1 << 16))
+#include <alsa/asoundlib.h>
 
 #include "ao.h"
 #include "internal.h"
@@ -54,7 +54,7 @@ struct priv {
     snd_pcm_format_t alsa_fmt;
     int can_pause;
     snd_pcm_sframes_t prepause_frames;
-    double delay_before_pause;
+    float delay_before_pause;
     int buffersize; // in frames
     int outburst; // in frames
 
@@ -78,12 +78,10 @@ struct priv {
         } \
     } while (0)
 
-#define CHECK_ALSA_WARN(message) \
-    do { \
-        if (err < 0) \
-            MP_WARN(ao, "%s: %s\n", (message), snd_strerror(err)); \
-    } while (0)
+static float get_delay(struct ao *ao);
+static void uninit(struct ao *ao);
 
+/* to set/get/query special features/parameters */
 static int control(struct ao *ao, enum aocontrol cmd, void *arg)
 {
     struct priv *p = ao->priv;
@@ -105,8 +103,10 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
         if (AF_FORMAT_IS_SPECIAL(ao->format))
             return CONTROL_FALSE;
 
+        //allocate simple id
         snd_mixer_selem_id_alloca(&sid);
 
+        //sets simple-mixer index and name
         snd_mixer_selem_id_set_index(sid, p->cfg_mixer_index);
         snd_mixer_selem_id_set_name(sid, p->cfg_mixer_name);
 
@@ -138,6 +138,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
             ao_control_vol_t *vol = arg;
             set_vol = vol->left / f_multi + pmin + 0.5;
 
+            //setting channels
             err = snd_mixer_selem_set_playback_volume
                     (elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol);
             CHECK_ALSA_ERROR("Error setting left channel");
@@ -149,7 +150,8 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
                     (elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol);
             CHECK_ALSA_ERROR("Error setting right channel");
             MP_DBG(ao, "right=%li, pmin=%li, pmax=%li, mult=%f\n",
-                   set_vol, pmin, pmax, f_multi);
+                   set_vol, pmin, pmax,
+                   f_multi);
             break;
         }
         case AOCONTROL_GET_VOLUME: {
@@ -229,98 +231,6 @@ static int find_alsa_format(int af_format)
     return SND_PCM_FORMAT_UNKNOWN;
 }
 
-#if HAVE_CHMAP_API
-
-static const int alsa_to_mp_channels[][2] = {
-    {SND_CHMAP_FL,      MP_SP(FL)},
-    {SND_CHMAP_FR,      MP_SP(FR)},
-    {SND_CHMAP_RL,      MP_SP(BL)},
-    {SND_CHMAP_RR,      MP_SP(BR)},
-    {SND_CHMAP_FC,      MP_SP(FC)},
-    {SND_CHMAP_LFE,     MP_SP(LFE)},
-    {SND_CHMAP_SL,      MP_SP(SL)},
-    {SND_CHMAP_SR,      MP_SP(SR)},
-    {SND_CHMAP_RC,      MP_SP(BC)},
-    {SND_CHMAP_FLC,     MP_SP(FLC)},
-    {SND_CHMAP_FRC,     MP_SP(FRC)},
-    {SND_CHMAP_FLW,     MP_SP(WL)},
-    {SND_CHMAP_FRW,     MP_SP(WR)},
-    {SND_CHMAP_TC,      MP_SP(TC)},
-    {SND_CHMAP_TFL,     MP_SP(TFL)},
-    {SND_CHMAP_TFR,     MP_SP(TFR)},
-    {SND_CHMAP_TFC,     MP_SP(TFC)},
-    {SND_CHMAP_TRL,     MP_SP(TBL)},
-    {SND_CHMAP_TRR,     MP_SP(TBR)},
-    {SND_CHMAP_TRC,     MP_SP(TBC)},
-    {SND_CHMAP_MONO,    MP_SP(FC)},
-    {SND_CHMAP_LAST,    MP_SPEAKER_ID_COUNT}
-};
-
-static int find_mp_channel(int alsa_channel)
-{
-    for (int i = 0; alsa_to_mp_channels[i][1] != MP_SPEAKER_ID_COUNT; i++) {
-        if (alsa_to_mp_channels[i][0] == alsa_channel)
-            return alsa_to_mp_channels[i][1];
-    }
-
-    return MP_SPEAKER_ID_COUNT;
-}
-
-static int find_alsa_channel(int mp_channel)
-{
-    for (int i = 0; alsa_to_mp_channels[i][1] != MP_SPEAKER_ID_COUNT; i++) {
-        if (alsa_to_mp_channels[i][1] == mp_channel)
-            return alsa_to_mp_channels[i][0];
-    }
-
-    return SND_CHMAP_UNKNOWN;
-}
-
-static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
-{
-    struct priv *p = ao->priv;
-    struct mp_chmap_sel chmap_sel = {.tmp = p};
-
-    snd_pcm_chmap_query_t **maps = snd_pcm_query_chmaps(p->alsa);
-    if (!maps)
-        return false;
-
-    for (int i = 0; maps[i] != NULL; i++) {
-        if (maps[i]->map.channels > MP_NUM_CHANNELS) {
-            MP_VERBOSE(ao, "skipping ALSA channel map with too many channels.\n");
-            continue;
-        }
-
-        struct mp_chmap entry = {.num = maps[i]->map.channels};
-        for (int c = 0; c < entry.num; c++)
-            entry.speaker[c] = find_mp_channel(maps[i]->map.pos[c]);
-
-        if (mp_chmap_is_valid(&entry)) {
-            MP_VERBOSE(ao, "Got supported channel map: %s (type %s)\n",
-                       mp_chmap_to_str(&entry),
-                       snd_pcm_chmap_type_name(maps[i]->type));
-            mp_chmap_sel_add_map(&chmap_sel, &entry);
-        } else {
-            char tmp[128];
-            if (snd_pcm_chmap_print(&maps[i]->map, sizeof(tmp), tmp) > 0)
-                MP_VERBOSE(ao, "skipping unknown ALSA channel map: %s\n", tmp);
-        }
-    }
-
-    snd_pcm_free_chmaps(maps);
-
-    return ao_chmap_sel_adjust(ao, &chmap_sel, chmap);
-}
-
-#else /* HAVE_CHMAP_API */
-
-static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
-{
-    return false;
-}
-
-#endif /* else HAVE_CHMAP_API */
-
 // Lists device names and their implied channel map.
 // The second item must be resolvable with mp_chmap_from_str().
 // Source: http://www.alsa-project.org/main/index.php/DeviceNames
@@ -338,9 +248,11 @@ static const char *const device_channel_layouts[][2] = {
     {"surround71",      "fl-fr-bl-br-fc-lfe-sl-sr"},
 };
 
-#define NUM_ALSA_CHMAPS MP_ARRAY_SIZE(device_channel_layouts)
+#define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0]))
 
-static const char *select_chmap(struct ao *ao, struct mp_chmap *chmap)
+#define NUM_ALSA_CHMAPS ARRAY_LEN(device_channel_layouts)
+
+static const char *select_chmap(struct ao *ao)
 {
     struct mp_chmap_sel sel = {0};
     struct mp_chmap maps[NUM_ALSA_CHMAPS];
@@ -349,16 +261,18 @@ static const char *select_chmap(struct ao *ao, struct mp_chmap *chmap)
         mp_chmap_sel_add_map(&sel, &maps[n]);
     };
 
-    if (!ao_chmap_sel_adjust(ao, &sel, chmap))
-        return "default";
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        return NULL;
 
     for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
-        if (mp_chmap_equals(chmap, &maps[n]))
+        if (mp_chmap_equals(&ao->channels, &maps[n]))
             return device_channel_layouts[n][0];
     }
 
+    char *name = mp_chmap_to_str(&ao->channels);
     MP_ERR(ao, "channel layout %s (%d ch) not supported.\n",
-           mp_chmap_to_str(chmap), chmap->num);
+           name, ao->channels.num);
+    talloc_free(name);
     return "default";
 }
 
@@ -379,44 +293,34 @@ static int map_iec958_srate(int srate)
     }
 }
 
-// ALSA device strings can have parameters. They are usually appended to the
-// device name. Since there can be various forms, and we (sometimes) want to
-// append them to unknown device strings, which possibly already include params.
-static char *append_params(void *ta_parent, const char *device, const char *p)
-{
-    if (!p || !p[0])
-        return talloc_strdup(ta_parent, device);
-
-    int len = strlen(device);
-    char *end = strchr(device, ':');
-    if (!end) {
-        /* no existing parameters: add it behind device name */
-        return talloc_asprintf(ta_parent, "%s:%s", device, p);
-    } else if (end[1] == '\0') {
-        /* ":" but no parameters */
-        return talloc_asprintf(ta_parent, "%s%s", device, p);
-    } else if (end[1] == '{' && device[len - 1] == '}') {
-        /* parameters in config syntax: add it inside the { } block */
-        return talloc_asprintf(ta_parent, "%.*s %s}", len - 1, device, p);
-    } else {
-        /* a simple list of parameters: add it at the end of the list */
-        return talloc_asprintf(ta_parent, "%s,%s", device, p);
-    }
-    abort();
-}
-
 static int try_open_device(struct ao *ao, const char *device, int open_mode)
 {
     struct priv *p = ao->priv;
 
     if (AF_FORMAT_IS_IEC61937(ao->format)) {
         void *tmp = talloc_new(NULL);
+        /* to set the non-audio bit, use AES0=6 */
         char *params = talloc_asprintf(tmp,
                         "AES0=%d,AES1=%d,AES2=0,AES3=%d",
                         IEC958_AES0_NONAUDIO | IEC958_AES0_PRO_EMPHASIS_NONE,
                         IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
                         map_iec958_srate(ao->samplerate));
-        const char *ac3_device = append_params(tmp, device, params);
+        const char *ac3_device = device;
+        int len = strlen(device);
+        char *end = strchr(device, ':');
+        if (!end) {
+            /* no existing parameters: add it behind device name */
+            ac3_device = talloc_asprintf(tmp, "%s:%s", device, params);
+        } else if (end[1] == '\0') {
+            /* ":" but no parameters */
+            ac3_device = talloc_asprintf(tmp, "%s%s", device, params);
+        } else if (end[1] == '{' && device[len - 1] == '}') {
+            /* parameters in config syntax: add it inside the { } block */
+            ac3_device = talloc_asprintf(tmp, "%.*s %s}", len - 1, device, params);
+        } else {
+            /* a simple list of parameters: add it at the end of the list */
+            ac3_device = talloc_asprintf(tmp, "%s,%s", device, params);
+        }
         int err = snd_pcm_open
                     (&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, open_mode);
         talloc_free(tmp);
@@ -427,56 +331,51 @@ static int try_open_device(struct ao *ao, const char *device, int open_mode)
     return snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, open_mode);
 }
 
-static void uninit(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    if (p->alsa) {
-        int err;
-
-        err = snd_pcm_close(p->alsa);
-        CHECK_ALSA_ERROR("pcm close error");
-    }
-
-alsa_error: ;
-}
-
+/*
+    open & setup audio device
+    return: 0=success -1=fail
+ */
 static int init(struct ao *ao)
 {
-    struct priv *p = ao->priv;
     int err;
+    snd_pcm_uframes_t chunk_size;
+    snd_pcm_uframes_t bufsize;
+    snd_pcm_uframes_t boundary;
+
+    struct priv *p = ao->priv;
 
     if (!p->cfg_ni)
         ao->format = af_fmt_from_planar(ao->format);
 
-    struct mp_chmap implied_chmap = ao->channels;
+    /* switch for spdif
+     * sets opening sequence for SPDIF
+     * sets also the playback and other switches 'on the fly'
+     * while opening the abstract alias for the spdif subdevice
+     * 'iec958'
+     */
     const char *device;
     if (AF_FORMAT_IS_IEC61937(ao->format)) {
         device = "iec958";
         MP_VERBOSE(ao, "playing AC3/iec61937/iec958, %i channels\n",
                    ao->channels.num);
     } else {
-        device = select_chmap(ao, &implied_chmap);
-        // Not-default likely means a hw device - enable software conversions.
-        if (strcmp(device, "default") != 0)
+        device = select_chmap(ao);
+        if (strcmp(device, "default") != 0 && (ao->format & AF_FORMAT_F)) {
+            // hack - use the converter plugin (why the heck?)
             device = talloc_asprintf(ao, "plug:%s", device);
+        }
     }
-    const char *old_dev = device;
-    if (ao->device)
-        device = ao->device;
     if (p->cfg_device && p->cfg_device[0])
         device = p->cfg_device;
-    bool user_set_device = device != old_dev; // not strcmp()
 
     MP_VERBOSE(ao, "using device: %s\n", device);
     MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());
 
-    err = try_open_device(ao, device, p->cfg_block ? 0 : SND_PCM_NONBLOCK);
+    int open_mode = p->cfg_block ? 0 : SND_PCM_NONBLOCK;
+    //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
+    err = try_open_device(ao, device, open_mode);
     if (err < 0) {
-        if (err == -EBUSY && !user_set_device && strcmp(device, "default") != 0) {
-            MP_WARN(ao, "Device '%s' busy, retrying default.\n", device);
-            err = try_open_device(ao, "default", 0);
-        } else if (err != -EBUSY && !p->cfg_block) {
+        if (err != -EBUSY && !p->cfg_block) {
             MP_WARN(ao, "Open in nonblock-mode "
                     "failed, trying to open in block-mode.\n");
             err = try_open_device(ao, device, 0);
@@ -497,6 +396,7 @@ static int init(struct ao *ao)
     snd_pcm_hw_params_alloca(&alsa_hwparams);
     snd_pcm_sw_params_alloca(&alsa_swparams);
 
+    // setting hw-parameters
     err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
     CHECK_ALSA_ERROR("Unable to get initial parameters");
 
@@ -538,24 +438,11 @@ static int init(struct ao *ao)
     }
     CHECK_ALSA_ERROR("Unable to set access type");
 
-    struct mp_chmap dev_chmap = ao->channels;
-    if (query_chmaps(ao, &dev_chmap)) {
-        ao->channels = dev_chmap;
-    } else {
-        dev_chmap.num = 0;
-        ao->channels = implied_chmap;
-    }
-
     int num_channels = ao->channels.num;
     err = snd_pcm_hw_params_set_channels_near
             (p->alsa, alsa_hwparams, &num_channels);
     CHECK_ALSA_ERROR("Unable to set channels");
 
-    if (num_channels > MP_NUM_CHANNELS) {
-        MP_FATAL(ao, "Too many audio channels (%d).\n", num_channels);
-        goto alsa_error;
-    }
-
     if (num_channels != ao->channels.num) {
         MP_ERR(ao, "Couldn't get requested number of channels.\n");
         mp_chmap_from_channels_alsa(&ao->channels, num_channels);
@@ -574,82 +461,25 @@ static int init(struct ao *ao)
 
     err = snd_pcm_hw_params_set_buffer_time_near
             (p->alsa, alsa_hwparams, &(unsigned int){BUFFER_TIME}, NULL);
-    CHECK_ALSA_WARN("Unable to set buffer time near");
+    CHECK_ALSA_ERROR("Unable to set buffer time near");
 
     err = snd_pcm_hw_params_set_periods_near
             (p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
-    CHECK_ALSA_WARN("Unable to set periods");
+    CHECK_ALSA_ERROR("Unable to set periods");
 
     /* finally install hardware parameters */
     err = snd_pcm_hw_params(p->alsa, alsa_hwparams);
     CHECK_ALSA_ERROR("Unable to set hw-parameters");
 
-    /* end setting hw-params */
+    // end setting hw-params
 
-#if HAVE_CHMAP_API
-    if (mp_chmap_is_valid(&dev_chmap)) {
-        snd_pcm_chmap_t *alsa_chmap =
-            calloc(1, sizeof(*alsa_chmap) +
-                      sizeof(alsa_chmap->pos[0]) * dev_chmap.num);
-        if (!alsa_chmap)
-            goto alsa_error;
-
-        alsa_chmap->channels = dev_chmap.num;
-        for (int c = 0; c < dev_chmap.num; c++)
-            alsa_chmap->pos[c] = find_alsa_channel(dev_chmap.speaker[c]);
-
-        char tmp[128];
-        if (snd_pcm_chmap_print(alsa_chmap, sizeof(tmp), tmp) > 0)
-            MP_VERBOSE(ao, "trying to set ALSA channel map: %s\n", tmp);
-
-        err = snd_pcm_set_chmap(p->alsa, alsa_chmap);
-        if (err == -ENXIO) {
-            MP_WARN(ao, "Device does not support requested channel map\n");
-        } else {
-            CHECK_ALSA_WARN("Channel map setup failed");
-        }
-    }
-
-    snd_pcm_chmap_t *alsa_chmap = snd_pcm_get_chmap(p->alsa);
-    if (alsa_chmap) {
-        char tmp[128];
-        if (snd_pcm_chmap_print(alsa_chmap, sizeof(tmp), tmp) > 0)
-            MP_VERBOSE(ao, "channel map reported by ALSA: %s\n", tmp);
-
-        struct mp_chmap chmap = {.num = alsa_chmap->channels};
-        for (int c = 0; c < chmap.num; c++)
-            chmap.speaker[c] = find_mp_channel(alsa_chmap->pos[c]);
-
-        MP_VERBOSE(ao, "which we understand as: %s\n", mp_chmap_to_str(&chmap));
-
-        if (mp_chmap_is_valid(&chmap)) {
-            if (mp_chmap_equals(&chmap, &ao->channels)) {
-                MP_VERBOSE(ao, "which is what we requested.\n");
-            } else if (chmap.num == ao->channels.num) {
-                MP_VERBOSE(ao, "using the ALSA channel map.\n");
-                ao->channels = chmap;
-            } else {
-                MP_WARN(ao, "ALSA channel map conflicts with channel count!\n");
-            }
-        } else {
-            MP_WARN(ao, "Got unknown channel map from ALSA.\n");
-        }
-
-        if (ao->channels.num == 1)
-            ao->channels.speaker[0] = MP_SP(FC);
-
-        free(alsa_chmap);
-    }
-#endif
-
-    snd_pcm_uframes_t bufsize;
+    // gets buffersize for control
     err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize);
     CHECK_ALSA_ERROR("Unable to get buffersize");
 
     p->buffersize = bufsize;
     MP_VERBOSE(ao, "got buffersize=%i samples\n", p->buffersize);
 
-    snd_pcm_uframes_t chunk_size;
     err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL);
     CHECK_ALSA_ERROR("Unable to get period size");
 
@@ -660,7 +490,6 @@ static int init(struct ao *ao)
     err = snd_pcm_sw_params_current(p->alsa, alsa_swparams);
     CHECK_ALSA_ERROR("Unable to get sw-parameters");
 
-    snd_pcm_uframes_t boundary;
     err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary);
     CHECK_ALSA_ERROR("Unable to get boundary");
 
@@ -680,17 +509,40 @@ static int init(struct ao *ao)
     CHECK_ALSA_ERROR("Unable to set silence size");
 
     err = snd_pcm_sw_params(p->alsa, alsa_swparams);
-    CHECK_ALSA_ERROR("Unable to set sw-parameters");
+    CHECK_ALSA_ERROR("Unable to get sw-parameters");
 
     /* end setting sw-params */
 
     p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
 
+    MP_VERBOSE(ao, "opened: %d Hz/%d channels/%d bps/%d samples buffer/%s\n",
+               ao->samplerate, ao->channels.num, af_fmt2bits(ao->format),
+               p->buffersize, snd_pcm_format_description(p->alsa_fmt));
+
     return 0;
 
 alsa_error:
     uninit(ao);
     return -1;
+} // end init
+
+
+/* close audio device */
+static void uninit(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    if (p->alsa) {
+        int err;
+
+        err = snd_pcm_close(p->alsa);
+        CHECK_ALSA_ERROR("pcm close error");
+
+        MP_VERBOSE(ao, "uninit: pcm closed\n");
+    }
+
+alsa_error:
+    p->alsa = NULL;
 }
 
 static void drain(struct ao *ao)
@@ -699,46 +551,6 @@ static void drain(struct ao *ao)
     snd_pcm_drain(p->alsa);
 }
 
-static int get_space(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-    snd_pcm_status_t *status;
-    int err;
-
-    snd_pcm_status_alloca(&status);
-
-    err = snd_pcm_status(p->alsa, status);
-    CHECK_ALSA_ERROR("cannot get pcm status");
-
-    unsigned space = snd_pcm_status_get_avail(status);
-    if (space > p->buffersize) // Buffer underrun?
-        space = p->buffersize;
-    return space / p->outburst * p->outburst;
-
-alsa_error:
-    return 0;
-}
-
-/* delay in seconds between first and last sample in buffer */
-static double get_delay(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-    snd_pcm_sframes_t delay;
-
-    if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED)
-        return p->delay_before_pause;
-
-    if (snd_pcm_delay(p->alsa, &delay) < 0)
-        return 0;
-
-    if (delay < 0) {
-        /* underrun - move the application pointer forward to catch up */
-        snd_pcm_forward(p->alsa, -delay);
-        delay = 0;
-    }
-    return delay / (double)ao->samplerate;
-}
-
 static void audio_pause(struct ao *ao)
 {
     struct priv *p = ao->priv;
@@ -755,7 +567,7 @@ static void audio_pause(struct ao *ao)
         if (snd_pcm_delay(p->alsa, &p->prepause_frames) < 0
             || p->prepause_frames < 0)
             p->prepause_frames = 0;
-        p->delay_before_pause = p->prepause_frames / (double)ao->samplerate;
+        p->delay_before_pause = p->prepause_frames / (float)ao->samplerate;
 
         err = snd_pcm_drop(p->alsa);
         CHECK_ALSA_ERROR("pcm drop error");
@@ -792,6 +604,7 @@ static void audio_resume(struct ao *ao)
 alsa_error: ;
 }
 
+/* stop playing and empty buffers (for seeking/pause) */
 static void reset(struct ao *ao)
 {
     struct priv *p = ao->priv;
@@ -824,7 +637,8 @@ static int play(struct ao *ao, void **data, int samples, int flags)
             res = snd_pcm_writei(p->alsa, data[0], samples);
         }
 
-        if (res == -EINTR || res == -EAGAIN) { /* retry */
+        if (res == -EINTR) {
+            /* nothing to do */
             res = 0;
         } else if (res == -ESTRPIPE) {  /* suspend */
             audio_resume(ao);
@@ -843,6 +657,46 @@ alsa_error:
     return -1;
 }
 
+static int get_space(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    snd_pcm_status_t *status;
+    int err;
+
+    snd_pcm_status_alloca(&status);
+
+    err = snd_pcm_status(p->alsa, status);
+    CHECK_ALSA_ERROR("cannot get pcm status");
+
+    unsigned space = snd_pcm_status_get_avail(status);
+    if (space > p->buffersize) // Buffer underrun?
+        space = p->buffersize;
+    return space / p->outburst * p->outburst;
+
+alsa_error:
+    return 0;
+}
+
+/* delay in seconds between first and last sample in buffer */
+static float get_delay(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    snd_pcm_sframes_t delay;
+
+    if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED)
+        return p->delay_before_pause;
+
+    if (snd_pcm_delay(p->alsa, &delay) < 0)
+        return 0;
+
+    if (delay < 0) {
+        /* underrun - move the application pointer forward to catch up */
+        snd_pcm_forward(p->alsa, -delay);
+        delay = 0;
+    }
+    return (float)delay / (float)ao->samplerate;
+}
+
 #define MAX_POLL_FDS 20
 static int audio_wait(struct ao *ao, pthread_mutex_t *lock)
 {
@@ -877,34 +731,10 @@ alsa_error:
     return -1;
 }
 
-static void list_devs(struct ao *ao, struct ao_device_list *list)
-{
-    void **hints;
-    if (snd_device_name_hint(-1, "pcm", &hints) < 0)
-        return;
-
-    for (int n = 0; hints[n]; n++) {
-        char *name = snd_device_name_get_hint(hints[n], "NAME");
-        char *desc = snd_device_name_get_hint(hints[n], "DESC");
-        char *io = snd_device_name_get_hint(hints[n], "IOID");
-        if (io && strcmp(io, "Output") != 0)
-            continue;
-        char desc2[1024];
-        snprintf(desc2, sizeof(desc2), "%s", desc ? desc : "");
-        for (int i = 0; desc2[i]; i++) {
-            if (desc2[i] == '\n')
-                desc2[i] = '/';
-        }
-        ao_device_list_add(list, ao, &(struct ao_device_desc){name, desc2});
-    }
-
-    snd_device_name_free_hint(hints);
-}
-
 #define OPT_BASE_STRUCT struct priv
 
 const struct ao_driver audio_out_alsa = {
-    .description = "ALSA audio output",
+    .description = "ALSA-0.9.x-1.x audio output",
     .name      = "alsa",
     .init      = init,
     .uninit    = uninit,
@@ -918,7 +748,6 @@ const struct ao_driver audio_out_alsa = {
     .drain     = drain,
     .wait      = audio_wait,
     .wakeup    = ao_wakeup_poll,
-    .list_devs = list_devs,
     .priv_size = sizeof(struct priv),
     .priv_defaults = &(const struct priv) {
         .cfg_block = 1,

audio/out/ao_coreaudio.c

@@ -33,6 +33,10 @@ struct priv {
     AudioUnit audio_unit;
 
     uint64_t hw_latency_us;
+
+    // options
+    int opt_device_id;
+    int opt_list;
 };
 
 bool ca_layout_to_mp_chmap(struct ao *ao, AudioChannelLayout *layout,
@@ -145,15 +149,11 @@ static bool init_audiounit(struct ao *ao, AudioStreamBasicDescription asbd);
 
 static int init(struct ao *ao)
 {
-    if (AF_FORMAT_IS_IEC61937(ao->format)) {
-        MP_WARN(ao, "detected IEC61937, redirecting to coreaudio_exclusive\n");
-        ao->redirect = "coreaudio_exclusive";
-        return CONTROL_ERROR;
-    }
-
     struct priv *p = ao->priv;
 
-    OSStatus err = ca_select_device(ao, ao->device, &p->device);
+    if (p->opt_list) ca_print_device_list(ao);
+
+    OSStatus err = ca_select_device(ao, p->opt_device_id, &p->device);
     CHECK_CA_ERROR("failed to select device");
 
     if (!init_chmap(ao))
@@ -185,7 +185,7 @@ static bool init_chmap(struct ao *ao)
                        &layouts, &n_layouts);
     CHECK_CA_ERROR("could not get audio device prefered layouts");
 
-    struct mp_chmap_sel chmap_sel = {.tmp = p};
+    struct mp_chmap_sel chmap_sel = {0};
     for (int i = 0; i < n_layouts; i++) {
         struct mp_chmap chmap = {0};
         if (ca_layout_to_mp_chmap(ao, &layouts[i], &chmap))
@@ -221,9 +221,9 @@ static bool init_audiounit(struct ao *ao, AudioStreamBasicDescription asbd)
 
     AudioComponentDescription desc = (AudioComponentDescription) {
         .componentType         = kAudioUnitType_Output,
-        .componentSubType      = (ao->device) ?
-                                    kAudioUnitSubType_HALOutput :
-                                    kAudioUnitSubType_DefaultOutput,
+        .componentSubType      = (p->opt_device_id < 0) ?
+                                    kAudioUnitSubType_DefaultOutput :
+                                    kAudioUnitSubType_HALOutput,
         .componentManufacturer = kAudioUnitManufacturer_Apple,
         .componentFlags        = 0,
         .componentFlagsMask    = 0,
@@ -354,28 +354,22 @@ static void ca_log_layout(struct ao *ao, AudioChannelLayout *layout)
 
     AudioChannelDescription *descs = layout->mChannelDescriptions;
 
-    MP_VERBOSE(ao, "layout: tag: <%u>, bitmap: <%u>, "
-                   "descriptions <%u>\n",
-                   (unsigned) layout->mChannelLayoutTag,
-                   (unsigned) layout->mChannelBitmap,
-                   (unsigned) layout->mNumberChannelDescriptions);
+    MP_VERBOSE(ao, "layout: tag: <%d>, bitmap: <%d>, "
+                   "descriptions <%d>\n",
+                   layout->mChannelLayoutTag,
+                   layout->mChannelBitmap,
+                   layout->mNumberChannelDescriptions);
 
     for (int i = 0; i < layout->mNumberChannelDescriptions; i++) {
         AudioChannelDescription d = descs[i];
-        MP_VERBOSE(ao, " - description %d: label <%u, %u>, "
-                       " flags: <%u>, coords: <%f, %f, %f>\n", i,
-                       (unsigned) d.mChannelLabel,
-                       (unsigned) ca_label_to_mp_speaker_id(d.mChannelLabel),
-                       (unsigned) d.mChannelFlags,
+        MP_VERBOSE(ao, " - description %d: label <%d, %d>, flags: <%u>, "
+                       "coords: <%f, %f, %f>\n", i,
+                       d.mChannelLabel,
+                       ca_label_to_mp_speaker_id(d.mChannelLabel),
+                       d.mChannelFlags,
                        d.mCoordinates[0],
                        d.mCoordinates[1],
                        d.mCoordinates[2]);
-
-        if (i >= 32) {
-            MP_VERBOSE(ao, " detected more than 32 channel descriptions, "
-                       "skipping output");
-            break;
-        }
     }
 }
 
@@ -416,22 +410,14 @@ bool ca_layout_to_mp_chmap(struct ao *ao, AudioChannelLayout *layout,
     //   to the waveextensible definition: this is the kind of
     //   descriptions we process here.
 
-    if (layout->mNumberChannelDescriptions > MP_NUM_CHANNELS) {
-        MP_VERBOSE(ao, "layout has too many descriptions (%u, max: %d)\n",
-                   (unsigned) layout->mNumberChannelDescriptions,
-                   MP_NUM_CHANNELS);
-        return false;
-    }
-
     for (int n = 0; n < layout->mNumberChannelDescriptions; n++) {
         AudioChannelLabel label = layout->mChannelDescriptions[n].mChannelLabel;
         uint8_t speaker = ca_label_to_mp_speaker_id(label);
         if (label == kAudioChannelLabel_Unknown)
             continue;
         if (speaker < 0) {
-            MP_VERBOSE(ao, "channel label=%u unusable to build channel "
-                           "bitmap, skipping layout\n", (unsigned) label);
-            goto coreaudio_error;
+            MP_VERBOSE(ao, "channel label=%d unusable to build channel "
+                           "bitmap, skipping layout\n", label);
         } else {
             chmap->speaker[n] = speaker;
             chmap->num = n + 1;
@@ -454,6 +440,10 @@ const struct ao_driver audio_out_coreaudio = {
     .control   = control,
     .pause     = stop,
     .resume    = start,
-    .list_devs = ca_get_device_list,
     .priv_size = sizeof(struct priv),
+    .options = (const struct m_option[]) {
+        OPT_INT("device_id", opt_device_id, 0, OPTDEF_INT(-1)),
+        OPT_FLAG("list", opt_list, 0),
+        {0}
+    },
 };

audio/out/ao_coreaudio_device.c

@@ -333,6 +333,10 @@ struct priv {
     bool changed_mixing;
     int stream_asbd_changed;
     bool muted;
+
+    // options
+    int opt_device_id;
+    int opt_list;
 };
 
 static int get_ring_size(struct ao *ao)
@@ -398,7 +402,9 @@ static int init(struct ao *ao)
 {
     struct priv *p = ao->priv;
 
-    OSStatus err = ca_select_device(ao, ao->device, &p->device);
+    if (p->opt_list) ca_print_device_list(ao);
+
+    OSStatus err = ca_select_device(ao, p->opt_device_id, &p->device);
     CHECK_CA_ERROR("failed to select device");
 
     ao->format = af_fmt_from_planar(ao->format);
@@ -581,12 +587,12 @@ static int get_space(struct ao *ao)
     return mp_ring_available(p->buffer) / ao->sstride;
 }
 
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     // FIXME: should also report the delay of coreaudio itself (hardware +
     // internal buffers)
     struct priv *p = ao->priv;
-    return mp_ring_buffered(p->buffer) / (double)ao->bps;
+    return mp_ring_buffered(p->buffer) / (float)ao->bps;
 }
 
 static void uninit(struct ao *ao)
@@ -654,7 +660,6 @@ const struct ao_driver audio_out_coreaudio_exclusive = {
     .reset     = reset,
     .pause     = audio_pause,
     .resume    = audio_resume,
-    .list_devs = ca_get_device_list,
     .priv_size = sizeof(struct priv),
     .priv_defaults = &(const struct priv){
         .muted = false,
@@ -664,4 +669,9 @@ const struct ao_driver audio_out_coreaudio_exclusive = {
         .stream_idx = -1,
         .changed_mixing = false,
     },
+    .options = (const struct m_option[]) {
+        OPT_INT("device_id", opt_device_id, 0, OPTDEF_INT(-1)),
+        OPT_FLAG("list", opt_list, 0),
+        {0}
+    },
 };

audio/out/ao_coreaudio_properties.c

@@ -62,7 +62,7 @@ OSStatus ca_get_ary(AudioObjectID id, ca_scope scope, ca_sel selector,
     err = AudioObjectGetPropertyDataSize(id, &p_addr, 0, NULL, &p_size);
     CHECK_CA_ERROR_SILENT_L(coreaudio_error);
 
-    *data = talloc_zero_size(NULL, p_size);
+    *data = talloc_size(NULL, p_size);
     *elements = p_size / element_size;
 
     err = ca_get(id, scope, selector, p_size, *data);
@@ -87,7 +87,7 @@ OSStatus ca_get_str(AudioObjectID id, ca_scope scope, ca_sel selector,
         CFStringGetMaximumSizeForEncoding(
             CFStringGetLength(string), CA_CFSTR_ENCODING) + 1;
 
-    *data = talloc_zero_size(NULL, size);
+    *data = talloc_size(NULL, size);
     CFStringGetCString(string, *data, size, CA_CFSTR_ENCODING);
     CFRelease(string);
 coreaudio_error:

audio/out/ao_coreaudio_utils.c

@@ -28,48 +28,42 @@
 #include "osdep/endian.h"
 #include "audio/format.h"
 
-static bool ca_is_output_device(struct ao *ao, AudioDeviceID dev)
+void ca_print_device_list(struct ao *ao)
 {
-    size_t n_buffers;
-    AudioBufferList *buffers;
-    const ca_scope scope = kAudioDevicePropertyStreamConfiguration;
-    CA_GET_ARY_O(dev, scope, &buffers, &n_buffers);
-    talloc_free(buffers);
-    return n_buffers > 0;
-}
+    char *help = talloc_strdup(NULL, "Available output devices:\n");
 
-void ca_get_device_list(struct ao *ao, struct ao_device_list *list)
-{
     AudioDeviceID *devs;
     size_t n_devs;
+
     OSStatus err =
         CA_GET_ARY(kAudioObjectSystemObject, kAudioHardwarePropertyDevices,
                    &devs, &n_devs);
+
     CHECK_CA_ERROR("Failed to get list of output devices.");
+
     for (int i = 0; i < n_devs; i++) {
-        if (!ca_is_output_device(ao, devs[i]))
-            continue;
-        char name[32];
-        char *desc;
-        sprintf(name, "%d", devs[i]);
-        err = CA_GET_STR(devs[i], kAudioObjectPropertyName, &desc);
-        if (err != noErr)
-            desc = "Unknown";
-        ao_device_list_add(list, ao, &(struct ao_device_desc){name, desc});
+        char *name;
+        err = CA_GET_STR(devs[i], kAudioObjectPropertyName, &name);
+
+        if (err == noErr)
+            talloc_steal(devs, name);
+        else
+            name = "Unknown";
+
+        help = talloc_asprintf_append(
+                help, "  * %s (id: %" PRIu32 ")\n", name, devs[i]);
     }
+
     talloc_free(devs);
+
 coreaudio_error:
-    return;
+    MP_INFO(ao, "%s", help);
+    talloc_free(help);
 }
 
-OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device)
+OSStatus ca_select_device(struct ao *ao, int selection, AudioDeviceID *device)
 {
     OSStatus err = noErr;
-    int selection = name ? strtol(name, (char **)NULL, 10) : -1;
-    if (errno == EINVAL || errno == ERANGE) {
-        selection = -1;
-        MP_WARN(ao, "device identifier '%s' is invalid\n", name);
-    }
     *device = 0;
     if (selection < 0) {
         // device not set by user, get the default one
@@ -82,14 +76,14 @@ OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device)
     }
 
     if (mp_msg_test(ao->log, MSGL_V)) {
-        char *desc;
-        err = CA_GET_STR(*device, kAudioObjectPropertyName, &desc);
+        char *name;
+        err = CA_GET_STR(*device, kAudioObjectPropertyName, &name);
         CHECK_CA_ERROR("could not get selected audio device name");
 
         MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
-                       desc, *device);
+                       name, *device);
 
-        talloc_free(desc);
+        talloc_free(name);
     }
 
 coreaudio_error:

audio/out/ao_coreaudio_utils.h

@@ -46,8 +46,8 @@ bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message);
         if (err != noErr) goto label; \
     } while (0)
 
-void ca_get_device_list(struct ao *ao, struct ao_device_list *list);
-OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device);
+void ca_print_device_list(struct ao *ao);
+OSStatus ca_select_device(struct ao *ao, int selection, AudioDeviceID *device);
 
 void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd);
 void ca_print_asbd(struct ao *ao, const char *description,

audio/out/ao_dsound.c

@@ -41,7 +41,6 @@
 #include "internal.h"
 #include "common/msg.h"
 #include "osdep/timer.h"
-#include "osdep/io.h"
 #include "options/m_option.h"
 
 /**
@@ -100,10 +99,10 @@ struct priv {
 
     int cfg_device;
     int cfg_buffersize;
-
-    struct ao_device_list *listing; ///temporary during list_devs()
 };
 
+static float get_delay(struct ao *ao);
+
 /***************************************************************************************/
 
 /**
@@ -173,84 +172,11 @@ static BOOL CALLBACK DirectSoundEnum(LPGUID guid, LPCSTR desc, LPCSTR module,
         if (guid)
             memcpy(&p->device, guid, sizeof(GUID));
     }
-    char *guidstr = talloc_strdup(NULL, "");
-    if (guid) {
-        wchar_t guidwstr[80] = {0};
-        StringFromGUID2(guid, guidwstr, MP_ARRAY_SIZE(guidwstr));
-        char *nstr = mp_to_utf8(NULL, guidwstr);
-        if (nstr) {
-            talloc_free(guidstr);
-            guidstr = nstr;
-        }
-    }
-    if (p->device_num < 0 && ao->device) {
-        if (strcmp(ao->device, guidstr) == 0) {
-            MP_VERBOSE(ao, "<--");
-            p->device_num = p->device_index;
-            if (guid)
-                memcpy(&p->device, guid, sizeof(GUID));
-        }
-    }
-    if (p->listing) {
-        struct ao_device_desc e = {guidstr, desc};
-        ao_device_list_add(p->listing, ao, &e);
-    }
-    talloc_free(guidstr);
-
     MP_VERBOSE(ao, "\n");
     p->device_index++;
     return TRUE;
 }
 
-static void EnumDevs(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    p->device_index = 0;
-    p->device_num = p->cfg_device;
-
-    HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
-    OurDirectSoundEnumerate = (void *)GetProcAddress(p->hdsound_dll,
-                                                     "DirectSoundEnumerateA");
-
-    if (OurDirectSoundEnumerate == NULL) {
-        MP_ERR(ao, "GetProcAddress FAILED\n");
-        return;
-    }
-
-    // Enumerate all directsound p->devices
-    MP_VERBOSE(ao, "Output Devices:\n");
-    OurDirectSoundEnumerate(DirectSoundEnum, ao);
-}
-
-static int LoadDirectSound(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    // initialize directsound
-    p->hdsound_dll = LoadLibrary("DSOUND.DLL");
-    if (p->hdsound_dll == NULL) {
-        MP_ERR(ao, "cannot load DSOUND.DLL\n");
-        return 0;
-    }
-
-    return 1;
-}
-
-static void list_devs(struct ao *ao, struct ao_device_list *list)
-{
-    struct priv *p = ao->priv;
-    bool need_init = !p->hdsound_dll;
-    if (need_init && !LoadDirectSound(ao))
-        return;
-
-    p->listing = list;
-    EnumDevs(ao);
-    p->listing = NULL;
-
-    if (need_init)
-        UninitDirectSound(ao);
-}
 
 /**
 \brief initilize direct sound
@@ -262,23 +188,34 @@ static int InitDirectSound(struct ao *ao)
 
     DSCAPS dscaps;
 
-    if (!LoadDirectSound(ao))
-        return 0;
-
+    // initialize directsound
     HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
-    OurDirectSoundCreate =
-        (void *)GetProcAddress(p->hdsound_dll, "DirectSoundCreate");
+    HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
+    p->device_index = 0;
+    p->device_num = p->cfg_device;
 
-    if (OurDirectSoundCreate == NULL) {
+    p->hdsound_dll = LoadLibrary("DSOUND.DLL");
+    if (p->hdsound_dll == NULL) {
+        MP_ERR(ao, "cannot load DSOUND.DLL\n");
+        return 0;
+    }
+    OurDirectSoundCreate = (void *)GetProcAddress(p->hdsound_dll,
+                                                  "DirectSoundCreate");
+    OurDirectSoundEnumerate = (void *)GetProcAddress(p->hdsound_dll,
+                                                     "DirectSoundEnumerateA");
+
+    if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) {
         MP_ERR(ao, "GetProcAddress FAILED\n");
         FreeLibrary(p->hdsound_dll);
         return 0;
     }
 
-    EnumDevs(ao);
+    // Enumerate all directsound p->devices
+    MP_VERBOSE(ao, "Output Devices:\n");
+    OurDirectSoundEnumerate(DirectSoundEnum, ao);
 
     // Create the direct sound object
-    if (FAILED(OurDirectSoundCreate((p->device_num > 0) ? &p->device : NULL,
+    if (FAILED(OurDirectSoundCreate((p->device_num) ? &p->device : NULL,
                                     &p->hds, NULL)))
     {
         MP_ERR(ao, "cannot create a DirectSound device\n");
@@ -676,12 +613,12 @@ static int play(struct ao *ao, void **data, int samples, int flags)
 \brief get the delay between the first and last sample in the buffer
 \return delay in seconds
 */
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     struct priv *p = ao->priv;
 
     int space = check_free_buffer_size(ao);
-    return (p->buffer_size - space) / (double)ao->bps;
+    return (float)(p->buffer_size - space) / (float)ao->bps;
 }
 
 #define OPT_BASE_STRUCT struct priv
@@ -698,10 +635,9 @@ const struct ao_driver audio_out_dsound = {
     .pause     = audio_pause,
     .resume    = audio_resume,
     .reset     = reset,
-    .list_devs = list_devs,
     .priv_size = sizeof(struct priv),
     .options = (const struct m_option[]) {
-        OPT_INT("device", cfg_device, 0, OPTDEF_INT(-1)),
+        OPT_INT("device", cfg_device, 0),
         OPT_INTRANGE("buffersize", cfg_buffersize, 0, 1, 10000, OPTDEF_INT(200)),
         {0}
     },

audio/out/ao_lavc.c

@@ -56,8 +56,6 @@ struct priv {
 
     AVRational worst_time_base;
     int worst_time_base_is_stream;
-
-    bool shutdown;
 };
 
 static void select_format(struct ao *ao, AVCodec *codec)
@@ -176,7 +174,6 @@ static int init(struct ao *ao)
 
 fail:
     pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
-    ac->shutdown = true;
     return -1;
 }
 
@@ -187,9 +184,6 @@ static void uninit(struct ao *ao)
     struct priv *ac = ao->priv;
     struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
 
-    if (!ac || ac->shutdown)
-        return;
-
     pthread_mutex_lock(&ectx->lock);
 
     if (!encode_lavc_start(ectx)) {
@@ -207,8 +201,6 @@ static void uninit(struct ao *ao)
     }
 
     pthread_mutex_unlock(&ectx->lock);
-
-    ac->shutdown = true;
 }
 
 // return: how many bytes can be played without blocking
@@ -242,9 +234,8 @@ static int encode(struct ao *ao, double apts, void **data)
         frame->format = af_to_avformat(ao->format);
         frame->nb_samples = ac->aframesize;
 
-        size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;
-        assert(num_planes <= AV_NUM_DATA_POINTERS);
-        for (int n = 0; n < num_planes; n++)
+        assert(ao->channels.num <= AV_NUM_DATA_POINTERS);
+        for (int n = 0; n < ao->channels.num; n++)
             frame->extended_data[n] = data[n];
 
         frame->linesize[0] = frame->nb_samples * ao->sstride;
@@ -354,12 +345,12 @@ static int play(struct ao *ao, void **data, int samples, int flags)
     size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;
 
     void *tempdata = NULL;
-    void *padded[MP_NUM_CHANNELS];
 
     if ((flags & AOPLAY_FINAL_CHUNK) && (samples % ac->aframesize)) {
        tempdata = talloc_new(NULL);
        size_t bytelen = samples * ao->sstride;
        size_t extralen = (ac->aframesize - 1) * ao->sstride;
+       void *padded[MP_NUM_CHANNELS];
        for (int n = 0; n < num_planes; n++) {
            padded[n] = talloc_size(tempdata, bytelen + extralen);
            memcpy(padded[n], data[n], bytelen);
@@ -439,7 +430,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
     outpts += encode_lavc_getoffset(ectx, ac->stream);
 
     while (samples - bufpos >= ac->aframesize) {
-        void *start[MP_NUM_CHANNELS] = {0};
+        void *start[MP_NUM_CHANNELS];
         for (int n = 0; n < num_planes; n++)
             start[n] = (char *)data[n] + bufpos * ao->sstride;
         encode(ao, outpts + bufpos / (double) ao->samplerate, start);

audio/out/ao_null.c

@@ -176,7 +176,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
     return accepted;
 }
 
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     struct priv *priv = ao->priv;
 

audio/out/ao_openal.c

@@ -101,17 +101,6 @@ static int validate_device_opt(struct mp_log *log, const m_option_t *opt,
     return 0;
 }
 
-static void list_devs(struct ao *ao, struct ao_device_list *list)
-{
-    if (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") != AL_TRUE)
-        return;
-    const char *devs = alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER);
-    while (devs && *devs) {
-        ao_device_list_add(list, ao, &(struct ao_device_desc){devs, devs});
-        devs = devs + strlen(devs) + 1;
-    }
-}
-
 struct speaker {
     int id;
     float pos[3];
@@ -162,10 +151,7 @@ static int init(struct ao *ao)
             goto err_out;
         }
     }
-    char *dev_name = p->cfg_device;
-    if (!dev_name || !dev_name[0])
-        dev_name = ao->device;
-    dev = alcOpenDevice(dev_name && dev_name[0] ? dev_name : NULL);
+    dev = alcOpenDevice(p->cfg_device && p->cfg_device[0] ? p->cfg_device : NULL);
     if (!dev) {
         MP_FATAL(ao, "could not open device\n");
         goto err_out;
@@ -293,12 +279,12 @@ static int play(struct ao *ao, void **data, int samples, int flags)
     return num * CHUNK_SAMPLES;
 }
 
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     ALint queued;
     unqueue_buffers();
     alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
-    return queued * CHUNK_SAMPLES / (double)ao->samplerate;
+    return queued * CHUNK_SAMPLES / (float)ao->samplerate;
 }
 
 #define OPT_BASE_STRUCT struct priv
@@ -316,7 +302,6 @@ const struct ao_driver audio_out_openal = {
     .resume    = audio_resume,
     .reset     = reset,
     .drain     = drain,
-    .list_devs = list_devs,
     .priv_size = sizeof(struct priv),
     .options = (const struct m_option[]) {
         OPT_STRING_VALIDATE("device", cfg_device, 0, validate_device_opt),

audio/out/ao_oss.c

@@ -322,8 +322,7 @@ static int reopen_device(struct ao *ao, bool allow_format_changes)
 #endif
 
     if (AF_FORMAT_IS_IEC61937(format)) {
-        if (ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
-            goto fail;
+        ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate);
         // Probably could be fixed by setting number of channels; needs testing.
         if (channels.num != 2) {
             MP_ERR(ao, "Format %s not implemented.\n", af_fmt_to_str(format));
@@ -375,8 +374,7 @@ static int reopen_device(struct ao *ao, bool allow_format_changes)
         MP_VERBOSE(ao, "using %d channels (requested: %d)\n",
                    channels.num, reqchannels);
         // set rate
-        if (ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
-            goto fail;
+        ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate);
         MP_VERBOSE(ao, "using %d Hz samplerate\n", samplerate);
     }
 
@@ -548,7 +546,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
 }
 
 // return: delay in seconds between first and last sample in buffer
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     struct priv *p = ao->priv;
     if (p->audio_fd < 0) {
@@ -562,18 +560,18 @@ static double get_delay(struct ao *ao)
 #ifdef SNDCTL_DSP_GETODELAY
         int r = 0;
         if (ioctl(p->audio_fd, SNDCTL_DSP_GETODELAY, &r) != -1)
-            return r / (double)ao->bps;
+            return ((float)r) / (float)ao->bps;
 #endif
         p->audio_delay_method = 1; // fallback if not supported
     }
     if (p->audio_delay_method == 1) {
         audio_buf_info zz = {0};
         if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &zz) != -1) {
-            return (p->buffersize - zz.bytes) / (double)ao->bps;
+            return ((float)(p->buffersize - zz.bytes)) / (float)ao->bps;
         }
         p->audio_delay_method = 0; // fallback if not supported
     }
-    return p->buffersize / (double)ao->bps;
+    return ((float)p->buffersize) / (float)ao->bps;
 }
 
 
@@ -618,17 +616,6 @@ static void audio_resume(struct ao *ao)
         ao_play_silence(ao, p->prepause_samples);
 }
 
-static int audio_wait(struct ao *ao, pthread_mutex_t *lock)
-{
-    struct priv *p = ao->priv;
-
-    struct pollfd fd = {.fd = p->audio_fd, .events = POLLOUT};
-    int r = ao_wait_poll(ao, &fd, 1, lock);
-    if (fd.revents & (POLLERR | POLLNVAL))
-        return -1;
-    return r;
-}
-
 #define OPT_BASE_STRUCT struct priv
 
 const struct ao_driver audio_out_oss = {
@@ -644,8 +631,6 @@ const struct ao_driver audio_out_oss = {
     .resume    = audio_resume,
     .reset     = reset,
     .drain     = drain,
-    .wait      = audio_wait,
-    .wakeup    = ao_wakeup_poll,
     .priv_size = sizeof(struct priv),
     .priv_defaults = &(const struct priv) {
         .audio_fd = -1,

audio/out/ao_pcm.c

@@ -72,35 +72,39 @@ static void fput32le(uint32_t val, FILE *fp)
 
 static void write_wave_header(struct ao *ao, FILE *fp, uint64_t data_length)
 {
+    bool use_waveex = true;
     uint16_t fmt = ao->format == AF_FORMAT_FLOAT ? WAV_ID_FLOAT_PCM : WAV_ID_PCM;
+    uint32_t fmt_chunk_size = use_waveex ? 40 : 16;
     int bits = af_fmt2bits(ao->format);
 
     // Master RIFF chunk
     fput32le(WAV_ID_RIFF, fp);
-    // RIFF chunk size: 'WAVE' + 'fmt ' + 4 + 40 +
+    // RIFF chunk size: 'WAVE' + 'fmt ' + 4 + fmt_chunk_size +
     // data chunk hdr (8) + data length
-    fput32le(12 + 40 + 8 + data_length, fp);
+    fput32le(12 + fmt_chunk_size + 8 + data_length, fp);
     fput32le(WAV_ID_WAVE, fp);
 
     // Format chunk
     fput32le(WAV_ID_FMT, fp);
-    fput32le(40, fp);
-    fput16le(WAV_ID_FORMAT_EXTENSIBLE, fp);
+    fput32le(fmt_chunk_size, fp);
+    fput16le(use_waveex ? WAV_ID_FORMAT_EXTENSIBLE : fmt, fp);
     fput16le(ao->channels.num, fp);
     fput32le(ao->samplerate, fp);
     fput32le(ao->bps, fp);
     fput16le(ao->channels.num * (bits / 8), fp);
     fput16le(bits, fp);
 
-    // Extension chunk
-    fput16le(22, fp);
-    fput16le(bits, fp);
-    fput32le(mp_chmap_to_waveext(&ao->channels), fp);
-    // 2 bytes format + 14 bytes guid
-    fput32le(fmt, fp);
-    fput32le(0x00100000, fp);
-    fput32le(0xAA000080, fp);
-    fput32le(0x719B3800, fp);
+    if (use_waveex) {
+        // Extension chunk
+        fput16le(22, fp);
+        fput16le(bits, fp);
+        fput32le(mp_chmap_to_waveext(&ao->channels), fp);
+        // 2 bytes format + 14 bytes guid
+        fput32le(fmt, fp);
+        fput32le(0x00100000, fp);
+        fput32le(0xAA000080, fp);
+        fput32le(0x719B3800, fp);
+    }
 
     // Data chunk
     fput32le(WAV_ID_DATA, fp);

audio/out/ao_portaudio.c

@@ -125,24 +125,6 @@ static int validate_device_opt(struct mp_log *log, const m_option_t *opt,
     return 0;
 }
 
-static void list_devs(struct ao *ao, struct ao_device_list *list)
-{
-    if (Pa_Initialize() < 0)
-        return;
-    int count = Pa_GetDeviceCount(); // negative on error
-    for (int n = 0; n < count; n++) {
-        const PaDeviceInfo* info = Pa_GetDeviceInfo(n);
-        char desc[1024];
-        snprintf(desc, sizeof(desc), "%s, %d channels, latency: %.2f "
-                 "ms, sample rate: %.0f", info->name,
-                 info->maxOutputChannels,
-                 info->defaultHighOutputLatency * 1000,
-                 info->defaultSampleRate);
-        ao_device_list_add(list, ao, &(struct ao_device_desc){info->name, desc});
-    }
-    Pa_Terminate();
-}
-
 static int stream_callback(const void *input,
                            void *output_v,
                            unsigned long frameCount,
@@ -185,11 +167,8 @@ static int init(struct ao *ao)
         return -1;
 
     int pa_device = Pa_GetDefaultOutputDevice();
-    char *dev_name = priv->cfg_device;
-    if (!dev_name || !dev_name[0])
-        dev_name = ao->device;
-    if (dev_name && dev_name[0])
-        pa_device = find_device(ao->log, dev_name);
+    if (priv->cfg_device && priv->cfg_device[0])
+        pa_device = find_device(ao->log, priv->cfg_device);
     if (pa_device == paNoDevice)
         goto error_exit;
 
@@ -267,7 +246,6 @@ const struct ao_driver audio_out_portaudio = {
     .uninit    = uninit,
     .reset     = reset,
     .resume    = resume,
-    .list_devs = list_devs,
     .priv_size = sizeof(struct priv),
     .options = (const struct m_option[]) {
         OPT_STRING_VALIDATE("device", cfg_device, 0, validate_device_opt),

audio/out/ao_pulse.c

@@ -35,6 +35,8 @@
 #include "ao.h"
 #include "internal.h"
 
+#define PULSE_CLIENT_NAME "mpv"
+
 #define VOL_PA2MP(v) ((v) * 100 / PA_VOLUME_NORM)
 #define VOL_MP2PA(v) ((v) * PA_VOLUME_NORM / 100)
 
@@ -285,11 +287,14 @@ static void uninit(struct ao *ao)
     pthread_mutex_destroy(&priv->wakeup_lock);
 }
 
-static int pa_init_boilerplate(struct ao *ao)
+static int init(struct ao *ao)
 {
+    struct pa_channel_map map;
+    pa_proplist *proplist = NULL;
+    pa_format_info *format = NULL;
     struct priv *priv = ao->priv;
     char *host = priv->cfg_host && priv->cfg_host[0] ? priv->cfg_host : NULL;
-    bool locked = false;
+    char *sink = priv->cfg_sink && priv->cfg_sink[0] ? priv->cfg_sink : NULL;
 
     pthread_mutex_init(&priv->wakeup_lock, NULL);
     pthread_cond_init(&priv->wakeup, NULL);
@@ -299,15 +304,8 @@ static int pa_init_boilerplate(struct ao *ao)
         goto fail;
     }
 
-    if (pa_threaded_mainloop_start(priv->mainloop) < 0)
-        goto fail;
-
-    pa_threaded_mainloop_lock(priv->mainloop);
-    locked = true;
-
     if (!(priv->context = pa_context_new(pa_threaded_mainloop_get_api(
-                                         priv->mainloop), ao->client_name)))
-    {
+                                 priv->mainloop), PULSE_CLIENT_NAME))) {
         MP_ERR(ao, "Failed to allocate context\n");
         goto fail;
     }
@@ -323,47 +321,17 @@ static int pa_init_boilerplate(struct ao *ao)
     if (pa_context_connect(priv->context, host, 0, NULL) < 0)
         goto fail;
 
-    /* Wait until the context is ready */
-    while (1) {
-        int state = pa_context_get_state(priv->context);
-        if (state == PA_CONTEXT_READY)
-            break;
-        if (!PA_CONTEXT_IS_GOOD(state))
-            goto fail;
-        pa_threaded_mainloop_wait(priv->mainloop);
-    }
-
-    pa_threaded_mainloop_unlock(priv->mainloop);
-    return 0;
-
-fail:
-    if (locked)
-        pa_threaded_mainloop_unlock(priv->mainloop);
-
-    if (priv->context) {
-        pa_threaded_mainloop_lock(priv->mainloop);
-        if (!(pa_context_errno(priv->context) == PA_ERR_CONNECTIONREFUSED
-              && ao->probing))
-            GENERIC_ERR_MSG("Init failed");
-        pa_threaded_mainloop_unlock(priv->mainloop);
-    }
-    uninit(ao);
-    return -1;
-}
-
-static int init(struct ao *ao)
-{
-    struct pa_channel_map map;
-    pa_proplist *proplist = NULL;
-    pa_format_info *format = NULL;
-    struct priv *priv = ao->priv;
-    char *sink = priv->cfg_sink && priv->cfg_sink[0] ? priv->cfg_sink : ao->device;
-
-    if (pa_init_boilerplate(ao) < 0)
-        return -1;
-
     pa_threaded_mainloop_lock(priv->mainloop);
 
+    if (pa_threaded_mainloop_start(priv->mainloop) < 0)
+        goto unlock_and_fail;
+
+    /* Wait until the context is ready */
+    pa_threaded_mainloop_wait(priv->mainloop);
+
+    if (pa_context_get_state(priv->context) != PA_CONTEXT_READY)
+        goto unlock_and_fail;
+
     if (!(format = pa_format_info_new()))
         goto unlock_and_fail;
 
@@ -394,7 +362,8 @@ static int init(struct ao *ao)
         goto unlock_and_fail;
     }
     (void)pa_proplist_sets(proplist, PA_PROP_MEDIA_ROLE, "video");
-    (void)pa_proplist_sets(proplist, PA_PROP_MEDIA_ICON_NAME, ao->client_name);
+    (void)pa_proplist_sets(proplist, PA_PROP_MEDIA_ICON_NAME,
+                           PULSE_CLIENT_NAME);
 
     pa_format_info_set_rate(format, ao->samplerate);
     pa_format_info_set_channels(format, ao->channels.num);
@@ -450,13 +419,22 @@ static int init(struct ao *ao)
     }
 
     pa_threaded_mainloop_unlock(priv->mainloop);
+
     return 0;
 
 unlock_and_fail:
-    pa_threaded_mainloop_unlock(priv->mainloop);
 
+    if (priv->mainloop)
+        pa_threaded_mainloop_unlock(priv->mainloop);
+
+fail:
     if (format)
         pa_format_info_free(format);
+    if (priv->context) {
+        if (!(pa_context_errno(priv->context) == PA_ERR_CONNECTIONREFUSED
+              && ao->probing))
+            GENERIC_ERR_MSG("Init failed");
+    }
 
     if (proplist)
         pa_proplist_free(proplist);
@@ -530,7 +508,7 @@ static int get_space(struct ao *ao)
     return space / ao->sstride;
 }
 
-static double get_delay_hackfixed(struct ao *ao)
+static float get_delay_hackfixed(struct ao *ao)
 {
     /* This code basically does what pa_stream_get_latency() _should_
      * do, but doesn't due to multiple known bugs in PulseAudio (at
@@ -585,7 +563,7 @@ static double get_delay_hackfixed(struct ao *ao)
     return latency / 1e6;
 }
 
-static double get_delay_pulse(struct ao *ao)
+static float get_delay_pulse(struct ao *ao)
 {
     struct priv *priv = ao->priv;
     pa_usec_t latency = (pa_usec_t) -1;
@@ -603,7 +581,7 @@ static double get_delay_pulse(struct ao *ao)
 }
 
 // Return the current latency in seconds
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     struct priv *priv = ao->priv;
     if (priv->cfg_latency_hacks) {
@@ -725,42 +703,6 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
     }
 }
 
-struct sink_cb_ctx {
-    struct ao *ao;
-    struct ao_device_list *list;
-};
-
-static void sink_info_cb(pa_context *c, const pa_sink_info *i, int eol, void *ud)
-{
-    struct sink_cb_ctx *ctx = ud;
-    struct priv *priv = ctx->ao->priv;
-
-    if (eol) {
-        pa_threaded_mainloop_signal(priv->mainloop, 0); // wakeup waitop()
-        return;
-    }
-
-    struct ao_device_desc entry = {.name = i->name, .desc = i->description};
-    ao_device_list_add(ctx->list, ctx->ao, &entry);
-}
-
-static void list_devs(struct ao *ao, struct ao_device_list *list)
-{
-    struct priv *priv = ao->priv;
-    bool need_uninit = !priv->mainloop;
-    struct sink_cb_ctx ctx = {ao, list};
-
-    if (need_uninit && pa_init_boilerplate(ao) < 0)
-        return;
-
-    pa_threaded_mainloop_lock(priv->mainloop);
-    waitop(priv, pa_context_get_sink_info_list(priv->context, sink_info_cb, &ctx));
-
-    if (need_uninit)
-        uninit(ao);
-}
-
-
 #define OPT_BASE_STRUCT struct priv
 
 const struct ao_driver audio_out_pulse = {
@@ -778,7 +720,6 @@ const struct ao_driver audio_out_pulse = {
     .drain     = drain,
     .wait      = wait_audio,
     .wakeup    = wakeup,
-    .list_devs = list_devs,
     .priv_size = sizeof(struct priv),
     .priv_defaults = &(const struct priv) {
         .cfg_buffer = 250,

audio/out/ao_rsound.c

@@ -153,7 +153,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
     return rsd_write(priv->rd, data[0], samples * ao->sstride) / ao->sstride;
 }
 
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     struct priv *priv = ao->priv;
     return rsd_delay_ms(priv->rd) / 1000.0;

audio/out/ao_sdl.c

@@ -38,7 +38,7 @@ struct priv
     float buflen;
 };
 
-static const int fmtmap[][2] = {
+static int fmtmap[][2] = {
     {AF_FORMAT_U8,      AUDIO_U8},
     {AF_FORMAT_S8,      AUDIO_S8},
     {AF_FORMAT_U16,     AUDIO_U16SYS},

audio/out/ao_sndio.c

@@ -111,8 +111,7 @@ static int init(struct ao *ao)
 
     struct af_to_par {
         int format, bits, sig;
-    };
-    static const struct af_to_par af_to_par[] = {
+    } static const af_to_par[] = {
         {AF_FORMAT_U8,   8, 0},
         {AF_FORMAT_S8,   8, 1},
         {AF_FORMAT_U16, 16, 0},
@@ -121,8 +120,7 @@ static int init(struct ao *ao)
         {AF_FORMAT_S24, 24, 1},
         {AF_FORMAT_U32, 32, 0},
         {AF_FORMAT_S32, 32, 1},
-    };
-    const struct af_to_par *ap;
+    }, *ap;
     int i;
 
     p->hdl = sio_open(p->dev, SIO_PLAY, 1);
@@ -282,7 +280,7 @@ static int get_space(struct ao *ao)
 /*
  * return: delay in seconds between first and last sample in buffer
  */
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     struct priv *p = ao->priv;
 

audio/out/ao_wasapi.c

@@ -36,6 +36,11 @@
 #include "osdep/timer.h"
 #include "osdep/io.h"
 
+#define EXIT_ON_ERROR(hres)  \
+              do { if (FAILED(hres)) { goto exit_label; } } while(0)
+#define SAFE_RELEASE(unk, release) \
+              do { if ((unk) != NULL) { release; (unk) = NULL; } } while(0)
+
 static double get_device_delay(struct wasapi_state *state) {
     UINT64 sample_count = atomic_load(&state->sample_count);
     UINT64 position, qpc_position;
@@ -45,8 +50,7 @@ static double get_device_delay(struct wasapi_state *state) {
         case S_OK: case S_FALSE:
             break;
         default:
-            MP_ERR(state, "IAudioClock::GetPosition returned %s (0x%"PRIx32")\n",
-                   wasapi_explain_err(hr), (uint32_t)hr);
+            MP_ERR(state, "IAudioClock::GetPosition returned %s\n", wasapi_explain_err(hr));
     }
 
     LARGE_INTEGER qpc_count;
@@ -61,7 +65,7 @@ static double get_device_delay(struct wasapi_state *state) {
     double diff = sample_count - position;
     double delay = diff / state->format.Format.nSamplesPerSec;
 
-    MP_TRACE(state, "Device delay: %g samples (%g ms)\n", diff, delay * 1000);
+    MP_TRACE(state, "device delay: %g samples (%g ms)\n", diff, delay * 1000);
 
     return delay;
 }
@@ -79,7 +83,6 @@ static void thread_feed(struct ao *ao)
         EXIT_ON_ERROR(hr);
 
         frame_count -= padding;
-        MP_TRACE(ao, "Frame to fill: %"PRIu32". Padding: %"PRIu32"\n", frame_count, padding);
     }
 
     BYTE *pData;
@@ -100,10 +103,7 @@ static void thread_feed(struct ao *ao)
 
     return;
 exit_label:
-    MP_ERR(state, "Error feeding audio: %s (0x%"PRIx32")\n",
-           wasapi_explain_err(hr), (uint32_t)hr);
-    MP_VERBOSE(ao, "Requesting ao reload\n");
-    ao_request_reload(ao);
+    MP_ERR(state, "thread_feed fails with %"PRIx32"!\n", (uint32_t)hr);
     return;
 }
 
@@ -113,27 +113,20 @@ static DWORD __stdcall ThreadLoop(void *lpParameter)
     if (!ao || !ao->priv)
         return -1;
     struct wasapi_state *state = (struct wasapi_state *)ao->priv;
-    int thread_ret;
-    CoInitializeEx(NULL, COINIT_APARTMENTTHREADED);
-
-    state->init_ret = wasapi_thread_init(ao);
-    SetEvent(state->init_done);
-    if (state->init_ret != S_OK) {
-        thread_ret = -1;
+    if (wasapi_thread_init(ao))
         goto exit_label;
-    }
 
-
-    DWORD waitstatus;
+    MSG msg;
+    DWORD waitstatus = WAIT_FAILED;
     HANDLE playcontrol[] =
         {state->hUninit, state->hFeed, state->hForceFeed, NULL};
-    MP_DBG(ao, "Entering dispatch loop\n");
+    MP_VERBOSE(ao, "Entering dispatch loop!\n");
     while (1) { /* watch events */
         waitstatus = MsgWaitForMultipleObjects(3, playcontrol, FALSE, INFINITE,
                                                QS_POSTMESSAGE | QS_SENDMESSAGE);
         switch (waitstatus) {
         case WAIT_OBJECT_0: /*shutdown*/
-            thread_ret = 0;
+            wasapi_thread_uninit(state);
             goto exit_label;
         case (WAIT_OBJECT_0 + 1): /* feed */
             thread_feed(ao);
@@ -143,52 +136,51 @@ static DWORD __stdcall ThreadLoop(void *lpParameter)
             SetEvent(state->hFeedDone);
             break;
         case (WAIT_OBJECT_0 + 3): /* messages to dispatch (COM marshalling) */
-            wasapi_dispatch();
+            while (PeekMessage(&msg, NULL, 0, 0, PM_REMOVE)) {
+                DispatchMessage(&msg);
+            }
             break;
-        default:
-            MP_ERR(ao, "Unhandled case in thread loop");
-            thread_ret = -1;
-            goto exit_label;
+        case WAIT_FAILED: /* ??? */
+            return -1;
         }
     }
 exit_label:
-    wasapi_thread_uninit(ao);
-
-    CoUninitialize();
-    return thread_ret;
+    /* dispatch any possible pending messages */
+    while (PeekMessage(&msg, NULL, 0, 0, PM_REMOVE)) {
+        DispatchMessage(&msg);
+    }
+    return state->init_ret;
 }
 
 static void closehandles(struct ao *ao)
 {
     struct wasapi_state *state = (struct wasapi_state *)ao->priv;
-    if (state->init_done)  CloseHandle(state->init_done);
-    if (state->hUninit)    CloseHandle(state->hUninit);
-    if (state->hFeed)      CloseHandle(state->hFeed);
-    if (state->hForceFeed) CloseHandle(state->hForceFeed);
-    if (state->hFeedDone)  CloseHandle(state->hFeedDone);
-    if (state->threadLoop) CloseHandle(state->threadLoop);
+    if (state->init_done)
+        CloseHandle(state->init_done);
+    if (state->hUninit)
+        CloseHandle(state->hUninit);
+    if (state->hFeed)
+        CloseHandle(state->hFeed);
 }
 
 static void uninit(struct ao *ao)
 {
-    MP_DBG(ao, "Uninit wasapi\n");
+    MP_VERBOSE(ao, "uninit!\n");
     struct wasapi_state *state = (struct wasapi_state *)ao->priv;
     wasapi_release_proxies(state);
     SetEvent(state->hUninit);
     /* wait up to 10 seconds */
-    if (WaitForSingleObject(state->threadLoop, 10000) == WAIT_TIMEOUT){
-        MP_ERR(ao, "Audio loop thread refuses to abort");
-        return;
-    }
+    if (WaitForSingleObject(state->threadLoop, 10000) == WAIT_TIMEOUT)
+        MP_ERR(ao, "audio loop thread refuses to abort!");
     if (state->VistaBlob.hAvrt)
         FreeLibrary(state->VistaBlob.hAvrt);
     closehandles(ao);
-    MP_DBG(ao, "Uninit wasapi done\n");
+    MP_VERBOSE(ao, "uninit END!\n");
 }
 
 static int init(struct ao *ao)
 {
-    MP_DBG(ao, "Init wasapi\n");
+    MP_VERBOSE(ao, "init!\n");
     ao->format = af_fmt_from_planar(ao->format);
     struct mp_chmap_sel sel = {0};
     mp_chmap_sel_add_waveext(&sel);
@@ -199,7 +191,7 @@ static int init(struct ao *ao)
     wasapi_fill_VistaBlob(state);
 
     if (state->opt_list) {
-        wasapi_enumerate_devices(state->log, NULL, NULL);
+        wasapi_enumerate_devices(state->log);
     }
 
     if (state->opt_exclusive) {
@@ -220,26 +212,23 @@ static int init(struct ao *ao)
         /* failed to init events */
         return -1;
     }
-
-    state->init_ret = E_FAIL;
+    state->init_ret = -1;
     state->threadLoop = (HANDLE)CreateThread(NULL, 0, &ThreadLoop, ao, 0, NULL);
     if (!state->threadLoop) {
         /* failed to init thread */
-        MP_ERR(ao, "Failed to create thread\n");
+        MP_ERR(ao, "fail to create thread!\n");
         return -1;
     }
-
     WaitForSingleObject(state->init_done, INFINITE); /* wait on init complete */
-    if (state->init_ret != S_OK) {
-        if (!ao->probing)
-            MP_ERR(ao, "Received failure from audio thread\n");
-        uninit(ao);
-        return -1;
-    }
+    if (state->init_ret) {
+        if (!ao->probing) {
+            MP_ERR(ao, "thread_init failed!\n");
+        }
+    } else
+        MP_VERBOSE(ao, "Init Done!\n");
 
     wasapi_setup_proxies(state);
-    MP_DBG(ao, "Init wasapi done\n");
-    return 0;
+    return state->init_ret;
 }
 
 static int control(struct ao *ao, enum aocontrol cmd, void *arg)
@@ -260,7 +249,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
         /* check to see if user manually changed volume through mixer;
            this information is used in exclusive mode for restoring the mixer volume on uninit */
         if (state->audio_volume != state->previous_volume) {
-            MP_VERBOSE(state, "Mixer difference: %.2g now, expected %.2g\n",
+            MP_VERBOSE(state, "mixer difference: %.2g now, expected %.2g\n",
                        state->audio_volume, state->previous_volume);
             state->initial_volume = state->audio_volume;
         }
@@ -340,11 +329,6 @@ static void audio_resume(struct ao *ao)
     IAudioClient_Start(state->pAudioClientProxy);
 }
 
-static void list_devs(struct ao *ao, struct ao_device_list *list)
-{
-    wasapi_enumerate_devices(mp_null_log, ao, list);
-}
-
 #define OPT_BASE_STRUCT struct wasapi_state
 
 const struct ao_driver audio_out_wasapi = {
@@ -353,9 +337,8 @@ const struct ao_driver audio_out_wasapi = {
     .init      = init,
     .uninit    = uninit,
     .control   = control,
-    .reset     = audio_reset,
+    //.reset     = audio_reset, <- doesn't wait for audio callback to return
     .resume    = audio_resume,
-    .list_devs = list_devs,
     .priv_size = sizeof(wasapi_state),
     .options   = (const struct m_option[]) {
         OPT_FLAG("exclusive", opt_exclusive, 0),

audio/out/ao_wasapi.h

@@ -30,26 +30,12 @@
 
 #include "osdep/atomics.h"
 
-typedef struct change_notify {
-    IMMNotificationClient client; /* this must be first in the structure! */
-    LPWSTR monitored; /* Monitored device */
-    struct ao *ao;
-} change_notify;
-
-HRESULT wasapi_change_init(struct ao* ao);
-void wasapi_change_uninit(struct ao* ao);
-
-#define EXIT_ON_ERROR(hres)  \
-              do { if (FAILED(hres)) { goto exit_label; } } while(0)
-#define SAFE_RELEASE(unk, release) \
-              do { if ((unk) != NULL) { release; (unk) = NULL; } } while(0)
-
 typedef struct wasapi_state {
     struct mp_log *log;
     HANDLE threadLoop;
 
     /* Init phase */
-    HRESULT init_ret;
+    int init_ret;
     HANDLE init_done;
     int share_mode;
 
@@ -63,6 +49,10 @@ typedef struct wasapi_state {
 
     /* Buffers */
     size_t buffer_block_size; /* Size of each block in bytes */
+    REFERENCE_TIME
+        minRequestedDuration; /* minimum wasapi buffer block size, in 100-nanosecond units */
+    REFERENCE_TIME
+        defaultRequestedDuration; /* default wasapi default block size, in 100-nanosecond units */
     UINT32 bufferFrameCount; /* wasapi buffer block size, number of frames, frame size at format.nBlockAlign */
 
     /* WASAPI handles, owned by other thread */
@@ -72,8 +62,6 @@ typedef struct wasapi_state {
     ISimpleAudioVolume *pAudioVolume;
     IAudioEndpointVolume *pEndpointVolume;
     IAudioSessionControl *pSessionControl;
-    IMMDeviceEnumerator *pEnumerator;
-
     HANDLE hFeed; /* wasapi event */
     HANDLE hForceFeed; /* forces writing a buffer (e.g. before audio_resume) */
     HANDLE hFeedDone; /* set only after a hForceFeed */
@@ -112,8 +100,6 @@ typedef struct wasapi_state {
         HANDLE (WINAPI *pAvSetMmThreadCharacteristicsW)(LPCWSTR, LPDWORD);
         WINBOOL (WINAPI *pAvRevertMmThreadCharacteristics)(HANDLE);
     } VistaBlob;
-
-    change_notify change;
 } wasapi_state;
 
 #endif

audio/out/ao_wasapi_changenotify.c

@@ -1,284 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * Original author: Jonathan Yong <10walls@gmail.com>
- *
- * 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/>.
- */
-
-#define COBJMACROS 1
-#define _WIN32_WINNT 0x600
-
-#include <initguid.h>
-#include <audioclient.h>
-#include <endpointvolume.h>
-#include <mmdeviceapi.h>
-#include <wchar.h>
-#include <stdlib.h>
-
-#include "ao_wasapi.h"
-#include "ao_wasapi_utils.h"
-
-static int GUID_compare(const GUID *l, const GUID *r)
-{
-    unsigned int i;
-    if (l->Data1 != r->Data1) return 1;
-    if (l->Data2 != r->Data2) return 1;
-    if (l->Data3 != r->Data3) return 1;
-    for (i = 0; i < 8; i++) {
-        if (l->Data4[i] != r->Data4[i]) return 1;
-    }
-    return 0;
-}
-
-static int PKEY_compare(const PROPERTYKEY *l, const PROPERTYKEY *r)
-{
-    if (GUID_compare(&l->fmtid, &r->fmtid)) return 1;
-    if (l->pid != r->pid) return 1;
-    return 0;
-}
-
-static char *GUID_to_str_buf(char *buf, size_t buf_size, const GUID *guid)
-{
-    snprintf(buf, buf_size,
-             "{%8.8x-%4.4x-%4.4x-%2.2x%2.2x-%2.2x%2.2x%2.2x%2.2x%2.2x%2.2x}",
-             (unsigned) guid->Data1, guid->Data2, guid->Data3,
-             guid->Data4[0], guid->Data4[1],
-             guid->Data4[2], guid->Data4[3],
-             guid->Data4[4], guid->Data4[5],
-             guid->Data4[6], guid->Data4[7]);
-    return buf;
-}
-
-static char *PKEY_to_str_buf(char *buf, size_t buf_size, const PROPERTYKEY *pkey)
-{
-    buf = GUID_to_str_buf(buf, buf_size, &pkey->fmtid);
-    size_t guid_len = strnlen(buf, buf_size);
-    snprintf(buf + guid_len, buf_size - guid_len, ",%"PRIu32, (uint32_t)pkey->pid );
-    return buf;
-}
-
-#define PKEY_to_str(pkey) PKEY_to_str_buf((char[42]){0}, 42, (pkey))
-
-static char* ERole_to_str(ERole role)
-{
-    switch(role){
-    case eConsole:        return "console";
-    case eMultimedia:     return "multimedia";
-    case eCommunications: return "communications";
-    default:              return "<Unknown>";
-    }
-}
-
-static char* EDataFlow_to_str(EDataFlow flow)
-{
-    switch(flow){
-    case eRender:  return "render";
-    case eCapture: return "capture";
-    case eAll:     return "all";
-    default:       return "<Unknown>";
-    }
-}
-
-static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_QueryInterface(
-    IMMNotificationClient* This, REFIID riid, void **ppvObject)
-{
-    /* Compatible with IMMNotificationClient and IUnknown */
-    if (!GUID_compare(&IID_IMMNotificationClient, riid) ||
-        !GUID_compare(&IID_IUnknown, riid)) {
-        *ppvObject = (void *)This;
-        return S_OK;
-    } else {
-        *ppvObject = NULL;
-        return E_NOINTERFACE;
-    }
-}
-
-/* these are required, but not actually used */
-static ULONG STDMETHODCALLTYPE sIMMNotificationClient_AddRef(
-    IMMNotificationClient *This)
-{
-    return 1;
-}
-
-/* MSDN says it should free itself, but we're static */
-static ULONG STDMETHODCALLTYPE sIMMNotificationClient_Release(
-    IMMNotificationClient *This)
-{
-    return 1;
-}
-
-static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnDeviceStateChanged(
-    IMMNotificationClient *This,
-    LPCWSTR pwstrDeviceId,
-    DWORD dwNewState)
-{
-    change_notify *change = (change_notify *)This;
-    struct ao *ao = change->ao;
-
-    if (pwstrDeviceId && !wcscmp(change->monitored, pwstrDeviceId)){
-        switch (dwNewState) {
-        case DEVICE_STATE_DISABLED:
-        case DEVICE_STATE_NOTPRESENT:
-        case DEVICE_STATE_UNPLUGGED:
-            MP_VERBOSE(ao,
-                       "OnDeviceStateChange triggered - requesting ao reload\n");
-            ao_request_reload(ao);
-        case DEVICE_STATE_ACTIVE:
-        default:
-            return S_OK;
-        }
-    }
-    return S_OK;
-}
-
-static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnDeviceAdded(
-    IMMNotificationClient *This,
-    LPCWSTR pwstrDeviceId)
-{
-    change_notify *change = (change_notify *)This;
-    struct ao *ao = change->ao;
-
-    MP_VERBOSE(ao, "OnDeviceAdded triggered\n");
-    if(pwstrDeviceId)
-        MP_VERBOSE(ao, "New device %S\n",pwstrDeviceId);
-    return S_OK;
-}
-
-/* maybe MPV can go over to the prefered device once it is plugged in? */
-static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnDeviceRemoved(
-    IMMNotificationClient *This,
-    LPCWSTR pwstrDeviceId)
-{
-    change_notify *change = (change_notify *)This;
-    struct ao *ao = change->ao;
-
-    if (pwstrDeviceId && !wcscmp(change->monitored, pwstrDeviceId)) {
-        MP_VERBOSE(ao, "OnDeviceRemoved triggered - requesting ao reload\n");
-        ao_request_reload(ao);
-    }
-    return S_OK;
-}
-
-static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnDefaultDeviceChanged(
-    IMMNotificationClient *This,
-    EDataFlow flow,
-    ERole role,
-    LPCWSTR pwstrDeviceId)
-{
-    change_notify *change = (change_notify *)This;
-    struct ao *ao = change->ao;
-    struct wasapi_state *state = (struct wasapi_state *)ao->priv;
-
-    MP_VERBOSE(ao, "OnDefaultDeviceChanged triggered for role:%s flow:%s\n",
-               ERole_to_str(role), EDataFlow_to_str(flow));
-    if(pwstrDeviceId)
-        MP_VERBOSE(ao, "New default device %S\n", pwstrDeviceId);
-
-    /* don't care about "eCapture" or non-"eMultimedia" roles  */
-    if ( flow == eCapture ||
-         role != eMultimedia ) return S_OK;
-
-    /* stay on the device the user specified */
-    if (state->opt_device) {
-        MP_VERBOSE(ao, "Staying on specified device \"%s\"", state->opt_device);
-        return S_OK;
-    }
-
-    /* don't reload if already on the new default */
-    if ( pwstrDeviceId && !wcscmp(change->monitored, pwstrDeviceId) ){
-        MP_VERBOSE(ao, "Already using default device, no reload required\n");
-        return S_OK;
-    }
-
-    /* if we got here, we need to reload */
-    ao_request_reload(ao);
-    MP_VERBOSE(ao, "Requesting ao reload\n");
-    return S_OK;
-}
-
-static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnPropertyValueChanged(
-    IMMNotificationClient *This,
-    LPCWSTR pwstrDeviceId,
-    const PROPERTYKEY key)
-{
-    change_notify *change = (change_notify *)This;
-    struct ao *ao = change->ao;
-
-    if (pwstrDeviceId && !wcscmp(change->monitored, pwstrDeviceId)) {
-        MP_VERBOSE(ao, "OnPropertyValueChanged triggered\n");
-        MP_VERBOSE(ao, "Changed property: ");
-        if (!PKEY_compare(&PKEY_AudioEngine_DeviceFormat, &key)) {
-            MP_VERBOSE(change->ao,
-                       "PKEY_AudioEngine_DeviceFormat - requesting ao reload\n");
-            ao_request_reload(change->ao);
-        } else {
-            MP_VERBOSE(ao, "%s\n", PKEY_to_str(&key));
-        }
-    }
-    return S_OK;
-}
-
-static CONST_VTBL IMMNotificationClientVtbl sIMMDeviceEnumeratorVtbl_vtbl = {
-    .QueryInterface = sIMMNotificationClient_QueryInterface,
-    .AddRef = sIMMNotificationClient_AddRef,
-    .Release = sIMMNotificationClient_Release,
-    .OnDeviceStateChanged = sIMMNotificationClient_OnDeviceStateChanged,
-    .OnDeviceAdded = sIMMNotificationClient_OnDeviceAdded,
-    .OnDeviceRemoved = sIMMNotificationClient_OnDeviceRemoved,
-    .OnDefaultDeviceChanged = sIMMNotificationClient_OnDefaultDeviceChanged,
-    .OnPropertyValueChanged = sIMMNotificationClient_OnPropertyValueChanged,
-};
-
-
-HRESULT wasapi_change_init(struct ao *ao)
-{
-    MP_DBG(ao, "Setting up monitoring on playback device\n");
-    struct wasapi_state *state = (struct wasapi_state *)ao->priv;
-    struct change_notify *change = &state->change;
-    /* COM voodoo to emulate c++ class */
-    change->client.lpVtbl = &sIMMDeviceEnumeratorVtbl_vtbl;
-
-    /* so the callbacks can access the ao */
-    change->ao = ao;
-
-    /* get the device string to compare with the pwstrDeviceId argument in callbacks */
-    HRESULT hr = IMMDevice_GetId(state->pDevice, &change->monitored);
-    EXIT_ON_ERROR(hr);
-
-    /* register the change notification client */
-    hr = IMMDeviceEnumerator_RegisterEndpointNotificationCallback(
-        state->pEnumerator, (IMMNotificationClient *)change);
-    EXIT_ON_ERROR(hr);
-
-    MP_VERBOSE(state, "Monitoring changes in device: %S\n", state->change.monitored);
-    return hr;
-exit_label:
-    MP_ERR(state, "Error setting up device change monitoring: %s\n",
-           wasapi_explain_err(hr));
-    wasapi_change_uninit(ao);
-    return hr;
-}
-
-void wasapi_change_uninit(struct ao *ao)
-{
-    struct wasapi_state *state = (struct wasapi_state *)ao->priv;
-    struct change_notify *change = &state->change;
-
-    if( state->pEnumerator && change->client.lpVtbl )
-        IMMDeviceEnumerator_UnregisterEndpointNotificationCallback(
-            state->pEnumerator, (IMMNotificationClient *)change);
-
-    if (change->monitored) CoTaskMemFree(change->monitored);
-}

audio/out/ao_wasapi_utils.c

@@ -20,7 +20,6 @@
 #define COBJMACROS 1
 #define _WIN32_WINNT 0x600
 
-#include <math.h>
 #include <initguid.h>
 #include <audioclient.h>
 #include <endpointvolume.h>
@@ -31,10 +30,14 @@
 
 #include "audio/format.h"
 #include "osdep/io.h"
-#include "osdep/timer.h"
 
 #define MIXER_DEFAULT_LABEL L"mpv - video player"
 
+#define EXIT_ON_ERROR(hres)  \
+              do { if (FAILED(hres)) { goto exit_label; } } while(0)
+#define SAFE_RELEASE(unk, release) \
+              do { if ((unk) != NULL) { release; (unk) = NULL; } } while(0)
+
 #ifndef PKEY_Device_FriendlyName
 DEFINE_PROPERTYKEY(PKEY_Device_FriendlyName,
                    0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20,
@@ -82,10 +85,6 @@ const char *wasapi_explain_err(const HRESULT hr)
 #define E(x) case x : return # x ;
     switch (hr) {
     E(S_OK)
-    E(E_FAIL)
-    E(E_OUTOFMEMORY)
-    E(E_POINTER)
-    E(E_INVALIDARG)
     E(AUDCLNT_E_NOT_INITIALIZED)
     E(AUDCLNT_E_ALREADY_INITIALIZED)
     E(AUDCLNT_E_WRONG_ENDPOINT_TYPE)
@@ -161,7 +160,7 @@ static int set_ao_format(struct wasapi_state *state,
                          WAVEFORMATEXTENSIBLE wformat)
 {
     if (wformat.SubFormat.Data1 != 1 && wformat.SubFormat.Data1 != 3) {
-        MP_ERR(ao, "Unknown SubFormat %"PRIu32"\n",
+        MP_ERR(ao, "unknown SubFormat %"PRIu32"\n",
                (uint32_t)wformat.SubFormat.Data1);
         return 0;
     }
@@ -197,7 +196,7 @@ static int try_format(struct wasapi_state *state,
     if (!af_format)
         return 0;
 
-    MP_VERBOSE(ao, "Trying %dch %s @ %dhz\n",
+    MP_VERBOSE(ao, "trying %dch %s @ %dhz\n",
                channels.num, af_fmt_to_str(af_format), samplerate);
 
     union WAVEFMT u;
@@ -221,7 +220,7 @@ static int try_format(struct wasapi_state *state,
 
     if (hr == S_FALSE) {
         if (set_ao_format(state, ao, wformat)) {
-            MP_VERBOSE(ao, "Accepted as %dch %s @ %dhz\n",
+            MP_VERBOSE(ao, "accepted as %dch %s @ %dhz\n",
                        ao->channels.num, af_fmt_to_str(ao->format), ao->samplerate);
 
             return 1;
@@ -284,7 +283,7 @@ static int try_passthrough(struct wasapi_state *state,
     union WAVEFMT u;
     u.extensible = &wformat;
 
-    MP_VERBOSE(ao, "Trying passthrough for %s...\n", af_fmt_to_str(ao->format));
+    MP_VERBOSE(ao, "trying passthrough for %s...\n", af_fmt_to_str(ao->format));
 
     HRESULT hr = IAudioClient_IsFormatSupported(state->pAudioClient,
                                                 state->share_mode,
@@ -305,7 +304,7 @@ static int find_formats(struct ao *const ao)
         if (try_passthrough(state, ao))
             return 0;
 
-        MP_ERR(ao, "Couldn't use passthrough");
+        MP_ERR(ao, "couldn't use passthrough!");
         if (!state->opt_exclusive)
             MP_ERR(ao, " (try exclusive mode)");
         MP_ERR(ao, "\n");
@@ -327,7 +326,7 @@ static int find_formats(struct ao *const ao)
             return 0;
         }
 
-        MP_WARN(ao, "Couldn't use default mix format\n");
+        MP_WARN(ao, "couldn't use default mix format!\n");
     }
 
     /* Exclusive mode, we have to guess. */
@@ -398,13 +397,13 @@ static int find_formats(struct ao *const ao)
             bits = start_bits;
             mp_chmap_from_channels(&ao->channels, 2);
         } else {
-            MP_ERR(ao, "Couldn't find acceptable audio format\n");
+            MP_ERR(ao, "couldn't find acceptable audio format!\n");
             return -1;
         }
     }
 }
 
-static HRESULT init_clock(struct wasapi_state *state) {
+static int init_clock(struct wasapi_state *state) {
     HRESULT hr;
 
     hr = IAudioClient_GetService(state->pAudioClient,
@@ -418,17 +417,16 @@ static HRESULT init_clock(struct wasapi_state *state) {
 
     atomic_store(&state->sample_count, 0);
 
-    MP_VERBOSE(state, "IAudioClock::GetFrequency gave a frequency of %"PRIu64".\n",
-               (uint64_t) state->clock_frequency);
+    MP_VERBOSE(state, "IAudioClock::GetFrequency gave a frequency of %"PRIu64".\n", (uint64_t) state->clock_frequency);
 
-    return S_OK;
+    return 0;
 exit_label:
-    MP_ERR(state, "Error obtaining the audio device's timing: %s (0x%"PRIx32")\n",
-           wasapi_explain_err(hr), (uint32_t)hr);
-    return hr;
+    MP_ERR(state, "init_clock failed with %s, unable to obtain the audio device's timing!\n",
+           wasapi_explain_err(hr));
+    return 1;
 }
 
-static HRESULT init_session_display(struct wasapi_state *state) {
+static int init_session_display(struct wasapi_state *state) {
     HRESULT hr;
     wchar_t path[MAX_PATH+12] = {0};
 
@@ -445,59 +443,44 @@ static HRESULT init_session_display(struct wasapi_state *state) {
     hr = IAudioSessionControl_SetIconPath(state->pSessionControl, path, NULL);
     EXIT_ON_ERROR(hr);
 
-    return S_OK;
+    return 0;
+
 exit_label:
-    MP_ERR(state, "Error setting audio session display name: %s (0x%"PRIx32")\n",
-           wasapi_explain_err(hr), (uint32_t)hr);
-    return hr;
+    MP_ERR(state, "init_session_display failed with %s.\n",
+           wasapi_explain_err(hr));
+    return 1;
 }
 
-static HRESULT fix_format(struct ao *ao)
+static int fix_format(struct wasapi_state *state)
 {
-    struct wasapi_state *state = (struct wasapi_state *)ao->priv;
     HRESULT hr;
+    double offset = 0.5;
 
-    REFERENCE_TIME devicePeriod, bufferDuration, bufferPeriod;
-    MP_DBG(state, "IAudioClient::GetDevicePeriod\n");
-    hr = IAudioClient_GetDevicePeriod(state->pAudioClient,&devicePeriod, NULL);
-    MP_VERBOSE(state, "Device period: %.2g ms\n", (double) devicePeriod / 10000.0 );
-
-    /* integer multiple of device period close to 50ms */
-    bufferPeriod = bufferDuration = ceil( 50.0 * 10000.0 / devicePeriod ) * devicePeriod;
-    if (state->share_mode == AUDCLNT_SHAREMODE_SHARED)
-        bufferPeriod = 0;
-
-    /* handle unsupported buffer size */
-    /* hopefully this shouldn't happen because of the above integer device period */
-    /* http://msdn.microsoft.com/en-us/library/windows/desktop/dd370875%28v=vs.85%29.aspx */
-    int retries=0;
+    /* cargo cult code to negotiate buffer block size, affected by hardware/drivers combinations,
+       gradually grow it to 10s, by 0.5s, consider failure if it still doesn't work
+     */
+    hr = IAudioClient_GetDevicePeriod(state->pAudioClient,
+                                      &state->defaultRequestedDuration,
+                                      &state->minRequestedDuration);
 reinit:
-    MP_DBG(state, "IAudioClient::Initialize\n");
     hr = IAudioClient_Initialize(state->pAudioClient,
                                  state->share_mode,
                                  AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
-                                 bufferDuration,
-                                 bufferPeriod,
+                                 state->defaultRequestedDuration,
+                                 state->defaultRequestedDuration,
                                  &(state->format.Format),
                                  NULL);
-    /* something about buffer sizes on Win7 */
+    /* something about buffer sizes on Win7, fixme might loop forever */
     if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) {
-        MP_VERBOSE(state, "IAudioClient::Initialize negotiation failed with %s (0x%"PRIx32"), used %lld * 100ns\n",
-                   wasapi_explain_err(hr), (uint32_t)hr, bufferDuration);
-        if (retries > 0) {
-            hr = E_FAIL;
-            EXIT_ON_ERROR(hr);
-        } else {
-            retries ++;
-        }
-
+        MP_VERBOSE(state, "IAudioClient::Initialize negotiation failed with %s, used %lld * 100ns\n",
+                   wasapi_explain_err(hr), state->defaultRequestedDuration);
+        if (offset > 10.0)
+            goto exit_label;                /* is 10 enough to break out of the loop?*/
         IAudioClient_GetBufferSize(state->pAudioClient, &state->bufferFrameCount);
-        bufferPeriod = bufferDuration =
+        state->defaultRequestedDuration =
             (REFERENCE_TIME)((10000.0 * 1000 / state->format.Format.nSamplesPerSec *
-                              state->bufferFrameCount) + 0.5);
-        if (state->share_mode == AUDCLNT_SHAREMODE_SHARED)
-            bufferPeriod = 0;
-
+                              state->bufferFrameCount) + offset);
+        offset += 0.5;
         IAudioClient_Release(state->pAudioClient);
         state->pAudioClient = NULL;
         hr = IMMDeviceActivator_Activate(state->pDevice,
@@ -506,54 +489,40 @@ reinit:
         goto reinit;
     }
     EXIT_ON_ERROR(hr);
-
-    MP_DBG(state, "IAudioClient::Initialize pRenderClient\n");
     hr = IAudioClient_GetService(state->pAudioClient,
                                  &IID_IAudioRenderClient,
                                  (void **)&state->pRenderClient);
     EXIT_ON_ERROR(hr);
-
-    MP_DBG(state, "IAudioClient::Initialize pAudioVolume\n");
     hr = IAudioClient_GetService(state->pAudioClient,
                                  &IID_ISimpleAudioVolume,
                                  (void **) &state->pAudioVolume);
     EXIT_ON_ERROR(hr);
 
-    MP_DBG(state, "IAudioClient::Initialize IAudioClient_SetEventHandle\n");
+    if (!state->hFeed)
+        goto exit_label;
     hr = IAudioClient_SetEventHandle(state->pAudioClient, state->hFeed);
     EXIT_ON_ERROR(hr);
-
-    MP_DBG(state, "IAudioClient::Initialize IAudioClient_GetBufferSize\n");
     hr = IAudioClient_GetBufferSize(state->pAudioClient,
                                     &state->bufferFrameCount);
     EXIT_ON_ERROR(hr);
-
-    ao->device_buffer = state->bufferFrameCount;
     state->buffer_block_size = state->format.Format.nChannels *
                                state->format.Format.wBitsPerSample / 8 *
                                state->bufferFrameCount;
-    bufferDuration =
-        (REFERENCE_TIME)((10000.0 * 1000 / state->format.Format.nSamplesPerSec *
-                          state->bufferFrameCount) + 0.5);
-    MP_VERBOSE(state, "Buffer frame count: %"PRIu32" (%.2g ms)\n",
-               state->bufferFrameCount, (double) bufferDuration / 10000.0 );
 
-    hr = init_clock(state);
-    EXIT_ON_ERROR(hr);
-
-    hr = init_session_display(state);
-    EXIT_ON_ERROR(hr);
+    if (init_clock(state))
+        return 1;
+    if (init_session_display(state))
+        return 1;
 
     state->hTask =
         state->VistaBlob.pAvSetMmThreadCharacteristicsW(L"Pro Audio", &state->taskIndex);
-    MP_VERBOSE(state, "Format fixed. Using %lld byte buffer block size\n",
+    MP_VERBOSE(state, "fix_format OK, using %lld byte buffer block size!\n",
                (long long) state->buffer_block_size);
-
-    return S_OK;
+    return 0;
 exit_label:
-    MP_ERR(state, "Error initializing device: %s (0x%"PRIx32")\n",
-           wasapi_explain_err(hr), (uint32_t)hr);
-    return hr;
+    MP_ERR(state, "fix_format fails with %s, failed to determine buffer block size!\n",
+           wasapi_explain_err(hr));
+    return 1;
 }
 
 static char* get_device_id(IMMDevice *pDevice) {
@@ -650,10 +619,8 @@ end:
     return found;
 }
 
-// Warning: ao and list are NULL in the "--ao=wasapi:device=help" path!
-static HRESULT enumerate_with_state(struct mp_log *log, struct ao *ao,
-                                    struct ao_device_list *list,
-                                    char *header, int status, int with_id)
+static HRESULT enumerate_with_state(struct mp_log *log, char *header,
+                                    int status, int with_id)
 {
     HRESULT hr;
     IMMDeviceEnumerator *pEnumerator = NULL;
@@ -666,7 +633,7 @@ static HRESULT enumerate_with_state(struct mp_log *log, struct ao *ao,
     EXIT_ON_ERROR(hr);
 
     hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint(pEnumerator,
-                                                     eRender, eMultimedia,
+                                                     eRender, eConsole,
                                                      &pDevice);
     EXIT_ON_ERROR(hr);
 
@@ -701,12 +668,6 @@ static HRESULT enumerate_with_state(struct mp_log *log, struct ao *ao,
             mp_info(log, "%s, ID: %s%s\n", name, id, mark);
         }
 
-        if (ao) {
-            char *desc = talloc_asprintf(NULL, "%s, ID: %s%s", name, id, mark);
-            struct ao_device_desc e = {id, desc};
-            ao_device_list_add(list, ao, &e);
-        }
-
         talloc_free(name);
         talloc_free(id);
         SAFE_RELEASE(pDevice, IMMDevice_Release(pDevice));
@@ -714,8 +675,8 @@ static HRESULT enumerate_with_state(struct mp_log *log, struct ao *ao,
     talloc_free(defid);
     SAFE_RELEASE(pDevices, IMMDeviceCollection_Release(pDevices));
     SAFE_RELEASE(pEnumerator, IMMDeviceEnumerator_Release(pEnumerator));
+    return hr;
 
-    return S_OK;
 exit_label:
     talloc_free(defid);
     SAFE_RELEASE(pDevice, IMMDevice_Release(pDevice));
@@ -724,50 +685,29 @@ exit_label:
     return hr;
 }
 
-int wasapi_enumerate_devices(struct mp_log *log, struct ao *ao,
-                             struct ao_device_list *list)
+int wasapi_enumerate_devices(struct mp_log *log)
 {
     HRESULT hr;
     CoInitializeEx(NULL, COINIT_APARTMENTTHREADED);
 
-    hr = enumerate_with_state(log, ao, list, "Active devices:",
-                              DEVICE_STATE_ACTIVE, 1);
+    hr = enumerate_with_state(log, "Active devices:", DEVICE_STATE_ACTIVE, 1);
     EXIT_ON_ERROR(hr);
-    hr = enumerate_with_state(log, ao, list, "Unplugged devices:",
-                              DEVICE_STATE_UNPLUGGED, 0);
+    hr = enumerate_with_state(log, "Unplugged devices:", DEVICE_STATE_UNPLUGGED, 0);
     EXIT_ON_ERROR(hr);
     CoUninitialize();
     return 0;
 exit_label:
-    mp_err(log, "Error enumerating devices: %s (0x%"PRIx32")\n",
-           wasapi_explain_err(hr), (uint32_t)hr);
+    mp_err(log, "Error enumerating devices: HRESULT %08"PRIx32" \"%s\"\n",
+           (uint32_t)hr, wasapi_explain_err(hr));
     CoUninitialize();
     return 1;
 }
 
-static HRESULT load_default_device(struct ao *ao, IMMDeviceEnumerator* pEnumerator,
-                                   IMMDevice **ppDevice)
-{
-    HRESULT hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint(pEnumerator,
-                                                             eRender, eMultimedia,
-                                                             ppDevice);
-    EXIT_ON_ERROR(hr);
-
-    char *id = get_device_id(*ppDevice);
-    MP_VERBOSE(ao, "Default device ID: %s\n", id);
-    talloc_free(id);
-
-    return S_OK;
-exit_label:
-    MP_ERR(ao , "Error loading default device: %s (0x%"PRIx32")\n",
-           wasapi_explain_err(hr), (uint32_t)hr);
-    return hr;
-}
-
-static HRESULT find_and_load_device(struct ao *ao, IMMDeviceEnumerator* pEnumerator,
-                                    IMMDevice **ppDevice, char *search)
+static HRESULT find_and_load_device(struct ao *ao, IMMDevice **ppDevice,
+                                    char *search)
 {
     HRESULT hr;
+    IMMDeviceEnumerator *pEnumerator = NULL;
     IMMDeviceCollection *pDevices = NULL;
     IMMDevice *pTempDevice = NULL;
     LPWSTR deviceID = NULL;
@@ -780,6 +720,10 @@ static HRESULT find_and_load_device(struct ao *ao, IMMDeviceEnumerator* pEnumera
         devid = search;
     }
 
+    hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
+                          &IID_IMMDeviceEnumerator, (void**)&pEnumerator);
+    EXIT_ON_ERROR(hr);
+
     int search_err = 0;
 
     if (devid == NULL) {
@@ -791,16 +735,16 @@ static HRESULT find_and_load_device(struct ao *ao, IMMDeviceEnumerator* pEnumera
         IMMDeviceCollection_GetCount(pDevices, &count);
 
         if (devno >= count) {
-            MP_ERR(ao, "No device #%d\n", devno);
+            MP_ERR(ao, "no device #%d!\n", devno);
         } else {
-            MP_VERBOSE(ao, "Finding device #%d\n", devno);
+            MP_VERBOSE(ao, "finding device #%d\n", devno);
             hr = IMMDeviceCollection_Item(pDevices, devno, &pTempDevice);
             EXIT_ON_ERROR(hr);
 
             hr = IMMDevice_GetId(pTempDevice, &deviceID);
             EXIT_ON_ERROR(hr);
 
-            MP_VERBOSE(ao, "Found device #%d\n", devno);
+            MP_VERBOSE(ao, "found device #%d\n", devno);
         }
     } else {
         hr = IMMDeviceEnumerator_EnumAudioEndpoints(pEnumerator, eRender,
@@ -811,7 +755,7 @@ static HRESULT find_and_load_device(struct ao *ao, IMMDeviceEnumerator* pEnumera
         int count;
         IMMDeviceCollection_GetCount(pDevices, &count);
 
-        MP_VERBOSE(ao, "Finding device %s\n", devid);
+        MP_VERBOSE(ao, "finding device %s\n", devid);
 
         IMMDevice *prevDevice = NULL;
 
@@ -829,7 +773,7 @@ static HRESULT find_and_load_device(struct ao *ao, IMMDeviceEnumerator* pEnumera
                 if (deviceID) {
                     char *name;
                     if (!search_err) {
-                        MP_ERR(ao, "Multiple matching devices found\n");
+                        MP_ERR(ao, "multiple matching devices found!\n");
                         name = get_device_name(prevDevice);
                         MP_ERR(ao, "%s\n", name);
                         talloc_free(name);
@@ -848,7 +792,7 @@ static HRESULT find_and_load_device(struct ao *ao, IMMDeviceEnumerator* pEnumera
         }
 
         if (deviceID == NULL) {
-            MP_ERR(ao, "Could not find device %s\n", devid);
+            MP_ERR(ao, "could not find device %s!\n", devid);
         }
     }
 
@@ -858,20 +802,19 @@ static HRESULT find_and_load_device(struct ao *ao, IMMDeviceEnumerator* pEnumera
     if (deviceID == NULL || search_err) {
         hr = E_NOTFOUND;
     } else {
-        MP_VERBOSE(ao, "Loading device %S\n", deviceID);
+        MP_VERBOSE(ao, "loading device %S\n", deviceID);
 
         hr = IMMDeviceEnumerator_GetDevice(pEnumerator, deviceID, ppDevice);
 
         if (FAILED(hr)) {
-            MP_ERR(ao, "Could not load requested device\n");
+            MP_ERR(ao, "could not load requested device!\n");
         }
     }
 
 exit_label:
     SAFE_RELEASE(pTempDevice, IMMDevice_Release(pTempDevice));
     SAFE_RELEASE(pDevices, IMMDeviceCollection_Release(pDevices));
-
-    CoTaskMemFree(deviceID);
+    SAFE_RELEASE(pEnumerator, IMMDeviceEnumerator_Release(pEnumerator));
     return hr;
 }
 
@@ -879,11 +822,11 @@ int wasapi_validate_device(struct mp_log *log, const m_option_t *opt,
                            struct bstr name, struct bstr param)
 {
     if (bstr_equals0(param, "help")) {
-        wasapi_enumerate_devices(log, NULL, NULL);
+        wasapi_enumerate_devices(log);
         return M_OPT_EXIT;
     }
 
-    mp_dbg(log, "Validating device=%s\n", param.start);
+    mp_dbg(log, "validating device=%s\n", param.start);
 
     char *end;
     int devno = (int) strtol(param.start, &end, 10);
@@ -916,8 +859,8 @@ HRESULT wasapi_setup_proxies(struct wasapi_state *state) {
 
 exit_label:
     if (hr != S_OK) {
-        MP_ERR(state, "Error reading COM proxy: %s (0x%"PRIx32")\n",
-               wasapi_explain_err(hr), (uint32_t)hr);
+        MP_ERR(state, "error reading COM proxy: %08x %s\n", (unsigned int)hr,
+               wasapi_explain_err(hr));
     }
     return hr;
 }
@@ -952,129 +895,80 @@ exit_label:
     return hr;
 }
 
-static void destroy_proxies(struct wasapi_state *state) {
-    SAFE_RELEASE(state->sAudioClient, IUnknown_Release(state->sAudioClient));
-    SAFE_RELEASE(state->sAudioVolume, IUnknown_Release(state->sAudioVolume));
-    SAFE_RELEASE(state->sEndpointVolume, IUnknown_Release(state->sEndpointVolume));
-    SAFE_RELEASE(state->sSessionControl, IUnknown_Release(state->sSessionControl));
-}
-
-void wasapi_dispatch(void)
-{
-    /* dispatch any possible pending messages */
-    MSG msg;
-    while (PeekMessage(&msg, NULL, 0, 0, PM_REMOVE))
-        DispatchMessage(&msg);
-}
-
-HRESULT wasapi_thread_init(struct ao *ao)
+int wasapi_thread_init(struct ao *ao)
 {
     struct wasapi_state *state = (struct wasapi_state *)ao->priv;
     HRESULT hr;
-    MP_DBG(ao, "Init wasapi thread\n");
-    int64_t retry_wait = 1;
-retry:
-    state->initial_volume = -1.0;
+    CoInitializeEx(NULL, COINIT_APARTMENTTHREADED);
 
-    hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
-                          &IID_IMMDeviceEnumerator, (void**)&state->pEnumerator);
-    EXIT_ON_ERROR(hr);
+    if (!state->opt_device) {
+        IMMDeviceEnumerator *pEnumerator;
+        hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
+                              &IID_IMMDeviceEnumerator, (void**)&pEnumerator);
+        EXIT_ON_ERROR(hr);
 
-    char *device = state->opt_device;
-    if (!device || !device[0])
-        device = ao->device;
+        hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint(pEnumerator,
+                                                         eRender, eConsole,
+                                                         &state->pDevice);
+        SAFE_RELEASE(pEnumerator, IMMDeviceEnumerator_Release(pEnumerator));
 
-
-    if (!device || !device[0])
-        hr = load_default_device(ao, state->pEnumerator, &state->pDevice);
-    else
-        hr = find_and_load_device(ao, state->pEnumerator, &state->pDevice, device);
+        char *id = get_device_id(state->pDevice);
+        MP_VERBOSE(ao, "default device ID: %s\n", id);
+        talloc_free(id);
+    } else {
+        hr = find_and_load_device(ao, &state->pDevice, state->opt_device);
+    }
     EXIT_ON_ERROR(hr);
 
     char *name = get_device_name(state->pDevice);
-    MP_VERBOSE(ao, "Device loaded: %s\n", name);
+    MP_VERBOSE(ao, "device loaded: %s\n", name);
     talloc_free(name);
 
-    MP_DBG(ao, "Activating pAudioClient interface\n");
     hr = IMMDeviceActivator_Activate(state->pDevice, &IID_IAudioClient,
                                      CLSCTX_ALL, NULL, (void **)&state->pAudioClient);
     EXIT_ON_ERROR(hr);
 
-    MP_DBG(ao, "Activating pEndpointVolume interface\n");
     hr = IMMDeviceActivator_Activate(state->pDevice, &IID_IAudioEndpointVolume,
                                      CLSCTX_ALL, NULL,
                                      (void **)&state->pEndpointVolume);
     EXIT_ON_ERROR(hr);
+    IAudioEndpointVolume_QueryHardwareSupport(state->pEndpointVolume,
+                                              &state->vol_hw_support);
 
-    MP_DBG(ao, "Query hardware volume support\n");
-    hr = IAudioEndpointVolume_QueryHardwareSupport(state->pEndpointVolume,
-                                                   &state->vol_hw_support);
-    if ( hr != S_OK )
-        MP_WARN(ao, "Query hardware volume control: %s (0x%"PRIx32")\n",
-                wasapi_explain_err(hr), (uint32_t)hr);
+    state->init_ret = find_formats(ao); /* Probe support formats */
+    if (state->init_ret)
+        goto exit_label;
+    if (!fix_format(state)) { /* now that we're sure what format to use */
+        EXIT_ON_ERROR(create_proxies(state));
 
-    MP_DBG(ao, "Probing formats\n");
-    if (find_formats(ao)){
-        hr = E_FAIL;
-        EXIT_ON_ERROR(hr);
+        if (state->opt_exclusive)
+            IAudioEndpointVolume_GetMasterVolumeLevelScalar(state->pEndpointVolume,
+                                                            &state->initial_volume);
+        else
+            ISimpleAudioVolume_GetMasterVolume(state->pAudioVolume,
+                                               &state->initial_volume);
+
+        state->previous_volume = state->initial_volume;
+
+        MP_VERBOSE(ao, "thread_init OK!\n");
+        SetEvent(state->init_done);
+        return state->init_ret;
     }
-
-    MP_DBG(ao, "Fixing format\n");
-    hr = fix_format(ao);
-    if ( (hr == AUDCLNT_E_DEVICE_IN_USE ||
-          hr == AUDCLNT_E_DEVICE_INVALIDATED) &&
-         retry_wait <= 8 ) {
-        wasapi_thread_uninit(ao);
-        MP_WARN(ao, "Retrying in %"PRId64" us\n", retry_wait);
-        mp_sleep_us(retry_wait);
-        retry_wait *= 2;
-        goto retry;
-    }
-    EXIT_ON_ERROR(hr);
-
-    MP_DBG(ao, "Creating proxies\n");
-    hr = create_proxies(state);
-    EXIT_ON_ERROR(hr);
-
-    MP_DBG(ao, "Read volume levels\n");
-    if (state->opt_exclusive)
-        IAudioEndpointVolume_GetMasterVolumeLevelScalar(state->pEndpointVolume,
-                                                        &state->initial_volume);
-    else
-        ISimpleAudioVolume_GetMasterVolume(state->pAudioVolume,
-                                           &state->initial_volume);
-
-    state->previous_volume = state->initial_volume;
-
-    wasapi_change_init(ao);
-
-    MP_DBG(ao, "Init wasapi thread done\n");
-    return S_OK;
 exit_label:
-    MP_ERR(state, "Error setting up audio thread: %s (0x%"PRIx32")\n",
-           wasapi_explain_err(hr), (uint32_t)hr);
-    return hr;
+    state->init_ret = -1;
+    SetEvent(state->init_done);
+    return -1;
 }
 
-void wasapi_thread_uninit(struct ao *ao)
+void wasapi_thread_uninit(wasapi_state *state)
 {
-    struct wasapi_state *state = (struct wasapi_state *)ao->priv;
-
-    wasapi_dispatch();
-
     if (state->pAudioClient)
         IAudioClient_Stop(state->pAudioClient);
 
-    wasapi_change_uninit(ao);
-
-    if (state->opt_exclusive &&
-        state->pEndpointVolume &&
-        state->initial_volume > 0 )
+    if (state->opt_exclusive)
         IAudioEndpointVolume_SetMasterVolumeLevelScalar(state->pEndpointVolume,
                                                         state->initial_volume, NULL);
 
-    destroy_proxies(state);
-
     SAFE_RELEASE(state->pRenderClient,   IAudioRenderClient_Release(state->pRenderClient));
     SAFE_RELEASE(state->pAudioClock,     IAudioClock_Release(state->pAudioClock));
     SAFE_RELEASE(state->pAudioVolume,    ISimpleAudioVolume_Release(state->pAudioVolume));
@@ -1082,9 +976,9 @@ void wasapi_thread_uninit(struct ao *ao)
     SAFE_RELEASE(state->pSessionControl, IAudioSessionControl_Release(state->pSessionControl));
     SAFE_RELEASE(state->pAudioClient,    IAudioClient_Release(state->pAudioClient));
     SAFE_RELEASE(state->pDevice,         IMMDevice_Release(state->pDevice));
-    SAFE_RELEASE(state->pEnumerator,     IMMDeviceEnumerator_Release(state->pEnumerator));
 
     if (state->hTask)
         state->VistaBlob.pAvRevertMmThreadCharacteristics(state->hTask);
-    MP_DBG(ao, "Thread uninit done\n");
+    CoUninitialize();
+    ExitThread(0);
 }

audio/out/ao_wasapi_utils.h

@@ -31,16 +31,13 @@ int wasapi_fill_VistaBlob(wasapi_state *state);
 
 const char *wasapi_explain_err(const HRESULT hr);
 
-int wasapi_enumerate_devices(struct mp_log *log, struct ao *ao,
-                             struct ao_device_list *list);
+int wasapi_enumerate_devices(struct mp_log *log);
 
 int wasapi_validate_device(struct mp_log *log, const m_option_t *opt,
                            struct bstr name, struct bstr param);
 
-
-void wasapi_dispatch(void);
-HRESULT wasapi_thread_init(struct ao *ao);
-void wasapi_thread_uninit(struct ao *ao);
+int wasapi_thread_init(struct ao *ao);
+void wasapi_thread_uninit(wasapi_state *state);
 
 HRESULT wasapi_setup_proxies(wasapi_state *state);
 void wasapi_release_proxies(wasapi_state *state);

audio/out/internal.h

@@ -22,8 +22,8 @@
 #include <stdbool.h>
 #include <pthread.h>
 
-#include "osdep/atomics.h"
-#include "audio/out/ao.h"
+#include "audio/chmap.h"
+#include "audio/chmap_sel.h"
 
 /* global data used by ao.c and ao drivers */
 struct ao {
@@ -45,19 +45,6 @@ struct ao {
     struct input_ctx *input_ctx;
     struct mp_log *log; // Using e.g. "[ao/coreaudio]" as prefix
 
-    // The device as selected by the user, usually using ao_device_desc.name
-    // from an entry from the list returned by driver->list_devices. If the
-    // default device should be used, this is set to NULL.
-    char *device;
-
-    // Application name to report to the audio API.
-    char *client_name;
-
-    // Used during init: if init fails, redirect to this ao
-    char *redirect;
-
-    atomic_bool request_reload;
-
     int buffer;
     double def_buffer;
     void *api_priv;
@@ -135,7 +122,7 @@ struct ao_driver {
     // push based: see ao_play()
     int (*play)(struct ao *ao, void **data, int samples, int flags);
     // push based: see ao_get_delay()
-    double (*get_delay)(struct ao *ao);
+    float (*get_delay)(struct ao *ao);
     // push based: block until all queued audio is played (optional)
     void (*drain)(struct ao *ao);
     // Optional. Return true if audio has stopped in any way.
@@ -155,20 +142,6 @@ struct ao_driver {
     // In combination with wait(). Lock may or may not be held.
     void (*wakeup)(struct ao *ao);
 
-    // Return the list of devices currently available in the system. Use
-    // ao_device_list_add() to add entries. The selected device will be set as
-    // ao->device (using ao_device_desc.name).
-    // Warning: the ao struct passed doesn't necessarily have ao_driver->init()
-    //          called on it - in this case, ->uninit() won't be called either
-    //          after this function. The idea is that list_devs can be called
-    //          both when no audio or when audio is active. the latter can
-    //          happen if the audio config change at runtime, and in this case
-    //          we don't want to force a new connection to the audio server
-    //          just to update the device list. For runtime updates, ->init()
-    //          will have been called. In both cases, ao->priv is properly
-    //          allocated. (Runtime updates are not used/supported yet.)
-    void (*list_devs)(struct ao *ao, struct ao_device_list *list);
-
     // For option parsing (see vo.h)
     int priv_size;
     const void *priv_defaults;
@@ -189,9 +162,4 @@ bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
 bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
                           struct mp_chmap *map, int num);
 
-// Add a deep copy of e to the list.
-// Call from ao_driver->list_devs callback only.
-void ao_device_list_add(struct ao_device_list *list, struct ao *ao,
-                        struct ao_device_desc *e);
-
 #endif

audio/out/pull.c

@@ -43,11 +43,8 @@ enum {
                     // finished, but device is open)
     AO_STATE_WAIT,  // wait for callback to go into AO_STATE_NONE state
     AO_STATE_PLAY,  // play the buffer
-    AO_STATE_BUSY,  // like AO_STATE_PLAY, but ao_read_data() is being called
 };
 
-#define IS_PLAYING(st) ((st) == AO_STATE_PLAY || (st) == AO_STATE_BUSY)
-
 struct ao_pull_state {
     // Be very careful with the order when accessing planes.
     struct mp_ring *buffers[MP_NUM_CHANNELS];
@@ -59,21 +56,6 @@ struct ao_pull_state {
     atomic_llong end_time_us;
 };
 
-static void set_state(struct ao *ao, int new_state)
-{
-    struct ao_pull_state *p = ao->api_priv;
-    while (1) {
-        int old = atomic_load(&p->state);
-        if (old == AO_STATE_BUSY) {
-            // A spinlock, because some audio APIs don't want us to use mutexes.
-            mp_sleep_us(1);
-            continue;
-        }
-        if (atomic_compare_exchange_strong(&p->state, &old, new_state))
-            break;
-    }
-}
-
 static int get_space(struct ao *ao)
 {
     struct ao_pull_state *p = ao->api_priv;
@@ -97,10 +79,8 @@ static int play(struct ao *ao, void **data, int samples, int flags)
         int r = mp_ring_write(p->buffers[n], data[n], write_bytes);
         assert(r == write_bytes);
     }
-
-    int state = atomic_load(&p->state);
-    if (!IS_PLAYING(state)) {
-        set_state(ao, AO_STATE_PLAY);
+    if (atomic_load(&p->state) != AO_STATE_PLAY) {
+        atomic_store(&p->state, AO_STATE_PLAY);
         ao->driver->resume(ao);
     }
 
@@ -121,13 +101,14 @@ int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us)
 
     struct ao_pull_state *p = ao->api_priv;
     int full_bytes = samples * ao->sstride;
-    bool need_wakeup = false;
+    int state = atomic_load(&p->state);
     int bytes = 0;
 
-    // Play silence in states other than AO_STATE_PLAY.
-    if (!atomic_compare_exchange_strong(&p->state, &(int){AO_STATE_PLAY},
-                                        AO_STATE_BUSY))
+    if (state != AO_STATE_PLAY) {
+        if (state == AO_STATE_WAIT)
+            atomic_store(&p->state, AO_STATE_NONE);
         goto end;
+    }
 
     // Since the writer will write the first plane last, its buffered amount
     // of data is the minimum amount across all planes.
@@ -143,16 +124,10 @@ int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us)
     }
 
     // Half of the buffer played -> request more.
-    need_wakeup = buffered_bytes - bytes <= mp_ring_size(p->buffers[0]) / 2;
-
-    // Should never fail.
-    atomic_compare_exchange_strong(&p->state, &(int){AO_STATE_BUSY}, AO_STATE_PLAY);
-
-end:
-
-    if (need_wakeup)
+    if (buffered_bytes - bytes <= mp_ring_size(p->buffers[0]) / 2)
         mp_input_wakeup_nolock(ao->input_ctx);
 
+end:
     // pad with silence (underflow/paused/eof)
     for (int n = 0; n < ao->num_planes; n++)
         af_fill_silence(data[n], full_bytes - bytes, ao->format);
@@ -172,7 +147,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
 // it takes until the last sample in the buffer reaches the speakers. This is
 // used for audio/video synchronization, so it's very important to implement
 // this correctly.
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     struct ao_pull_state *p = ao->api_priv;
 
@@ -185,9 +160,19 @@ static double get_delay(struct ao *ao)
 static void reset(struct ao *ao)
 {
     struct ao_pull_state *p = ao->api_priv;
-    if (ao->driver->reset)
+    if (ao->driver->reset) {
         ao->driver->reset(ao); // assumes the audio callback thread is stopped
-    set_state(ao, AO_STATE_NONE);
+        atomic_store(&p->state, AO_STATE_NONE);
+    } else {
+        // The thread keeps running. Wait until the audio callback gets into
+        // a defined state where it won't touch the ringbuffer. We must do
+        // this, because emptying the ringbuffer is not an atomic operation.
+        if (atomic_load(&p->state) != AO_STATE_NONE) {
+            atomic_store(&p->state, AO_STATE_WAIT);
+            while (atomic_load(&p->state) != AO_STATE_NONE)
+                mp_sleep_us(1);
+        }
+    }
     for (int n = 0; n < ao->num_planes; n++)
         mp_ring_reset(p->buffers[n]);
     atomic_store(&p->end_time_us, 0);
@@ -195,22 +180,23 @@ static void reset(struct ao *ao)
 
 static void pause(struct ao *ao)
 {
+    struct ao_pull_state *p = ao->api_priv;
     if (ao->driver->reset)
         ao->driver->reset(ao);
-    set_state(ao, AO_STATE_NONE);
+    atomic_store(&p->state, AO_STATE_NONE);
 }
 
 static void resume(struct ao *ao)
 {
-    set_state(ao, AO_STATE_PLAY);
+    struct ao_pull_state *p = ao->api_priv;
+    atomic_store(&p->state, AO_STATE_PLAY);
     ao->driver->resume(ao);
 }
 
 static void drain(struct ao *ao)
 {
     struct ao_pull_state *p = ao->api_priv;
-    int state = atomic_load(&p->state);
-    if (IS_PLAYING(state))
+    if (atomic_load(&p->state) == AO_STATE_PLAY)
         mp_sleep_us(get_delay(ao) * 1000000);
     reset(ao);
 }

audio/out/push.c

@@ -93,11 +93,11 @@ static double unlocked_get_delay(struct ao *ao)
     return driver_delay + mp_audio_buffer_seconds(p->buffer);
 }
 
-static double get_delay(struct ao *ao)
+static float get_delay(struct ao *ao)
 {
     struct ao_push_state *p = ao->api_priv;
     pthread_mutex_lock(&p->lock);
-    double delay = unlocked_get_delay(ao);
+    float delay = unlocked_get_delay(ao);
     pthread_mutex_unlock(&p->lock);
     return delay;
 }
@@ -234,10 +234,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
 
     p->final_chunk = is_final;
     p->paused = false;
-    if (got_data) {
-        p->still_playing = true;
-        p->expected_end_time = 0;
-    }
+    p->still_playing |= write_samples > 0;
 
     // If we don't have new data, the decoder thread basically promises it
     // will send new data as soon as it's available.
@@ -303,7 +300,6 @@ static void *playthread(void *arg)
 {
     struct ao *ao = arg;
     struct ao_push_state *p = ao->api_priv;
-    mpthread_set_name("ao");
     pthread_mutex_lock(&p->lock);
     while (!p->terminate) {
         if (!p->paused)
@@ -352,8 +348,7 @@ static void *playthread(void *arg)
                     if (ao->driver->get_delay)
                         timeout = ao->driver->get_delay(ao);
                     timeout *= 0.25; // wake up if 25% played
-                    if (!p->need_wakeup)
-                        mpthread_cond_timedwait_rel(&p->wakeup, &p->lock, timeout);
+                    mpthread_cond_timedwait_rel(&p->wakeup, &p->lock, timeout);
                 }
             }
             MP_STATS(ao, "end audio wait");
@@ -464,7 +459,7 @@ int ao_wait_poll(struct ao *ao, struct pollfd *fds, int num_fds,
     assert(ao->api == &ao_api_push);
     assert(&p->lock == lock);
 
-    if (num_fds >= MAX_POLL_FDS || p->wakeup_pipe[0] < 0)
+    if (num_fds > MAX_POLL_FDS || p->wakeup_pipe[0] < 0)
         return -1;
 
     struct pollfd p_fds[MAX_POLL_FDS];

bootstrap.py

@@ -5,9 +5,9 @@
 from __future__ import print_function
 import os, sys, stat, hashlib, subprocess
 
-WAFRELEASE = "waf-1.8.1"
+WAFRELEASE = "waf-1.7.16"
 WAFURL     = "http://ftp.waf.io/pub/release/" + WAFRELEASE
-SHA256HASH = "ec658116ba0b96629d91fde0b32321849e866e0819f1e835c4c2c7f7ffe1a21d"
+SHA256HASH = "b64dc26c882572415fd450b745006107965f3fe17b357e3eb43d6676c9635a61"
 
 if os.path.exists("waf"):
     wafver = subprocess.check_output(['./waf', '--version']).decode()
@@ -31,8 +31,6 @@ if SHA256HASH == hashlib.sha256(waf).hexdigest():
     print("Checksum verified.")
 else:
     print("The checksum of the downloaded file does not match!")
-    print(" - got:      {}".format(hashlib.sha256(waf).hexdigest()))
-    print(" - expected: {}".format(SHA256HASH))
     print("Please download and verify the file manually.")
 
     sys.exit(1)

common/common.c

@@ -171,7 +171,7 @@ static bool mp_parse_escape(void *talloc_ctx, bstr *dst, bstr *code)
     if (code->start[0] == 'x' && code->len >= 3) {
         bstr num = bstr_splice(*code, 1, 3);
         char c = bstrtoll(num, &num, 16);
-        if (num.len)
+        if (!num.len)
             return false;
         bstr_xappend(talloc_ctx, dst, (bstr){&c, 1});
         *code = bstr_cut(*code, 3);

common/common.h

@@ -60,8 +60,8 @@ enum stream_type {
     STREAM_TYPE_COUNT,
 };
 
-extern const char *const mpv_version;
-extern const char *const mpv_builddate;
+extern const char *mpv_version;
+extern const char *mpv_builddate;
 
 char *mp_format_time(double time, bool fractions);
 char *mp_format_time_fmt(const char *fmt, double time);

common/encode_lavc.c

@@ -1145,7 +1145,6 @@ void encode_lavc_fail(struct encode_lavc_context *ctx, const char *format, ...)
     va_list va;
     va_start(va, format);
     mp_msg_va(ctx->log, MSGL_ERR, format, va);
-    va_end(va);
     if (ctx->failed)
         return;
     ctx->failed = true;

common/msg.c

@@ -37,8 +37,6 @@
 #include "osdep/io.h"
 #include "osdep/timer.h"
 
-#include "libmpv/client.h"
-
 #include "msg.h"
 #include "msg_control.h"
 
@@ -53,6 +51,7 @@ struct mp_log_root {
     bool module;
     bool show_time;
     bool termosd;       // use terminal control codes for status line
+    bool header;        // indicate that message header should be printed
     int blank_lines;    // number of lines useable by status
     int status_lines;   // number of current status lines
     bool color;
@@ -68,8 +67,6 @@ struct mp_log_root {
      * (This is perhaps better than maintaining a globally accessible and
      * synchronized mp_log tree.) */
     atomic_ulong reload_counter;
-    // --- protected by mp_msg_lock
-    char buffer[MSGSIZE_MAX];
 };
 
 struct mp_log {
@@ -158,9 +155,6 @@ static void prepare_status_line(struct mp_log_root *root, char *new_status)
     }
 
     size_t old_lines = root->status_lines;
-    if (!new_status[0] && old_lines == 0)
-        return; // nothing to clear
-
     size_t clear_lines = MPMIN(MPMAX(new_lines, old_lines), root->blank_lines);
 
     // clear the status line itself
@@ -235,42 +229,70 @@ static void pretty_print_module(FILE* stream, const char *prefix, bool use_color
         set_msg_color(stream, lev);
 }
 
-static bool test_terminal_level(struct mp_log *log, int lev)
+static void print_msg_on_terminal(struct mp_log *log, int lev, char *text)
 {
-    return lev <= log->terminal_level &&
-           !(lev == MSGL_STATUS && terminal_in_background());
-}
-
-static void print_terminal_line(struct mp_log *log, int lev, char *text)
-{
-    if (!test_terminal_level(log, lev))
-        return;
-
     struct mp_log_root *root = log->root;
     FILE *stream = (root->force_stderr || lev == MSGL_STATUS) ? stderr : stdout;
 
-    if (lev != MSGL_STATUS)
+    if (!(lev <= log->terminal_level))
+        return;
+
+    bool header = root->header;
+    const char *prefix = log->prefix;
+    char *terminate = NULL;
+
+    if ((lev >= MSGL_V) || root->verbose || root->module)
+        prefix = log->verbose_prefix;
+
+    if (lev == MSGL_STATUS) {
+        // skip status line output if stderr is a tty but in background
+        if (terminal_in_background())
+            return;
+        // don't clear if we don't have to
+        if (!text[0] && !root->status_lines)
+            return;
+        if (root->termosd) {
+            prepare_status_line(root, text);
+            terminate = "\r";
+        } else {
+            terminate = "\n";
+        }
+        root->header = true;
+    } else {
         flush_status_line(root);
+        size_t len = strlen(text);
+        root->header = len && text[len - 1] == '\n';
+        if (lev == MSGL_STATS)
+            terminate = "\n";
+    }
 
     if (root->color)
         set_msg_color(stream, lev);
 
-    if (root->show_time)
-        fprintf(stream, "[%" PRId64 "] ", mp_time_us());
+    do {
+        if (header) {
+            if (root->show_time)
+                fprintf(stream, "[%" PRId64 "] ", mp_time_us());
 
-    const char *prefix = log->prefix;
-    if ((lev >= MSGL_V) || root->verbose || root->module)
-        prefix = log->verbose_prefix;
-
-    if (prefix) {
-        if (root->module) {
-            pretty_print_module(stream, prefix, root->color, lev);
-        } else {
-            fprintf(stream, "[%s] ", prefix);
+            if (prefix) {
+                if (root->module) {
+                    pretty_print_module(stream, prefix, root->color, lev);
+                } else {
+                    fprintf(stream, "[%s] ", prefix);
+                }
+            }
         }
-    }
 
-    fprintf(stream, "%s", text);
+        char *next = strchr(text, '\n');
+        int len = next ? next - text + 1 : strlen(text);
+        fprintf(stream, "%.*s", len, text);
+        text = text + len;
+
+        header = true;
+    } while (text[0]);
+
+    if (terminate)
+        fprintf(stream, "%s", terminate);
 
     if (root->color)
         set_term_color(stream, -1);
@@ -282,7 +304,7 @@ static void write_msg_to_buffers(struct mp_log *log, int lev, char *text)
     struct mp_log_root *root = log->root;
     for (int n = 0; n < root->num_buffers; n++) {
         struct mp_log_buffer *buffer = root->buffers[n];
-        if (lev <= buffer->level && lev != MSGL_STATUS) {
+        if (lev <= buffer->level) {
             // Assuming a single writer (serialized by msg lock)
             int avail = mp_ring_available(buffer->ring) / sizeof(void *);
             if (avail < 1)
@@ -299,7 +321,7 @@ static void write_msg_to_buffers(struct mp_log *log, int lev, char *text)
                 *entry = (struct mp_log_buffer_entry) {
                     .prefix = "overflow",
                     .level = MSGL_FATAL,
-                    .text = "log message buffer overflow\n",
+                    .text = "",
                 };
             }
             mp_ring_write(buffer->ring, (unsigned char *)&entry, sizeof(entry));
@@ -326,54 +348,15 @@ void mp_msg_va(struct mp_log *log, int lev, const char *format, va_list va)
     pthread_mutex_lock(&mp_msg_lock);
 
     char tmp[MSGSIZE_MAX];
-    bool use_tmp = lev == MSGL_STATUS || lev == MSGL_STATS;
+    if (vsnprintf(tmp, MSGSIZE_MAX, format, va) < 0)
+        snprintf(tmp, MSGSIZE_MAX, "[fprintf error]\n");
+    tmp[MSGSIZE_MAX - 2] = '\n';
+    tmp[MSGSIZE_MAX - 1] = 0;
+    char *text = tmp;
 
-    struct mp_log_root *root = log->root;
-    char *text = use_tmp ? tmp : root->buffer;
-    int len = use_tmp ? 0 : strlen(text);
-
-    if (vsnprintf(text + len, MSGSIZE_MAX - len, format, va) < 0)
-        snprintf(text + len, MSGSIZE_MAX - len, "[fprintf error]\n");
-    text[MSGSIZE_MAX - 2] = '\n';
-    text[MSGSIZE_MAX - 1] = 0;
-
-    if (lev == MSGL_STATS) {
-        dump_stats(log, lev, text);
-    } else if (lev == MSGL_STATUS && !test_terminal_level(log, lev)) {
-        /* discard */
-    } else {
-        if (lev == MSGL_STATUS && root->termosd)
-            prepare_status_line(root, text);
-
-        // Split away each line. Normally we require full lines; buffer partial
-        // lines if they happen.
-        while (1) {
-            char *end = strchr(text, '\n');
-            if (!end)
-                break;
-            char *next = &end[1];
-            char saved = next[0];
-            next[0] = '\0';
-            print_terminal_line(log, lev, text);
-            write_msg_to_buffers(log, lev, text);
-            next[0] = saved;
-            text = next;
-        }
-
-        if (lev == MSGL_STATUS) {
-            if (text[0]) {
-                len = strlen(text);
-                if (len < MSGSIZE_MAX - 1) {
-                    text[len] = root->termosd ? '\r' : '\n';
-                    text[len + 1] = '\0';
-                }
-                print_terminal_line(log, lev, text);
-            }
-        } else {
-            int leftover = strlen(text);
-            memmove(root->buffer, text, leftover + 1);
-        }
-    }
+    print_msg_on_terminal(log, lev, text);
+    write_msg_to_buffers(log, lev, text);
+    dump_stats(log, lev, text);
 
     pthread_mutex_unlock(&mp_msg_lock);
 }
@@ -427,6 +410,7 @@ void mp_msg_init(struct mpv_global *global)
     struct mp_log_root *root = talloc_zero(NULL, struct mp_log_root);
     *root = (struct mp_log_root){
         .global = global,
+        .header = true,
         .reload_counter = ATOMIC_VAR_INIT(1),
     };
 
@@ -604,18 +588,6 @@ const char *const mp_log_levels[MSGL_MAX + 1] = {
     [MSGL_STATS]        = "stats",
 };
 
-const int mp_mpv_log_levels[MSGL_MAX + 1] = {
-    [MSGL_FATAL]        = MPV_LOG_LEVEL_FATAL,
-    [MSGL_ERR]          = MPV_LOG_LEVEL_ERROR,
-    [MSGL_WARN]         = MPV_LOG_LEVEL_WARN,
-    [MSGL_INFO]         = MPV_LOG_LEVEL_INFO,
-    [MSGL_STATUS]       = 0, // never used
-    [MSGL_V]            = MPV_LOG_LEVEL_V,
-    [MSGL_DEBUG]        = MPV_LOG_LEVEL_DEBUG,
-    [MSGL_TRACE]        = MPV_LOG_LEVEL_TRACE,
-    [MSGL_STATS]        = 0, // never used
-};
-
 int mp_msg_split_msglevel(struct bstr *s, struct bstr *out_mod, int *out_level)
 {
     if (s->len == 0)

common/msg_control.h

@@ -32,6 +32,5 @@ struct bstr;
 int mp_msg_split_msglevel(struct bstr *s, struct bstr *out_mod, int *out_level);
 
 extern const char *const mp_log_levels[MSGL_MAX + 1];
-extern const int mp_mpv_log_levels[MSGL_MAX + 1];
 
 #endif

common/version.c

@@ -16,7 +16,6 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "common.h"
 #include "version.h"
 #ifdef NO_BUILD_TIMESTAMPS
 #undef BUILDDATE

demux/codec_tags.c

@@ -352,8 +352,8 @@ static const char *lookup_tag(const struct mp_codec_tag *mp_table,
     return id == AV_CODEC_ID_NONE ? NULL : mp_codec_from_av_codec_id(id);
 }
 
-static const char *const pcm_le[] = {"pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"};
-static const char *const pcm_be[] = {"pcm_s8", "pcm_s16be", "pcm_s24be", "pcm_s32be"};
+static const char *pcm_le[] = {"pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"};
+static const char *pcm_be[] = {"pcm_s8", "pcm_s16be", "pcm_s24be", "pcm_s32be"};
 
 static const char *map_audio_pcm_tag(uint32_t tag, int bits)
 {

demux/demux.c

@@ -33,7 +33,6 @@
 #include "talloc.h"
 #include "common/msg.h"
 #include "common/global.h"
-#include "osdep/threads.h"
 
 #include "stream/stream.h"
 #include "demux.h"
@@ -123,15 +122,12 @@ struct demux_internal {
     double seek_pts;
 
     // Cached state.
-    bool force_cache_update;
     double time_length;
     struct mp_tags *stream_metadata;
     int64_t stream_size;
     int64_t stream_cache_size;
     int64_t stream_cache_fill;
     int stream_cache_idle;
-    // Updated during init only.
-    char *stream_base_filename;
 };
 
 struct demux_stream {
@@ -149,11 +145,9 @@ struct demux_stream {
     struct demux_packet *tail;
 };
 
-// Return "a", or if that is NOPTS, return "def".
-#define PTS_OR_DEF(a, def) ((a) == MP_NOPTS_VALUE ? (def) : (a))
 // If one of the values is NOPTS, always pick the other one.
-#define MP_PTS_MIN(a, b) MPMIN(PTS_OR_DEF(a, b), PTS_OR_DEF(b, a))
-#define MP_PTS_MAX(a, b) MPMAX(PTS_OR_DEF(a, b), PTS_OR_DEF(b, a))
+#define MP_PTS_MIN(a, b) ((a) == MP_NOPTS_VALUE || ((a) > (b)) ? (b) : (a))
+#define MP_PTS_MAX(a, b) ((a) == MP_NOPTS_VALUE || ((a) < (b)) ? (b) : (a))
 
 static void demuxer_sort_chapters(demuxer_t *demuxer);
 static void *demux_thread(void *pctx);
@@ -195,7 +189,6 @@ struct sh_stream *new_sh_stream(demuxer_t *demuxer, enum stream_type type)
     *sh = (struct sh_stream) {
         .type = type,
         .index = demuxer->num_streams,
-        .ff_index = demuxer->num_streams,
         .demuxer_id = demuxer_id, // may be overwritten by demuxer
         .ds = talloc(sh, struct demux_stream),
     };
@@ -390,9 +383,10 @@ static bool read_packet(struct demux_internal *in)
     pthread_mutex_unlock(&in->lock);
     struct demuxer *demux = in->d_thread;
     bool eof = !demux->desc->fill_buffer || demux->desc->fill_buffer(demux) <= 0;
-    update_cache(in);
     pthread_mutex_lock(&in->lock);
 
+    update_cache(in);
+
     if (eof) {
         for (int n = 0; n < in->d_buffer->num_streams; n++) {
             struct demux_stream *ds = in->d_buffer->streams[n]->ds;
@@ -461,7 +455,6 @@ static void execute_seek(struct demux_internal *in)
 static void *demux_thread(void *pctx)
 {
     struct demux_internal *in = pctx;
-    mpthread_set_name("demux");
     pthread_mutex_lock(&in->lock);
     while (!in->thread_terminate) {
         in->thread_paused = in->thread_request_pause > 0;
@@ -482,13 +475,7 @@ static void *demux_thread(void *pctx)
             if (read_packet(in))
                 continue; // read_packet unlocked, so recheck conditions
         }
-        if (in->force_cache_update) {
-            pthread_mutex_unlock(&in->lock);
-            update_cache(in);
-            pthread_mutex_lock(&in->lock);
-            in->force_cache_update = false;
-            continue;
-        }
+        update_cache(in);
         pthread_cond_signal(&in->wakeup);
         pthread_cond_wait(&in->wakeup, &in->lock);
     }
@@ -766,10 +753,10 @@ void demux_changed(demuxer_t *demuxer, int events)
 
     demuxer->events |= events;
 
-    update_cache(in);
-
     pthread_mutex_lock(&in->lock);
 
+    update_cache(in);
+
     if (demuxer->events & DEMUX_EVENT_INIT)
         demuxer_sort_chapters(demuxer);
     if (demuxer->events & (DEMUX_EVENT_METADATA | DEMUX_EVENT_STREAMS))
@@ -787,26 +774,15 @@ void demux_update(demuxer_t *demuxer)
     assert(demuxer == demuxer->in->d_user);
     struct demux_internal *in = demuxer->in;
 
+    pthread_mutex_lock(&in->lock);
     if (!in->threading)
         update_cache(in);
-
-    pthread_mutex_lock(&in->lock);
     demux_copy(demuxer, in->d_buffer);
     if (in->stream_metadata && (demuxer->events & DEMUX_EVENT_METADATA))
         mp_tags_merge(demuxer->metadata, in->stream_metadata);
     pthread_mutex_unlock(&in->lock);
 }
 
-static void demux_init_cache(struct demuxer *demuxer)
-{
-    struct demux_internal *in = demuxer->in;
-    struct stream *stream = demuxer->stream;
-
-    char *base = NULL;
-    stream_control(stream, STREAM_CTRL_GET_BASE_FILENAME, &base);
-    in->stream_base_filename = talloc_steal(demuxer, base);
-}
-
 static struct demuxer *open_given_type(struct mpv_global *global,
                                        struct mp_log *log,
                                        const struct demuxer_desc *desc,
@@ -854,18 +830,13 @@ static struct demuxer *open_given_type(struct mpv_global *global,
     in->d_user->metadata = talloc_zero(in->d_user, struct mp_tags);
     in->d_buffer->metadata = talloc_zero(in->d_buffer, struct mp_tags);
 
+    int64_t start_pos = stream_tell(stream);
+
     mp_verbose(log, "Trying demuxer: %s (force-level: %s)\n",
                desc->name, d_level(check));
 
     in->d_thread->params = params; // temporary during open()
 
-    if (stream->seekable) // not for DVD/BD/DVB in particular
-        stream_seek(stream, 0);
-
-    // Peek this much data to avoid that stream_read() run by some demuxers
-    // or stream filters will flush previous peeked data.
-    stream_peek(stream, STREAM_BUFFER_SIZE);
-
     int ret = demuxer->desc->open(in->d_thread, check);
     if (ret >= 0) {
         in->d_thread->params = NULL;
@@ -874,20 +845,19 @@ static struct demuxer *open_given_type(struct mpv_global *global,
                        in->d_thread->filetype, desc->desc);
         else
             mp_verbose(log, "Detected file format: %s\n", desc->desc);
-        if (!in->d_thread->seekable)
-            mp_warn(log, "Stream is not seekable.\n");
         // Pretend we can seek if we can't seek, but there's a cache.
         if (!in->d_thread->seekable && stream->uncached_stream) {
-            mp_warn(log, "Enabling seeking because stream cache is active.\n");
+            mp_warn(log,
+                    "File is not seekable, but there's a cache: enabling seeking.\n");
             in->d_thread->seekable = true;
         }
-        demux_init_cache(demuxer);
         demux_changed(in->d_thread, DEMUX_EVENT_ALL);
         demux_update(demuxer);
         return demuxer;
     }
 
     free_demuxer(demuxer);
+    stream_seek(stream, start_pos);
     return NULL;
 }
 
@@ -923,6 +893,10 @@ struct demuxer *demux_open(struct stream *stream, char *force_format,
         }
     }
 
+    // Peek this much data to avoid that stream_read() run by some demuxers
+    // or stream filters will flush previous peeked data.
+    stream_peek(stream, STREAM_BUFFER_SIZE);
+
     // Test demuxers from first to last, one pass for each check_levels[] entry
     for (int pass = 0; check_levels[pass] != -1; pass++) {
         enum demux_check level = check_levels[pass];
@@ -963,7 +937,7 @@ void demux_flush(demuxer_t *demuxer)
     pthread_mutex_unlock(&demuxer->in->lock);
 }
 
-int demux_seek(demuxer_t *demuxer, double rel_seek_secs, int flags)
+int demux_seek(demuxer_t *demuxer, float rel_seek_secs, int flags)
 {
     struct demux_internal *in = demuxer->in;
     assert(demuxer == in->d_user);
@@ -973,22 +947,9 @@ int demux_seek(demuxer_t *demuxer, double rel_seek_secs, int flags)
         return 0;
     }
 
-    if ((flags & SEEK_FACTOR) && !(flags & SEEK_ABSOLUTE)) {
-        MP_WARN(demuxer, "Invalid seek flags.\n");
-        return 0;
-    }
-
     if (rel_seek_secs == MP_NOPTS_VALUE && (flags & SEEK_ABSOLUTE))
         return 0;
 
-    if (!(flags & (SEEK_BACKWARD | SEEK_FORWARD))) {
-        if (flags & SEEK_ABSOLUTE || rel_seek_secs < 0) {
-            flags |= SEEK_BACKWARD;
-        } else {
-            flags |= SEEK_FORWARD;
-        }
-    }
-
     pthread_mutex_lock(&in->lock);
 
     flush_locked(demuxer);
@@ -1086,9 +1047,9 @@ static int chapter_compare(const void *p1, const void *p2)
     struct demux_chapter *c1 = (void *)p1;
     struct demux_chapter *c2 = (void *)p2;
 
-    if (c1->pts > c2->pts)
+    if (c1->start > c2->start)
         return 1;
-    else if (c1->pts < c2->pts)
+    else if (c1->start < c2->start)
         return -1;
     return c1->original_index > c2->original_index ? 1 :-1; // never equal
 }
@@ -1100,11 +1061,12 @@ static void demuxer_sort_chapters(demuxer_t *demuxer)
 }
 
 int demuxer_add_chapter(demuxer_t *demuxer, struct bstr name,
-                        double pts, uint64_t demuxer_id)
+                        uint64_t start, uint64_t end, uint64_t demuxer_id)
 {
     struct demux_chapter new = {
         .original_index = demuxer->num_chapters,
-        .pts = pts,
+        .start = start,
+        .end = end,
         .name = name.len ? bstrdup0(demuxer, name) : NULL,
         .metadata = talloc_zero(demuxer, struct mp_tags),
         .demuxer_id = demuxer_id,
@@ -1122,53 +1084,42 @@ double demuxer_get_time_length(struct demuxer *demuxer)
     return -1;
 }
 
-// must be called not locked
+// must be called locked
 static void update_cache(struct demux_internal *in)
 {
     struct demuxer *demuxer = in->d_thread;
     struct stream *stream = demuxer->stream;
 
-    // Don't lock while querying the stream.
-    double time_length = -1;
-    struct mp_tags *stream_metadata = NULL;
-    int64_t stream_size = -1;
-    int64_t stream_cache_size = -1;
-    int64_t stream_cache_fill = -1;
-    int stream_cache_idle = -1;
-
+    in->time_length = -1;
     if (demuxer->desc->control) {
         demuxer->desc->control(demuxer, DEMUXER_CTRL_GET_TIME_LENGTH,
-                               &time_length);
+                               &in->time_length);
     }
 
-    stream_control(stream, STREAM_CTRL_GET_METADATA, &stream_metadata);
-    stream_control(stream, STREAM_CTRL_GET_SIZE, &stream_size);
-    stream_control(stream, STREAM_CTRL_GET_CACHE_SIZE, &stream_cache_size);
-    stream_control(stream, STREAM_CTRL_GET_CACHE_FILL, &stream_cache_fill);
-    stream_control(stream, STREAM_CTRL_GET_CACHE_IDLE, &stream_cache_idle);
-
-    pthread_mutex_lock(&in->lock);
-    in->time_length = time_length;
-    in->stream_size = stream_size;
-    in->stream_cache_size = stream_cache_size;
-    in->stream_cache_fill = stream_cache_fill;
-    in->stream_cache_idle = stream_cache_idle;
-    if (stream_metadata) {
+    struct mp_tags *s_meta = NULL;
+    stream_control(stream, STREAM_CTRL_GET_METADATA, &s_meta);
+    if (s_meta) {
         talloc_free(in->stream_metadata);
-        in->stream_metadata = talloc_steal(in, stream_metadata);
+        in->stream_metadata = talloc_steal(in, s_meta);
         in->d_buffer->events |= DEMUX_EVENT_METADATA;
     }
-    pthread_mutex_unlock(&in->lock);
+
+    in->stream_size = -1;
+    stream_control(stream, STREAM_CTRL_GET_SIZE, &in->stream_size);
+    in->stream_cache_size = -1;
+    stream_control(stream, STREAM_CTRL_GET_CACHE_SIZE, &in->stream_cache_size);
+    in->stream_cache_fill = -1;
+    stream_control(stream, STREAM_CTRL_GET_CACHE_FILL, &in->stream_cache_fill);
+    in->stream_cache_idle = -1;
+    stream_control(stream, STREAM_CTRL_GET_CACHE_IDLE, &in->stream_cache_idle);
 }
 
 // must be called locked
 static int cached_stream_control(struct demux_internal *in, int cmd, void *arg)
 {
     // If the cache is active, wake up the thread to possibly update cache state.
-    if (in->stream_cache_size >= 0) {
-        in->force_cache_update = true;
+    if (in->stream_cache_size >= 0)
         pthread_cond_signal(&in->wakeup);
-    }
 
     switch (cmd) {
     case STREAM_CTRL_GET_CACHE_SIZE:
@@ -1191,11 +1142,6 @@ static int cached_stream_control(struct demux_internal *in, int cmd, void *arg)
             return STREAM_UNSUPPORTED;
         *(int64_t *)arg = in->stream_size;
         return STREAM_OK;
-    case STREAM_CTRL_GET_BASE_FILENAME:
-        if (!in->stream_base_filename)
-            return STREAM_UNSUPPORTED;
-        *(char **)arg = talloc_strdup(NULL, in->stream_base_filename);
-        return STREAM_OK;
     }
     return STREAM_ERROR;
 }
@@ -1228,24 +1174,18 @@ static int cached_demux_control(struct demux_internal *in, int cmd, void *arg)
             .ts_range = {MP_NOPTS_VALUE, MP_NOPTS_VALUE},
             .ts_duration = -1,
         };
-        int num_packets = 0;
         for (int n = 0; n < in->d_user->num_streams; n++) {
             struct demux_stream *ds = in->d_user->streams[n]->ds;
             if (ds->active) {
-                r->underrun |= !ds->head && !ds->eof;
-                if (!ds->eof) {
-                    r->ts_range[0] = MP_PTS_MAX(r->ts_range[0], ds->base_ts);
-                    r->ts_range[1] = MP_PTS_MIN(r->ts_range[1], ds->last_ts);
-                }
-                num_packets += ds->packs;
+                r->underrun |= !ds->head;
+                r->ts_range[0] = MP_PTS_MAX(r->ts_range[0], ds->base_ts);
+                r->ts_range[1] = MP_PTS_MIN(r->ts_range[1], ds->last_ts);
             }
         }
         r->idle = (in->idle && !r->underrun) || r->eof;
         r->underrun &= !r->idle;
         if (r->ts_range[0] != MP_NOPTS_VALUE && r->ts_range[1] != MP_NOPTS_VALUE)
-            r->ts_duration = MPMAX(0, r->ts_range[1] - r->ts_range[0]);
-        if (!num_packets || in->seeking)
-            r->ts_duration = 0;
+            r->ts_duration = r->ts_range[1] - r->ts_range[0];
         return DEMUXER_CTRL_OK;
     }
     }
@@ -1326,14 +1266,3 @@ void demux_unpause(demuxer_t *demuxer)
     pthread_cond_signal(&in->wakeup);
     pthread_mutex_unlock(&in->lock);
 }
-
-struct demux_chapter *demux_copy_chapter_data(struct demux_chapter *c, int num)
-{
-    struct demux_chapter *new = talloc_array(NULL, struct demux_chapter, num);
-    for (int n = 0; n < num; n++) {
-        new[n] = c[n];
-        new[n].name = talloc_strdup(new, new[n].name);
-        new[n].metadata = mp_tags_dup(new, new[n].metadata);
-    }
-    return new;
-}

demux/demux.h

@@ -123,7 +123,7 @@ typedef struct demuxer_desc {
 typedef struct demux_chapter
 {
     int original_index;
-    double pts;
+    uint64_t start, end;
     char *name;
     struct mp_tags *metadata;
     uint64_t demuxer_id; // for mapping to internal demuxer data structures
@@ -254,7 +254,7 @@ void demux_stop_thread(struct demuxer *demuxer);
 void demux_set_wakeup_cb(struct demuxer *demuxer, void (*cb)(void *ctx), void *ctx);
 
 void demux_flush(struct demuxer *demuxer);
-int demux_seek(struct demuxer *demuxer, double rel_seek_secs, int flags);
+int demux_seek(struct demuxer *demuxer, float rel_seek_secs, int flags);
 
 int demux_control(struct demuxer *demuxer, int cmd, void *arg);
 
@@ -268,8 +268,8 @@ void demuxer_help(struct mp_log *log);
 
 int demuxer_add_attachment(struct demuxer *demuxer, struct bstr name,
                            struct bstr type, struct bstr data);
-int demuxer_add_chapter(demuxer_t *demuxer, struct bstr name,
-                        double pts, uint64_t demuxer_id);
+int demuxer_add_chapter(struct demuxer *demuxer, struct bstr name,
+                        uint64_t start, uint64_t end, uint64_t demuxer_id);
 
 double demuxer_get_time_length(struct demuxer *demuxer);
 
@@ -284,8 +284,6 @@ void demux_update(demuxer_t *demuxer);
 struct sh_stream *demuxer_stream_by_demuxer_id(struct demuxer *d,
                                                enum stream_type t, int id);
 
-struct demux_chapter *demux_copy_chapter_data(struct demux_chapter *c, int num);
-
 bool demux_matroska_uid_cmp(struct matroska_segment_uid *a,
                             struct matroska_segment_uid *b);
 

demux/demux_cue.c

@@ -35,9 +35,13 @@ static int try_open_file(struct demuxer *demuxer, enum demux_check check)
 {
     struct stream *s = demuxer->stream;
     if (check >= DEMUX_CHECK_UNSAFE) {
-        bstr d = stream_peek(s, PROBE_SIZE);
-        if (d.len < 1 || !mp_probe_cue(d))
+        char buf[PROBE_SIZE];
+        int len = stream_read(s, buf, sizeof(buf));
+        if (len <= 0)
             return -1;
+        if (!mp_probe_cue((struct bstr) { buf, len }))
+            return -1;
+        stream_seek(s, 0);
     }
     demuxer->file_contents = stream_read_complete(s, demuxer, 1000000);
     if (demuxer->file_contents.start == NULL)

demux/demux_disc.c

@@ -275,7 +275,7 @@ static void add_stream_chapters(struct demuxer *demuxer)
         double p = n;
         if (stream_control(demuxer->stream, STREAM_CTRL_GET_CHAPTER_TIME, &p) < 1)
             continue;
-        demuxer_add_chapter(demuxer, bstr0(""), p, 0);
+        demuxer_add_chapter(demuxer, bstr0(""), p * 1e9, 0, 0);
     }
 }
 

demux/demux_lavf.c

@@ -101,7 +101,6 @@ typedef struct lavf_priv {
     AVFormatContext *avfc;
     AVIOContext *pb;
     int64_t last_pts;
-    bool init_pts;
     struct sh_stream **streams; // NULL for unknown streams
     int num_streams;
     int cur_program;
@@ -190,11 +189,8 @@ static int64_t mp_read_seek(void *opaque, int stream_idx, int64_t ts, int flags)
         .flags = flags,
     };
 
-    if (stream_control(stream, STREAM_CTRL_AVSEEK, &cmd) == STREAM_OK) {
-        stream_drop_buffers(stream);
-        return 0;
-    }
-    return AVERROR(ENOSYS);
+    int ret = stream_control(stream, STREAM_CTRL_AVSEEK, &cmd);
+    return ret < 1 ? AVERROR(ENOSYS) : 0;
 }
 
 static void list_formats(struct demuxer *demuxer)
@@ -286,8 +282,6 @@ static int lavf_check_file(demuxer_t *demuxer, enum demux_check check)
         .buf_size = 0,
         .buf = av_mallocz(PROBE_BUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE),
     };
-    if (!avpd.buf)
-        return -1;
 
     bool final_probe = false;
     do {
@@ -593,7 +587,6 @@ static void handle_stream(demuxer_t *demuxer, int i)
     MP_TARRAY_APPEND(priv, priv->streams, priv->num_streams, sh);
 
     if (sh) {
-        sh->ff_index = st->index;
         sh->codec = mp_codec_from_av_codec_id(codec->codec_id);
         sh->lav_headers = codec;
 
@@ -681,8 +674,6 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
         return -1;
 
     avfc = avformat_alloc_context();
-    if (!avfc)
-        return -1;
 
     if (lavfdopts->cryptokey)
         parse_cryptokey(avfc, lavfdopts->cryptokey);
@@ -715,13 +706,9 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
                    "analyzeduration to %f\n", analyze_duration);
     }
 
-    AVDictionary *dopts = NULL;
-
     if ((priv->avif->flags & AVFMT_NOFILE) ||
-        demuxer->stream->type == STREAMTYPE_AVDEVICE ||
-        matches_avinputformat_name(priv, "hls"))
+        demuxer->stream->type == STREAMTYPE_AVDEVICE)
     {
-        mp_setup_av_network_options(&dopts, demuxer->global, demuxer->log, opts);
         // This might be incorrect.
         demuxer->seekable = true;
     } else {
@@ -737,10 +724,10 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
         priv->pb->read_seek = mp_read_seek;
         priv->pb->seekable = demuxer->seekable ? AVIO_SEEKABLE_NORMAL : 0;
         avfc->pb = priv->pb;
-        if (stream_control(demuxer->stream, STREAM_CTRL_HAS_AVSEEK, NULL) > 0)
-            demuxer->seekable = true;
     }
 
+    AVDictionary *dopts = NULL;
+
     if (matches_avinputformat_name(priv, "rtsp")) {
         const char *transport = NULL;
         switch (opts->network_rtsp_transport) {
@@ -774,9 +761,13 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
 
     for (i = 0; i < avfc->nb_chapters; i++) {
         AVChapter *c = avfc->chapters[i];
+        uint64_t start = av_rescale_q(c->start, c->time_base,
+                                      (AVRational){1, 1000000000});
+        uint64_t end   = av_rescale_q(c->end, c->time_base,
+                                      (AVRational){1, 1000000000});
         t = av_dict_get(c->metadata, "title", NULL, 0);
         int index = demuxer_add_chapter(demuxer, t ? bstr0(t->value) : bstr0(""),
-                                        c->start * av_q2d(c->time_base), i);
+                                        start, end, i);
         mp_tags_copy_from_av_dictionary(demuxer->chapters[index].metadata, c->metadata);
     }
 
@@ -837,12 +828,6 @@ static int demux_lavf_fill_buffer(demuxer_t *demux)
         return 1;
     }
 
-    if (!priv->init_pts && (priv->avfc->flags & AVFMT_NOTIMESTAMPS)) {
-        if (pkt->pts == AV_NOPTS_VALUE && pkt->dts == AV_NOPTS_VALUE)
-            pkt->dts = 0;
-        priv->init_pts = true;
-    }
-
     if (pkt->pts != AV_NOPTS_VALUE)
         dp->pts = pkt->pts * av_q2d(st->time_base);
     if (pkt->dts != AV_NOPTS_VALUE)
@@ -867,8 +852,6 @@ static void demux_seek_lavf(demuxer_t *demuxer, double rel_seek_secs, int flags)
     lavf_priv_t *priv = demuxer->priv;
     int avsflags = 0;
 
-    priv->init_pts = false;
-
     if (flags & SEEK_ABSOLUTE)
         priv->last_pts = 0;
     else if (rel_seek_secs < 0)

demux/demux_mkv.c

@@ -112,7 +112,6 @@ typedef struct mkv_track {
     float a_osfreq;
 
     double default_duration;
-    double codec_delay;
 
     int default_track;
 
@@ -131,13 +130,13 @@ typedef struct mkv_track {
     double ra_pts;              /* previous audio timestamp */
 
   /** realaudio descrambling */
-    uint32_t sub_packet_size;   ///< sub packet size, per stream
-    uint32_t sub_packet_h;      ///< number of coded frames per block
+    uint16_t sub_packet_size;   ///< sub packet size, per stream
+    uint16_t sub_packet_h;      ///< number of coded frames per block
     uint32_t coded_framesize;   ///< coded frame size, per stream
-    uint32_t audiopk_size;      ///< audio packet size
+    uint16_t audiopk_size;      ///< audio packet size
     unsigned char *audio_buf;   ///< place to store reordered audio data
     double *audio_timestamp;    ///< timestamp for each audio packet
-    uint32_t sub_packet_cnt;    ///< number of subpacket already received
+    uint16_t sub_packet_cnt;    ///< number of subpacket already received
     int audio_filepos;          ///< file position of first audio packet in block
 
     /* generic content encoding support */
@@ -150,8 +149,7 @@ typedef struct mkv_track {
 
 typedef struct mkv_index {
     int tnum;
-    uint64_t timecode, duration;
-    uint64_t filepos; // position of the cluster which contains the packet
+    uint64_t timecode, filepos;
 } mkv_index_t;
 
 typedef struct mkv_demuxer {
@@ -184,10 +182,7 @@ typedef struct mkv_demuxer {
 
     uint64_t skip_to_timecode;
     int v_skip_to_keyframe, a_skip_to_keyframe;
-    int a_skip_preroll;
     int subtitle_preroll;
-
-    bool index_has_durations;
 } mkv_demuxer_t;
 
 #define REALHEADER_SIZE    16
@@ -199,8 +194,6 @@ typedef struct mkv_demuxer {
 // (Subtitle packets added before first A/V keyframe packet is found with seek.)
 #define NUM_SUB_PREROLL_PACKETS 500
 
-static void probe_last_timestamp(struct demuxer *demuxer);
-
 #define AAC_SYNC_EXTENSION_TYPE 0x02b7
 static int aac_get_sample_rate_index(uint32_t sample_rate)
 {
@@ -345,10 +338,6 @@ static int demux_mkv_read_info(demuxer_t *demuxer)
     if (info.n_timecode_scale) {
         mkv_d->tc_scale = info.timecode_scale;
         MP_VERBOSE(demuxer, "| + timecode scale: %" PRIu64 "\n", mkv_d->tc_scale);
-        if (mkv_d->tc_scale < 1 || mkv_d->tc_scale > INT_MAX) {
-            res = -1;
-            goto out;
-        }
     }
     if (info.n_duration) {
         mkv_d->duration = info.duration * mkv_d->tc_scale / 1e9;
@@ -637,9 +626,6 @@ static void parse_trackentry(struct demuxer *demuxer,
     if (entry->n_content_encodings)
         parse_trackencodings(demuxer, track, &entry->content_encodings);
 
-    if (entry->n_codec_delay)
-        track->codec_delay = entry->codec_delay / 1e9;
-
     mkv_d->tracks[mkv_d->num_tracks++] = track;
 }
 
@@ -666,40 +652,34 @@ static int demux_mkv_read_tracks(demuxer_t *demuxer)
 }
 
 static void cue_index_add(demuxer_t *demuxer, int track_id, uint64_t filepos,
-                          uint64_t timecode, uint64_t duration)
+                          uint64_t timecode)
 {
     mkv_demuxer_t *mkv_d = (mkv_demuxer_t *) demuxer->priv;
 
     MP_TARRAY_GROW(mkv_d, mkv_d->indexes, mkv_d->num_indexes);
 
-    mkv_d->indexes[mkv_d->num_indexes] = (mkv_index_t) {
-        .tnum = track_id,
-        .filepos = filepos,
-        .timecode = timecode,
-        .duration = duration,
-    };
-
+    mkv_d->indexes[mkv_d->num_indexes].tnum = track_id;
+    mkv_d->indexes[mkv_d->num_indexes].timecode = timecode;
+    mkv_d->indexes[mkv_d->num_indexes].filepos = filepos;
     mkv_d->num_indexes++;
 }
 
 static void add_block_position(demuxer_t *demuxer, struct mkv_track *track,
-                               uint64_t filepos,
-                               uint64_t timecode, uint64_t duration)
+                               uint64_t filepos, uint64_t timecode)
 {
     mkv_demuxer_t *mkv_d = (mkv_demuxer_t *) demuxer->priv;
 
     if (mkv_d->index_complete || !track)
         return;
-
-    mkv_d->index_has_durations = true;
-
     if (track->last_index_entry != (size_t)-1) {
         mkv_index_t *index = &mkv_d->indexes[track->last_index_entry];
-        // Never add blocks which are already covered by the index.
-        if (index->timecode >= timecode)
+        // filepos is always the cluster position, which can contain multiple
+        // blocks with different timecodes - one is enough.
+        // Also, never add block which are already covered by the index.
+        if (index->filepos == filepos || index->timecode >= timecode)
             return;
     }
-    cue_index_add(demuxer, track->tnum, filepos, timecode, duration);
+    cue_index_add(demuxer, track->tnum, filepos, timecode);
     track->last_index_entry = mkv_d->num_indexes - 1;
 }
 
@@ -723,7 +703,6 @@ static int demux_mkv_read_cues(demuxer_t *demuxer)
         return -1;
 
     mkv_d->num_indexes = 0;
-    mkv_d->index_has_durations = false;
 
     for (int i = 0; i < cues.n_cue_point; i++) {
         struct ebml_cue_point *cuepoint = &cues.cue_point[i];
@@ -736,14 +715,10 @@ static int demux_mkv_read_cues(demuxer_t *demuxer)
             struct ebml_cue_track_positions *trackpos =
                 &cuepoint->cue_track_positions[c];
             uint64_t pos = mkv_d->segment_start + trackpos->cue_cluster_position;
-            cue_index_add(demuxer, trackpos->cue_track, pos,
-                          time, trackpos->cue_duration);
-            mkv_d->index_has_durations |= trackpos->n_cue_duration > 0;
+            cue_index_add(demuxer, trackpos->cue_track, pos, time);
             MP_DBG(demuxer, "|+ found cue point for track %" PRIu64
-                   ": timecode %" PRIu64 ", filepos: %" PRIu64
-                   " offset %" PRIu64 ", duration %" PRIu64 "\n",
-                   trackpos->cue_track, time, pos,
-                   trackpos->cue_relative_position, trackpos->cue_duration);
+                   ": timecode %" PRIu64 ", filepos: %" PRIu64 "\n",
+                   trackpos->cue_track, time, pos);
         }
     }
 
@@ -904,7 +879,7 @@ static int demux_mkv_read_chapters(struct demuxer *demuxer)
                    BSTR_P(name));
 
             if (idx == selected_edition) {
-                demuxer_add_chapter(demuxer, name, chapter.start / 1e9,
+                demuxer_add_chapter(demuxer, name, chapter.start, chapter.end,
                                     ca->chapter_uid);
             }
             if (m_chapters) {
@@ -1174,7 +1149,7 @@ static const videocodec_info_t vinfo[] = {
 
 static int demux_mkv_open_video(demuxer_t *demuxer, mkv_track_t *track)
 {
-    unsigned char *extradata = NULL;
+    unsigned char *extradata;
     unsigned int extradata_size = 0;
     bool raw = false;
     struct sh_stream *sh = new_sh_stream(demuxer, STREAM_VIDEO);
@@ -1418,22 +1393,20 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)
         sh_a->bitrate = 16000 * 8;
         sh_a->block_align = 1024;
 
-        if (!strcmp(track->codec_id, MKV_A_AAC)) {
-            if (track->private_data && !extradata_len) {
+        if (!strcmp(track->codec_id, MKV_A_AAC) && track->private_data) {
+            if (!extradata_len) {
                 extradata = track->private_data;
                 extradata_len = track->private_size;
             }
         } else {
             /* Recreate the 'private data' */
+            /* which faad2 uses in its initialization */
             srate_idx = aac_get_sample_rate_index(track->a_sfreq);
-            const char *tail = "";
-            if (strlen(track->codec_id) >= 12)
-                tail = &track->codec_id[12];
-            if (!strncmp(tail, "MAIN", 4))
+            if (!strncmp(&track->codec_id[12], "MAIN", 4))
                 profile = 0;
-            else if (!strncmp(tail, "LC", 2))
+            else if (!strncmp(&track->codec_id[12], "LC", 2))
                 profile = 1;
-            else if (!strncmp(tail, "SSR", 3))
+            else if (!strncmp(&track->codec_id[12], "SSR", 3))
                 profile = 2;
             else
                 profile = 3;
@@ -1477,7 +1450,7 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)
         /* Common initialization for all RealAudio codecs */
         unsigned char *src = track->private_data;
         int codecdata_length, version;
-        unsigned int flavor;
+        int flavor;
 
         sh_a->bitrate = 0;  /* FIXME !? */
 
@@ -1513,20 +1486,14 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)
 
         switch (track->a_formattag) {
         case MP_FOURCC('a', 't', 'r', 'c'):
-            if (flavor >= MP_ARRAY_SIZE(atrc_fl2bps))
-                goto error;
             sh_a->bitrate = atrc_fl2bps[flavor] * 8;
             sh_a->block_align = track->sub_packet_size;
             goto audiobuf;
         case MP_FOURCC('c', 'o', 'o', 'k'):
-            if (flavor >= MP_ARRAY_SIZE(cook_fl2bps))
-                goto error;
             sh_a->bitrate = cook_fl2bps[flavor] * 8;
             sh_a->block_align = track->sub_packet_size;
             goto audiobuf;
         case MP_FOURCC('s', 'i', 'p', 'r'):
-            if (flavor >= MP_ARRAY_SIZE(sipr_fl2bps))
-                goto error;
             sh_a->bitrate = sipr_fl2bps[flavor] * 8;
             sh_a->block_align = track->coded_framesize;
             goto audiobuf;
@@ -1771,13 +1738,6 @@ static int demux_mkv_open(demuxer_t *demuxer, enum demux_check check)
     int64_t start_pos;
     int64_t end_pos;
 
-    bstr start = stream_peek(s, 4);
-    uint32_t start_id = 0;
-    for (int n = 0; n < start.len; n++)
-        start_id = (start_id << 8) | start.start[n];
-    if (start_id != EBML_ID_EBML)
-        return -1;
-
     if (!read_ebml_header(demuxer))
         return -1;
     MP_VERBOSE(demuxer, "Found the head...\n");
@@ -1790,7 +1750,6 @@ static int demux_mkv_open(demuxer_t *demuxer, enum demux_check check)
     mkv_d->tc_scale = 1000000;
     mkv_d->segment_start = stream_tell(s);
     mkv_d->segment_end = end_pos;
-    mkv_d->a_skip_preroll = 1;
 
     if (demuxer->params && demuxer->params->matroska_was_valid)
         *demuxer->params->matroska_was_valid = true;
@@ -1859,9 +1818,6 @@ static int demux_mkv_open(demuxer_t *demuxer, enum demux_check check)
     process_tags(demuxer);
     display_create_tracks(demuxer);
 
-    if (demuxer->opts->mkv_probe_duration)
-        probe_last_timestamp(demuxer);
-
     return 0;
 }
 
@@ -2019,11 +1975,11 @@ static void handle_realaudio(demuxer_t *demuxer, mkv_track_t *track,
                              bstr data, bool keyframe)
 {
     mkv_demuxer_t *mkv_d = (mkv_demuxer_t *) demuxer->priv;
-    uint32_t sps = track->sub_packet_size;
-    uint32_t sph = track->sub_packet_h;
+    uint16_t sps = track->sub_packet_size;
+    uint16_t sph = track->sub_packet_h;
     uint32_t cfs = track->coded_framesize; // restricted to [1,0x40000000]
-    uint32_t w = track->audiopk_size;
-    uint32_t spc = track->sub_packet_cnt;
+    uint16_t w = track->audiopk_size;
+    uint16_t spc = track->sub_packet_cnt;
     uint8_t *buffer = data.start;
     uint32_t size = data.len;
     demux_packet_t *dp;
@@ -2302,7 +2258,7 @@ static bool mkv_parse_packet(mkv_track_t *track, bstr *raw, bstr *out)
 }
 
 struct block_info {
-    uint64_t duration, discardpadding;
+    uint64_t duration;
     bool simple, keyframe;
     uint64_t timecode;
     mkv_track_t *track;
@@ -2323,8 +2279,7 @@ static void index_block(demuxer_t *demuxer, struct block_info *block)
     mkv_demuxer_t *mkv_d = (mkv_demuxer_t *) demuxer->priv;
     if (block->keyframe) {
         add_block_position(demuxer, block->track, mkv_d->cluster_start,
-                           block->timecode / mkv_d->tc_scale,
-                           block->duration / mkv_d->tc_scale);
+                           block->timecode / mkv_d->tc_scale);
     }
 }
 
@@ -2403,7 +2358,7 @@ static int handle_block(demuxer_t *demuxer, struct block_info *block_info)
         return 0;
     }
 
-    current_pts = tc / 1e9 - track->codec_delay;
+    current_pts = tc / 1e9;
 
     if (track->type == MATROSKA_TRACK_AUDIO) {
         if (mkv_d->a_skip_to_keyframe)
@@ -2452,7 +2407,6 @@ static int handle_block(demuxer_t *demuxer, struct block_info *block_info)
                     if (!dp)
                         break;
                     dp->keyframe = keyframe;
-                    dp->pos = mkv_d->last_filepos;
                     /* If default_duration is 0, assume no pts value is known
                      * for packets after the first one (rather than all pts
                      * values being the same). Also, don't use it for extra
@@ -2463,13 +2417,6 @@ static int handle_block(demuxer_t *demuxer, struct block_info *block_info)
                         MPSWAP(double, dp->pts, dp->dts);
                     if (p == 0)
                         dp->duration = block_duration / 1e9;
-                    if (stream->type == STREAM_AUDIO) {
-                        unsigned int srate = track->a_sfreq;
-                        demux_packet_set_padding(dp,
-                            mkv_d->a_skip_preroll ? track->codec_delay * srate : 0,
-                            block_info->discardpadding / 1e9 * srate);
-                        mkv_d->a_skip_preroll = 0;
-                    }
                     demux_add_packet(stream, dp);
                     p++;
                 }
@@ -2508,12 +2455,6 @@ static int read_block_group(demuxer_t *demuxer, int64_t end,
             block->duration *= mkv_d->tc_scale;
             break;
 
-        case MATROSKA_ID_DISCARDPADDING:
-            block->discardpadding = ebml_read_uint(s);
-            if (block->discardpadding == EBML_UINT_INVALID)
-                goto error;
-            break;
-
         case MATROSKA_ID_BLOCK:
             if (read_block(demuxer, end, block) < 0)
                 goto error;
@@ -2739,39 +2680,14 @@ static struct mkv_index *seek_with_cues(struct demuxer *demuxer, int seek_id,
     if (index) {        /* We've found an entry. */
         uint64_t seek_pos = index->filepos;
         if (flags & SEEK_SUBPREROLL) {
-            // Find the cluster with the highest filepos, that has a timestamp
-            // still lower than min_tc.
-            double secs = demuxer->opts->mkv_subtitle_preroll_secs;
-            uint64_t pre = MPMIN(INT64_MAX, secs * 1e9 / mkv_d->tc_scale);
-            uint64_t min_tc = pre < index->timecode ? index->timecode - pre : 0;
             uint64_t prev_target = 0;
-            uint64_t prev_tc = 0;
             for (size_t i = 0; i < mkv_d->num_indexes; i++) {
                 if (seek_id < 0 || mkv_d->indexes[i].tnum == seek_id) {
-                    struct mkv_index *cur = &mkv_d->indexes[i];
-                    if (cur->timecode <= min_tc && cur->timecode >= prev_tc) {
-                        prev_tc = cur->timecode;
-                        prev_target = cur->filepos;
-                    }
+                    uint64_t index_pos = mkv_d->indexes[i].filepos;
+                    if (index_pos > prev_target && index_pos < seek_pos)
+                        prev_target = index_pos;
                 }
             }
-            if (mkv_d->index_has_durations) {
-                // Find the earliest cluster that is not before prev_target,
-                // but contains subtitle packets overlapping with the cluster
-                // at seek_pos.
-                uint64_t target = seek_pos;
-                for (size_t i = 0; i < mkv_d->num_indexes; i++) {
-                    struct mkv_index *cur = &mkv_d->indexes[i];
-                    if (cur->timecode <= index->timecode &&
-                        cur->timecode + cur->duration > index->timecode &&
-                        cur->filepos >= prev_target &&
-                        cur->filepos < target)
-                    {
-                        target = cur->filepos;
-                    }
-                }
-                prev_target = target;
-            }
             if (prev_target)
                 seek_pos = prev_target;
         }
@@ -2803,7 +2719,13 @@ static void demux_mkv_seek(demuxer_t *demuxer, double rel_seek_secs, int flags)
     mkv_d->subtitle_preroll = NUM_SUB_PREROLL_PACKETS;
     if (!st_active[STREAM_SUB] || !st_active[STREAM_VIDEO])
         flags &= ~SEEK_SUBPREROLL;
-
+    if (!(flags & (SEEK_BACKWARD | SEEK_FORWARD))) {
+        if (flags & SEEK_ABSOLUTE || rel_seek_secs < 0) {
+            flags |= SEEK_BACKWARD;
+        } else {
+            flags |= SEEK_FORWARD;
+        }
+    }
     // Adjust the target a little bit to catch cases where the target position
     // specifies a keyframe with high, but not perfect, precision.
     rel_seek_secs += flags & SEEK_FORWARD ? -0.005 : 0.005;
@@ -2826,115 +2748,58 @@ static void demux_mkv_seek(demuxer_t *demuxer, double rel_seek_secs, int flags)
         if (!index)
             stream_seek(demuxer->stream, old_pos);
 
+        mkv_d->v_skip_to_keyframe = st_active[STREAM_VIDEO];
+        mkv_d->a_skip_to_keyframe = st_active[STREAM_AUDIO];
+
         if (flags & SEEK_FORWARD) {
             mkv_d->skip_to_timecode = target_timecode;
         } else {
             mkv_d->skip_to_timecode = index ? index->timecode * mkv_d->tc_scale
                                             : 0;
         }
-    } else {
+
+        demux_mkv_fill_buffer(demuxer);
+    } else if (!(flags & SEEK_ABSOLUTE))
+        MP_VERBOSE(demuxer, "seek unsupported flags\n");
+    else {
         stream_t *s = demuxer->stream;
+        uint64_t target_filepos;
+        mkv_index_t *index = NULL;
 
         read_deferred_cues(demuxer);
 
+        if (!mkv_d->index_complete) {   /* not implemented without index */
+            MP_VERBOSE(demuxer, "seek unsupported flags\n");
+            stream_seek(s, old_pos);
+            return;
+        }
+
         int64_t size = 0;
         stream_control(s, STREAM_CTRL_GET_SIZE, &size);
 
-        int64_t target_filepos = size * MPCLAMP(rel_seek_secs, 0, 1);
+        target_filepos = (uint64_t) (size * rel_seek_secs);
+        for (size_t i = 0; i < mkv_d->num_indexes; i++)
+            if (mkv_d->indexes[i].tnum == v_tnum)
+                if ((index == NULL)
+                    || ((mkv_d->indexes[i].filepos >= target_filepos)
+                        && ((index->filepos < target_filepos)
+                            || (mkv_d->indexes[i].filepos < index->filepos))))
+                    index = &mkv_d->indexes[i];
 
-        mkv_index_t *index = NULL;
-        if (mkv_d->index_complete) {
-            for (size_t i = 0; i < mkv_d->num_indexes; i++) {
-                if (mkv_d->indexes[i].tnum == v_tnum) {
-                    if ((index == NULL)
-                        || ((mkv_d->indexes[i].filepos >= target_filepos)
-                            && ((index->filepos < target_filepos)
-                                || (mkv_d->indexes[i].filepos < index->filepos))))
-                        index = &mkv_d->indexes[i];
-                }
-            }
+        if (!index) {
+            stream_seek(s, old_pos);
+            return;
         }
 
         mkv_d->cluster_end = 0;
+        stream_seek(s, index->filepos);
 
-        if (index) {
-            stream_seek(s, index->filepos);
-            mkv_d->skip_to_timecode = index->timecode * mkv_d->tc_scale;
-        } else {
-            stream_seek(s, target_filepos);
-            if (ebml_resync_cluster(mp_null_log, s) < 0) {
-                // Assume EOF
-                mkv_d->cluster_end = size;
-            }
-        }
+        mkv_d->v_skip_to_keyframe = st_active[STREAM_VIDEO];
+        mkv_d->a_skip_to_keyframe = st_active[STREAM_AUDIO];
+        mkv_d->skip_to_timecode = index->timecode * mkv_d->tc_scale;
+
+        demux_mkv_fill_buffer(demuxer);
     }
-
-    mkv_d->v_skip_to_keyframe = st_active[STREAM_VIDEO];
-    mkv_d->a_skip_to_keyframe = st_active[STREAM_AUDIO];
-    mkv_d->a_skip_preroll = mkv_d->a_skip_to_keyframe;
-
-    demux_mkv_fill_buffer(demuxer);
-}
-
-static void probe_last_timestamp(struct demuxer *demuxer)
-{
-    mkv_demuxer_t *mkv_d = demuxer->priv;
-    int64_t old_pos = stream_tell(demuxer->stream);
-
-    if (!demuxer->seekable)
-        return;
-
-    // Pick some arbitrary video track
-    int v_tnum = -1;
-    for (int n = 0; n < mkv_d->num_tracks; n++) {
-        if (mkv_d->tracks[n]->type == MATROSKA_TRACK_VIDEO) {
-            v_tnum = mkv_d->tracks[n]->tnum;
-            break;
-        }
-    }
-    if (v_tnum < 0)
-        return;
-
-    read_deferred_cues(demuxer);
-
-    if (!mkv_d->index_complete)
-        return;
-
-    // Find last cluster that still has video packets
-    int64_t target = 0;
-    for (size_t i = 0; i < mkv_d->num_indexes; i++) {
-        struct mkv_index *cur = &mkv_d->indexes[i];
-        if (cur->tnum == v_tnum)
-            target = MPMAX(target, cur->filepos);
-    }
-    if (!target)
-        return;
-
-    if (!stream_seek(demuxer->stream, target))
-        return;
-
-    int64_t last_ts[STREAM_TYPE_COUNT] = {0};
-    while (1) {
-        struct block_info block;
-        int res = read_next_block(demuxer, &block);
-        if (res < 0)
-            break;
-        if (res > 0) {
-            if (block.track && block.track->stream) {
-                enum stream_type type = block.track->stream->type;
-                uint64_t endtime = block.timecode + block.duration;
-                if (last_ts[type] < endtime)
-                    last_ts[type] = endtime;
-            }
-            free_block(&block);
-        }
-    }
-
-    if (last_ts[STREAM_VIDEO])
-        mkv_d->duration = last_ts[STREAM_VIDEO] / 1e9;
-
-    stream_seek(demuxer->stream, old_pos);
-    mkv_d->cluster_start = mkv_d->cluster_end = 0;
 }
 
 static int demux_mkv_control(demuxer_t *demuxer, int cmd, void *arg)

demux/demux_playlist.c

@@ -27,17 +27,6 @@
 
 #define PROBE_SIZE (8 * 1024)
 
-static bool check_mimetype(struct stream *s, const char *const *list)
-{
-    if (s->mime_type) {
-        for (int n = 0; list && list[n]; n++) {
-            if (strcmp(s->mime_type, list[n]) == 0)
-                return true;
-        }
-    }
-    return false;
-}
-
 struct pl_parser {
     struct mp_log *log;
     struct stream *s;
@@ -86,7 +75,7 @@ static int parse_m3u(struct pl_parser *p)
         return -1;
     if (p->probing)
         return 0;
-    while (line.len || !pl_eof(p)) {
+    while (!pl_eof(p)) {
         if (line.len > 0 && !bstr_startswith0(line, "#"))
             pl_add(p, line);
         line = bstr_strip(pl_get_line(p));
@@ -99,22 +88,6 @@ static int parse_ref_init(struct pl_parser *p)
     bstr line = bstr_strip(pl_get_line(p));
     if (!bstr_equals0(line, "[Reference]"))
         return -1;
-
-    // ASF http streaming redirection - this is needed because ffmpeg http://
-    // and mmsh:// can not automatically switch automatically between each
-    // others. Both protocols use http - MMSH requires special http headers
-    // to "activate" it, and will in other cases return this playlist.
-    static const char *const mmsh_types[] = {"audio/x-ms-wax",
-        "audio/x-ms-wma", "video/x-ms-asf", "video/x-ms-afs", "video/x-ms-wmv",
-        "video/x-ms-wma", "application/x-mms-framed",
-        "application/vnd.ms.wms-hdr.asfv1", NULL};
-    bstr burl = bstr0(p->s->url);
-    if (bstr_eatstart0(&burl, "http://") && check_mimetype(p->s, mmsh_types)) {
-        MP_INFO(p, "Redirecting to mmsh://\n");
-        playlist_add_file(p->pl, talloc_asprintf(p, "mmsh://%.*s", BSTR_P(burl)));
-        return 0;
-    }
-
     while (!pl_eof(p)) {
         line = bstr_strip(pl_get_line(p));
         if (bstr_case_startswith(line, bstr0("Ref"))) {
@@ -180,15 +153,15 @@ static int parse_txt(struct pl_parser *p)
     return 0;
 }
 
-#define MIME_TYPES(...) \
-    .mime_types = (const char*const[]){__VA_ARGS__, NULL}
-
 struct pl_format {
     const char *name;
     int (*parse)(struct pl_parser *p);
     const char *const *mime_types;
 };
 
+#define MIME_TYPES(...) \
+    .mime_types = (const char*const[]){__VA_ARGS__, NULL}
+
 static const struct pl_format formats[] = {
     {"m3u", parse_m3u,
      MIME_TYPES("audio/mpegurl", "audio/x-mpegurl", "application/x-mpegurl")},
@@ -199,6 +172,17 @@ static const struct pl_format formats[] = {
     {"txt", parse_txt},
 };
 
+static bool check_mimetype(struct stream *s, const char *const *list)
+{
+    if (s->mime_type) {
+        for (int n = 0; list && list[n]; n++) {
+            if (strcmp(s->mime_type, list[n]) == 0)
+                return true;
+        }
+    }
+    return false;
+}
+
 static const struct pl_format *probe_pl(struct pl_parser *p)
 {
     int64_t start = stream_tell(p->s);

demux/ebml.c

@@ -173,7 +173,7 @@ uint64_t ebml_read_uint(stream_t *s)
  */
 int64_t ebml_read_int(stream_t *s)
 {
-    uint64_t value = 0;
+    int64_t value = 0;
     uint64_t len;
     int l;
 
@@ -189,7 +189,7 @@ int64_t ebml_read_int(stream_t *s)
     while (len--)
         value = (value << 8) | stream_read_char(s);
 
-    return (int64_t)value; // assume complement of 2
+    return value;
 }
 
 /*
@@ -328,12 +328,12 @@ static uint64_t ebml_parse_uint(uint8_t *data, int length)
 static int64_t ebml_parse_sint(uint8_t *data, int length)
 {
     assert(length >=1 && length <= 8);
-    uint64_t r = 0;
+    int64_t r = 0;
     if (*data & 0x80)
         r = -1;
     while (length--)
         r = (r << 8) | *data++;
-    return (int64_t)r; // assume complement of 2
+    return r;
 }
 
 static double ebml_parse_float(uint8_t *data, int length)

demux/packet.c

@@ -20,9 +20,6 @@
 #include <assert.h>
 
 #include <libavcodec/avcodec.h>
-#include <libavutil/intreadwrite.h>
-
-#include "config.h"
 
 #include "common/av_common.h"
 #include "common/common.h"
@@ -116,20 +113,3 @@ struct demux_packet *demux_copy_packet(struct demux_packet *dp)
     new->duration = dp->duration;
     return new;
 }
-
-int demux_packet_set_padding(struct demux_packet *dp, int start, int end)
-{
-#if HAVE_AVFRAME_SKIP_SAMPLES
-    if (!start  && !end)
-        return 0;
-    if (!dp->avpacket)
-        return -1;
-    uint8_t *p = av_packet_new_side_data(dp->avpacket, AV_PKT_DATA_SKIP_SAMPLES, 10);
-    if (!p)
-        return -1;
-
-    AV_WL32(p + 0, start);
-    AV_WL32(p + 4, end);
-#endif
-    return 0;
-}

demux/packet.h

@@ -44,6 +44,4 @@ void demux_packet_shorten(struct demux_packet *dp, size_t len);
 void free_demux_packet(struct demux_packet *dp);
 struct demux_packet *demux_copy_packet(struct demux_packet *dp);
 
-int demux_packet_set_padding(struct demux_packet *dp, int start, int end);
-
 #endif /* MPLAYER_DEMUX_PACKET_H */

demux/stheader.h

@@ -36,8 +36,6 @@ struct sh_stream {
     // Demuxer/format specific ID. Corresponds to the stream IDs as encoded in
     // some file formats (e.g. MPEG), or an index chosen by demux.c.
     int demuxer_id;
-    // FFmpeg stream index (AVFormatContext.streams[index]), or equivalent.
-    int ff_index;
     // One of these is non-NULL, the others are NULL, depending on the stream
     // type.
     struct sh_audio *audio;
@@ -50,7 +48,8 @@ struct sh_stream {
     // Usually a FourCC, exact meaning depends on codec.
     unsigned int format;
 
-    // Codec specific header data (set by demux_lavf.c only)
+    // Codec specific header data (set by demux_lavf.c)
+    // Other demuxers use sh_audio->wf and sh_video->bih instead.
     struct AVCodecContext *lav_headers;
 
     char *title;

etc/input.conf

@@ -55,9 +55,6 @@
 #Shift+LEFT  no-osd seek -1 - exact
 #Shift+UP    no-osd seek  5 - exact
 #Shift+DOWN  no-osd seek -5 - exact
-# Skip to previous/next subtitle (subject to some restrictions; see manpage)
-#Ctrl+LEFT   no-osd sub_seek -1
-#Ctrl+RIGHT  no-osd sub_seek  1
 #PGUP add chapter 1                     # skip to next chapter
 #PGDWN add chapter -1                   # skip to previous chapter
 #Shift+PGUP seek 600
@@ -77,7 +74,7 @@
 #, frame_back_step                      # go back by one frame and pause
 #SPACE cycle pause
 #> playlist_next                        # skip to next file
-#ENTER playlist_next                    # skip to next file
+#ENTER playlist_next force              # skip to next file or quit
 #< playlist_prev                        # skip to previous file
 #O osd                                  # cycle through OSD mode
 #o show_progress
@@ -135,7 +132,6 @@
 #CLOSE_WIN quit
 #CLOSE_WIN {encode} quit
 #E cycle edition                        # next edition
-#l ab_loop                              # Set/clear A-B loop points
 #ctrl+c quit
 
 # Apple Remote section

etc/restore-old-bindings.conf

@@ -9,10 +9,6 @@
 #
 # Older installations use ~/.mpv/input.conf instead.
 
-# changed in mpv 0.7.0
-
-ENTER playlist_next force
-
 # changed in mpv 0.6.0
 
 ESC quit

input/cmd_list.c

@@ -51,7 +51,6 @@
 #define OARG_DOUBLE(def)        OPT_DOUBLE(ARG(d), 0, OPTDEF_DOUBLE(def))
 #define OARG_INT(def)           OPT_INT(ARG(i), 0, OPTDEF_INT(def))
 #define OARG_CHOICE(def, c)     OPT_CHOICE(ARG(i), 0, c, OPTDEF_INT(def))
-#define OARG_STRING(def)        OPT_STRING(ARG(s), 0, OPTDEF_STR(def))
 
 #define OARG_CYCLEDIR(def)      OPT_CYCLEDIR(ARG(d), 0, OPTDEF_DOUBLE(def))
 
@@ -73,8 +72,7 @@ const struct mp_cmd_def mp_cmds[] = {
   { MP_CMD_QUIT, "quit", { OARG_INT(0) } },
   { MP_CMD_QUIT_WATCH_LATER, "quit_watch_later", { OARG_INT(0) } },
   { MP_CMD_STOP, "stop", },
-  { MP_CMD_FRAME_STEP, "frame_step", .allow_auto_repeat = true,
-    .on_updown = true },
+  { MP_CMD_FRAME_STEP, "frame_step", .allow_auto_repeat = true },
   { MP_CMD_FRAME_BACK_STEP, "frame_back_step", .allow_auto_repeat = true },
   { MP_CMD_PLAYLIST_NEXT, "playlist_next", {
       OARG_CHOICE(0, ({"weak", 0},              {"0", 0},
@@ -91,9 +89,7 @@ const struct mp_cmd_def mp_cmds[] = {
   { MP_CMD_SHOW_TEXT, "show_text", { ARG_STRING, OARG_INT(-1), OARG_INT(0) },
     .allow_auto_repeat = true},
   { MP_CMD_SHOW_PROGRESS, "show_progress",  .allow_auto_repeat = true},
-  { MP_CMD_SUB_ADD, "sub_add", { ARG_STRING,
-      OARG_CHOICE(0, ({"select", 0}, {"auto", 1}, {"cached", 2})),
-      OARG_STRING(""), OARG_STRING("") } },
+  { MP_CMD_SUB_ADD, "sub_add", { ARG_STRING } },
   { MP_CMD_SUB_REMOVE, "sub_remove", { OARG_INT(-1) } },
   { MP_CMD_SUB_RELOAD, "sub_reload", { OARG_INT(-1) } },
 
@@ -156,20 +152,13 @@ const struct mp_cmd_def mp_cmds[] = {
 
   { MP_CMD_DISCNAV, "discnav", { ARG_STRING } },
 
-  { MP_CMD_AB_LOOP, "ab_loop", },
-
-  { MP_CMD_DROP_BUFFERS, "drop_buffers", },
-
   { MP_CMD_AF, "af", { ARG_STRING, ARG_STRING } },
-  { MP_CMD_AO_RELOAD, "ao_reload", },
 
   { MP_CMD_VF, "vf", { ARG_STRING, ARG_STRING } },
 
   { MP_CMD_VO_CMDLINE, "vo_cmdline", { ARG_STRING } },
 
-  { MP_CMD_SCRIPT_BINDING, "script_binding", { ARG_STRING },
-    .allow_auto_repeat = true, .on_updown = true},
-
+  { MP_CMD_SCRIPT_DISPATCH, "script_dispatch", { ARG_STRING, ARG_INT } },
   { MP_CMD_SCRIPT_MESSAGE, "script_message", { ARG_STRING }, .vararg = true },
   { MP_CMD_SCRIPT_MESSAGE_TO, "script_message_to", { ARG_STRING, ARG_STRING },
     .vararg = true },
@@ -181,9 +170,6 @@ const struct mp_cmd_def mp_cmds[] = {
 
   { MP_CMD_WRITE_WATCH_LATER_CONFIG, "write_watch_later_config", },
 
-  { MP_CMD_HOOK_ADD, "hook_add", { ARG_STRING, ARG_INT, ARG_INT } },
-  { MP_CMD_HOOK_ACK, "hook_ack", { ARG_STRING } },
-
   {0}
 };
 
@@ -300,8 +286,7 @@ bool mp_input_is_abort_cmd(struct mp_cmd *cmd)
 bool mp_input_is_repeatable_cmd(struct mp_cmd *cmd)
 {
     return (cmd->def && cmd->def->allow_auto_repeat) ||
-           cmd->id == MP_CMD_COMMAND_LIST ||
-           (cmd->flags & MP_ALLOW_REPEAT);
+           cmd->id == MP_CMD_COMMAND_LIST;
 }
 
 void mp_print_cmd_list(struct mp_log *out)