Release 0.15.0

next_scanline is usually an unsigned int.

Fixes #2635 (again).

.gitignore

@@ -15,6 +15,7 @@
 /tags
 /TAGS
 /video/out/x11_icon.inc
+/video/out/opengl/nnedi3_weights.inc
 /demux/ebml_defs.c
 /demux/ebml_types.h
 /sub/osd_font.h

CONTRIBUTING.md

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

Copyright

@@ -12,6 +12,9 @@ the code to LGPLv2.1+ at a later point without asking the contributor. (This
 is a safeguard for making potential relicensing of the project to LGPLv2.1+
 easier.) Using a more liberal license compatible to LGPLv2.1+ is also ok.
 
+Some GPLv3 or LGPLv3 code in mpv can be enabled by building with --enable-gpl3,
+and in this case, the resulting binaries must be distributed under GPLv3.
+
 For information about authors and contributors, consult the git log, which
 contains the complete SVN and CVS history as well.
 

DOCS/client-api-changes.rst

@@ -32,7 +32,12 @@ API changes
 
 ::
 
+ --- mpv 0.12.0 ---
+ 1.20   - deprecate "GL_MP_D3D_interfaces"/"glMPGetD3DInterface", and introduce
+          "GL_MP_MPGetNativeDisplay"/"glMPGetNativeDisplay" (this is a
+          backwards-compatible rename)
  --- mpv 0.11.0 ---
+ --- mpv 0.10.0 ---
  1.19   - add "GL_MP_D3D_interfaces" pseudo extension to make it possible to
           use DXVA2 in OpenGL fullscreen mode in some situations
         - mpv_request_log_messages() now accepts "terminal-default" as parameter

DOCS/client_api_examples/README.md

@@ -1,35 +1,114 @@
-Client API examples
-===================
+# Client API examples
 
 All these examples use the mpv client API through libmpv.
 
-cocoa
------
+## Where are the docs?
+
+The libmpv C API is documented directly in the header files (on normal Unix
+systems, this is in `/usr/include/mpv/client.h`.
+
+libmpv merely gives you access to mpv's command interface, which is documented
+here:
+* Options (`mpv_set_option()` and friends): http://mpv.io/manual/master/#options
+* Commands (`mpv_command()` and friends): http://mpv.io/manual/master/#list-of-input-commands
+* Properties (`mpv_set_property()` and friends): http://mpv.io/manual/master/#properties
+
+Essentially everything is done with them, including loading a file, retrieving
+playback progress, and so on.
+
+## Methods of embedding the video window
+
+All of these examples concentrate on how to integrate mpv's video renderers
+with your own GUI. This is generally the hardest part. libmpv enforces a
+somewhat efficient video output method, rather than e.g. returning a RGBA
+surface in memory for each frame. The latter would be prohibitively inefficient,
+because it would require conversion on the CPU. The goal is also not requiring
+the API users to reinvent their own video rendering/scaling/presentation
+mechanisms.
+
+There are currently 2 methods of embedding video.
+
+### Native window embedding
+
+This uses the platform's native method of nesting multiple windows. For example,
+Linux/X11 can nest a window from a completely different process. The nested
+window can redraw contents on its own, and receive user input if the user
+interacts with this window.
+
+libmpv lets you specify a parent window for its own video window via the `wid`
+option. Then libmpv will create its window with your window as parent, and
+render its video inside of it.
+
+This method is highly OS-dependent. Some behavior is OS-specific. There are
+problems with focusing on X11 (the ancient X11 focus policy mismatches with
+that of modern UI toolkits - it's normally worked around, but this is not
+easily possible with raw window embedding). It seems to have stability problems
+on OSX when using the Qt toolkit.
+
+### OpenGL embedding
+
+This method lets you use libmpv's OpenGL renderer directly. You create an
+OpenGL context, and then use `mpv_opengl_cb_draw()` to render the video on
+each frame.
+
+This is more complicated, because libmpv will work directly on your own OpenGL
+state. It's also not possible to have mpv automatically receive user input.
+You will have to simulate this with the `mouse`/`keypress`/`keydown`/`keyup`
+commands.
+
+You also get much more flexibility. For example, you can actually render your
+own OSD on top of the video, something that is not possible with raw window
+embedding.
+
+### Which one to use?
+
+Due to the various platform-specific behavior and problems (in particular on
+OSX), OpenGL embedding is recommended. If you're not comfortable with requiring
+OpenGL, or want to support "direct" video output such as vdpau (which might
+win when it comes to performance and energy-saving), you should probably
+support both methods if possible.
+
+## List of examples
+
+### simple
+
+Very primitive terminal-only example. Shows some most basic API usage.
+
+### cocoa
 
 Shows how to embed the mpv video window in Objective-C/Cocoa.
 
-qt
---
+### cocoa-openglcb
+
+Similar to the cocoa sample, but shows how to integrate mpv's OpenGL renderer
+using libmpv's opengl-cb API. Since it does not require complicated interaction
+with Cocoa elements from different libraries, it's more robust.
+
+### qt
 
 Shows how to embed the mpv video window in Qt (using normal desktop widgets).
 
-qml
----
+### qt_opengl
 
-Shows how to use mpv's OpenGL video renderer in QtQuick2 with QML.
+Shows how to use mpv's OpenGL video renderer in Qt. This uses the opengl-cb API
+for video. Since it does not rely on embedding "foreign" native Windows, it's
+usually more robust, potentially faster, and it's easier to control how your
+GUI interacts with the video. You can do your own OpenGL rendering on top of
+the video as well.
 
-qml_direct
-----------
+### qml
+
+Shows how to use mpv's OpenGL video renderer in QtQuick2 with QML. Uses the
+opengl-cb API for video. Since the video is a normal QML element, it's trivial
+to create OSD overlays with QML-native graphical elements as well.
+
+### qml_direct
 
 Alternative example, which typically avoids a FBO indirection. Might be
-slightly faster, but is less flexible and harder to use.
+slightly faster, but is less flexible and harder to use. In particular, the
+video is not a normal QML element. Uses the opengl-cb API for video.
 
-sdl
----
+### sdl
 
-Show how to embed the mpv OpenGL renderer in SDL.
-
-simple
-------
-
-Very primitive terminal-only example. Shows some most basic API usage.
+Show how to embed the mpv OpenGL renderer in SDL. Uses the opengl-cb API for
+video.

DOCS/client_api_examples/qml/main.qml

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

DOCS/client_api_examples/qt_opengl/main.cpp

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

DOCS/client_api_examples/qt_opengl/mainwindow.cpp

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

DOCS/client_api_examples/qt_opengl/mainwindow.h

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

DOCS/client_api_examples/qt_opengl/mpvwidget.cpp

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

DOCS/client_api_examples/qt_opengl/mpvwidget.h

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

DOCS/client_api_examples/qt_opengl/qt_opengl.pro

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

DOCS/compile-windows.md

@@ -68,7 +68,7 @@ echo "MXE_TARGETS := i686-w64-mingw32.static" >> settings.mk
 # Build required packages. The following provide a minimum required to build
 # a reasonable mpv binary (though not an absolute minimum).
 
-make gcc ffmpeg libass jpeg pthreads lua
+make gcc ffmpeg libass jpeg lua
 
 # Add MXE binaries to $PATH
 export PATH=/opt/mxe/usr/bin/:$PATH
@@ -173,40 +173,3 @@ mv -f /mingw64/bin/libmpv.dll.a /mingw64/lib/
 sed -i 's_/mingw64/bin_/mingw64/lib_' /mingw64/lib/pkgconfig/mpv.pc
 rmdir /mingw64/bin/pkgconfig
 ```
-
-Additional dependencies
-=======================
-
-pthreads
---------
-
-mpv will use a pthreads wrapper by default. Either pthreads-win32 or
-winpthreads should work. The latter is packaged with most MinGW-w64
-environments, including MSYS2, so it shouldn't be a problem. If you don't have
-a pthreads wrapper or you want to build mpv without one, configure with:
-
-```bash
-./waf configure --enable-win32-internal-pthreads
-```
-
-libwaio
--------
-
-If you want to use ``--input-file``, you need libwaio. It's available from
-git://midipix.org/waio
-
-To compile libwaio in MSYS2, run:
-
-```bash
-git clone git://midipix.org/waio && cd waio
-
-# 32-bit build, run from mingw32_shell.bat
-./build-mingw-nt32 lib-static CC=gcc AR=ar
-cp -r include/waio /mingw32/include
-cp lib32/libwaio.a /mingw32/lib
-
-# 64-bit build, run from mingw64_shell.bat
-./build-mingw-nt64 lib-static CC=gcc AR=ar
-cp -r include/waio /mingw64/include
-cp lib64/libwaio.a /mingw64/lib
-```

DOCS/contribute.md

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

DOCS/interface-changes.rst

@@ -19,6 +19,31 @@ Interface changes
 
 ::
 
+ --- mpv 0.15.0 ---
+    - change "yadif" video filter defaults
+ --- mpv 0.14.0 ---
+    - vo_opengl interpolation now requires --video-sync=display-... to be set
+    - change some vo_opengl defaults (including changing tscale)
+    - add "vsync-ratio", "estimated-display-fps" properties
+    - add --rebase-start-time option
+      This is a breaking change to start time handling. Instead of making start
+      time handling an aspect of different options and properties (like
+      "time-pos" vs. "playback-time"), make it dependent on the new option. For
+      compatibility, the "time-start" property now always returns 0, so code
+      which attempted to handle rebasing manually will not break.
+ --- mpv 0.13.0 ---
+    - remove VO opengl-cb frame queue suboptions (no replacement)
+ --- mpv 0.12.0 ---
+    - remove --use-text-osd (useless; fontconfig isn't a requirement anymore,
+      and text rendering is also lazily initialized)
+    - some time properties (at least "playback-time", "time-pos",
+      "time-remaining", "playtime-remaining") now are unavailable if the time
+      is unknown, instead of just assuming that the internal playback position
+      is 0
+    - add --audio-fallback-to-null option
+    - replace vf_format outputlevels suboption with "video-output-levels" global
+      property/option; also remove "colormatrix-output-range" property
+    - vo_opengl: remove sharpen3/sharpen5 scale filters, add sharpen sub-option
  --- mpv 0.11.0 ---
     - add "af-metadata" property
  --- mpv 0.10.0 ---

DOCS/man/af.rst

@@ -164,7 +164,7 @@ Available filters are:
 
     .. admonition:: Examples
 
-        ``mpv --af=channels=4:[0-1,1-0,0-2,1-3] media.avi``
+        ``mpv --af=channels=4:[0-1,1-0,2-2,3-3] media.avi``
             Would change the number of channels to 4 and set up 4 routes that
             swap channel 0 and channel 1 and leave channel 2 and 3 intact.
             Observe that if media containing two channels were played back,

DOCS/man/ao.rst

@@ -20,10 +20,8 @@ normal driver parameters.
 
     See ``--ao=help`` for a list of compiled-in audio output drivers. The
     driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
-    where PulseAudio is used. On Windows, ``--ao=wasapi`` is preferred,
-    though it might cause trouble sometimes, in which case ``--ao=dsound``
-    should be used. On BSD systems, ``--ao=oss`` or `--ao=sndio`` may work
-    (the latter being experimental). On OS X systems, use ``--ao=coreaudio``.
+    where PulseAudio is used. On BSD systems, ``--ao=oss`` or ``--ao=sndio``
+    may work (the latter being experimental).
 
 .. admonition:: Examples
 
@@ -139,16 +137,13 @@ Available audio output drivers are:
         Automatically create connections to output ports (default: enabled).
         When enabled, the maximum number of output channels will be limited to
         the number of available output ports.
-    ``std-channel-layout=alsa|waveext|any``
-        Select the standard channel layout (default: alsa). JACK itself has no
+    ``std-channel-layout=waveext|any``
+        Select the standard channel layout (default: waveext). JACK itself has no
         notion of channel layouts (i.e. assigning which speaker a given
         channel is supposed to map to) - it just takes whatever the application
         outputs, and reroutes it to whatever the user defines. This means the
         user and the application are in charge of dealing with the channel
-        layout. ``alsa`` uses the old MPlayer layout, which is inspired by
-        ALSA's standard layouts. In this mode, ao_jack will refuse to play 3
-        or 7 channels (because these do not really have a defined meaning in
-        MPlayer). ``waveext`` uses WAVE_FORMAT_EXTENSIBLE order, which, even
+        layout. ``waveext`` uses WAVE_FORMAT_EXTENSIBLE order, which, even
         though it was defined by Microsoft, is the standard on many systems.
         The value ``any`` makes JACK accept whatever comes from the audio
         filter chain, regardless of channel layout and without reordering. This
@@ -168,7 +163,7 @@ Available audio output drivers are:
         will actually work. The disadvantage is that it will change the
         system-wide audio settings. This is equivalent to changing the ``Format``
         setting in the ``Audio Devices`` dialog in the ``Audio MIDI Setup``
-        utility. Note that this does not effect the selected speaker setup.
+        utility. Note that this does not affect the selected speaker setup.
 
     ``exclusive``
         Use exclusive mode access. This merely redirects to
@@ -208,19 +203,7 @@ Available audio output drivers are:
         (it used to be required to get good behavior on old PulseAudio versions).
 
         If you have stuttering video when using pulse, try to enable this
-        option. (Or alternatively, try to update PulseAudio.)
-
-``dsound`` (Windows only)
-    DirectX DirectSound audio output driver
-
-    .. note:: This driver is for compatibility with old systems.
-
-    ``device=<devicenum>``
-        Sets the device number to use. Playing a file with ``-v`` will show a
-        list of available devices.
-
-    ``buffersize=<ms>``
-        DirectSound buffer size in milliseconds (default: 200).
+        option. (Or try to update PulseAudio.)
 
 ``sdl``
     SDL 1.2+ audio output driver. Should work on any platform supported by SDL
@@ -259,7 +242,7 @@ Available audio output drivers are:
     ``speed``
         Simulated audio playback speed as a multiplier. Usually, a real audio
         device will not go exactly as fast as the system clock. It will deviate
-        just a little, and this option helps simulating this.
+        just a little, and this option helps to simulate this.
 
     ``latency``
         Simulated device latency. This is additional to EOF.
@@ -328,7 +311,7 @@ Available audio output drivers are:
         String are valid; the GUID string is guaranteed to not change
         unless the driver is uninstalled.
 
-        Also supports searching active devices by human readable name. If more
+        Also supports searching active devices by human-readable name. If more
         than one device matches the name, refuses loading it.
 
         This option is mostly deprecated in favour of the more general

DOCS/man/encode.rst

@@ -93,8 +93,8 @@ You can encode files from one format/codec to another using this facility.
         Completely empties the options list.
 
 ``--oafirst``
-    Force the audio stream to become the first stream in the output. By default
-    the order is unspecified.
+    Force the audio stream to become the first stream in the output.
+    By default, the order is unspecified.
 
 ``--ovc=<codec>``
     Specifies the output video codec. This can be a comma separated list of
@@ -134,8 +134,8 @@ You can encode files from one format/codec to another using this facility.
         Completely empties the options list.
 
 ``--ovfirst``
-    Force the video stream to become the first stream in the output. By default
-    the order is unspecified.
+    Force the video stream to become the first stream in the output.
+    By default, the order is unspecified.
 
 ``--ocopyts``
     Copies input pts to the output video (not supported by some output

DOCS/man/input.rst

@@ -329,9 +329,9 @@ List of Input Commands
     This is similar to ``sub-step``, except that it seeks video and audio
     instead of adjusting the subtitle delay.
 
-    Like with ``sub-step``, this works with external text subtitles only. For
-    embedded text subtitles (like with Matroska), this works only with subtitle
-    events that have already been displayed.
+    For embedded subtitles (like with Matroska), this works only with subtitle
+    events that have already been displayed, or are within a short prefetch
+    range.
 
 ``osd [<level>]``
     Toggle OSD level. If ``<level>`` is specified, set the OSD mode
@@ -418,7 +418,7 @@ List of Input Commands
 
     <reselect> (default)
         Select the default audio and subtitle streams, which typically selects
-        external files with highest preference. (The implementation is not
+        external files with the highest preference. (The implementation is not
         perfect, and could be improved on request.)
 
     <keep-selection>
@@ -546,6 +546,12 @@ Input Commands that are Possibly Subject to Change
         Always bind a key. (The input section that was made active most recently
         wins if there are ambiguities.)
 
+    This command can be used to dispatch arbitrary keys to a script or a client
+    API user. If the input section defines ``script-binding`` commands, it is
+    also possible to get separate events on key up/down, and relatively detailed
+    information about the key state. The special key name ``unmapped`` can be
+    used to match any unmapped key.
+
 ``overlay-add <id> <x> <y> "<file>" <offset> "<fmt>" <w> <h> <stride>``
     Add an OSD overlay sourced from raw data. This might be useful for scripts
     and applications controlling mpv, and which want to display things on top
@@ -647,14 +653,20 @@ Input Commands that are Possibly Subject to Change
     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).
+    included), with the following arguments:
+
+    1. The string ``key-binding``.
+    2. The name of the binding (as established above).
+    3. The key state as string (see below).
+    4. The key name (since mpv 0.15.0).
+
+    The key state consists of 2 letters:
+
+    1. 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).
+    2. 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
@@ -782,7 +794,7 @@ Input sections group a set of bindings, and enable or disable them at once.
 In ``input.conf``, each key binding is assigned to an input section, rather
 than actually having explicit text sections.
 
-Also see ``enable_section`` and ``disable_section`` commands.
+See also: ``enable_section`` and ``disable_section`` commands.
 
 Predefined bindings:
 
@@ -860,7 +872,11 @@ Property list
               quantities: fps and possibly rounded timestamps.)
 
 ``path``
-    Full path of the currently played file.
+    Full path of the currently played file. Usually this is exactly the same
+    string you pass on the mpv command line or the ``loadfile`` command, even
+    if it's a relative path. If you expect an absolute path, you will have to
+    determine it yourself, for example by using the ``working-directory``
+    property.
 
 ``media-title``
     If the currently played file has a ``title`` tag, use that.
@@ -913,6 +929,26 @@ Property list
 ``vo-drop-frame-count``
     Frames dropped by VO (when using ``--framedrop=vo``).
 
+``mistimed-frame-count``
+    Number of video frames that were not timed correctly in display-sync mode
+    for the sake of keeping A/V sync. This does not include external
+    circumstances, such as video rendering being too slow or the graphics
+    driver somehow skipping a vsync. It does not include rounding errors either
+    (which can happen especially with bad source timestamps). For example,
+    using the ``display-desync`` mode should never change this value from 0.
+
+``vsync-ratio``
+    For how many vsyncs a frame is displayed on average. This is available if
+    display-sync is active only. For 30 FPS video on a 60 Hz screen, this will
+    be 2. This is the moving average of what actually has been scheduled, so
+    24 FPS on 60 Hz will never remain exactly on 2.5, but jitter depending on
+    the last frame displayed.
+
+``vo-delayed-frame-count``
+    Estimated number of frames delayed due to external circumstances in
+    display-sync mode. Note that in general, mpv has to guess that this is
+    happening, and the guess can be inaccurate.
+
 ``percent-pos`` (RW)
     Position in current file (0-100). The advantage over using this instead of
     calculating it out of other properties is that it properly falls back to
@@ -923,8 +959,9 @@ Property list
     Position in current file in seconds.
 
 ``time-start``
-    Return the start time of the file. (Usually 0, but some kind of files,
-    especially transport streams, can have a different start time.)
+    Deprecated. Always returns 0. Before mpv 0.14, this used to return the start
+    time of the file (could affect e.g. transport streams). See
+    ``--rebase-start-time`` option.
 
 ``time-remaining``
     Remaining length of the file in seconds. Note that the file duration is not
@@ -934,9 +971,10 @@ Property list
     ``time-remaining`` scaled by the current ``speed``.
 
 ``playback-time`` (RW)
-    The playback time, which is the time relative to playback start. (This can
-    be different from the ``time-pos`` property if the file does not start at
-    position ``0``, in which case ``time-pos`` is the source timestamp.)
+    Position in current file in seconds. Unlike ``time-pos``, the time is
+    clamped to the range of the file. (Inaccurate file durations etc. could
+    make it go out of range. Useful on attempts to seek outside of the file,
+    as the seek target time is considered the current position during seeking.)
 
 ``chapter`` (RW)
     Current chapter number. The number of the first chapter is 0.
@@ -1082,13 +1120,13 @@ Property list
 
 ``vf-metadata/<filter-label>``
     Metadata added by video filters. Accessed by the filter label,
-    which if not explicitly specified using the ``@filter-label:`` syntax,
+    which, if not explicitly specified using the ``@filter-label:`` syntax,
     will be ``<filter-name>NN``.
 
     Works similar to ``metadata`` property. It allows the same access
     methods (using sub-properties).
 
-    An example of these kind of metadata are the cropping parameters
+    An example of this kind of metadata are the cropping parameters
     added by ``--vf=lavfi=cropdetect``.
 
 ``af-metadata/<filter-label>``
@@ -1115,7 +1153,7 @@ Property list
 
 ``cache-size`` (RW)
     Network cache size in KB. This is similar to ``--cache``. This allows
-    to set the cache size at runtime. Currently, it's not possible to enable
+    setting the cache size at runtime. Currently, it's not possible to enable
     or disable the cache at runtime using this property, just to resize an
     existing cache.
 
@@ -1171,9 +1209,6 @@ Property list
     is loaded, or when switching ordered chapter segments. This is because
     the same underlying code is used for seeking and resyncing.)
 
-``pts-association-mode`` (RW)
-    See ``--pts-association-mode``.
-
 ``hr-seek`` (RW)
     See ``--hr-seek``.
 
@@ -1260,8 +1295,8 @@ Property list
 ``colormatrix-input-range`` (R)
     See ``colormatrix``.
 
-``colormatrix-output-range`` (R)
-    See ``colormatrix``.
+``video-output-levels`` (RW)
+    See ``--video-output-levels``,
 
 ``colormatrix-primaries`` (R)
     See ``colormatrix``.
@@ -1421,6 +1456,19 @@ Property list
 
     Has the same sub-properties as ``video-params``.
 
+``video-frame-info``
+    Approximate information of the current frame. Note that if any of these
+    are used on OSD, the information might be off by a few frames due to OSD
+    redrawing and frame display being somewhat disconnected, and you might
+    have to pause and force a redraw.
+
+    Sub-properties:
+
+    ``video-frame-info/picture-type``
+    ``video-frame-info/interlaced``
+    ``video-frame-info/tff``
+    ``video-frame-info/repeat``
+
 ``fps``
     Container FPS. This can easily contain bogus values. For videos that use
     modern container formats or video codecs, this will often be incorrect.
@@ -1447,13 +1495,21 @@ Property list
     Names of the displays that the mpv window covers. On X11, these
     are the xrandr names (LVDS1, HDMI1, DP1, VGA1, etc.).
 
-``display-fps``
+``display-fps`` (RW)
     The refresh rate of the current display. Currently, this is the lowest FPS
     of any display covered by the video, as retrieved by the underlying system
     APIs (e.g. xrandr on X11). It is not the measured FPS. It's not necessarily
     available on all platforms. Note that any of the listed facts may change
     any time without a warning.
 
+``estimated-display-fps``
+    Only available if display-sync mode (as selected by ``--video-sync``) is
+    active. Returns the actual rate at which display refreshes seem to occur,
+    measured by system time.
+
+``vsync-jitter``
+    Estimated deviation factor of the vsync duration.
+
 ``video-aspect`` (RW)
     Video aspect, see ``--video-aspect``.
 
@@ -1486,6 +1542,15 @@ Property list
 ``program`` (W)
     Switch TS program (write-only).
 
+``dvb-channel`` (W)
+    Pair of integers: card,channel of current DVB stream.
+    Can be switched to switch to another channel on the same card. 
+
+``dvb-channel-name`` (RW)
+    Name of current DVB program.
+    On write, a channel-switch to the named channel on the same
+    card is performed. Can also be used for channel switching. 
+
 ``sid`` (RW)
     Current subtitle track (similar to ``--sid``).
 

DOCS/man/ipc.rst

@@ -111,7 +111,7 @@ rely on this.
 Commands
 --------
 
-Additionally to  the commands described in `List of Input Commands`_, a few
+In addition to the commands described in `List of Input Commands`_, a few
 extra commands can also be used as part of the protocol:
 
 ``client_name``
@@ -189,7 +189,7 @@ extra commands can also be used as part of the protocol:
 
 ``unobserve_property``
     Undo ``observe_property`` or ``observe_property_string``. This requires the
-    numeric id passed to the observe command as argument.
+    numeric id passed to the observed command as argument.
 
     Example:
 
@@ -228,7 +228,9 @@ extra commands can also be used as part of the protocol:
 
 ``get_version``
     Returns the client API version the C API of the remote mpv instance
-    provides. (Also see ``DOCS/client-api-changes.rst``.)
+    provides.
+
+    See also: ``DOCS/client-api-changes.rst``.
 
 UTF-8
 -----

DOCS/man/lua.rst

@@ -198,7 +198,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
     be called (unless the user remapped the key with another binding).
 
     The ``name`` argument should be a short symbolic string. It allows the user
-    to remap the key binding via input.conf using the ``script_message``
+    to remap the key binding via input.conf using the ``script-message``
     command, and the name of the key binding (see below for
     an example). The name should be unique across other bindings in the same
     script - if not, the previous binding with the same name will be
@@ -220,8 +220,8 @@ The ``mp`` module is preloaded, although it can be loaded manually with
             has an ``is_mouse`` entry, which tells whether the event was caused
             by a mouse button.
 
-    Internally, key bindings are dispatched via the ``script_message_to`` or
-    ``script_binding`` input commands and ``mp.register_script_message``.
+    Internally, key bindings are dispatched via the ``script-message-to`` or
+    ``script-binding`` input commands and ``mp.register_script_message``.
 
     Trying to map multiple commands to a key will essentially prefer a random
     binding, while the other bindings are not called. It is guaranteed that
@@ -244,7 +244,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 
     ::
 
-        y script_binding something
+        y script-binding something
 
 
     This will print the message when the key ``y`` is pressed. (``x`` will
@@ -255,7 +255,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 
     ::
 
-        y script_binding fooscript.something
+        y script-binding fooscript.something
 
 ``mp.add_forced_key_binding(...)``
     This works almost the same as ``mp.add_key_binding``, but registers the
@@ -389,7 +389,7 @@ The ``mp`` module is preloaded, although it can be loaded manually with
 ``mp.get_script_name()``
     Return the name of the current script. The name is usually made of the
     filename of the script, with directory and file extension removed. If
-    there are several script which would have the same name, it's made unique
+    there are several scripts which would have the same name, it's made unique
     by appending a number.
 
     .. admonition:: Example

DOCS/man/mpv.rst

@@ -50,7 +50,7 @@ UP and DOWN
 
 Ctrl+LEFT and Ctrl+RIGHT
     Seek to the previous/next subtitle. Subject to some restrictions and
-    might not work always; see ``sub_seek`` command.
+    might not always work; see ``sub_seek`` command.
 
 [ and ]
     Decrease/increase current playback speed by 10%.
@@ -128,7 +128,7 @@ x and z
     Adjust subtitle delay by +/- 0.1 seconds.
 
 l
-    Set/clear A-B loop points. See ``ab_loop`` command for details.
+    Set/clear A-B loop points. See ``ab-loop`` command for details.
 
 L
     Toggle infinite looping.
@@ -192,23 +192,19 @@ corresponding adjustment, or the software equalizer (``--vf=eq``).)
 7 and 8
     Adjust saturation.
 
-(The following keys are valid only on OS X.)
-
-command + 0
+Alt+0 (and command+0 on OSX)
     Resize video window to half its original size.
-    (On other platforms, you can bind keys to change the ``window-scale``
-    property.)
 
-command + 1
+Alt+1 (and command+1 on OSX)
     Resize video window to its original size.
 
-command + 2
+Alt+2 (and command+2 on OSX)
     Resize video window to double its original size.
 
-command + f
+command + f (OSX only)
     Toggle fullscreen (see also ``--fs``).
 
-command + [ and command + ]
+command + [ and command + ] (OSX only)
     Set video window alpha.
 
 (The following keys are valid if you have a keyboard with multimedia keys.)
@@ -226,7 +222,10 @@ PREVIOUS and NEXT
 support.)
 
 h and k
-    Select previous/next channel.
+    Select previous/next tv-channel.
+
+H and K
+    Select previous/next dvb-channel.
 
 Mouse Control
 -------------
@@ -262,7 +261,7 @@ parser puts several options into a single string, and passes them to a
 component at once, instead of using multiple options on the level of the
 command line.
 
-The suboption parser can quote strings with ``"``, ``'``, and ``[...]``.
+The suboption parser can quote strings with ``"`` and ``[...]``.
 Additionally, there is a special form of quoting with ``%n%`` described below.
 
 For example, the ``opengl`` VO can take multiple options:
@@ -392,10 +391,11 @@ You can put all of the options in configuration files which will be read every
 time mpv is run. The system-wide configuration file 'mpv.conf' is in your
 configuration directory (e.g. ``/etc/mpv`` or ``/usr/local/etc/mpv``), the
 user-specific one is ``~/.config/mpv/mpv.conf``. For details and platform
-specifics see the `FILES`_ section.
+specifics (in particular Windows paths) see the `FILES`_ section.
+
 User-specific options override system-wide options and options given on the
 command line override either. The syntax of the configuration files is
-``option=<value>``; everything after a *#* is considered a comment. Options
+``option=value``. Everything after a *#* is considered a comment. Options
 that work without values can be enabled by setting them to *yes* and disabled by
 setting them to *no*. Even suboptions can be specified in this way.
 
@@ -413,7 +413,7 @@ Escaping spaces and special characters
 
 This is done like with command line options. The shell is not involved here,
 but option values still need to be quoted as a whole if it contains certain
-characters like spaces. A config entry can be quoted with ``"`` and ``'``,
+characters like spaces. A config entry can be quoted with ``"``,
 as well as with the fixed-length syntax (``%n%``) mentioned before. This is like
 passing the exact contents of the quoted string as command line option. C-style
 escapes are currently _not_ interpreted on this level, although some options do
@@ -460,7 +460,38 @@ description (shown by ``--profile=help``) can be defined with the
 ``profile-desc`` option. To end the profile, start another one or use the
 profile name ``default`` to continue with normal options.
 
-.. admonition:: Example mpv profile
+.. admonition:: Example mpv config file with profiles
+
+    ::
+
+        # normal top-level option
+        fullscreen=yes
+
+        # a profile that can be enabled with --profile=big-cache
+        [big-cache]
+        cache=123400
+        demuxer-readahead-secs=20
+
+        [slow]
+        profile-desc="some profile name"
+        vo=opengl:scale=ewa_lanczos:scale-radius=16
+
+        [fast]
+        vo=vdpau
+
+        # using a profile again extends it
+        [slow]
+        framedrop=no
+        # you can also include other profiles
+        profile=big-cache
+
+
+Auto profiles
+-------------
+
+Some profiles are loaded automatically. The following example demonstrates this:
+
+.. admonition:: Auto profile loading
 
     ::
 
@@ -479,14 +510,22 @@ profile name ``default`` to continue with normal options.
         [ao.alsa]
         device=spdif
 
+The profile name follows the schema ``type.name``, where type can be ``vo``
+to match the value the ``--vo`` option is set to, ``ao`` for ``--ao``,
+``protocol`` for the input/output protocol in use (see ``--list-protocols``),
+and ``extension`` for the extension of the path of the currently played file
+(*not* the file format).
+
+This feature is very limited, and there are no other auto profiles.
 
 TAKING SCREENSHOTS
 ==================
 
 Screenshots of the currently played file can be taken using the 'screenshot'
 input mode command, which is by default bound to the ``s`` key. Files named
-``shotNNNN.jpg`` will be saved in the working directory, using the first
-available number - no files will be overwritten.
+``mpv-shotNNNN.jpg`` will be saved in the working directory, using the first
+available number - no files will be overwritten. In pseudo-GUI mode, the
+screenshot will be saved somewhere else. See `PSEUDO GUI MODE`_.
 
 A screenshot will usually contain the unscaled video contents at the end of the
 video filter chain and subtitles. By default, ``S`` takes screenshots without
@@ -527,16 +566,16 @@ listed.
   this will indicate a problem. (``total-avsync-change`` property.)
 - Encoding state in ``{...}``, only shown in encoding mode.
 - Display sync state. If display sync is active (``display-sync-active``
-  property), this shows ``DS: +0.02598%``, where the number is the speed change
-  factor applied to audio to achieve sync to display, expressed in percent
-  deviation from 1.0 (``audio-speed-correction`` property). In sync modes which
-  don't resample, this will always be ``+0.00000%``.
-- Missed frames, e.g. ``Missed: 4``. (``vo-missed-frame-count`` property.) Shows
-  up in display sync mode only. This is incremented each time a frame took
-  longer to display than intended.
+  property), this shows ``DS: 2.500/13``, where the first number is average
+  number of vsyncs per video frame (e.g. 2.5 when playing 24Hz videos on 60Hz
+  screens), which might jitter if the ratio doesn't round off, or there are
+  mistimed frames (``vsync-ratio``), and the second number of estimated number
+  of vsyncs which took too long (``vo-delayed-frame-count`` property). The
+  latter is a heuristic, as it's generally not possible to determine this with
+  certainty.
 - Dropped frames, e.g. ``Dropped: 4``. Shows up only if the count is not 0. Can
   grow if the video framerate is higher than that of the display, or if video
-  rendering is too slow. Also can be incremented on "hiccups" and when the video
+  rendering is too slow. May also be incremented on "hiccups" and when the video
   frame couldn't be displayed on time. (``vo-drop-frame-count`` property.)
   If the decoder drops frames, the number of decoder-dropped frames is appended
   to the display as well, e.g.: ``Dropped: 4/34``. This happens only if
@@ -567,7 +606,7 @@ PROTOCOLS
     Play a path from  Samba share.
 
 ``bd://[title][/device]`` ``--bluray-device=PATH``
-    Play a Blu-Ray disc. Currently, this does not accept ISO files. Instead,
+    Play a Blu-ray disc. Currently, this does not accept ISO files. Instead,
     you must mount the ISO file as filesystem, and point ``--bluray-device``
     to the mounted directory directly.
 
@@ -619,9 +658,8 @@ PROTOCOLS
     absolute path.
 
 ``fd://123``
-    Read data from the given UNIX FD (for example 123). This is similar to
-    piping data to stdin via ``-``, but can use an arbitrary file descriptor.
-    Will not work correctly on MS Windows.
+    Read data from the given file descriptor (for example 123). This is similar
+    to piping data to stdin via ``-``, but can use an arbitrary file descriptor.
 
 ``edl://[edl specification as in edl-mpv.rst]``
     Stitch together parts of multiple files and play them.
@@ -688,6 +726,15 @@ The profile always overrides other settings in ``mpv.conf``.
 
 .. include:: changes.rst
 
+
+EMBEDDING INTO OTHER PROGRAMS (LIBMPV)
+======================================
+
+mpv can be embedded into other programs as video/audio playback backend. The
+recommended way to do so is using libmpv. See ``libmpv/client.h`` in the mpv
+source code repository. This provides a C API. Bindings for other languages
+might be available (see wiki).
+
 ENVIRONMENT VARIABLES
 =====================
 
@@ -759,7 +806,7 @@ libdvdcss:
 
         key
            is the default method. libdvdcss will use a set of calculated
-           player keys to try and get the disc key. This can fail if the drive
+           player keys to try to get the disc key. This can fail if the drive
            does not recognize any of the player keys.
 
         disc
@@ -875,7 +922,7 @@ locations are different. They are generally located under ``%APPDATA%/mpv/``.
 For example, the path to mpv.conf is ``%APPDATA%/mpv/mpv.conf``, which maps to
 a system and user-specific path, for example
 
-    ``C:\users\USERNAME\Application Data\mpv\mpv.conf``
+    ``C:\users\USERNAME\AppData\Roaming\mpv\mpv.conf``
 
 You can find the exact path by running ``echo %APPDATA%\mpv\mpv.conf`` in cmd.exe.
 
@@ -898,56 +945,10 @@ 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
+A third config directory with the lowest priority is the directory named ``mpv``
+in the same directory as ``mpv.exe``. This used to be the directory with the
 highest priority, but is now discouraged to use and might be removed in the
 future.
 
 Note that mpv likes to mix ``/`` and ``\`` path separators for simplicity.
 kernel32.dll accepts this, but cmd.exe does not.
-
-EXAMPLES OF MPV USAGE
-=====================
-
-Blu-ray playback:
-    - ``mpv bd:////path/to/disc``
-    - ``mpv bd:// --bluray-device=/path/to/disc``
-
-Play in Japanese with English subtitles:
-    ``mpv dvd://1 --alang=ja --slang=en``
-
-Play only chapters 5, 6, 7:
-    ``mpv dvd://1 --chapter=5-7``
-
-Play only titles 5, 6, 7:
-    ``mpv dvd://5-7``
-
-Play a multi-angle DVD:
-    ``mpv dvd://1 --dvd-angle=2``
-
-Play from a different DVD device:
-    ``mpv dvd://1 --dvd-device=/dev/dvd2``
-
-Play DVD video from a directory with VOB files:
-    ``mpv dvd://1 --dvd-device=/path/to/directory/``
-
-Stream from HTTP:
-    ``mpv http://example.com/example.avi``
-
-Stream using RTSP:
-    ``mpv rtsp://server.example.com/streamName``
-
-Play a libavfilter graph:
-    ``mpv avdevice://lavfi:mandelbrot``
-
-AUTHORS
-=======
-
-mpv is a MPlayer fork based on mplayer2, which in turn is a fork of MPlayer.
-
-MPlayer was initially written by Arpad Gereoffy. See the ``AUTHORS`` file for
-a list of some of the many other contributors.
-
-MPlayer is (C) 2000-2013 The MPlayer Team
-
-This man page was written mainly by Gabucino, Jonas Jermann and Diego Biurrun.

DOCS/man/options.rst

@@ -69,7 +69,8 @@ Playback Control
 
     The general format for absolute times is ``[[hh:]mm:]ss[.ms]``. If the time
     is given with a prefix of ``+`` or ``-``, the seek is relative from the start
-    or end of the file.
+    or end of the file. (Since mpv 0.14, the start of the file is always
+    considered 0.)
 
     ``pp%`` seeks to percent position pp (0-100).
 
@@ -101,6 +102,14 @@ Playback Control
     Stop after a given time relative to the start time.
     See ``--start`` for valid option values and examples.
 
+``--rebase-start-time=<yes|no>``
+    Whether to move the file start time to ``00:00:00`` (default: yes). This
+    is less awkward for files which start at a random timestamp, such as
+    transport streams. On the other hand, if there are timestamp resets, the
+    resulting behavior can be rather weird. For this reason, and in case you
+    are actually interested in the real timestamps, this behavior can be
+    disabled with ``no``.
+
 ``--speed=<0.01-100>``
     Slow down or speed up playback by the factor given as parameter.
 
@@ -127,7 +136,9 @@ Playback Control
 
 ``--chapter=<start[-end]>``
     Specify which chapter to start playing at. Optionally specify which
-    chapter to end playing at. Also see ``--start``.
+    chapter to end playing at.
+
+    See also: ``--start``.
 
 ``--playlist-pos=<no|index>``
     Set which file on the internal playlist to start playback with. The index
@@ -312,7 +323,7 @@ Program Behavior
         Files explicitly requested by command line options, like
         ``--include`` or ``--use-filedir-conf``, will still be loaded.
 
-    Also see ``--config-dir``.
+    See also: ``--config-dir``.
 
 ``--list-options``
     Prints all available options.
@@ -456,16 +467,13 @@ Program Behavior
 
     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``)
+    available aliases.
+    (Default: youtube-dl's default, currently ``bestvideo+bestaudio/best``)
 
 ``--ytdl-raw-options=<key>=<value>[,<key>=<value>[,...]]``
     Pass arbitrary options to youtube-dl. Parameter and argument should be
@@ -551,11 +559,10 @@ Video
         differences to other VOs are possible.
 
 ``--display-fps=<fps>``
-    Set the maximum assumed display FPS used with ``--framedrop``. By default
-    a detected value is used (X11 only, not correct on multi-monitor systems),
-    or infinite display FPS if that fails. Infinite FPS means only frames too
-    late are dropped. If a correct FPS is provided, frames that are predicted
-    to be too late are dropped too.
+    Set the display FPS used with the ``--video-sync=display-*`` modes. By
+    default, a detected value is used (X11 only, not correct on multi-monitor
+    systems). Keep in mind that setting an incorrect value (even if slightly
+    incorrect) can ruin video playback.
 
 ``--hwdec=<api>``
     Specify the hardware video decoding API that should be used if possible.
@@ -567,12 +574,11 @@ Video
     :no:        always use software decoding (default)
     :auto:      see below
     :vdpau:     requires ``--vo=vdpau`` or ``--vo=opengl`` (Linux only)
-    :vaapi:     requires ``--vo=opengl`` or ``--vo=vaapi`` (Linux with Intel GPUs only)
+    :vaapi:     requires ``--vo=opengl`` or ``--vo=vaapi`` (Linux only)
     :vaapi-copy: copies video back into system RAM (Linux with Intel GPUs only)
-    :vda:       requires ``--vo=opengl`` (OS X only)
-    :videotoolbox: requires ``--vo=opengl`` (newer OS X only)
+    :videotoolbox: requires ``--vo=opengl`` (OS X 10.8 and up only)
     :dxva2-copy: copies video back to system RAM (Windows only)
-    :rpi:      requires ``--vo=rpi`` (Raspberry Pi only - default if available)
+    :rpi:       requires ``--vo=rpi`` (Raspberry Pi only - default if available)
 
     ``auto`` tries to automatically enable hardware decoding using the first
     available method. This still depends what VO you are using. For example,
@@ -581,6 +587,12 @@ Video
     work, it will always fall back to software decoding, instead of trying the
     next method (might matter on some Linux systems).
 
+    The ``vaapi`` mode, if used with ``--vo=opengl``, requires Mesa 11 and most
+    likely works with Intel GPUs only. It also requires the opengl EGL backend
+    (automatically used if available). You can also try the old GLX backend by
+    forcing it with ``--vo=opengl:backend=x11``, but the vaapi/GLX interop is
+    said to be slower than ``vaapi-copy``.
+
     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.
@@ -603,10 +615,18 @@ Video
 
     For ``opengl-cb``, if set, load the interop context as soon as the OpenGL
     context is created. Since ``opengl-cb`` has no on-demand loading, this
-    allows enabling hardware decoding at runtime at all, without having to
+    allows enabling hardware decoding at runtime at all, without having
     to temporarily set the ``hwdec`` option just during OpenGL context
     initialization with ``mpv_opengl_cb_init_gl()``.
 
+``--videotoolbox-format=<name>``
+    Set the internal pixel format used by ``--hwdec=videotoolbox`` on OSX. The
+    choice of the format can influence performance considerably. On the other
+    hand, there doesn't appear to be a good way to detect the best format for
+    the given hardware. ``nv12``, the default, works better on modern hardware,
+    while ``uyvy422`` appears to be better for old hardware. ``rgb0`` also
+    works.
+
 ``--panscan=<0.0-1.0>``
     Enables pan-and-scan functionality (cropping the sides of e.g. a 16:9
     video to make it fit a 4:3 display without black bands). The range
@@ -657,7 +677,7 @@ Video
     example ``--video-zoom`` does nothing if this option is enabled.)
 
     The video and monitor aspects aspect will be ignored. Aspect correction
-    would require to scale the video in the X or Y direction, but this option
+    would require scaling the video in the X or Y direction, but this option
     disables scaling, disabling all aspect correction.
 
     Note that the scaler algorithm may still be used, even if the video isn't
@@ -677,7 +697,7 @@ Video
 
     This option is disabled if the ``--no-keepaspect`` option is used.
 
-``--video-rotate=<0-360|no>``
+``--video-rotate=<0-359|no>``
     Rotate the video clockwise, in degrees. Currently supports 90° steps only.
     If ``no`` is given, the video is never rotated, even if the file has
     rotation metadata. (The rotation value is added to the rotation metadata,
@@ -745,7 +765,7 @@ Video
     if supported.
 
     This behaves exactly like the ``deinterlace`` input property (usually
-    mapped to ``Shift+D``).
+    mapped to ``d``).
 
     ``auto`` is a technicality. Strictly speaking, the default for this option
     is deinterlacing disabled, but the ``auto`` case is needed if ``yadif`` was
@@ -775,6 +795,26 @@ Video
     For audio-only playback, any value greater than 0 will quit playback
     immediately after initialization. The value 0 works as with video.
 
+``--video-output-levels=<outputlevels>``
+    RGB color levels used with YUV to RGB conversion. Normally, output devices
+    such as PC monitors use full range color levels. However, some TVs and
+    video monitors expect studio RGB levels. Providing full range output to a
+    device expecting studio level input results in crushed blacks and whites,
+    the reverse in dim gray blacks and dim whites.
+
+    Not all VOs support this option. Some will silently ignore it.
+
+    Available color ranges are:
+
+    :auto:      automatic selection (equals to full range) (default)
+    :limited:   limited range (16-235 per component), studio levels
+    :full:      full range (0-255 per component), PC levels
+
+    .. note::
+
+        It is advisable to use your graphics driver's color range option
+        instead, if available.
+
 ``--hwdec-codecs=<codec1,codec2,...|all>``
     Allow hardware decoding for a given list of codecs only. The special value
     ``all`` always allows all codecs.
@@ -782,9 +822,9 @@ Video
     You can get the list of allowed codecs with ``mpv --vd=help``. Remove the
     prefix, e.g. instead of ``lavc:h264`` use ``h264``.
 
-    By default this is set to ``h264,vc1,wmv3,hevc``. Note that the hardware
-    acceleration special codecs like ``h264_vdpau`` are not relevant anymore,
-    and in fact have been removed from Libav in this form.
+    By default, this is set to ``h264,vc1,wmv3,hevc,mpeg2video``. Note that the
+    hardware acceleration special codecs like ``h264_vdpau`` are not relevant
+    anymore, and in fact have been removed from Libav in this form.
 
     This is usually only needed with broken GPUs, where a codec is reported
     as supported, but decoding causes more problems than it solves.
@@ -801,6 +841,11 @@ Video
     The result is most likely broken decoding, but may also help if the
     detected or reported profiles are somehow incorrect.
 
+``--vd-lavc-software-fallback=<yes|no|N>``
+    Fallback to software decoding if the hardware-accelerated decoder fails
+    (default: 3). If this is a number, then fallback will be triggered if
+    N frames fail to decode in a row. 1 is equivalent to ``yes``.
+
 ``--vd-lavc-bitexact``
     Only use bit-exact algorithms in all decoding steps (for codec testing).
 
@@ -821,7 +866,7 @@ Video
 
     .. admonition:: Example
 
-        ``--vd--lavc-o=debug=pict``
+        ``--vd-lavc-o=debug=pict``
 
 ``--vd-lavc-show-all=<yes|no>``
     Show even broken/corrupt frames (default: no). If this option is set to
@@ -892,6 +937,14 @@ Audio
 
     Currently not implemented for most AOs.
 
+``--audio-fallback-to-null=<yes|no>``
+    If no audio device can be opened, behave as if ``--ao=null`` was given. This
+    is useful in combination with ``--audio-device``: instead of causing an
+    error if the selected device does not exist, the client API user (or a
+    Lua script) could let playback continue normally, and check the
+    ``current-ao`` and ``audio-device-list`` properties to make high-level
+    decisions about how to continue.
+
 ``--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
@@ -976,7 +1029,9 @@ Audio
 
 ``--mute=<auto|yes|no>``
     Set startup audio mute status. ``auto`` (default) will not change the mute
-    status. Also see ``--volume``.
+    status.
+
+    See also: ``--volume``.
 
 ``--softvol=<mode>``
     Control whether to use the volume controls of the audio output driver or
@@ -1101,8 +1156,8 @@ Audio
     point of file change. Default: ``weak``.
 
     :no:    Disable gapless audio.
-    :yes:   The audio device is opened using parameters chosen according to the
-            first file played and is then kept open for gapless playback. This
+    :yes:   The audio device is opened using parameters chosen for the first
+            file played and is then kept open for gapless playback. This
             means that if the first file for example has a low sample rate, then
             the following files may get resampled to the same low sample rate,
             resulting in reduced sound quality. If you play files with different
@@ -1149,7 +1204,11 @@ Audio
     :no:    Don't automatically load external audio files.
     :exact: Load the media filename with audio file extension (default).
     :fuzzy: Load all audio files containing media filename.
-    :all:   Load all audio files in the current directory.
+    :all:   Load all aufio files in the current and ``--audio-file-paths``
+            directories.
+
+``--audio-file-paths=<path1:path2:...>``
+    Equivalent to ``--sub-paths`` option, but for auto-loaded audio files.
 
 ``--audio-client-name=<name>``
     The application name the player reports to the audio API. Can be useful
@@ -1186,6 +1245,14 @@ Audio
 Subtitles
 ---------
 
+.. note::
+
+    Changing styling and position does not work with all subtitles. Image-based
+    subtitles (DVD, Bluray/PGS, DVB) can not changed for fundamental reasons.
+    Subtitles in ASS format are normally not changed intentionally, but
+    overriding them can be controlled with ``--ass-style-override``.
+
+
 ``--no-sub``
     Do not select any subtitle when the file is loaded.
 
@@ -1283,8 +1350,8 @@ Subtitles
 
 ``--sub-speed=<0.1-10.0>``
     Multiply the subtitle event timestamps with the given value. Can be used
-    to fix the playback speed for frame-based subtitle formats. Works for
-    external text subtitles only.
+    to fix the playback speed for frame-based subtitle formats. Affects text
+    subtitles only.
 
     .. admonition:: Example
 
@@ -1532,27 +1599,27 @@ Subtitles
     This mode doesn't take language or fallback codepage.
 
 ``--sub-fix-timing``, ``--no-sub-fix-timing``
-    By default, external text subtitles are preprocessed to remove minor gaps
-    or overlaps between subtitles (if the difference is smaller than 200 ms,
-    the gap or overlap is removed). This does not affect image subtitles,
-    subtitles muxed with audio/video, or subtitles in the ASS format.
+    By default, subtitle timing is adjusted to remove minor gaps or overlaps
+    between subtitles (if the difference is smaller than 210 ms, the gap or
+    overlap is removed).
 
 ``--sub-forced-only``
     Display only forced subtitles for the DVD subtitle stream selected by e.g.
     ``--slang``.
 
 ``--sub-fps=<rate>``
-    Specify the framerate of the subtitle file (default: video fps).
+    Specify the framerate of the subtitle file (default: video fps). Affects
+    text subtitles only.
 
     .. note::
 
         ``<rate>`` > video fps speeds the subtitles up for frame-based
         subtitle files and slows them down for time-based ones.
 
-    Also see ``--sub-speed`` option.
+    See also: ``--sub-speed``.
 
 ``--sub-gauss=<0.0-3.0>``
-    Apply Gaussian blur to image subtitles (default: 0). This can help making
+    Apply Gaussian blur to image subtitles (default: 0). This can help to make
     pixelated DVD/Vobsubs look nicer. A value other than 0 also switches to
     software subtitle scaling. Might be slow.
 
@@ -1561,7 +1628,7 @@ Subtitles
         Never applied to text subtitles.
 
 ``--sub-gray``
-    Convert image subtitles to grayscale. Can help making yellow DVD/Vobsubs
+    Convert image subtitles to grayscale. Can help to make yellow DVD/Vobsubs
     look nicer.
 
     .. note::
@@ -1702,6 +1769,10 @@ Window
 ``--ontop``
     Makes the player window stay on top of other windows.
 
+    On Windows, if combined with fullscreen mode, this causes mpv to be
+    treated as exclusive fullscreen window that bypasses the Desktop Window
+    Manager.
+
 ``--border``, ``--no-border``
     Play video with window border and decorations. Since this is on by
     default, use ``--no-border`` to disable the standard window decorations.
@@ -1929,8 +2000,8 @@ Window
     always re-enabled when the player is paused.
 
     This is not supported on all video outputs or platforms. Sometimes it is
-    implemented, but does not work (happens often on GNOME). You might be able
-    to to work this around using ``--heartbeat-cmd`` instead.
+    implemented, but does not work (known to happen with GNOME). You might be
+    able to work around this using ``--heartbeat-cmd`` instead.
 
 ``--wid=<ID>``
     This tells mpv to attach to an existing window. If a VO is selected that
@@ -1949,7 +2020,7 @@ Window
     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
+    to ``intptr_t``. mpv will create 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.
 
@@ -1980,6 +2051,10 @@ Window
 
     This option might be removed in the future.
 
+``--x11-bypass-compositor=<yes|no>``
+    If set to ``yes`` (default), then ask the compositor to unredirect the
+    mpv window. This uses the ``_NET_WM_BYPASS_COMPOSITOR`` hint.
+
 
 Disc Devices
 ------------
@@ -2154,7 +2229,7 @@ Demuxer
     Encryption key the demuxer should use. This is the raw binary data of
     the key converted to a hexadecimal string.
 
-``--demuxer-mkv-subtitle-preroll``, ``--mkv-subtitle-preroll``
+``--demuxer-mkv-subtitle-preroll=<yes|index|no>``, ``--mkv-subtitle-preroll``
     Try harder to show embedded soft subtitles when seeking somewhere. Normally,
     it can happen that the subtitle at the seek target is not shown due to how
     some container file formats are designed. The subtitles appear only if
@@ -2186,7 +2261,11 @@ Demuxer
     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.)
+    target.) If the ``index`` choice (which is the default) is specified, then
+    prerolling will be done only if this information is actually available. If
+    this method is used, the maximum amount of data to skip can be additionally
+    controlled by ``--demuxer-mkv-subtitle-preroll-secs-index`` (it still uses
+    the value of the option without ``-index`` if that is higher).
 
     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,
@@ -2198,6 +2277,9 @@ Demuxer
 ``--demuxer-mkv-subtitle-preroll-secs=<value>``
     See ``--demuxer-mkv-subtitle-preroll``.
 
+``--demuxer-mkv-subtitle-preroll-secs-index=<value>``
+    See ``--demuxer-mkv-subtitle-preroll``.
+
 ``--demuxer-mkv-probe-video-duration=<yes|no|full>``
     When opening the file, seek to the end of it, and check what timestamp the
     last video packet has, and report that as file duration. This is strictly
@@ -2210,18 +2292,6 @@ Demuxer
     file and can make a reliable estimate even without an index present (such
     as partial files).
 
-``--demuxer-mkv-fix-timestamps=<yes|no>``
-    Fix rounded Matroska timestamps (disabled by default). Matroska usually
-    stores timestamps rounded to milliseconds. This means timestamps jitter
-    by some amount around the intended timestamp. mpv can correct the timestamps
-    based on the framerate value stored in the file: the timestamp is rounded
-    to the next frame (according to the framerate), unless the new timestamp
-    would deviate more than 1ms from the old one. This should undo the rounding
-    done by the muxer.
-
-    (The allowed deviation can be less than 1ms if the file uses a non-standard
-    timecode scale.)
-
 ``--demuxer-rawaudio-channels=<value>``
     Number of channels (or channel layout) if ``--demuxer=rawaudio`` is used
     (default: stereo).
@@ -2334,7 +2404,7 @@ Input
 
 ``--input-key-fifo-size=<2-65000>``
     Specify the size of the FIFO that buffers key events (default: 7). If it
-    is too small some events may be lost. The main disadvantage of setting it
+    is too small, some events may be lost. The main disadvantage of setting it
     to a very large value is that if you hold down a key triggering some
     particularly slow command then the player may be unresponsive while it
     processes all the queued commands.
@@ -2401,7 +2471,7 @@ Input
 
     On X11, a sub-window with input enabled grabs all keyboard input as long
     as it is 1. a child of a focused window, and 2. the mouse is inside of
-    the sub-window. The can steal away all keyboard input from the
+    the sub-window. It can steal away all keyboard input from the
     application embedding the mpv window, and on the other hand, the mpv
     window will receive no input if the mouse is outside of the mpv window,
     even though mpv has focus. Modern toolkits work around this weird X11
@@ -2441,14 +2511,14 @@ OSD
 ``--osd-duration=<time>``
     Set the duration of the OSD messages in ms (default: 1000).
 
-``--osd-font=<pattern>``, ``--sub-text-font=<pattern>``
+``--osd-font=<name>``, ``--sub-text-font=<name>``
     Specify font to use for OSD and for subtitles that do not themselves
     specify a particular font. The default is ``sans-serif``.
 
     .. admonition:: Examples
 
         - ``--osd-font='Bitstream Vera Sans'``
-        - ``--osd-font='Bitstream Vera Sans:style=Bold'`` (fontconfig pattern)
+        - ``--osd-font='MS Comic Sans'``
 
     .. note::
 
@@ -2456,6 +2526,9 @@ OSD
         options) are ignored when ASS-subtitles are rendered, unless the
         ``--no-sub-ass`` option is specified.
 
+        This used to support fontconfig patterns. Starting with libass 0.13.0,
+        this stopped working.
+
 ``--osd-font-size=<size>``, ``--sub-text-font-size=<size>``
     Specify the OSD/sub font size. The unit is the size in scaled pixels at a
     window height of 720. The actual pixel size is scaled with the window
@@ -2636,17 +2709,6 @@ 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
 ----------
 
@@ -2679,7 +2741,7 @@ Screenshot
     Specify the filename template used to save screenshots. The template
     specifies the filename without file extension, and can contain format
     specifiers, which will be substituted when taking a screenshot.
-    By default the template is ``mpv-shot%n``, which results in filenames like
+    By default, the template is ``mpv-shot%n``, which results in filenames like
     ``mpv-shot0012.png`` for example.
 
     The template can start with a relative or absolute path, in order to
@@ -2835,7 +2897,7 @@ Terminal
     Particularly useful on slow terminals or broken ones which do not properly
     handle carriage return (i.e. ``\r``).
 
-    Also see ``--really-quiet`` and ``--msg-level``.
+    See also: ``--really-quiet`` and ``--msg-level``.
 
 ``--really-quiet``
     Display even less output and status messages than with ``--quiet``.
@@ -2968,8 +3030,9 @@ TV
     maximum size of the capture buffer in megabytes (default: dynamical)
 
 ``--tv-norm=<value>``
-    See the console output for a list of all available norms, also see the
-    ``normid`` option below.
+    See the console output for a list of all available norms.
+
+    See also: ``--tv-normid``.
 
 ``--tv-normid=<value> (v4l2 only)``
     Sets the TV norm to the given numeric ID. The TV norm depends on the
@@ -3179,7 +3242,7 @@ Cache
        multiple cache streams, and using the same file for them obviously
        clashes.
 
-    Also see ``--cache-file-size``.
+    See also: ``--cache-file-size``.
 
 ``--cache-file-size=<kBytes>``
     Maximum size of the file created with ``--cache-file``. For read accesses
@@ -3245,6 +3308,13 @@ Network
     Verify peer certificates when using TLS (e.g. with ``https://...``).
     (Silently fails with older FFmpeg or Libav versions.)
 
+``--tls-cert-file``
+    A file containing a certificate to use in the handshake with the
+    peer.
+
+``--tls-key-file``
+    A file containing the private key for the certificate.
+
 ``--referrer=<string>``
     Specify a referrer path or URL for HTTP requests.
 
@@ -3300,72 +3370,18 @@ DVB
 
 ``--dvbin-full-transponder=<yes|no>``
     Apply no filters on program PIDs, only tune to frequency and pass full
-    transponder to demuxer. This is useful to record multiple programs
-    on a single transponder, or to work around issues in the ``channels.conf``.
+    transponder to demuxer.
+    The player frontend selects the streams from the full TS in this case,
+    so the program which is shown initially may not match the chosen channel.
+    Switching between the programs is possible by cycling the ``program``
+    property.
+    This is useful to record multiple programs on a single transponder,
+    or to work around issues in the ``channels.conf``.
     It is also recommended to use this for channels which switch PIDs
     on-the-fly, e.g. for regional news.
 
     Default: ``no``
 
-PVR
----
-
-``--pvr-...``
-    These options tune various encoding properties of the PVR capture module.
-    It has to be used with any hardware MPEG encoder based card supported by
-    the V4L2 driver. The Hauppauge WinTV PVR-150/250/350/500 and all IVTV
-    based cards are known as PVR capture cards. Be aware that only Linux
-    2.6.18 kernel and above is able to handle MPEG stream through V4L2 layer.
-    For hardware capture of an MPEG stream and watching it with mpv, use
-    ``pvr://`` as media URL.
-
-
-``--pvr-aspect=<0-3>``
-    Specify input aspect ratio:
-
-    :0: 1:1
-    :1: 4:3 (default)
-    :2: 16:9
-    :3: 2.21:1
-
-``--pvr-arate=<32000-48000>``
-    Specify encoding audio rate (default: 48000 Hz, available: 32000,
-    44100 and 48000 Hz).
-
-``--pvr-alayer=<1-3>``
-    Specify MPEG audio layer encoding (default: 2).
-
-``--pvr-abitrate=<32-448>``
-    Specify audio encoding bitrate in kbps (default: 384).
-
-``--pvr-amode=<value>``
-    Specify audio encoding mode. Available preset values are 'stereo',
-    'joint_stereo', 'dual' and 'mono' (default: stereo).
-
-``--pvr-vbitrate=<value>``
-    Specify average video bitrate encoding in Mbps (default: 6).
-
-``--pvr-vmode=<value>``
-    Specify video encoding mode:
-
-    :vbr: Variable Bit Rate (default)
-    :cbr: Constant Bit Rate
-
-``--pvr-vpeak=<value>``
-    Specify peak video bitrate encoding in Mbps (only useful for VBR
-    encoding, default: 9.6).
-
-``--pvr-fmt=<value>``
-    Choose an MPEG format for encoding:
-
-    :ps:    MPEG-2 Program Stream (default)
-    :ts:    MPEG-2 Transport Stream
-    :mpeg1: MPEG-1 System Stream
-    :vcd:   Video CD compatible stream
-    :svcd:  Super Video CD compatible stream
-    :dvd:   DVD compatible stream
-
-
 Miscellaneous
 -------------
 
@@ -3393,7 +3409,7 @@ Miscellaneous
     implement a perfect audio delay measurement. With this value, if large A/V
     sync offsets occur, they will only take about 1 or 2 seconds to settle
     out. This delay in reaction time to sudden A/V offsets should be the only
-    side-effect of turning this option on, for all sound drivers.
+    side effect of turning this option on, for all sound drivers.
 
 ``--video-sync=<audio|...>``
     How the player synchronizes audio and video.
@@ -3430,6 +3446,11 @@ Miscellaneous
     :display-vdrop:     Drop or repeat video frames to compensate desyncing
                         video. (Although it should have the same effects as
                         ``audio``, the implementation is very different.)
+    :display-adrop:     Drop or repeat audio data to compensate desyncing
+                        video. See ``--video-sync-adrop-size``. This mode will
+                        cause severe audio artifacts if the real monitor
+                        refresh rate is too different from the reported or
+                        forced rate.
     :display-desync:    Sync video to display, and let audio play on its own.
     :desync:            Sync video according to system clock, and let audio play
                         on its own.
@@ -3461,6 +3482,13 @@ Miscellaneous
     frame dropping due to the audio "overshooting" and skipping multiple video
     frames before the sync logic can react.
 
+``--video-sync-adrop-size=<value``
+    For the ``--video-sync=display-adrop`` mode. This mode duplicates/drops
+    audio data to keep audio in sync with video. To avoid audio artifacts on
+    jitter (which would add/remove samples all the time), this is done in
+    relatively large, fixed units, controlled by this option. The unit is
+    seconds.
+
 ``--mf-fps=<value>``
     Framerate used when decoding from multiple PNG or JPEG files with ``mf://``
     (default: 1).
@@ -3492,6 +3520,10 @@ Miscellaneous
     other options such as e.g. user agent are not available with all protocols,
     and printing errors for unknown options would end up being too noisy.)
 
+``--vo-mmcss-profile=<name>``
+    (Windows only.)
+    Set the MMCSS profile for the video renderer thread (default: ``Playback``).
+
 ``--priority=<prio>``
     (Windows only.)
     Set process priority for mpv according to the predefined priorities
@@ -3502,20 +3534,6 @@ Miscellaneous
 
     .. warning:: Using realtime priority can cause system lockup.
 
-``--pts-association-mode=<decode|sort|auto>``
-    Select the method used to determine which container packet timestamp
-    corresponds to a particular output frame from the video decoder. Normally
-    you should not need to change this option.
-
-    :decoder: Use decoder reordering functionality. Unlike in classic MPlayer
-              and mplayer2, this includes a DTS fallback. (Default.)
-    :sort:    Maintain a buffer of unused pts values and use the lowest value
-              for the frame.
-    :auto:    Try to pick a working mode from the ones above automatically.
-
-    You can also try to use ``--no-correct-pts`` for files with completely
-    broken timestamps.
-
 ``--force-media-title=<string>``
     Force the contents of the ``media-title`` property to this value. Useful
     for scripts which want to set a title, without overriding the user's

DOCS/man/vf.rst

@@ -151,15 +151,15 @@ Available filters are:
         :yes: Enable accurate rounding.
 
 ``dsize[=w:h:aspect-method:r:aspect]``
-    Changes the intended display size/aspect at an arbitrary point in the
+    Changes the intended display aspect at an arbitrary point in the
     filter chain. Aspect can be given as a fraction (4/3) or floating point
-    number (1.33). Alternatively, you may specify the exact display width and
-    height desired. Note that this filter does *not* do any scaling itself; it
+    number (1.33). Note that this filter does *not* do any scaling itself; it
     just affects what later scalers (software or hardware) will do when
     auto-scaling to the correct aspect.
 
     ``<w>,<h>``
-        New display width and height.
+        New aspect ratio given by a display width and height. Unlike older mpv
+        versions or MPlayer, this does not set the display size.
 
         Can also be these special values:
 
@@ -259,26 +259,6 @@ Available filters are:
         :limited:   limited range (16-235 for luma, 16-240 for chroma)
         :full:      full range (0-255 for both luma and chroma)
 
-    ``<outputlevels>``
-        RGB color levels used with YUV to RGB conversion. Normally, output devices
-        such as PC monitors use full range color levels. However, some TVs and
-        video monitors expect studio RGB levels. Providing full range output to a
-        device expecting studio level input results in crushed blacks and whites,
-        the reverse in dim gray blacks and dim whites.
-
-        The same limitations as with ``<colormatrix>`` apply.
-
-        Available color ranges are:
-
-        :auto:      automatic selection (equals to full range) (default)
-        :limited:   limited range (16-235 per component), studio levels
-        :full:      full range (0-255 per component), PC levels
-
-        .. note::
-
-            It is advisable to use your graphics driver's color range option
-            instead, if available.
-
     ``<primaries>``
         RGB primaries the source file was encoded with. Normally this should be set
         in the file header, but when playing broken or mistagged files this can be
@@ -458,9 +438,8 @@ Available filters are:
         generating an occasional mismatched frame, but it may also cause an
         excessive number of frames to be dropped during high motion sequences.
         Conversely, setting it to -1 will make ``pullup`` match fields more
-        easily. This may help processing of video where there is slight
-        blurring between the fields, but may also cause there to be interlaced
-        frames in the output.
+        easily. This may help process video with slight blurring between the
+        fields, but may also cause interlaced frames in the output.
 
     ``mp`` (metric plane)
         This option may be set to ``u`` or ``v`` to use a chroma plane instead of the
@@ -475,25 +454,24 @@ Available filters are:
 
     ``<mode>``
         :frame: Output 1 frame for each frame.
-        :field: Output 1 frame for each field.
+        :field: Output 1 frame for each field (default).
         :frame-nospatial: Like ``frame`` but skips spatial interlacing check.
         :field-nospatial: Like ``field`` but skips spatial interlacing check.
 
     ``<interlaced-only>``
-        :no:  Deinterlace all frames (default).
-        :yes: Only deinterlace frames marked as interlaced (default if this
-              filter is inserted via ``deinterlace`` property).
+        :no:  Deinterlace all frames.
+        :yes: Only deinterlace frames marked as interlaced (default).
 
-    This filter, is automatically inserted when using the ``D`` key (or any
+    This filter is automatically inserted when using the ``d`` key (or any
     other key that toggles the ``deinterlace`` property or when using the
     ``--deinterlace`` switch), assuming the video output does not have native
     deinterlacing support.
 
     If you just want to set the default mode, put this filter and its options
-    into ``--vf-defaults`` instead, and enable deinterlacing with ``D`` or
+    into ``--vf-defaults`` instead, and enable deinterlacing with ``d`` or
     ``--deinterlace``.
 
-    Also note that the ``D`` key is stupid enough to insert an interlacer twice
+    Also, note that the ``d`` key is stupid enough to insert a deinterlacer twice
     when inserting yadif with ``--vf``, so using the above methods is
     recommended.
 
@@ -665,10 +643,11 @@ Available filters are:
     ``buffered-frames``
         Maximum number of decoded video frames that should be buffered before
         the filter (default: 4). This specifies the maximum number of frames
-        the script can requests backwards. E.g. if ``buffered-frames=5``, and
-        the script just requested frame 15, it can still request frame 10, but
-        frame 9 is not available anymore. If it requests frame 30, mpv will
-        decode 15 more frames, and keep only frames 25-30.
+        the script can request in reverse direction.
+        E.g. if ``buffered-frames=5``, and the script just requested frame 15,
+        it can still request frame 10, but frame 9 is not available anymore.
+        If it requests frame 30, mpv will decode 15 more frames, and keep only
+        frames 25-30.
 
         The actual number of buffered frames also depends on the value of the
         ``concurrent-frames`` option. Currently, both option values are
@@ -741,7 +720,7 @@ Available filters are:
 ``vavpp``
     VA-AP-API video post processing. Works with ``--vo=vaapi`` and ``--vo=opengl``
     only. Currently deinterlaces. This filter is automatically inserted if
-    deinterlacing is requested (either using the ``D`` key, by default mapped to
+    deinterlacing is requested (either using the ``d`` key, by default mapped to
     the command ``cycle deinterlace``, or the ``--deinterlace`` option).
 
     ``deint=<method>``
@@ -765,7 +744,7 @@ Available filters are:
 ``vdpaupp``
     VDPAU video post processing. Works with ``--vo=vdpau`` and ``--vo=opengl``
     only. This filter is automatically inserted if deinterlacing is requested
-    (either using the ``D`` key, by default mapped to the command
+    (either using the ``d`` key, by default mapped to the command
     ``cycle deinterlace``, or the ``--deinterlace`` option). When enabling
     deinterlacing, it is always preferred over software deinterlacer filters
     if the ``vdpau`` VO is used, and also if ``opengl`` is used and hardware
@@ -823,6 +802,6 @@ Available filters are:
 
 ``buffer=<num>``
     Buffer ``<num>`` frames in the filter chain. This filter is probably pretty
-    useless, except for debugging. (Note that this won't help smoothing out
+    useless, except for debugging. (Note that this won't help to smooth out
     latencies with decoding, because the filter will never output a frame if
     the buffer isn't full, except on EOF.)

DOCS/man/vo.rst

@@ -20,13 +20,10 @@ normal driver parameters.
 
     See ``--vo=help`` for a list of compiled-in video output drivers.
 
-    The recommended output drivers are ``--vo=vdpau`` and ``--vo=opengl-hq``.
-    All other drivers are just for compatibility or special purposes.
-
-.. admonition:: Example
-
-    ``--vo=opengl,xv,``
-        Try the ``opengl`` driver, then the ``xv`` driver, then others.
+    The recommended output driver is ``--vo=opengl-hq``. All other drivers are
+    for compatibility or special purposes. By default, ``--vo=opengl`` is used,
+    but if that appears not to work, it fallback to other drivers (in the same
+    order as listed by ``--vo=help``).
 
 Available video output drivers are:
 
@@ -75,6 +72,12 @@ Available video output drivers are:
         not responding quickly enough if video FPS is close to or higher than
         the display refresh rate.
 
+``x11`` (X11 only)
+    Shared memory video output driver without hardware acceleration that works
+    whenever X11 is present.
+
+    .. note:: This is a fallback only, and should not be normally used.
+
 ``vdpau`` (X11 only)
     Uses the VDPAU interface to display and optionally also decode video.
     Hardware decoding is used with ``--hwdec=vdpau``.
@@ -102,7 +105,7 @@ Available video output drivers are:
         Select deinterlacing mode (default: 0). In older versions (as well as
         MPlayer/mplayer2) you could use this option to enable deinterlacing.
         This doesn't work anymore, and deinterlacing is enabled with either
-        the ``D`` key (by default mapped to the command ``cycle deinterlace``),
+        the ``d`` key (by default mapped to the command ``cycle deinterlace``),
         or the ``--deinterlace`` option. Also, to select the default deint mode,
         you should use something like ``--vf-defaults=vdpaupp:deint-mode=temporal``
         instead of this sub-option.
@@ -308,7 +311,10 @@ Available video output drivers are:
         of bad or old hardware.
 
         This mode is forced automatically if needed, and this option is mostly
-        useful for debugging.
+        useful for debugging. It's also enabled automatically if nothing uses
+        features which require FBOs.
+
+        This option might be silently removed in the future.
 
     ``scale=<filter>``
 
@@ -375,10 +381,6 @@ Available video output drivers are:
             Scale parameter (``t``). Increasing this makes the result blurrier.
             Defaults to 1.
 
-        sharpen3, sharpen5
-            Sharpening strength. Increasing this makes the image sharper but
-            adds more ringing and aliasing. Defaults to 0.5.
-
         oversample
             Minimum distance to an edge before interpolation is used. Setting
             this to 0 will always interpolate edges, whereas setting it to 0.5
@@ -408,8 +410,8 @@ Available video output drivers are:
         between 0.0 and 1.0. The default value of 0.0 disables antiringing
         entirely.
 
-        Note that this doesn't affect the special filters ``bilinear``,
-        ``bicubic_fast`` or ``sharpen``.
+        Note that this doesn't affect the special filters ``bilinear`` and
+        ``bicubic_fast``.
 
     ``scale-window=<window>``
         (Advanced users only) Choose a custom windowing function for the kernel.
@@ -429,6 +431,15 @@ Available video output drivers are:
             Scale parameter (t). Increasing this makes the window wider.
             Defaults to 1.
 
+    ``scaler-lut-size=<4..10>``
+        Set the size of the lookup texture for scaler kernels (default: 6).
+        The actual size of the texture is ``2^N`` for an option value of ``N``.
+        So the lookup texture with the default setting uses 64 samples.
+
+        All weights are bilinearly interpolated from those samples, so
+        increasing the size of lookup table might improve the accuracy of
+        scaler.
+
     ``scaler-resizes-only``
         Disable the scaler if the video image is not resized. In that case,
         ``bilinear`` is used instead whatever is set with ``scale``. Bilinear
@@ -436,10 +447,12 @@ Available video output drivers are:
         Note that this option never affects ``cscale``.
 
     ``pbo``
-        Enable use of PBOs. This is slightly faster, but can sometimes lead to
-        sporadic and temporary image corruption (in theory, because reupload
-        is not retried when it fails), and perhaps actually triggers slower
-        paths with drivers that don't support PBOs properly.
+        Enable use of PBOs. On some drivers this can be faster, especially if
+        the source video size is huge (e.g. so called "4K" video). On other
+        drivers it might be slower or cause latency issues.
+
+        In theory, this can sometimes lead to sporadic and temporary image
+        corruption (because reupload is not retried when it fails).
 
     ``dither-depth=<N|no|auto>``
         Set dither target depth to N. Default: no.
@@ -470,7 +483,7 @@ Available video output drivers are:
 
     ``temporal-dither``
         Enable temporal dithering. (Only active if dithering is enabled in
-        general.) This changes between 8 different dithering pattern on each
+        general.) This changes between 8 different dithering patterns on each
         frame by changing the orientation of the tiled dithering matrix.
         Unfortunately, this can lead to flicker on LCD displays, since these
         have a high reaction time.
@@ -481,7 +494,7 @@ Available video output drivers are:
         video frame, 2 on every other frame, etc.
 
     ``debug``
-        Check for OpenGL errors, i.e. call ``glGetError()``. Also request a
+        Check for OpenGL errors, i.e. call ``glGetError()``. Also, request a
         debug OpenGL context (which does nothing with current graphics drivers
         as of this writing).
 
@@ -489,6 +502,10 @@ Available video output drivers are:
         Reduce stuttering caused by mismatches in the video fps and display
         refresh rate (also known as judder).
 
+        .. warning:: This requires setting the ``--video-sync`` option to one
+                     of the ``display-`` modes, or it will be silently disabled.
+                     This was not required before mpv 0.14.0.
+
         This essentially attempts to interpolate the missing frames by
         convoluting the video along the temporal axis. The filter used can be
         controlled using the ``tscale`` setting.
@@ -520,7 +537,7 @@ Available video output drivers are:
         The filter used for interpolating the temporal axis (frames). This is
         only used if ``interpolation`` is enabled. The only valid choices
         for ``tscale`` are separable convolution filters (use ``tscale=help``
-        to get a list). The default is ``oversample``.
+        to get a list). The default is ``mitchell``.
 
         Note that the maximum supported filter radius is currently 3, due to
         limitations in the number of video textures that can be loaded
@@ -542,13 +559,82 @@ Available video output drivers are:
         Scale in linear light. It should only be used with a ``fbo-format``
         that has at least 16 bit precision.
 
-    ``fancy-downscaling``
+    ``correct-downscaling``
         When using convolution based filters, extend the filter size
-        when downscaling. Trades quality for reduced downscaling performance.
+        when downscaling. Increases quality, but reduces performance while
+        downscaling.
+
+        This will perform slightly sub-optimally for anamorphic video (but still
+        better than without it) since it will extend the size to match only the
+        milder of the scale factors between the axes.
+
+    ``prescale=<filter>``
+        This option provides non-convolution-based filters for upscaling. These
+        filters resize the video to multiple of the original size (all currently
+        supported prescalers can only perform image doubling in a single pass).
+        Generally another convolution based filter (the main scaler) will be
+        applied after prescaler to match the target display size.
+
+        ``none``
+            Disable all prescalers. This is the default.
+
+        ``superxbr``
+            A relatively fast prescaler originally developed for pixel art.
+
+            Some parameters can be tuned with ``superxbr-sharpness`` and
+            ``superxbr-edge-strength`` options.
+
+        ``nnedi3``
+            An artificial neural network based deinterlacer, which can be used
+            to upscale images.
+
+            Extremely slow and requires a recent mid or high end graphics card
+            to work smoothly (as of 2015).
+
+        Note that all the filters above are designed (or implemented) to process
+        luma plane only and probably won't work as intended for video in RGB
+        format.
+
+    ``prescale-passes=<1..5>``
+        The number of passes to apply the prescaler (defaults to be 1). Setting
+        it to 2 will perform a 4x upscaling.
+
+    ``prescale-downscaling-threshold=<0..32>``
+        This option prevents "overkill" use of prescalers, which can be caused
+        by misconfiguration, or user trying to play a video with much larger
+        size. With this option, user can specify the maximal allowed downscaling
+        ratio in both dimension. To satisfy it, the number of passes for
+        prescaler will be reduced, and if necessary prescaler could also be
+        disabled.
+
+        The default value is 2.0, and should be able to prevent most seemingly
+        unreasonable use of prescalers. Most user would probably want to set it
+        to a smaller value between 1.0 and 1.5 for better performance.
+
+        A value less than 1.0 will disable the check.
+
+    ``nnedi3-neurons=<16|32|64|128>``
+        Specify the neurons for nnedi3 prescaling (defaults to be 32). The
+        rendering time is expected to be linear to the number of neurons.
+
+    ``nnedi3-window=<8x4|8x6>``
+        Specify the size of local window for sampling in nnedi3 prescaling
+        (defaults to be ``8x4``). The ``8x6`` window produces sharper images,
+        but is also slower.
+
+    ``nnedi3-upload=<ubo|shader>``
+        Specify how to upload the NN weights to GPU. Depending on the graphics
+        card, driver, shader compiler and nnedi3 settings, both method can be
+        faster or slower.
+
+        ``ubo``
+            Upload these weights via uniform buffer objects. This is the
+            default. (requires OpenGL 3.1 / GLES 3.0)
+
+        ``shader``
+            Hard code all the weights into the shader source code. (requires
+            OpenGL 3.3 / GLES 3.0)
 
-        This is automatically disabled for anamorphic video, because this
-        feature doesn't work correctly with different scale factors in
-        different directions.
 
     ``pre-shaders=<files>``, ``post-shaders=<files>``, ``scale-shader=<file>``
         Custom GLSL fragment shaders.
@@ -612,11 +698,8 @@ Available video output drivers are:
     ``deband-iterations=<1..16>``
         The number of debanding steps to perform per sample. Each step reduces
         a bit more banding, but takes time to compute. Note that the strength
-        of each step falls off very quickly, so high numbers are practically
-        useless. (Default 4)
-
-        If the performance hit of debanding is too great, you can reduce this
-        to 2 or 1 with marginal visual quality loss.
+        of each step falls off very quickly, so high numbers (>4) are
+        practically useless. (Default 1)
 
     ``deband-threshold=<0..4096>``
         The debanding filter's cut-off threshold. Higher numbers increase the
@@ -626,7 +709,10 @@ Available video output drivers are:
     ``deband-range=<1..64>``
         The debanding filter's initial radius. The radius increases linearly
         for each iteration. A higher radius will find more gradients, but
-        a lower radius will smooth more aggressively. (Default 8)
+        a lower radius will smooth more aggressively. (Default 16)
+
+        If you increase the ``deband-iterations``, you should probably
+        decrease this to compensate.
 
     ``deband-grain=<0..4096>``
         Add some extra noise to the image. This significantly helps cover up
@@ -645,9 +731,19 @@ Available video output drivers are:
         The slope of the sigmoid curve used for ``sigmoid-upscaling``, must
         be a float between 1.0 and 20.0. Defaults to 6.5 if not specified.
 
+    ``sharpen=<value>``
+        If set to a value other than 0, enable an unsharp masking filter.
+        Positive values will sharpen the image (but add more ringing and
+        aliasing). Negative values will blur the image. If your GPU is powerful
+        enough, consider alternatives like the ``ewa_lanczossharp`` scale
+        filter, or the ``scale-blur`` sub-option.
+
+        (This feature is the replacement for the old ``sharpen3`` and
+        ``sharpen5`` scalers.)
+
     ``glfinish``
         Call ``glFinish()`` before and after swapping buffers (default: disabled).
-        Slower, but might help getting better results when doing framedropping.
+        Slower, but might improve results when doing framedropping.
         Can completely ruin performance. The details depend entirely on the
         OpenGL driver.
 
@@ -658,14 +754,27 @@ Available video output drivers are:
 
         X11/GLX only.
 
-    ``dwmflush=<no|windowed|yes>``
-        Calls ``DwmFlush`` after swapping buffers on Windows (default: no).
+    ``vsync-fences=<N>``
+        Synchronize the CPU to the Nth past frame using the ``GL_ARB_sync``
+        extension. A value of 0 disables this behavior (default). A value of
+        1 means it will synchronize to the current frame after rendering it.
+        Like ``glfinish`` and ``waitvsync``, this can lower or ruin performance.
+        Its advantage is that it can span multiple frames, and effectively limit
+        the number of frames the GPU queues ahead (which also has an influence
+        on vsync).
+
+    ``dwmflush=<no|windowed|yes|auto>``
+        Calls ``DwmFlush`` after swapping buffers on Windows (default: auto).
         It also sets ``SwapInterval(0)`` to ignore the OpenGL timing. Values
         are: no (disabled), windowed (only in windowed mode), yes (also in
         full screen).
-        This may help getting more consistent frame intervals, especially with
-        high-fps clips - which might also reduce dropped frames. Typically a
-        value of ``windowed`` should be enough since full screen may bypass the
+
+        The value ``auto`` will try to determine whether the compositor is
+        active, and calls ``DwmFlush`` only if it seems to be.
+
+        This may help to get more consistent frame intervals, especially with
+        high-fps clips - which might also reduce dropped frames. Typically, a
+        value of ``windowed`` should be enough, since full screen may bypass the
         DWM.
 
         Windows only.
@@ -684,22 +793,39 @@ Available video output drivers are:
             Cocoa/OS X
         win
             Win32/WGL
+        angle
+            Direct3D11 through the OpenGL ES translation layer ANGLE. This
+            supports almost everything the ``win`` backend does, except ICC
+            profiles, high bit depth video input, and the ``nnedi3`` prescaler.
+        dxinterop (experimental)
+            Win32, using WGL for rendering and Direct3D 9Ex for presentation.
+            Works on Nvidia and AMD only.
         x11
             X11/GLX
         wayland
             Wayland/EGL
+        drm-egl
+            DRM/EGL
         x11egl
             X11/EGL
 
-    ``es``
-        Force or prefer GLES2/3 over desktop OpenGL, if supported.
+    ``es=<mode>``
+        Select whether to use GLES:
+
+        yes
+            Try to prefer ES over Desktop GL
+        no
+            Try to prefer desktop GL over ES
+        auto
+            Use the default for each backend (default)
 
     ``fbo-format=<fmt>``
         Selects the internal format of textures used for FBOs. The format can
         influence performance and quality of the video output.
         ``fmt`` can be one of: rgb, rgba, rgb8, rgb10, rgb10_a2, rgb16, rgb16f,
         rgb32f, rgba12, rgba16, rgba16f, rgba32f.
-        Default: rgba16.
+        Default: ``auto``, which maps to rgba16 on desktop GL, and rgb10_a2 on
+        GLES (e.g. ANGLE).
 
     ``gamma=<0.1..2.0>``
         Set a gamma value (default: 1.0). If gamma is adjusted in other ways
@@ -779,7 +905,8 @@ Available video output drivers are:
         Automatically select the ICC display profile currently specified by
         the display settings of the operating system.
 
-        NOTE: Only implemented on OS X and X11
+        NOTE: On Windows, the default profile must be an ICC profile. WCS
+        profiles are not supported.
 
     ``icc-cache-dir=<dirname>``
         Store and load the 3D LUTs created from the ICC profile in this directory.
@@ -830,9 +957,11 @@ Available video output drivers are:
                      things like softsubbed ASS signs to match the video colors,
                      but may cause SRT subtitles or similar to look slightly off.
 
-    ``alpha=<blend|yes|no>``
-        Decides what to do if the input has an alpha component (default: blend).
+    ``alpha=<blend-tiles|blend|yes|no>``
+        Decides what to do if the input has an alpha component.
 
+        blend-tiles
+            Blend the frame against a 16x16 gray/white tiles background (default).
         blend
             Blend the frame against a black background.
         yes
@@ -859,7 +988,7 @@ Available video output drivers are:
 
     This is equivalent to::
 
-        --vo=opengl:scale=spline36:cscale=spline36:dscale=mitchell:dither-depth=auto:fancy-downscaling:sigmoid-upscaling:pbo:deband
+        --vo=opengl:scale=spline36:cscale=spline36:dscale=mitchell:dither-depth=auto:correct-downscaling:sigmoid-upscaling:deband:es=no
 
     Note that some cheaper LCDs do dithering that gravely interferes with
     ``opengl``'s dithering. Disabling dithering with ``dither-depth=no`` helps.
@@ -899,7 +1028,7 @@ Available video output drivers are:
 
     ``deint-mode=<mode>``
         Select deinterlacing algorithm. Note that by default deinterlacing is
-        initially always off, and needs to be enabled with the ``D`` key
+        initially always off, and needs to be enabled with the ``d`` key
         (default key binding for ``cycle deinterlace``).
 
         This option doesn't apply if libva supports video post processing (vpp).
@@ -997,27 +1126,6 @@ Available video output drivers are:
 ``opengl-cb``
     For use with libmpv direct OpenGL embedding; useless in any other contexts.
     (See ``<mpv/opengl_cb.h>``.)
-    Usually, ``opengl-cb`` renders frames asynchronously by client and this
-    can cause some frame drops. In order to provide a way to handle this
-    situation, ``opengl-cb`` has its own frame queue and calls update callback
-    more frequently if the queue is not empty regardless of existence of new frame.
-    Once the queue is filled, ``opengl-cb`` drops frames automatically.
-
-    With default options, ``opengl-cb`` renders only the latest frame and drops
-    all frames handed over while waiting render function after update callback.
-
-    ``frame-queue-size=<1..100>``
-        The maximum count of frames which the frame queue can hold (default: 1)
-
-    ``frame-drop-mode=<pop|clear|block>``
-        Select the behavior when the frame queue is full.
-
-        pop
-            Drop the oldest frame in the frame queue.
-        clear
-            Drop all frames in the frame queue.
-        block
-            Wait for a short time, behave like ``clear`` on timeout. (default)
 
     This also supports many of the suboptions the ``opengl`` VO has. Run
     ``mpv --vo=opengl-cb:help`` for a list.
@@ -1042,10 +1150,15 @@ Available video output drivers are:
         Normally it's better to kill the console framebuffer instead, which
         gives better performance.
 
+    ``osd=<yes|no>``
+        Enabled by default. If disabled with ``no``, no OSD layer is created.
+        This also means there will be no subtitles rendered.
+
 ``drm`` (Direct Rendering Manager)
     Video output driver using Kernel Mode Setting / Direct Rendering Manager.
-    Does not support hardware acceleration. Should be used when one doesn't
-    want to install full-blown graphical environment (e.g. no X).
+    Should be used when one doesn't want to install full-blown graphical
+    environment (e.g. no X). Does not support hardware acceleration (if you
+    need this, check the ``drm-egl`` backend for ``opengl`` VO).
 
     ``connector=<number>``
         Select the connector to use (usually this is a monitor.) If set to -1,

README.md

@@ -26,6 +26,21 @@ variety of video file formats, audio and video codecs, and subtitle types.
 
 Releases can be found on the [release list][releases].
 
+## System requirements
+
+- A not too ancient Linux, or Windows Vista or later, or OSX 10.8 or later.
+- A somewhat capable CPU. Hardware decoding might sometimes help if the CPU
+  is too slow to decode video realtime, but must be explicitly enabled with
+  the `--hwdec` option. On Windows, a CPU with SSE4 instruction set is required
+  to get decent hardware decoding performance.
+- A not too crappy GPU. mpv is not intended to be used with bad GPUs. There are
+  many caveats with drivers or system compositors causing tearing, stutter,
+  etc. On Windows, you might want to make sure the graphics drivers are
+  current, especially OpenGL. In some cases, ancient fallback video output
+  methods can help (such as `--vo=xv` on Linux), but this use is not
+  recommended or supported.
+
+
 ## Downloads
 
 
@@ -70,7 +85,7 @@ Essential dependencies (incomplete list):
 - libass (OSD, OSC, text subtitles)
 - Lua (optional, required for the OSC pseudo-GUI and youtube-dl integration)
 - libjpeg (optional, used for screenshots only)
-- Enca (optional, for subtitle charset detection)
+- uchardet (optional, for subtitle charset detection)
 - vdpau and vaapi libraries for hardware decoding on Linux (optional)
 
 Libass dependencies:
@@ -96,13 +111,10 @@ the separately available build wrapper ([mpv-build][mpv-build]) that first compi
 libraries and libass, and then compiles the player statically linked against
 those.
 
-If you are running Mac OSX and using homebrew we provide [homebrew-mpv][homebrew-mpv], an up
-to date formula that compiles mpv with sensible dependencies and defaults for
-OSX.
-
 If you want to build a Windows binary, you either have to use MSYS2 and MinGW,
 or cross-compile from Linux with MinGW. See [Windows compilation][windows_compilation].
 
+
 ## FFmpeg vs. Libav
 
 
@@ -187,7 +199,7 @@ only if discretion is required.
 [windows_compilation]: https://github.com/mpv-player/mpv/blob/master/DOCS/compile-windows.md
 [mplayer-changes]: https://github.com/mpv-player/mpv/blob/master/DOCS/mplayer-changes.rst
 
-## License 
+## License
 
 
-[GPLv2](https://github.com/mpv-player/mpv/blob/master/LICENSE)
+Mostly GPLv2 or later. See [details.](https://github.com/mpv-player/mpv/blob/master/Copyright)

RELEASE_NOTES

@@ -1,3 +1,340 @@
+Release 0.15.0
+==============
+
+Build System Changes
+--------------------
+
+- OS X bundle: remove git sha from the Info.plist version (issue #2677)
+- add "lua51" ("51obsd") to list of possible lua names
+- add option to customize config files system path (issue #2704)
+
+
+Features
+--------
+
+New
+~~~
+
+- vo_opengl: implement support for transparent video display on OS X (alpha=yes suboption)
+- vo_opengl: use a checkerboard pattern as background for transparent video by default
+
+
+Options and Commands
+--------------------
+
+Added
+~~~~~
+
+- add --audio-file-paths (issue #2632)
+- player, stream_dvb: implement dvb-channel-name property, add switch binding
+
+
+Changed
+~~~~~~~
+
+- vf_stereo3d: add alternating modes
+- vo_opengl: disable pbo by default for opengl-hq due to driver problems
+- vf_yadif: change defaults (issue #2539)
+- command: change heuristic for files with 1 chapter (issue #2550)
+- demux_mkv: adjust subtitle preroll defaults
+- exclude 360 from --video-rotate range (issue #2647)
+- osd: make osd-width/height properties watchable
+
+
+Fixes and Minor Enhancements
+----------------------------
+
+- ao_pulse: check for sample rate bounds, attempt fallback (issue #2654)
+- ao_wasapi: remove volume "restore" on exit
+- demux_cue: better error resilience
+- mixer: fix volume initialization with --af=volume
+- mpv.desktop: add audio/mp4 mime type
+- player: detect audio PTS jumps, make video PTS heuristic less aggressive
+- player: make watch later/resume work when "playing" directories
+- player: reset playback abort when reloading a file (issue #2568)
+- recognize frame sequenced 3D Matroska video
+- stream_dvb: fix channel switching
+- vaapi: add VP9 profile (requires VA-API 0.38.1 or newer)
+- vo_opengl: dxinterop: prevent crash after lost device
+- vo_opengl: enable brightness/contrast controls for RGB
+- vo_opengl: fix operation on GLES 2.0
+- vo_opengl: fix operation on GLSL versions earlier than 1.30
+- vo_opengl: flip screenshot image if backend uses flipped rendering (issue #2635)
+- vo_opengl: reset nnedi3 weights properly (issue #2661)
+- vo_rpi: handle rotation
+- vo_rpi: work around firmware oddness leading to incorrect video rect
+- windows: fix fd://
+- ytdl: Include Referer header as well
+- TOOLS/zsh.pl: add .opus extension in zsh completions
+
+
+This listing is not complete. Check DOCS/client-api-changes.rst for a history
+of changes to the client API. A complete changelog can be seen by running
+`git log v0.14.0..v0.15.0` in the git repository or by visiting either
+https://github.com/mpv-player/mpv/compare/v0.14.0...v0.15.0 or
+http://git.srsfckn.biz/mpv/log/?qt=range&q=v0.14.0..v0.15.0
+
+
+
+Release 0.14.0
+==============
+
+Build System Changes
+--------------------
+
+- build: install scalable svg icon as well
+- vo_opengl: require --enable-gpl3 for nnedi
+- win32: enable internal pthreads wrapper by default
+
+
+Features
+--------
+
+New
+~~~
+
+- vo_opengl: add experimental dxinterop backend (renders with OpenGL, displays through Direct3D; broken with Intel drivers)
+- vo_opengl: add initial ANGLE support
+- windows: implement icc-profile-auto
+- windows: support taskbar button progress indicator (issue #2399)
+
+Removed
+~~~~~~~
+
+- Windows XP support
+- demux: remove old subtitle parser (FFmpeg users are unaffected; Libav loses support for some minor formats)
+- demux_libass: remove this demuxer (libavformat takes its place)
+- stream: drop old Linux PVR support
+
+
+Options and Commands
+--------------------
+
+Added
+~~~~~
+
+- command, vo: add estimated-display-fps property
+- command: add vsync-ratio property
+- command: export some per-video-frame information (issue #2444)
+- vo_opengl: make LOOKUP_TEXTURE_SIZE configurable
+- windows: add option to set VO MMCSS profile
+
+
+Changed
+~~~~~~~
+
+- --sub-fix-timing now applies to ASS subtitles as well
+- command: rename vo-missed-frame-count property to vo-delayed-frame-count
+- input.conf: add default bindings for changing window scale (issue #2500)
+- videotoolbox: make decoder format customizable
+- vo_opengl: disable interpolation without display-sync
+- vo_opengl: make tscale=mitchell:tscale-clamp the default
+- vo_rpi: add an option to disable OSD
+
+
+Fixes and Minor Enhancements
+----------------------------
+
+- af_lavrresample: clamp float output to range
+- ao: disambiguate default device list entries
+- ao_alsa: filter audio device list
+- ao_alsa: list bidirectional devices too
+- ao_openal: accommodate more sample formats (issue #2494)
+- ao_openal: fix virtual speaker positioning
+- ao_wasapi: only report per-app volume in shared mode
+- ao_wasapi: work around DTS passthrough failure
+- build: make vaapi-wayland depend on gl-wayland (issue #2476)
+- demux: fix seeking in .ts
+- demux_lavf: mark ASS tracks as always UTF-8
+- demux_mkv: fix incremental indexing with single-keyframe files (issue #2498)
+- drm: fix setting up connectors
+- dxva2: reject 10 bit HEVC (issue #2516)
+- player: replace mistimed-frame-count with vsync-ratio on status line
+- sd_ass: fix secondary subtitle mode
+- various display-sync improvements
+- vo_opengl: enable NNEDI3 prescaler on OpenGL ES 3.0
+- vo_opengl: enable colormatrix even for RGB input
+- vo_opengl: fix backend autoprobing, attempt to improve GLX vs. EGL backend detection
+- vo_opengl: fix backend=x11 on Intel
+- vo_opengl: fix issues with some obscure pixel formats (e.g. rgb555)
+- vo_opengl: fix precision loss of fruit dithering matrix
+- vo_opengl: force dumb mode if RG textures are not available
+- vo_opengl: improve boundary check for polar filters
+- vo_opengl: various GLES compatibility improvements
+- vo_opengl: win32: test for exclusive mode
+- vo_opengl_cb: do not block on flipping when redrawing
+- vo_rpi: set aspect ratio
+- win32: fix console output with raw stdio functions
+- windows: try to avoid detection as exclusive fullscreen window (issue #2177)
+- x11: request bypassing compositor (issue #2502)
+
+
+This listing is not complete. Check DOCS/client-api-changes.rst for a history
+of changes to the client API. A complete changelog can be seen by running
+`git log v0.13.0..v0.14.0` in the git repository or by visiting either
+https://github.com/mpv-player/mpv/compare/v0.13.0...v0.14.0 or
+http://git.srsfckn.biz/mpv/log/?qt=range&q=v0.13.0..v0.14.0
+
+
+Release 0.13.0
+==============
+
+NOTE: The previous release changed the default format for youtube-dl and
+removed some workarounds related to MPEG DASH support. However, the required
+changes to FFmpeg code (FFmpeg commit 4ab56667594842283dc5ae07f0daba2a2cb4d3af)
+are not in any FFmpeg release yet (as of 2015-11-10), so for now, playing DASH
+streams requires using FFmpeg git master. You can work around this by using the
+--ytdl-format=best option.
+
+
+Features
+--------
+
+New
+~~~
+
+- SVG version of the icon (with symbolic counterpart)
+- stream_libarchive: add multivolume support
+- vo_opengl: add prescaling framework along with Super-xBR and NNEDI3 (currently very slow) prescalers (issue #2230)
+
+
+Options and Commands
+--------------------
+
+Added
+~~~~~
+
+- command: add mistimed-frame-count property
+- vo_opengl: add vsync-fences option
+
+
+Changed
+~~~~~~~
+
+- command: make display-fps property writable
+- options: enable mpeg2 hw decoding by default if hw decoding is requested
+- vo_opengl: rename fancy-downscaling to correct-downscaling
+- vo_opengl: correct-downscaling: enable also for anamorphic clips
+- vo_opengl: rename "drm_egl" to "drm-egl"
+- vo_opengl: disable drm-egl autopickup
+- vo_opengl: never load vaapi GLX interop by default
+
+
+Fixes and Minor Enhancements
+----------------------------
+
+- ao_alsa: fix 7.1 over HDMI
+- audio: do not require full audio chain reinit for speed changes
+- rpi: add support for codecs other than h264 (mpeg2 for now)
+- vd_lavc: make hwdec fallback more tolerant
+- video: fix playback of pal8
+- video: multiple display-sync fixes
+- vo: fix no-audio mode with interpolation enabled/display-sync disabled
+- vo_direct3d: fix operation (issue #2434)
+- vo_drm: handle possible errors from sigaction
+- vo_drm: show osd in audio only mode
+- vo_opengl: do not attempt to cache frames in FBO in dumb-mode (issue #2432)
+- vo_opengl: win32: always request MMCSS for DWM
+- vo_opengl: win32: try to enable DwmFlush by default
+- vo_vdpau: check VDP_RGBA_FORMAT_A8 support
+- win32: request MMCSS "Playback" profile
+
+
+This listing is not complete. Check DOCS/client-api-changes.rst for a history
+of changes to the client API. A complete changelog can be seen by running
+`git log v0.12.0..v0.13.0` in the git repository or by visiting either
+https://github.com/mpv-player/mpv/compare/v0.12.0...v0.13.0 or
+http://git.srsfckn.biz/mpv/log/?qt=range&q=v0.12.0..v0.13.0
+
+
+Release 0.12.0
+==============
+
+NOTE: This release changes the default format for youtube-dl and removes some
+workarounds related to MPEG DASH support. However, the required changes to
+FFmpeg code are not in any FFmpeg release yet (as of 2015-10-29), so for now,
+playing DASH streams requires using FFmpeg git master. You can work around
+this by using the --ytdl-format=best option.
+
+
+Features
+--------
+
+New
+~~~
+
+- vo_opengl: support new VAAPI EGL interop (requires Mesa 11)
+- vo_opengl: vaapi: add Wayland support
+- bring back the x11 video output
+- vo_opengl: support all kinds of GBRP formats
+
+Removed
+~~~~~~~
+
+- video: remove VDA support (VideoToolbox is preferred)
+
+Behavior
+--------
+
+- vo_opengl: enable X11 EGL backend by default (disabled for NVIDIA due to driver bugs)
+- ytdl: don't override user-set format in no-video mode
+- sub: adjust behavior on mismatching video/subtitle aspect ratio
+
+
+Options and Commands
+--------------------
+
+Added
+~~~~~
+
+- audio: add option for falling back to ao_null
+- options: add support for client certificate authentication
+- input: add key name for U+3000 IDEOGRAPHIC SPACE
+- player: add audio drop/duplicate mode to video-sync
+
+Changed
+~~~~~~~
+
+- vo_opengl: remove sharpen scalers, add sharpen sub-option
+- vo_opengl: make sw suboption work without explicit backend selection
+- command: make time properties unavailable if timestamp is unknown
+- command: do not return 0 for bitrate if unknown (make property unavailable instead)
+- vo_opengl: make the default debanding settings less excessive
+- ytdl: disable --all-subs if "sub-lang" is in raw-options
+
+Removed
+~~~~~~~
+
+- video: remove user-controllable PTS sorting (--pts-association-mode)
+
+
+Fixes and Minor Enhancements
+----------------------------
+
+- player: fix another --force-window bug
+- player: add wav to list of external audio file extensions
+- ao_alsa: fix failure to find any sample format
+- player: make stop command actually stop in all cases
+- audio: various fixes related to audio device hotplugging
+- vo_xv: fix crash with --wid
+- ytdl: Remove DASH hacks, use DASH by default
+- player: be slightly less prone to framedrop in display sync mode
+- sd_lavc: extend subtitle resolution if images go outside video frame
+- player: offset chapter display by start time
+- command: make bitrate properties work correctly for external tracks
+- w32_common: disable IME
+- player: fix display-sync A/V calculation on high playback speeds
+- player: fix display sync A/V difference estimation on drops
+- player: raise display sync desync tolerance
+
+
+This listing is not complete. Check DOCS/client-api-changes.rst for a history
+of changes to the client API. A complete changelog can be seen by running
+`git log v0.11.0..v0.12.0` in the git repository or by visiting either
+https://github.com/mpv-player/mpv/compare/v0.11.0...v0.12.0 or
+http://git.srsfckn.biz/mpv/log/?qt=range&q=v0.11.0..v0.12.0
+
+
 Release 0.11.0
 ==============
 

TOOLS/file2string.pl

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

TOOLS/gen-x11-icon.sh

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

TOOLS/lua/autoload.lua

@@ -30,7 +30,12 @@ function add_files_at(index, files)
 end
 
 function get_extension(path)
-    return string.match(path, "%.([^%.]+)$" )
+    match = string.match(path, "%.([^%.]+)$" )
+    if match == nil then
+        return "nomatch"
+    else
+        return match
+    end
 end
 
 table.filter = function(t, iter)

TOOLS/old-configure

@@ -1,998 +0,0 @@
-#! /bin/sh
-#
-# Original version (C) 2000 Pontscho/fresh!mindworkz pontscho@makacs.poliod.hu
-# Cleanups all over the place (c) 2001 pl
-# Rewritten for mpv in 2014.
-#
-#
-# Warning: this is not officially supported. Use on your own risk. Might require
-#          modifications when used on systems other than Linux. Contributors
-#          are not expected to update these files when making changes to the
-#          normal/preferred waf build system.
-#
-#
-# This configure script is *not* autoconf-based and has different semantics.
-# It attempts to autodetect all settings and options where possible. It is
-# possible to override autodetection with the --enable-option/--disable-option
-# command line parameters.  --enable-option forces the option on skipping
-# autodetection. Yes, this means that compilation may fail and yes, this is not
-# how autoconf-based configure scripts behave.
-#
-# configure generates a series of configuration files:
-#  - config.h contains #defines that are used in the C code.
-#  - config.mak is included from the Makefiles.
-#
-# If you want to add a new check for $feature, look at the existing checks
-# and try to use helper functions where you can.
-#############################################################################
-
-# Prevent locale nonsense from breaking basic text processing utils
-export LC_ALL=C
-
-# Store the configure line that was used
-configuration="$*"
-
-# Prefer these macros to full length text !
-# These macros only return an error code - NO display is done
-command_check() {
-  echo >> "$TMPLOG"
-  echo "$@" >> "$TMPLOG"
-  "$@" >> "$TMPLOG" 2>&1
-  TMPRES="$?"
-  echo >> "$TMPLOG"
-  return "$TMPRES"
-}
-
-compile_check() {
-  source="$1"
-  shift
-  echo >> "$TMPLOG"
-  cat "$source" >> "$TMPLOG"
-  echo >> "$TMPLOG"
-  echo "$_cc $OURCFLAGS $CFLAGS $source $extra_cflags $_ld_static $extra_ldflags $libs_mplayer -o $TMPEXE $@" >> "$TMPLOG"
-  rm -f "$TMPEXE"
-  $_cc $OURCFLAGS $CFLAGS "$source" $extra_cflags $_ld_static $extra_ldflags $libs_mplayer -o "$TMPEXE" $@ >> "$TMPLOG" 2>&1
-  TMPRES="$?"
-  echo >> "$TMPLOG"
-  echo >> "$TMPLOG"
-  return "$TMPRES"
-}
-
-cflag_check() {
-    echo "int main(void) { return 0; }" > $TMPC
-    compile_check $TMPC $@
-}
-
-statement_check() {
-    echo "" > $TMPC
-    for _header in $1 ; do echo "#include <$_header>" >> $TMPC ; done
-    echo "int main(void) { $2; return 0; }" >> $TMPC
-    shift 2
-    compile_check $TMPC $@
-}
-
-pkg_config_add() {
-  echo >> "$TMPLOG"
-  echo "$_pkg_config --cflags $@" >> "$TMPLOG"
-  ctmp=$($_pkg_config --cflags "$@" 2>> "$TMPLOG") || return $?
-  echo >> "$TMPLOG"
-  echo "$_pkg_config --libs $@" >> "$TMPLOG"
-  ltmp=$($_pkg_config --libs "$@" 2>> "$TMPLOG") || return $?
-  echo >> "$TMPLOG"
-  echo "cflags: $ctmp" >> "$TMPLOG"
-  echo "libs: $ltmp" >> "$TMPLOG"
-  echo >> "$TMPLOG"
-  extra_cflags="$extra_cflags $ctmp"
-  libs_mplayer="$libs_mplayer $ltmp"
-}
-
-# Display error message, flushes tempfile, exit
-die () {
-  echo
-  echo "Error: $@" >&2
-  echo >&2
-  rm -f "$TMPEXE" "$TMPC"
-  echo "Check \"$TMPLOG\" if you do not understand why it failed."
-  exit 1
-}
-
-# Use this before starting a check
-echocheck() {
-  echo "============ Checking for $@ ============" >> "$TMPLOG"
-  echo ${_echo_n} "Checking for $@ ... ${_echo_c}"
-}
-
-# Use this to echo the results of a check
-echores() {
-  test "$res_comment" && res_comment="($res_comment)"
-  echo "Result is: $@ $res_comment" >> "$TMPLOG"
-  echo "##########################################" >> "$TMPLOG"
-  echo "" >> "$TMPLOG"
-  echo "$@ $res_comment"
-  res_comment=""
-}
-
-# Check how echo works in this /bin/sh
-case $(echo -n) in
-  -n)   _echo_n=        _echo_c='\c'    ;;      # SysV echo
-  *)    _echo_n='-n '   _echo_c=        ;;      # BSD echo
-esac
-
-# setind $a b sets the variable named by the value of the variable a to b
-setind() { eval "$1=\"\$2\"" ; }
-
-# Generate --enable-NAME/--disable-NAME options, set $1 to the option value.
-# Since shell has no data structures, do a weird statemachine thing.
-# Arguments: "_name"($1) "description"($2) "default"($3)
-# If "default"($3) is empty, use "auto"
-# Option name: a leading "_" in name is stripped, further "_" are changed to "-"
-opt_yes_no() {
-    _name=$(echo "$1" | sed 's/^_//' | tr _ -)
-    _defval="$3"
-    test -z "$_defval" && _defval=auto
-
-    case "$_opt_state_mode" in
-    init)
-        setind "$1" "$_defval"
-        ;;
-    help)
-        if test "$_defval" = yes || test "$_defval" = auto ; then
-            _defdesc=enable
-            test "$_defval" = auto && _defdesc=auto
-            printf "  %-21s  disable $2 [$_defdesc]\n" "--disable-$_name"
-            unset _defdesc
-        else
-            printf "  %-21s  enable $2 [disable]\n" "--enable-$_name"
-        fi
-        ;;
-    parse)
-        if test "$_opt_state_name" = "--enable-$_name" ; then
-            setind "$1" yes
-            _opt_state_known=yes
-        elif test "$_opt_state_name" = "--disable-$_name" ; then
-            setind "$1" no
-            _opt_state_known=yes
-        elif test "$_opt_state_name" = "--auto-$_name" ; then
-            setind "$1" auto
-            _opt_state_known=yes
-        fi
-        ;;
-    esac
-    unset _name
-    unset _defval
-}
-
-options_state_machine() {
-    _opt_state_mode=$1
-    _opt_state_name=$2
-    _opt_state_known=no
-
-    opt_yes_no _gl          "OpenGL video output"
-    opt_yes_no _libguess    "libguess"
-    opt_yes_no _termios     "termios database for key codes"
-    opt_yes_no _iconv       "iconv for encoding conversion"
-    opt_yes_no _vm          "X video mode extensions"
-    opt_yes_no _dvb         "DVB input"
-    opt_yes_no _tv          "TV interface (TV/DVB grabbers)" yes
-    opt_yes_no _tv_v4l2     "Video4Linux2 TV interface"
-    opt_yes_no _libv4l2     "libv4l2"
-    opt_yes_no _pvr         "Video4Linux2 MPEG PVR"
-    opt_yes_no _smb         "Samba (SMB) input"
-    opt_yes_no _lcms2       "LCMS2 support"
-    opt_yes_no _bluray      "Blu-ray support"
-    opt_yes_no _dvdread     "libdvdread"
-    opt_yes_no _dvdnav      "libdvdnav"
-    opt_yes_no _enca        "ENCA charset oracle library"
-    opt_yes_no _uchardet    "uchardet charset detection library"
-    opt_yes_no _libass      "subtitle rendering with libass"
-    opt_yes_no _libavdevice "libavdevice demuxers"
-    opt_yes_no _libavfilter "libavfilter"
-    opt_yes_no _jpeg        "support for writing JPEG screenshots"
-    opt_yes_no _libcdio     "libcdio support"
-    opt_yes_no _librubberband "librubberband support"
-    opt_yes_no _ffmpeg      "skip FFmpeg/Libav autodetection"
-    opt_yes_no _libavresample "libavresample (preferred over libswresample)"
-    opt_yes_no _libswresample "libswresample"
-    opt_yes_no _caca        "CACA  video output"
-    opt_yes_no _sdl2        "SDL2 video and audio outputs" no
-    opt_yes_no _xv          "Xv video output"
-    opt_yes_no _vdpau       "VDPAU acceleration"
-    opt_yes_no _vaapi       "VAAPI acceleration"
-    opt_yes_no _xrandr      "Xrandr support (used for monitor FPS detection)"
-    opt_yes_no _xinerama    "Xinerama support"
-    opt_yes_no _x11         "X11 video output"
-    opt_yes_no _wayland     "Wayland video output"
-    opt_yes_no _xss         "support for disabling screensaver via xss"
-    opt_yes_no _alsa        "ALSA audio output"
-    opt_yes_no _ossaudio    "OSS audio output"
-    opt_yes_no _rsound      "RSound audio output"
-    opt_yes_no _sndio       "sndio audio output"
-    opt_yes_no _pulse       "Pulseaudio audio output"
-    opt_yes_no _jack        "JACK audio output"
-    opt_yes_no _openal      "OpenAL audio output"
-    opt_yes_no _shm         "X11/Xv shared memory"
-    opt_yes_no _lua         "Lua scripting"
-    opt_yes_no _vapoursynth "VapourSynth filter bridge (Python)"
-    opt_yes_no _vapoursynth_lazy "VapourSynth filter bridge (Lua)"
-    opt_yes_no _libarchive  "libarchive"
-    opt_yes_no _encoding    "encoding functionality" yes
-    opt_yes_no _build_man   "building manpage"
-}
-
-show_help(){
-cat << EOF
-Usage: $0 [OPTIONS]...
-
-Configuration:
-  -h, --help             display this help and exit
-
-Installation directories:
-  --prefix=DIR           prefix directory for installation [/usr/local]
-  --bindir=DIR           directory for installing binaries [PREFIX/bin]
-  --datadir=DIR          directory for installing machine independent
-                         data files (skins, etc) [PREFIX/share/mpv]
-  --mandir=DIR           directory for installing man pages [PREFIX/share/man]
-  --confdir=DIR          directory for installing configuration files
-                         [PREFIX/etc/mpv]
-
-Compilation options:
-  --cc=COMPILER          C compiler to build mpv [gcc]
-  --pkg-config=PKGCONFIG pkg-config to find some libraries [pkg-config]
-  --enable-static        build a statically linked binary
-  --disable-debug        compile-in debugging information [enable]
-  --disable-optimization compile without -O2 [enable]
-
-Use these options if autodetection fails:
-  --extra-cflags=FLAGS   extra CFLAGS
-  --extra-ldflags=FLAGS  extra LDFLAGS
-  --extra-libs=FLAGS     extra linker flags
-
-Features:
-EOF
-options_state_machine help
-cat << EOF
-
-This configure script is NOT autoconf-based, even though its output is similar.
-It will try to autodetect all configuration options. If you --enable an option
-it will be forcefully turned on, skipping autodetection. This can break
-compilation, so you need to know what you are doing.
-EOF
-exit 0
-} #show_help()
-
-# GOTCHA: the variables below defines the default behavior for autodetection
-# and have - unless stated otherwise - at least 2 states : yes no
-# If autodetection is available then the third state is: auto
-_pkg_config=auto
-_cc=auto
-test -n "$CC" && _cc="$CC"
-_opt=-O2
-_prefix="/usr/local"
-options_state_machine init
-for ac_option do
-  case "$ac_option" in
-  --help|-help|-h)
-    show_help
-    ;;
-  --prefix=*)
-    _prefix=$(echo $ac_option | cut -d '=' -f 2)
-    ;;
-  --bindir=*)
-    _bindir=$(echo $ac_option | cut -d '=' -f 2)
-    ;;
-  --mandir=*)
-    _mandir=$(echo $ac_option | cut -d '=' -f 2)
-    ;;
-  --confdir=*)
-    _confdir=$(echo $ac_option | cut -d '=' -f 2)
-    ;;
-  --extra-cflags=*)
-    extra_cflags="$extra_cflags $(echo $ac_option | cut -d '=' -f 2-)"
-    ;;
-  --extra-ldflags=*)
-    extra_ldflags="$extra_ldflags $(echo $ac_option | cut -d '=' -f 2-)"
-    ;;
-  --extra-libs=*)
-    libs_mplayer=$(echo $ac_option | cut -d '=' -f 2)
-    ;;
-  --cc=*)
-    _cc=$(echo $ac_option | cut -d '=' -f 2)
-    ;;
-  --pkg-config=*)
-    _pkg_config=$(echo $ac_option | cut -d '=' -f 2)
-    ;;
-  --enable-static)
-    _ld_static='-static'
-    ;;
-  --disable-static)
-    _ld_static=''
-    ;;
-  --enable-optimization)
-    _opt='-O2'
-    ;;
-  --enable-optimization=*)
-    _opt=$(echo $_echo_n '-O'$_echo_c; echo $ac_option | cut -d '=' -f 2)
-    ;;
-  --disable-optimization)
-    _opt=
-    ;;
-  *)
-    options_state_machine parse "$ac_option"
-    if test "$_opt_state_known" != yes ; then
-        echo "Unknown parameter: $ac_option" >&2
-        exit 1
-    fi
-    ;;
-
-  esac
-done
-
-test -z "$_bindir"  && _bindir="$_prefix/bin"
-test -z "$_mandir"  && _mandir="$_prefix/share/man"
-test -z "$_confdir" && _confdir="$_prefix/etc/mpv"
-
-mplayer_tmpdir=$(mktemp -d -p ${TMPDIR:=/tmp} mpv-configure-XXXXXX)
-test -n "$mplayer_tmpdir" || die "Unable to create tmpdir."
-trap 'rm -rf "$mplayer_tmpdir"' EXIT
-
-mkdir old_build 2> /dev/null
-
-TMPLOG="old_build/config.log"
-
-rm -f "$TMPLOG"
-echo Parameters configure was run with: > "$TMPLOG"
-echo CFLAGS="'$CFLAGS'" PKG_CONFIG_PATH="'$PKG_CONFIG_PATH'" ./configure $configuration >> "$TMPLOG"
-echo >> "$TMPLOG"
-
-TMPC="$mplayer_tmpdir/tmp.c"
-TMPEXE="$mplayer_tmpdir/tmp"
-CONFIG_MAK="$mplayer_tmpdir/config.mak"
-CONFIG_H="$mplayer_tmpdir/config.h"
-
-echo > $CONFIG_MAK
-echo > $CONFIG_H
-
-test "$_pkg_config" = auto && _pkg_config=pkg-config
-test "$_cc" = auto && _cc=cc
-
-extra_cflags="-I. -D_GNU_SOURCE $extra_cflags"
-
-_rst2man=rst2man
-test -f "$(which rst2man.py)" && _rst2man=rst2man.py
-
-echocheck "whether to build manpages with rst2man"
-if test "$_build_man" = auto ; then
-    _build_man=no
-    command_check "$_rst2man" --version && _build_man=yes
-fi
-echores "$_build_man"
-
-echocheck "working compiler"
-cflag_check "" || die "Compiler is not functioning correctly. Check your installation and custom CFLAGS $CFLAGS ."
-echores "yes"
-
-echocheck "perl"
-command_check perl -Mv5.8 -e';' || die "Perl is not functioning correctly or is ancient. Install the latest perl available."
-echores yes
-
-echocheck "compiler support of -pipe option"
-cflag_check -pipe -I. && _pipe="-pipe" && echores "yes" || echores "no"
-
-addcflags() { cflag_check "$@" && OURCFLAGS="$OURCFLAGS $@" ; }
-
-OURCFLAGS="-std=c99 -Wall $_opt"
-
-addcflags -g -g3 -ggdb
-addcflags -Wundef -Wmissing-prototypes -Wshadow -Wno-switch -Wparentheses -Wpointer-arith -Wno-redundant-decls -Wno-pointer-sign -Werror=implicit-function-declaration -Wno-error=deprecated-declarations -Wno-error=unused-function
-# clang
-addcflags -Wno-logical-op-parentheses -fcolor-diagnostics -Wno-tautological-compare -Wno-tautological-constant-out-of-range-compare
-# extra
-addcflags -Wno-format-zero-length -Wempty-body -Wdisabled-optimization -Wstrict-prototypes
-
-cflag_check -MD -MP && DEPFLAGS="-MD -MP"
-cflag_check -lm && _ld_lm="-lm"
-
-extra_ldflags="$extra_ldflags $LDFLAGS"
-extra_cflags="$extra_cflags $CPPFLAGS"
-
-# If $1 is "yes", define $2 as 1 in config.h, else define it as 0
-define_yes_no() {
-    if test "$1" = yes ; then
-        echo "#define $2 1" >> $CONFIG_H
-    else
-        echo "#define $2 0" >> $CONFIG_H
-    fi
-}
-
-# Write the results of a check to config.mak/.h.
-# Arguments: "yes/no"($1) "name"($2)
-check_yes_no() {
-    define_yes_no $1 "HAVE_$2"
-    echo "$2 = $1" >> $CONFIG_MAK
-}
-
-check_trivial() {
-    echocheck "$1"
-    check_yes_no $2 $3
-    echores "$2"
-}
-
-# Arguments: "message"($1) "setting"($2) "name"($3) "code"($4)
-# Also, $5 - $N can be libraries needed - it'll try each separately.
-# Use " " as first entry if you want to try with no libraries too.
-check_compile() {
-    _res="$2"
-    _name="$3"
-    _code="$4"
-    echocheck "$1"
-    if test $_res = auto ; then
-        _res=no
-        if test $# -gt 4 ; then
-            shift 4
-        else
-            shift $#
-        fi
-        while true ; do
-            compile_check "$_code" "$1" && libs_mplayer="$libs_mplayer $1" && _res=yes && break
-            test -z "$1" && break
-            shift
-        done
-    fi
-    check_yes_no $_res $_name
-    echores $_res
-    test $_res = yes && return 0 || return 1
-}
-
-# Arguments: "message"($1) "setting"($2) "name"($3) "include"($4) "statement"($5)
-# Also, $6 - $N can be libraries needed - it'll try each separately.
-# Use " " as first entry if you want to try with no libraries too.
-check_statement_libs() {
-    _res="$2"
-    _name="$3"
-    _inc="$4"
-    _st="$5"
-    echocheck "$1"
-    if test $_res = auto ; then
-        _res=no
-        if test $# -gt 5 ; then
-            shift 5
-        else
-            shift $#
-        fi
-        while true ; do
-            statement_check "$_inc" "$_st" "$1" && libs_mplayer="$libs_mplayer $1" && _res=yes && break
-            test -z "$1" && break
-            shift
-        done
-    fi
-    check_yes_no $_res $_name
-    echores $_res
-    test $_res = yes && return 0 || return 1
-}
-
-# Print "yes" if previous command succeeded, else "no"
-defretval() { # shell is retarded?
-    if test $? = 0 ; then
-        echo "yes"
-    else
-        echo "no"
-    fi
-}
-
-echocheck "dynamic loader"
-_dl=no
-for _ld_tmp in "" "-ldl"; do
-  statement_check dlfcn.h 'dlopen("", 0)' $_ld_tmp && _ld_dl="$_ld_tmp" && _dl=yes && break
-done
-define_yes_no $_dl HAVE_LIBDL
-echores "$_dl"
-
-echocheck "pthread"
-cflag_check -pthread && _ld_pthread="$_ld_pthread -pthread"
-cflag_check -lpthread && _ld_pthread="$_ld_pthread -lpthread"
-cflag_check -lrt && _ld_pthread="$_ld_pthread -lrt"
-extra_cflags="$extra_cflags -D_REENTRANT -D_THREAD_SAFE"
-compile_check waftools/fragments/pthreads.c "$_ld_pthread" || die "Unable to find pthreads support."
-echores "yes"
-
-check_statement_libs "support for stdatomic.h" auto STDATOMIC \
-    stdatomic.h 'atomic_int_least64_t test = ATOMIC_VAR_INIT(123); int test2 = atomic_load(&test)' \
-    " " "-latomic"
-_stdatomic=$(defretval)
-
-_atomic=auto
-test "$_stdatomic" = yes && _atomic=no
-check_statement_libs "compiler support for __atomic built-ins" $_atomic ATOMIC_BUILTINS \
-    stdint.h 'int64_t test = 0; test = __atomic_add_fetch(&test, 1, __ATOMIC_SEQ_CST)' \
-    " " "-latomic"
-_atomic=$(defretval)
-
-_sync=auto
-(test "$_atomic" = yes || test "$_stdatomic" = yes ) && _sync=no
-check_statement_libs "compiler support for __sync built-ins" $_sync SYNC_BUILTINS \
-    stdint.h 'int64_t test = 0; test = __sync_add_and_fetch(&test, 1)'
-_sync=$(defretval)
-
-_any_atomic=yes
-if test "$_atomic" = no && test "$_sync" = no && test "$_stdatomic" = no ; then
-  echo "your compiler must support either stdatomic.h, or __atomic, or __sync built-ins."
-  _any_atomic=no
-fi
-define_yes_no $_any_atomic HAVE_ATOMICS
-
-check_compile "iconv" $_iconv ICONV waftools/fragments/iconv.c " " "-liconv" "-liconv $_ld_dl"
-_iconv=$(defretval)
-if test "$_iconv" != yes ; then
-    die "Unable to find iconv which should be part of standard compilation environment. Aborting. If you really mean to compile without iconv support use --disable-iconv."
-fi
-
-_soundcard_header=sys/soundcard.h
-_check_snd=yes
-check_statement_libs "sys/soundcard.h" auto SYS_SOUNDCARD_H sys/soundcard.h
-test $(defretval) = yes && _soundcard_header=sys/soundcard.h && _check_snd=no
-check_statement_libs "soundcard.h" $_check_snd SOUNDCARD_H soundcard.h
-test $(defretval) = yes && _soundcard_header=soundcard.h
-
-check_statement_libs "sys/videoio.h" auto SYS_VIDEOIO_H sys/videoio.h
-
-_termios_ok=no
-check_statement_libs "termios.h" $_termios TERMIOS_H termios.h
-test $(defretval) = yes && _termios_ok=yes && _termios=no
-check_statement_libs "sys/termios.h" $_termios SYS_TERMIOS_H 'sys/termios.h'
-test $(defretval) = yes && _termios_ok=yes
-define_yes_no $_termios_ok HAVE_TERMIOS
-
-check_statement_libs "shm" $_shm SHM "sys/types.h sys/ipc.h sys/shm.h" \
-    "shmget(0, 0, 0); shmat(0, 0, 0); shmctl(0, 0, 0);"
-
-echocheck "pkg-config"
-if $($_pkg_config --version > /dev/null 2>&1); then
-  if test "$_ld_static"; then
-    _pkg_config="$_pkg_config --static"
-  fi
-  echores "yes"
-else
-  _pkg_config=false
-  echores "no"
-fi
-
-# Arguments: "message"($1) "setting"($2) "name"($3) "pkg-config string"($4)
-# The name will be used as "#define HAVE_$name 1/0" in config.h, and as
-# "$name = yes/no" in config.mak
-# "setting"($2) is yes/no/auto and represents the --enable/--disable option
-check_pkg_config() {
-    echocheck "$1"
-    _res=$2
-    if test "$2" = auto ; then
-        _res=no
-        if pkg_config_add "$4" ; then
-            _res=yes
-        fi
-    fi
-    check_yes_no $_res $3
-    echores "$_res"
-    test $_res = yes && return 0 || return 1
-}
-
-check_pkg_config "libguess support" $_libguess LIBGUESS 'libguess >= 1.0'
-
-check_pkg_config "Samba support (libsmbclient)" $_smb LIBSMBCLIENT 'smbclient >= 0.2.0'
-
-_wlver="1.6.0"
-check_pkg_config "Wayland" $_wayland WAYLAND "wayland-client >= $_wlver wayland-cursor >= $_wlver xkbcommon >= 0.3.0"
-_wayland=$(defretval)
-
-check_pkg_config "X11" $_x11 X11 "x11"
-_x11=$(defretval)
-
-# Disable X11 dependencies
-_xext=auto
-if test "$_x11" = no ; then
-    _xss=no
-    _xext=no
-    _xv=no
-    _vdpau=no
-    _vaapi=no
-    _xinerama=no
-    _vm=no
-fi
-
-check_pkg_config "Xss screensaver extensions" $_xss XSS "xscrnsaver"
-
-check_pkg_config "X extensions" $_xext XEXT "xext"
-
-check_pkg_config "Xv" $_xv XV "xv"
-
-check_pkg_config "VDPAU" $_vdpau VDPAU "vdpau >= 0.2"
-_vdpau=$(defretval)
-define_yes_no $_vdpau HAVE_VDPAU_HWACCEL
-
-check_pkg_config "VAAPI" $_vaapi VAAPI 'libva >= 0.32.0 libva-x11 >= 0.32.0'
-_vaapi=$(defretval)
-define_yes_no $_vaapi HAVE_VAAPI_HWACCEL
-
-if test "$_vaapi" = yes ; then
-    check_pkg_config "VAAPI VPP" auto VAAPI_VPP 'libva >= 0.34.0'
-else
-    check_yes_no no VAAPI_VPP
-fi
-
-check_pkg_config "Xinerama" $_xinerama XINERAMA 'xinerama'
-
-check_pkg_config "Xrandr" $_xrandr XRANDR 'xrandr >= 1.2.0'
-
-check_pkg_config "CACA" $_caca CACA 'caca >= 0.99.beta18'
-
-check_compile "DVB" $_dvb DVB waftools/fragments/dvb.c
-_dvbin=$(defretval)
-check_yes_no $_dvbin DVBIN
-
-check_statement_libs "JPEG support" $_jpeg JPEG "stdio.h jpeglib.h" "" "-ljpeg $_ld_lm"
-
-_gl_x11_egl=no
-(test "$_x11" = no && test "$_wayland" = no) && _gl=no
-echocheck "OpenGL"
-#Note: this test is run even with --enable-gl since we autodetect linker flags
-if test "$_gl" != no ; then
-  cat > $TMPC << EOF
-#if defined(GL_WAYLAND) || defined(EGL_X11)
-#include <EGL/egl.h>
-#else
-#include <X11/Xlib.h>
-#include <GL/glx.h>
-#endif
-#include <GL/gl.h>
-#include <GL/glext.h>
-int main(int argc, char *argv[]) {
-#if defined(GL_WAYLAND)
-  eglCreateContext(NULL, NULL, EGL_NO_CONTEXT, NULL);
-#else
-  glXCreateContext(NULL, NULL, NULL, True);
-#endif
-  glFinish();
-  return !GL_INVALID_FRAMEBUFFER_OPERATION; // check correct glext.h
-}
-EOF
-  _gl=no
-  if test "$_x11" = yes ; then
-    for _ld_tmp in "" -lGL "-lGL -lXdamage" "-lGL $_ld_pthread" ; do
-      if compile_check $TMPC $_ld_tmp $_ld_lm ; then
-        _gl=yes
-        _gl_x11=yes
-        libs_mplayer="$libs_mplayer $_ld_tmp $_ld_dl"
-        test "$_gl_x11" = yes && res_comment="$res_comment x11"
-        break
-      fi
-    done
-  fi
-  if test "$_wayland" = yes && compile_check $TMPC -DGL_WAYLAND -lGL -lEGL &&
-      pkg_config_add "wayland-egl >= 9.0.0"; then
-    _gl=yes
-    _gl_wayland=yes
-    libs_mplayer="$libs_mplayer -lGL -lEGL"
-    test "$_gl_wayland" = yes && res_comment="$res_comment wayland"
-  else
-    _gl_wayland=no
-  fi
-  if test "$_x11" = yes && test "$_gl" = yes && pkg_config_add "egl"; then
-    _gl_x11_egl=yes
-    res_comment="$res_comment x11egl"
-  fi
-else
-  _gl=no
-fi
-
-if test "$_gl" = no ; then
-    _gl_x11=no
-    _gl_wayland=no
-    _gl_x11_egl=no
-fi
-check_yes_no $_gl GL
-check_yes_no $_gl_x11 GL_X11
-check_yes_no $_gl_x11_egl EGL_X11
-check_yes_no $_gl_wayland GL_WAYLAND
-echores "$_gl"
-
-echocheck "VDPAU with OpenGL/X11"
-_vdpau_gl_x11=no
-(test "$_gl_x11" = yes && test "$_vdpau" = yes) && _vdpau_gl_x11=yes
-check_yes_no $_vdpau_gl_x11 VDPAU_GL_X11
-echores "$_vdpau_gl_x11"
-
-_vaapi_glx=no
-(test "$_gl_x11" = yes && test "$_vaapi" = yes) && _vaapi_glx=auto
-check_pkg_config "VAAPI with OpenGL/X11" $_vaapi_glx VAAPI_GLX 'libva-glx >= 0.32.0'
-
-check_pkg_config "SDL 2.0" $_sdl2 SDL2 'sdl2'
-
-check_statement_libs "OSS Audio" $_ossaudio OSS_AUDIO $_soundcard_header "int x = SNDCTL_DSP_SETFRAGMENT;"
-
-check_statement_libs "RSound" $_rsound RSOUND rsound.h 'rsd_init(NULL);' -lrsound
-
-check_statement_libs "sndio" $_sndio SNDIO sndio.h 'struct sio_par par; sio_initpar(&par); const char *s = SIO_DEVANY' -lsndio
-
-check_pkg_config "PulseAudio" $_pulse PULSE 'libpulse >= 1.0'
-
-check_pkg_config "JACK" $_jack JACK 'jack'
-
-check_pkg_config "OpenAL" $_openal OPENAL 'openal >= 1.13'
-
-check_pkg_config "ALSA audio" $_alsa ALSA 'alsa >= 1.0.9'
-
-check_pkg_config "Blu-ray support" $_bluray LIBBLURAY 'libbluray >= 0.3.0'
-
-check_pkg_config "dvdread" $_dvdread DVDREAD 'dvdread >= 4.1.0'
-
-check_pkg_config "dvdnav" $_dvdnav DVDNAV 'dvdnav >= 4.2.0'
-
-check_pkg_config "libcdio" $_libcdio CDDA 'libcdio_paranoia'
-
-check_pkg_config "rubberband" $_librubberband RUBBERBAND 'rubberband'
-
-_oldass=$_libass
-check_pkg_config "SSA/ASS support" $_libass LIBASS 'libass'
-_libass=$(defretval)
-if test $_oldass != no && test $_libass = no ; then
-    die "Unable to find development files for libass. Aborting. If you really mean to compile without libass support use --disable-libass."
-fi
-
-_dummy_osd=yes
-test $_libass = yes && _dummy_osd=no
-echo "LIBASS_OSD = $_libass" >> $CONFIG_MAK
-echo "DUMMY_OSD = $_dummy_osd" >> $CONFIG_MAK
-
-check_pkg_config "ENCA" $_enca ENCA 'enca'
-check_pkg_config "uchardet" $_uchardet UCHARDET 'uchardet'
-
-check_pkg_config "zlib" auto ZLIB 'zlib'
-test $(defretval) = no && die "Unable to find development files for zlib."
-
-check_pkg_config "LCMS2 support" $_lcms2 LCMS2 'lcms2 >= 2.6'
-
-check_pkg_config "FFmpeg/Libav" $_ffmpeg FFMPEG \
-    "libavutil >= 54.02.0 libavcodec >= 56.1.0 libavformat >= 56.01.0 libswscale >= 2.1.3"
-test $(defretval) = no && die "Unable to find development files for some of the required Libav libraries above. Aborting."
-
-check_pkg_config "Libswresample" $_libswresample LIBSWRESAMPLE 'libswresample >= 1.1.100'
-_libswresample=$(defretval)
-
-_libavresample=auto
-test $_libswresample = yes && _libavresample=no
-check_pkg_config "Libavresample" $_libavresample LIBAVRESAMPLE 'libavresample >= 2.1.0'
-_libavresample=$(defretval)
-
-if test "$_libswresample" = no && test "$_libavresample" = no ; then
-  die "No resampler found. Install libavresample or libswresample (FFmpeg)."
-fi
-
-# Arguments: "message"($1) "define"($2) "header"($3) "code"($4)
-api_statement_check() {
-    echocheck "$1"
-    _res=no
-    statement_check "$3" "$4" && _res=yes
-    define_yes_no $_res "$2"
-    echores "$_res"
-}
-
-api_statement_check \
-    "libavcodec avcodec_enum_to_chroma_pos API" \
-    HAVE_AVCODEC_CHROMA_POS_API \
-    libavcodec/avcodec.h \
-    'int x, y; avcodec_enum_to_chroma_pos(&x, &y, AVCHROMA_LOC_UNSPECIFIED)'
-
-api_statement_check \
-    "libavutil AVFrame metadata" \
-    HAVE_AVFRAME_METADATA \
-    libavutil/frame.h \
-    'av_frame_get_metadata(NULL)'
-
-api_statement_check \
-    "libavutil AVFrame skip samples metadata" \
-    HAVE_AVFRAME_SKIP_SAMPLES \
-    libavutil/frame.h \
-    'enum AVFrameSideDataType type = AV_FRAME_DATA_SKIP_SAMPLES'
-
-api_statement_check \
-    "libavutil av_version_info()" \
-    HAVE_AV_VERSION_INFO \
-    libavutil/avutil.h \
-    'const char *x = av_version_info()'
-
-api_statement_check \
-    "libavutil new pixdesc fields" \
-    HAVE_AV_NEW_PIXDESC \
-    libavutil/pixdesc.h \
-    'AVComponentDescriptor d; int x = d.depth'
-
-check_pkg_config "libavfilter" $_libavfilter LIBAVFILTER 'libavfilter >= 5.0.0'
-
-check_pkg_config "libavdevice" $_libavdevice LIBAVDEVICE 'libavdevice >= 55.0.0'
-
-check_trivial "TV interface" $_tv TV
-
-check_statement_libs "Video 4 Linux 2 TV interface" $_tv_v4l2 TV_V4L2 \
-    "sys/time.h linux/videodev2.h"
-_tv_v4l2=$(defretval)
-check_trivial "TV audio input" $_tv_v4l2 AUDIO_INPUT
-
-test $_tv_v4l2 = no && _libv4l2=no
-check_pkg_config "libv4l2 support" $_libv4l2 LIBV4L2 'libv4l2'
-
-test "$_tv_v4l2" = no && _pvr=no
-check_compile "Video 4 Linux 2 MPEG PVR interface" $_pvr PVR waftools/fragments/pvr.c
-
-# Note: Lua has no official .pc file, so there are different OS-specific ones.
-# Also, we support luajit, which is compatible to 5.1.
-
-test_lua() {
-    if test "$_lua" = auto && $_pkg_config "$1" ; then
-        check_pkg_config "Lua ($1)" $_lua LUA "$1"
-        _lua=$(defretval)
-    fi
-}
-
-test_lua "lua >= 5.1.0 lua < 5.2.0"
-test_lua "lua5.1 >= 5.1.0" # debian
-test_lua "lua51 >= 5.1.0" # OpenBSD
-test_lua "luajit >= 2.0.0"
-test_lua "lua5.2 >= 5.2.0" # debian
-test_lua "lua52 >= 5.2.0" # OpenBSD
-test_lua "lua >= 5.2.0"
-
-test "$_lua" != yes && check_yes_no no LUA
-
-if ! ( $_pkg_config 'vapoursynth >= 24' ) ; then
-    _vapoursynth=no
-    _vapoursynth_lazy=no
-fi
-check_pkg_config "VapourSynth support (Python)" $_vapoursynth VAPOURSYNTH 'vapoursynth >= 23 vapoursynth-script >= 23'
-_vapoursynth=$(defretval)
-if test "$_lua" = no ; then
-    _vapoursynth_lazy=no
-fi
-check_pkg_config "VapourSynth support (Lua)" $_vapoursynth_lazy VAPOURSYNTH_LAZY 'vapoursynth >= 23'
-_vapoursynth_lazy=$(defretval)
-
-_vapoursynth_core=yes
-if test "$_vapoursynth" = no && test "$_vapoursynth_lazy" = no ; then
-    _vapoursynth_core=no
-fi
-check_trivial "VapourSynth core" $_vapoursynth_core VAPOURSYNTH_CORE
-
-check_pkg_config "libarchive support" $_libarchive LIBARCHIVE 'libarchive >= 3.0.0'
-
-check_trivial "encoding" $_encoding ENCODING
-
-# needs dlopen on unix
-_dlopen="$_dl"
-check_yes_no $_dlopen DLOPEN
-
-extra_ldflags="$extra_ldflags $_ld_pthread"
-libs_mplayer="$libs_mplayer $_ld_dl"
-
-CFLAGS="$CFLAGS -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE64_SOURCE"
-
-# This is done so waf builds won't conflict with this. In fact, waf and old
-# build system can coexist in parallel, at the same time. This is because
-# waf always does out-of-tree builds, while this build system does always
-# in-tree builds.
-
-if test ! -f Makefile ; then
-    ln -s TOOLS/old-makefile Makefile
-fi
-
-cat > old_build/config.mak << EOF
-# -------- Generated by configure -----------
-export LC_ALL = C
-
-CONFIGURATION = $configuration
-
-prefix  = \$(DESTDIR)$_prefix
-BINDIR  = \$(DESTDIR)$_bindir
-MANDIR  = \$(DESTDIR)$_mandir
-CONFDIR = \$(DESTDIR)$_confdir
-
-CC      = $_cc
-INSTALL = install
-
-CFLAGS   = -Iold_build $OURCFLAGS $CFLAGS $extra_cflags
-DEPFLAGS = $DEPFLAGS
-
-EXTRALIBS = $extra_ldflags $_ld_static $_ld_lm $libs_mplayer $end_ldflags
-
-RST2MAN = $_rst2man
-BUILD_MAN = $_build_man
-
-# features
-EOF
-cat $CONFIG_MAK >> old_build/config.mak
-
-cat > $TMPC << EOF
-/*----------------------------------------------------------------------------
-** This file has been automatically generated by configure any changes in it
-** will be lost when you run configure again.
-** Instead of modifying definitions here, use the --enable/--disable options
-** of the configure script! See ./configure --help for details.
-*---------------------------------------------------------------------------*/
-
-#ifndef MPV_CONFIG_H
-#define MPV_CONFIG_H
-
-#define CONFIGURATION "$configuration"
-
-#define MPV_CONFDIR "$_confdir"
-
-/* we didn't bother to add actual config checks for this, or they are
-   for platforms not supported by this configure script */
-#define HAVE_BSD_FSTATFS 0
-#define HAVE_LINUX_FSTATFS 1
-#define HAVE_VDA_AV_VDA_ALLOC_CONTEXT 0
-#define HAVE_VDA_HWACCEL 0
-#define HAVE_VDA_LIBAVCODEC_REFCOUNTING 0
-#define HAVE_VDA_GL 0
-#define HAVE_GL_COCOA 0
-#define HAVE_COCOA 0
-#define HAVE_COCOA_APPLICATION 0
-#define HAVE_COREVIDEO 0
-#define HAVE_COREAUDIO 0
-#define HAVE_GL_WIN32 0
-#define HAVE_DIRECT3D 0
-#define HAVE_DSOUND 0
-#define HAVE_WASAPI 0
-#define HAVE_DOS_PATHS 0
-#define HAVE_PRIORITY 0
-#define HAVE_GLOB 1
-#define HAVE_NANOSLEEP 1
-#define HAVE_SDL1 0
-#define HAVE_WAIO 0
-#define HAVE_POSIX_SPAWN 1
-#define HAVE_GLIBC_THREAD_NAME (!!__GLIBC__)
-#define HAVE_OSX_THREAD_NAME 0
-#define HAVE_BSD_THREAD_NAME 0
-#define HAVE_NETBSD_THREAD_NAME 0
-#define HAVE_DXVA2_HWACCEL 0
-#define HAVE_FCHMOD 0
-#define HAVE_RPI 0
-#define HAVE_RPI_GLES 0
-#define HAVE_AV_PIX_FMT_MMAL 0
-#define HAVE_DRM 0
-#define HAVE_VIDEOTOOLBOX_HWACCEL 0
-#define HAVE_VIDEOTOOLBOX_GL 0
-#define HAVE_VIDEOTOOLBOX_VDA_GL 0
-
-#ifdef __OpenBSD__
-#define DEFAULT_CDROM_DEVICE "/dev/rcd0c"
-#define DEFAULT_DVD_DEVICE   "/dev/rcd0c"
-#else
-#define DEFAULT_CDROM_DEVICE "/dev/cdrom"
-#define DEFAULT_DVD_DEVICE   "/dev/dvd"
-#endif
-#define PATH_DEV_DSP "/dev/dsp"
-#define PATH_DEV_MIXER "/dev/mixer"
-
-EOF
-cat $CONFIG_H >> $TMPC
-echo '#endif /* MPV_CONFIG_H */' >> $TMPC
-
-# Do not overwrite an unchanged config.h to avoid superfluous rebuilds.
-cmp -s "$TMPC" old_build/config.h || mv -f "$TMPC" old_build/config.h
-
-cat <<EOF
-
-Configuration successful. See $TMPLOG for details.
-
-NOTE: The --enable-* parameters unconditionally force options on, completely
-skipping autodetection. This behavior is unlike what you may be used to from
-autoconf-based configure scripts that can decide to override you. This greater
-level of control comes at a price. You may have to provide the correct compiler
-and linker flags yourself.
-If you used one of these options and experience a compilation or
-linking failure, make sure you have passed the necessary compiler/linker flags
-to configure.
-
-WARNING: The ./old-configure + make build system you are using is deprecated in
-favour of waf and will be removed in a future version of mpv. Check the
-README for instructions on how to build mpv with the new build system.
-This will not work correctly on MinGW, Cygwin, OSX, and other systems.
-
-EOF

TOOLS/old-makefile

@@ -1,483 +0,0 @@
-# MPlayer Makefile
-#
-# copyright (c) 2008 Diego Biurrun
-# Rewritten entirely from a set of Makefiles written by Arpi and many others.
-#
-# This file is part of MPlayer.
-#
-# MPlayer is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 2 of the License, or
-# (at your option) any later version.
-#
-# MPlayer is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-# GNU General Public License for more details.
-#
-# You should have received a copy of the GNU General Public License along
-# with MPlayer; if not, write to the Free Software Foundation, Inc.,
-# 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
-
-include old_build/config.mak
-
-###### variable declarations #######
-
-SOURCES_AUDIO_INPUT-$(ALSA)     += stream/ai_alsa1x.c
-SOURCES_AUDIO_INPUT-$(OSS_AUDIO)+= stream/ai_oss.c
-SOURCES_AUDIO_INPUT-$(SNDIO)    += stream/ai_sndio.c
-SOURCES-$(AUDIO_INPUT)          += $(SOURCES_AUDIO_INPUT-yes)
-SOURCES-$(CDDA)                 += stream/stream_cdda.c
-SOURCES-$(DVBIN)                += stream/dvb_tune.c \
-                                   stream/stream_dvb.c
-SOURCES-$(DVDREAD)              += stream/stream_dvd.c \
-                                   stream/stream_dvd_common.c
-SOURCES-$(DVDNAV)               += stream/stream_dvdnav.c \
-                                   stream/stream_dvd_common.c
-
-SOURCES-$(RUBBERBAND)           += audio/filter/af_rubberband.c
-SOURCES-$(LIBASS)               += sub/ass_mp.c sub/sd_ass.c \
-                                   demux/demux_libass.c
-
-SOURCES-$(LIBBLURAY)            += stream/stream_bluray.c
-
-SOURCES-$(LIBSMBCLIENT)         += stream/stream_smb.c
-
-SOURCES-$(PVR)                  += stream/stream_pvr.c
-
-SOURCES-$(TV)                   += stream/stream_tv.c stream/tv.c \
-                                   stream/frequencies.c stream/tvi_dummy.c \
-                                   demux/demux_tv.c
-
-SOURCES-$(TV_V4L2)              += stream/tvi_v4l2.c stream/audio_in.c
-SOURCES-$(DUMMY_OSD)            += sub/osd_dummy.c
-SOURCES-$(LIBASS_OSD)           += sub/osd_libass.c
-
-SOURCES-$(ALSA)                 += audio/out/ao_alsa.c
-SOURCES-$(CACA)                 += video/out/vo_caca.c
-SOURCES-$(SDL2)                 += audio/out/ao_sdl.c video/out/vo_sdl.c
-SOURCES-$(GL)                   += video/out/opengl/common.c \
-                                   video/out/opengl/osd.c \
-                                   video/out/opengl/lcms.c \
-                                   video/out/opengl/video.c \
-                                   video/out/opengl/video_shaders.c \
-                                   video/out/dither.c \
-                                   video/out/opengl/hwdec.c \
-                                   video/out/opengl/utils.c \
-                                   video/out/vo_opengl.c \
-                                   video/out/vo_opengl_cb.c
-
-SOURCES-$(ENCODING)             += video/out/vo_lavc.c audio/out/ao_lavc.c \
-                                   common/encode_lavc.c
-
-SOURCES-$(GL_X11)               += video/out/x11_common.c video/out/opengl/x11.c
-SOURCES-$(EGL_X11)              += video/out/x11_common.c video/out/opengl/x11egl.c
-SOURCES-$(GL_WAYLAND)           += video/out/wayland_common.c \
-                                   video/out/opengl/wayland.c
-
-SOURCES-$(JACK)                 += audio/out/ao_jack.c
-SOURCES-$(OPENAL)               += audio/out/ao_openal.c
-SOURCES-$(OSS_AUDIO)            += audio/out/ao_oss.c
-SOURCES-$(PULSE)                += audio/out/ao_pulse.c
-SOURCES-$(RSOUND)               += audio/out/ao_rsound.c
-SOURCES-$(SNDIO)                += audio/out/ao_sndio.c
-SOURCES-$(VDPAU)                += video/vdpau.c video/vdpau_mixer.c \
-                                   video/out/vo_vdpau.c video/decode/vdpau.c \
-                                   video/filter/vf_vdpaupp.c \
-                                   video/filter/vf_vdpaurb.c
-SOURCES-$(VDPAU_GL_X11)         += video/out/opengl/hwdec_vdpau.c
-SOURCES-$(VAAPI)                += video/out/vo_vaapi.c \
-                                   video/decode/vaapi.c \
-                                   video/vaapi.c
-SOURCES-$(VAAPI_VPP)            += video/filter/vf_vavpp.c
-SOURCES-$(VAAPI_GLX)            += video/out/opengl/hwdec_vaglx.c
-
-SOURCES-$(XV)                   += video/out/x11_common.c video/out/vo_xv.c
-SOURCES-$(WAYLAND)              += video/out/vo_wayland.c \
-                                   video/out/wayland_common.c \
-                                   video/out/wayland/buffer.c \
-                                   video/out/wayland/memfile.c
-
-SOURCES-$(LIBAVFILTER)          += video/filter/vf_lavfi.c \
-                                   video/filter/vf_gradfun.c \
-                                   video/filter/vf_pullup.c \
-                                   video/filter/vf_rotate.c \
-                                   video/filter/vf_yadif.c \
-                                   audio/filter/af_lavfi.c
-
-SOURCES-$(LUA)                  += player/lua.c
-SOURCES-$(VAPOURSYNTH_CORE)     += video/filter/vf_vapoursynth.c
-SOURCES-$(LIBARCHIVE)           += demux/demux_libarchive.c \
-                                   stream/stream_libarchive.c
-SOURCES-$(DLOPEN)               += video/filter/vf_dlopen.c
-
-SOURCES = audio/audio.c \
-          audio/audio_buffer.c \
-          audio/chmap.c \
-          audio/chmap_sel.c \
-          audio/fmt-conversion.c \
-          audio/format.c \
-          audio/mixer.c \
-          audio/decode/ad_lavc.c \
-          audio/decode/ad_spdif.c      \
-          audio/decode/dec_audio.c \
-          audio/filter/af.c \
-          audio/filter/af_channels.c \
-          audio/filter/af_delay.c \
-          audio/filter/af_equalizer.c \
-          audio/filter/af_format.c \
-          audio/filter/af_lavcac3enc.c \
-          audio/filter/af_lavrresample.c \
-          audio/filter/af_pan.c \
-          audio/filter/af_scaletempo.c \
-          audio/filter/af_drc.c \
-          audio/filter/af_volume.c \
-          audio/filter/tools.c \
-          audio/out/ao.c \
-          audio/out/ao_null.c \
-          audio/out/ao_pcm.c \
-          audio/out/pull.c \
-          audio/out/push.c \
-          common/av_common.c \
-          common/av_log.c \
-          common/codecs.c \
-          common/common.c \
-          common/msg.c \
-          common/playlist.c \
-          common/tags.c \
-          common/version.c \
-          demux/codec_tags.c \
-          demux/cue.c \
-          demux/demux.c \
-          demux/demux_edl.c \
-          demux/demux_cue.c \
-          demux/demux_disc.c \
-          demux/demux_lavf.c \
-          demux/demux_mf.c \
-          demux/demux_mkv.c \
-          demux/demux_mkv_timeline.c \
-          demux/demux_playlist.c \
-          demux/demux_rar.c \
-          demux/demux_raw.c \
-          demux/demux_subreader.c \
-          demux/ebml.c \
-          demux/packet.c \
-          demux/timeline.c \
-          input/cmd_list.c \
-          input/cmd_parse.c \
-          input/event.c \
-          input/input.c \
-          input/ipc.c \
-          input/keycodes.c \
-          misc/bstr.c \
-          misc/charset_conv.c \
-          misc/dispatch.c \
-          misc/json.c \
-          misc/rendezvous.c \
-          misc/ring.c \
-          options/m_config.c \
-          options/m_option.c \
-          options/m_property.c \
-          options/options.c \
-          options/parse_commandline.c \
-          options/parse_configfile.c \
-          options/path.c \
-          osdep/io.c \
-          osdep/path-unix.c \
-          osdep/semaphore_osx.c \
-          osdep/subprocess.c \
-          osdep/subprocess-posix.c \
-          osdep/terminal-unix.c \
-          osdep/timer.c \
-          osdep/timer-linux.c \
-          osdep/threads.c \
-          player/audio.c \
-          player/client.c \
-          player/configfiles.c \
-          player/command.c \
-          player/external_files.c \
-          player/loadfile.c \
-          player/main.c \
-          player/misc.c \
-          player/osd.c \
-          player/playloop.c \
-          player/screenshot.c \
-          player/scripting.c \
-          player/sub.c \
-          player/video.c \
-          stream/cache.c \
-          stream/cache_file.c \
-          stream/cookies.c \
-          stream/rar.c \
-          stream/stream.c \
-          stream/stream_avdevice.c \
-          stream/stream_edl.c \
-          stream/stream_file.c \
-          stream/stream_lavf.c \
-          stream/stream_memory.c \
-          stream/stream_mf.c \
-          stream/stream_null.c \
-          stream/stream_rar.c \
-          sub/dec_sub.c \
-          sub/draw_bmp.c \
-          sub/img_convert.c \
-          sub/osd.c \
-          sub/sd_lavc.c \
-          sub/sd_lavc_conv.c \
-          sub/sd_lavf_srt.c \
-          sub/sd_microdvd.c \
-          sub/sd_movtext.c \
-          sub/sd_srt.c \
-          ta/ta.c \
-          ta/ta_utils.c \
-          ta/ta_talloc.c \
-          video/csputils.c \
-          video/fmt-conversion.c \
-          video/image_writer.c \
-          video/img_format.c \
-          video/mp_image.c \
-          video/mp_image_pool.c \
-          video/sws_utils.c \
-          video/decode/dec_video.c \
-          video/decode/vd_lavc.c \
-          video/filter/vf.c \
-          video/filter/vf_buffer.c \
-          video/filter/vf_crop.c \
-          video/filter/vf_dsize.c \
-          video/filter/vf_eq.c \
-          video/filter/vf_expand.c \
-          video/filter/vf_flip.c \
-          video/filter/vf_format.c \
-          video/filter/vf_mirror.c \
-          video/filter/vf_noformat.c \
-          video/filter/vf_scale.c \
-          video/filter/vf_stereo3d.c \
-          video/filter/vf_sub.c \
-          video/out/bitmap_packer.c \
-          video/out/aspect.c \
-          video/out/filter_kernels.c \
-          video/out/vo.c \
-          video/out/vo_null.c \
-          video/out/vo_image.c \
-          video/out/win_state.c \
-          $(SOURCES-yes)
-
-OBJECTS         += $(addsuffix .o, $(basename $(SOURCES)))
-OBJECTS         += $(OBJECTS-yes)
-
-DEP_FILES = $(patsubst %.S,%.d,$(patsubst %.cpp,%.d,$(patsubst %.c,%.d,$(SOURCES:.m=.d) $(SOURCES:.m=.d))))
-
-ALL_TARGETS     += mpv
-
-INSTALL_BIN     += install-mpv
-INSTALL_BIN_STRIP += install-mpv-strip
-INSTALL_MAN      =
-
-ifeq ($(BUILD_MAN),yes)
-    INSTALL_MAN += install-mpv-man
-    ALL_TARGETS += DOCS/man/mpv.1
-endif
-
-DIRS =  . \
-        audio \
-        audio/decode \
-        audio/filter \
-        audio/out \
-        common \
-        compat \
-        input \
-        player/timeline \
-        demux \
-        misc \
-        options \
-        osdep \
-        osdep/ar \
-        player \
-        stream \
-        sub \
-        ta \
-        video \
-        video/decode \
-        video/filter \
-        video/out
-
-
-ADDSUFFIXES     = $(foreach suf,$(1),$(addsuffix $(suf),$(2)))
-ADD_ALL_DIRS    = $(call ADDSUFFIXES,$(1),$(DIRS))
-
-###### brief build output #######
-
-ifndef V
-$(eval override CC = @printf "CC\t$$@\n"; $(CC))
-$(eval override RM = @$(RM))
-endif
-
-###### generic rules #######
-
-all: $(ALL_TARGETS)
-
-%.1: %.rst
-	$(RST2MAN) $< $@
-
-%.o: %.c
-	$(CC) $(DEPFLAGS) $(CFLAGS) -c -o $@ $<
-
-mpv: $(OBJECTS) osdep/main-fn-unix.o
-	$(CC) -o $@ $^ $(EXTRALIBS)
-
-input/input.c: input/input_conf.h
-input/input_conf.h: TOOLS/file2string.pl etc/input.conf
-	./$^ >$@
-
-MKVLIB_DEPS = TOOLS/lib/Parse/Matroska.pm \
-              TOOLS/lib/Parse/Matroska/Definitions.pm \
-              TOOLS/lib/Parse/Matroska/Element.pm \
-              TOOLS/lib/Parse/Matroska/Reader.pm \
-              TOOLS/lib/Parse/Matroska/Utils.pm \
-
-demux/ebml.c demux/demux_mkv.c: demux/ebml_types.h
-demux/ebml_types.h: TOOLS/matroska.pl $(MKVLIB_DEPS)
-	./$< --generate-header > $@
-
-demux/ebml.c: demux/ebml_defs.c
-demux/ebml_defs.c: TOOLS/matroska.pl $(MKVLIB_DEPS)
-	./$< --generate-definitions > $@
-
-video/out/x11_common.c: video/out/x11_icon.inc
-video/out/x11_icon.inc: TOOLS/file2string.pl video/out/x11_icon.bin
-	./$^ >$@
-
-sub/osd_libass.c: sub/osd_font.h
-sub/osd_font.h: TOOLS/file2string.pl sub/osd_font.otf
-	./$^ >$@
-
-player/lua/%.inc: TOOLS/file2string.pl player/lua/%.lua
-	./$^ >$@
-
-player/lua.c: player/lua/defaults.inc \
-              player/lua/assdraw.inc \
-              player/lua/osc.inc \
-              player/lua/ytdl_hook.inc \
-              player/lua/options.inc
-
-etc/_mpv: TOOLS/zsh.pl ./mpv
-	./$< > $@
-
-# ./configure must be rerun if it changed
-config.mak: configure
-	@echo "############################################################"
-	@echo "####### Please run ./configure again - it's changed! #######"
-	@echo "############################################################"
-
-old_build/version.h .version: version.sh
-	./version.sh --versionh=old_build/version.h
-
-# Force version.sh to run to potentially regenerate version.h
--include .version
-
-%: %.c
-	$(CC) $(CFLAGS) -o $@ $^
-
-
-###### dependency declarations / specific CFLAGS ######
-
-common/version.c: old_build/version.h
-
-DOCS/man/mpv.1: DOCS/man/af.rst \
-                   DOCS/man/ao.rst \
-                   DOCS/man/changes.rst \
-                   DOCS/man/encode.rst \
-                   DOCS/man/input.rst \
-                   DOCS/man/options.rst \
-                   DOCS/man/vf.rst \
-                   DOCS/man/vo.rst
-
-###### installation / clean / generic rules #######
-
-install:               $(INSTALL_BIN)       install-data $(INSTALL_MAN)
-install-no-man:        $(INSTALL_BIN)       install-data
-install-strip:         $(INSTALL_BIN_STRIP) install-data $(INSTALL_MAN)
-install-strip-no-man:  $(INSTALL_BIN_STRIP) install-data
-
-install-dirs:
-	if test ! -d $(BINDIR) ; then $(INSTALL) -d $(BINDIR) ; fi
-
-install-%: % install-dirs
-	$(INSTALL) -m 755 $< $(BINDIR)
-
-install-%-strip: % install-dirs
-	$(INSTALL) -m 755 -s $< $(BINDIR)
-
-install-mpv-man:  install-mpv-man-en
-
-install-mpv-man-en: DOCS/man/mpv.1
-	if test ! -d $(MANDIR)/man1 ; then $(INSTALL) -d $(MANDIR)/man1 ; fi
-	$(INSTALL) -m 644 DOCS/man/mpv.1 $(MANDIR)/man1/
-
-ICONSIZES = 16x16 32x32 64x64
-
-define ICON_INSTALL_RULE
-install-mpv-icon-$(size): etc/mpv-icon-8bit-$(size).png
-	$(INSTALL) -d $(prefix)/share/icons/hicolor/$(size)/apps
-	$(INSTALL) -m 644 etc/mpv-icon-8bit-$(size).png $(prefix)/share/icons/hicolor/$(size)/apps/mpv.png
-endef
-
-$(foreach size,$(ICONSIZES),$(eval $(ICON_INSTALL_RULE)))
-
-install-mpv-icons: $(foreach size,$(ICONSIZES),install-mpv-icon-$(size))
-
-install-mpv-desktop: etc/mpv.desktop
-	$(INSTALL) -d $(prefix)/share/applications
-	$(INSTALL) -m 644 etc/mpv.desktop $(prefix)/share/applications/
-
-install-mpv-config: etc/encoding-profiles.conf
-	$(INSTALL) -d $(CONFDIR)
-	$(INSTALL) -m 644 etc/encoding-profiles.conf $(CONFDIR)
-
-install-mpv-zsh: etc/_mpv
-	$(INSTALL) -d $(prefix)/share/zsh/vendor-completions
-	$(INSTALL) -m 644 etc/_mpv $(prefix)/share/zsh/vendor-completions/
-
-install-data: install-mpv-icons install-mpv-desktop install-mpv-config install-mpv-zsh
-
-uninstall:
-	$(RM) $(BINDIR)/mpv
-	$(RM) $(MANDIR)/man1/mpv.1 $(MANDIR)/man1/mpv.1
-	$(RM) $(prefix)/share/applications/mpv.desktop
-	$(RM) $(prefix)/share/zsh/vendor-completions/_mpv
-	$(RM) $(foreach size,$(ICONSIZES),$(prefix)/share/icons/hicolor/$(size)/apps/mpv.png)
-
-clean:
-	-$(RM) $(call ADD_ALL_DIRS,/*.o /*.d /*.a /*.ho /*~)
-	-$(RM) $(call ADD_ALL_DIRS,/*.o /*.a /*.ho /*~)
-	-$(RM) mpv
-	-$(RM) DOCS/man/*/mpv.1
-	-$(RM) old_build/version.h version.h
-	-$(RM) input/input_conf.h
-	-$(RM) video/out/vdpau_template.c
-	-$(RM) demux/ebml_types.h demux/ebml_defs.c
-	-$(RM) video/out/x11_icon.inc
-	-$(RM) sub/osd_font.h
-	-$(RM) player/lua/defaults.inc
-	-$(RM) player/lua/assdraw.inc
-	-$(RM) player/lua/osc.inc
-	-$(RM) player/lua/ytdl_hook.inc
-	-$(RM) player/lua/options.inc
-
-distclean: clean
-	-$(RM) config.log old_build/config.h old_build/config.mak Makefile
-	-rmdir old_build/
-
--include $(DEP_FILES)
-
-.PHONY: all *install* *clean .version
-
-# Disable suffix rules.  Most of the builtin rules are suffix rules,
-# so this saves some time on slow systems.
-.SUFFIXES:
-
-# If a command returns failure but changed its target file, delete the
-# (presumably malformed) file. Otherwise the file would be considered to
-# be up to date if make is restarted.
-
-.DELETE_ON_ERROR:

TOOLS/osxbundle.py

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

TOOLS/stats-conv.py

@@ -1,5 +1,6 @@
 #!/usr/bin/env python3
-import matplotlib.pyplot as plot
+from pyqtgraph.Qt import QtGui, QtCore
+import pyqtgraph as pg
 import sys
 import re
 
@@ -38,8 +39,8 @@ Currently, the following event types are supported:
 class G:
     events = {}
     start = None
-    # http://matplotlib.org/api/markers_api.html#module-matplotlib.markers
-    markers = ["o", "8", "s", "p", "*", "h", "+", "x", "D"]
+    markers = ["o", "s", "t", "d"]
+    curveno = {}
 
 def find_marker():
     if len(G.markers) == 0:
@@ -65,6 +66,10 @@ def get_event(event, evtype):
         G.events[event] = e
     return G.events[event]
 
+colors = [(0.0, 0.5, 0.0), (0.0, 0.0, 1.0), (0.0, 0.0, 0.0), (1.0, 0.0, 0.0), (0.75, 0.75, 0), (0.0, 0.75, 0.75), (0.75, 0, 0.75)]
+def mkColor(t):
+    return pg.mkColor(int(t[0] * 255), int(t[1] * 255), int(t[2] * 255))
+
 SCALE = 1e6 # microseconds to seconds
 
 for line in [line.split("#")[0].strip() for line in open(filename, "r")]:
@@ -128,20 +133,33 @@ for e, index in zip(G.sevents, range(len(G.sevents))):
     else:
         e.vals = [(x, y * (m - index) / m) for (x, y) in e.vals]
 
-fig = plot.figure()
-fig.hold(True)
+pg.setConfigOption('background', 'w')
+pg.setConfigOption('foreground', 'k')
+app = QtGui.QApplication([])
+win = pg.GraphicsWindow()
+#win.resize(1500, 900)
+
 ax = [None, None]
 plots = 2 if hasval else 1
-ax[0] = fig.add_subplot(plots, 1, 1)
+ax[0] = win.addPlot()
 if hasval:
-    ax[1] = fig.add_subplot(plots, 1, 2, sharex=ax[0])
-legends = [[], []]
+    win.nextRow()
+    ax[1] = win.addPlot()
+    ax[1].setXLink(ax[0])
+for cur in ax:
+   if cur is not None:
+       cur.addLegend(offset = (-1, 1))
 for e in G.sevents:
     cur = ax[1 if e.type == "value" else 0]
-    pl, = cur.plot([x for x,y in e.vals], [y for x,y in e.vals], label=e.name)
+    args = {'name': e.name,'antialias':True}
     if e.type == "event-signal":
-        plot.setp(pl, marker = e.marker, linestyle = "None")
-for cur in ax:
-    if cur is not None:
-        cur.legend()
-plot.show()
+        args['symbol'] = e.marker
+        args['pen'] = None
+    else:
+        if not cur in G.curveno:
+            G.curveno[cur] = 0
+        args['pen'] = pg.mkPen(mkColor(colors[G.curveno[cur] % len(colors)]), width=0)
+        G.curveno[cur] += 1
+    n = cur.plot([x for x,y in e.vals], [y for x,y in e.vals], **args)
+
+QtGui.QApplication.instance().exec_()

TOOLS/zsh.pl

@@ -120,7 +120,7 @@ $profile_comp
     _tags files urls
     while _tags; do
       _requested files expl 'media file' _files -g \\
-        "*.(#i)(asf|asx|avi|flac|flv|m1v|m2p|m2v|m4v|mjpg|mka|mkv|mov|mp3|mp4|mpe|mpeg|mpg|ogg|ogm|ogv|qt|rm|ts|vob|wav|webm|wma|wmv)(-.)" && rc=0
+        "*.(#i)(asf|asx|avi|flac|flv|m1v|m2p|m2v|m4v|mjpg|mka|mkv|mov|mp3|mp4|mpe|mpeg|mpg|ogg|ogm|ogv|opus|qt|rm|ts|vob|wav|webm|wma|wmv)(-.)" && rc=0
       if _requested urls; then
         while _next_label urls expl URL; do
           _urls "\$expl[@]" && rc=0

VERSION

@@ -1 +1 @@
-0.11.0
+0.15.0

audio/audio.c

@@ -24,7 +24,7 @@
 #include <libavutil/frame.h>
 #include <libavutil/version.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 #include "common/common.h"
 #include "fmt-conversion.h"
 #include "audio.h"
@@ -323,21 +323,23 @@ struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe)
     }
 
     // If we can't handle the format (e.g. too many channels), bail out.
-    if (!mp_audio_config_valid(new) || avframe->nb_extended_buf)
+    if (!mp_audio_config_valid(new))
         goto fail;
 
-    for (int n = 0; n < AV_NUM_DATA_POINTERS; n++) {
-        if (!avframe->buf[n])
+    for (int n = 0; n < AV_NUM_DATA_POINTERS + avframe->nb_extended_buf; n++) {
+        AVBufferRef *buf = n < AV_NUM_DATA_POINTERS ? avframe->buf[n]
+                            : avframe->extended_buf[n - AV_NUM_DATA_POINTERS];
+        if (!buf)
             break;
         if (n >= MP_NUM_CHANNELS)
             goto fail;
-        new->allocated[n] = av_buffer_ref(avframe->buf[n]);
+        new->allocated[n] = av_buffer_ref(buf);
         if (!new->allocated[n])
             goto fail;
     }
 
     for (int n = 0; n < new->num_planes; n++)
-        new->planes[n] = avframe->data[n];
+        new->planes[n] = avframe->extended_data[n];
     new->samples = avframe->nb_samples;
 
     return new;

audio/audio.h

@@ -37,6 +37,8 @@ struct mp_audio {
     int num_planes;     // number of planes
     int bps;            // size of sub-samples (af_fmt_to_bytes(format))
 
+    double pts;         // currently invalid within the filter chain
+
     // --- private
     // These do not necessarily map directly to planes[]. They can have
     // different order or count. There shouldn't be more buffers than planes.

audio/audio_buffer.c

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

audio/audio_buffer.h

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

audio/chmap.c

@@ -93,7 +93,7 @@ static const char *const std_layout_names[][2] = {
     {"7.1(alsa)",       "fl-fr-bl-br-fc-lfe-sl-sr"}, // not in lavc
     {"7.1(wide)",       "fl-fr-fc-lfe-bl-br-flc-frc"},
     {"7.1(wide-side)",  "fl-fr-fc-lfe-flc-frc-sl-sr"},
-    {"7.1(rear)",       "fl-fr-fc-lfe-bl-br-sdl-sdr"},
+    {"7.1(rear)",       "fl-fr-fc-lfe-bl-br-sdl-sdr"}, // not in lavc
     {"octagonal",       "fl-fr-fc-bl-br-bc-sl-sr"},
     {"downmix",         "dl-dr"},
     {"auto",            ""}, // not in lavc
@@ -112,19 +112,6 @@ static const struct mp_chmap default_layouts[] = {
     MP_CHMAP8(FL, FR, FC, LFE, BL, BR, SL, SR), // 7.1
 };
 
-// The channel order was lavc/waveex, but differs from lavc for 5, 6 and 8
-// channels. 3 and 7 channels were likely undefined (no ALSA support).
-// I'm not sure about the 4 channel case: ALSA uses "quad", while the ffmpeg
-// default layout is "4.0".
-static const char *const mplayer_layouts[MP_NUM_CHANNELS + 1] = {
-    [1] = "mono",
-    [2] = "stereo",
-    [4] = "quad",
-    [5] = "5.0(alsa)",
-    [6] = "5.1(alsa)",
-    [8] = "7.1(alsa)",
-};
-
 // Returns true if speakers are mapped uniquely, and there's at least 1 channel.
 bool mp_chmap_is_valid(const struct mp_chmap *src)
 {
@@ -227,19 +214,6 @@ void mp_chmap_from_channels(struct mp_chmap *dst, int num_channels)
         mp_chmap_set_unknown(dst, num_channels);
 }
 
-// Try to do what mplayer/mplayer2/mpv did before channel layouts were
-// introduced, i.e. get the old default channel order.
-void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels)
-{
-    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
-        *dst = (struct mp_chmap) {0};
-    } else {
-        mp_chmap_from_str(dst, bstr0(mplayer_layouts[num_channels]));
-        if (!dst->num)
-            mp_chmap_from_channels(dst, num_channels);
-    }
-}
-
 // Set *dst to an unknown layout for the given numbers of channels.
 // If the number of channels is invalid, an invalid map is set, and
 // mp_chmap_is_valid(dst) will return false.
@@ -298,12 +272,18 @@ void mp_chmap_remove_useless_channels(struct mp_chmap *map,
 // Speakers not representable by ffmpeg/libav are dropped.
 // Warning: this ignores the order of the channels, and will return a channel
 //          mask even if the order is different from libavcodec's.
+//          Also, "unknown" channel maps are translated to non-sense channel
+//          maps with the same number of channels.
 uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src)
 {
-    // lavc has no concept for unknown layouts yet, so pick a default
     struct mp_chmap t = *src;
+    if (t.num > 64)
+        return 0;
+    // lavc has no concept for unknown layouts yet, so pick something that does
+    // the job of signaling the number of channels, even if it makes no sense
+    // as a proper layout.
     if (mp_chmap_is_unknown(&t))
-        mp_chmap_from_channels(&t, t.num);
+        return t.num == 64 ? (uint64_t)-1 : (1ULL << t.num) - 1;
     uint64_t mask = 0;
     for (int n = 0; n < t.num; n++) {
         if (t.speaker[n] < 64) // ignore MP_SPEAKER_ID_NA etc.
@@ -359,6 +339,8 @@ bool mp_chmap_is_lavc(const struct mp_chmap *src)
     return true;
 }
 
+// Warning: for "unknown" channel maps, this returns something that may not
+//          make sense. Invalid channel maps are not changed.
 void mp_chmap_reorder_to_lavc(struct mp_chmap *map)
 {
     if (!mp_chmap_is_valid(map))

audio/chmap.h

@@ -22,7 +22,7 @@
 #include <stdbool.h>
 #include "misc/bstr.h"
 
-#define MP_NUM_CHANNELS 8
+#define MP_NUM_CHANNELS 16
 
 // Speaker a channel can be assigned to.
 // This corresponds to WAVEFORMATEXTENSIBLE channel mask bit indexes.
@@ -108,7 +108,6 @@ void mp_chmap_fill_na(struct mp_chmap *map, int num);
 
 void mp_chmap_from_channels(struct mp_chmap *dst, int num_channels);
 void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels);
-void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels);
 
 void mp_chmap_remove_useless_channels(struct mp_chmap *map,
                                       const struct mp_chmap *requested);

audio/chmap_sel.c

@@ -20,6 +20,7 @@
 #include <limits.h>
 
 #include "common/common.h"
+#include "common/msg.h"
 #include "chmap_sel.h"
 
 static const struct mp_chmap speaker_replacements[][2] = {
@@ -94,17 +95,6 @@ void mp_chmap_sel_add_waveext(struct mp_chmap_sel *s)
     s->allow_waveext = true;
 }
 
-// Classic ALSA-based MPlayer layouts.
-void mp_chmap_sel_add_alsa_def(struct mp_chmap_sel *s)
-{
-    for (int n = 1; n <= MP_NUM_CHANNELS; n++) {
-        struct mp_chmap t;
-        mp_chmap_from_channels_alsa(&t, n);
-        if (t.num)
-            mp_chmap_sel_add_map(s, &t);
-    }
-}
-
 // Add a channel map that should be allowed.
 void mp_chmap_sel_add_map(struct mp_chmap_sel *s, const struct mp_chmap *map)
 {
@@ -284,19 +274,25 @@ static bool mp_chmap_is_better(struct mp_chmap *req, struct mp_chmap *old,
     if (new_lost_r != old_lost_r)
         return new_lost_r < old_lost_r;
 
-    int old_lost = mp_chmap_diffn(req, old);
-    int new_lost = mp_chmap_diffn(req, new);
-
-    // If the situation is equal with replaced speakers, but one of them loses
-    // less if no replacements are performed, pick the better one, even if it
-    // means an upmix. This prefers exact supersets over inexact equivalents.
-    if (new_lost != old_lost)
-        return new_lost < old_lost;
-
     struct mp_chmap old_p = *old, new_p = *new;
     mp_chmap_remove_na(&old_p);
     mp_chmap_remove_na(&new_p);
 
+    // If the situation is equal with replaced speakers, but the replacement is
+    // perfect for only one of them, let the better one win. This prefers
+    // inexact equivalents over exact supersets.
+    bool perfect_r_new = !new_lost_r && new_p.num <= old_p.num;
+    bool perfect_r_old = !old_lost_r && old_p.num <= new_p.num;
+    if (perfect_r_new != perfect_r_old)
+        return perfect_r_new;
+
+    int old_lost = mp_chmap_diffn(req, old);
+    int new_lost = mp_chmap_diffn(req, new);
+    // If the situation is equal with replaced speakers, pick the better one,
+    // even if it means an upmix.
+    if (new_lost != old_lost)
+        return new_lost < old_lost;
+
     // Some kind of upmix. If it's perfect, prefer the smaller one. Even if not,
     // both have equal loss, so also prefer the smaller one.
     // Drop padding channels (NA) for the sake of this check, as the number of
@@ -357,3 +353,24 @@ bool mp_chmap_sel_get_def(const struct mp_chmap_sel *s, struct mp_chmap *map,
     }
     return map->num > 0;
 }
+
+// Print the set of allowed channel layouts.
+void mp_chmal_sel_log(const struct mp_chmap_sel *s, struct mp_log *log, int lev)
+{
+    if (!mp_msg_test(log, lev))
+        return;
+
+    for (int i = 0; i < s->num_chmaps; i++)
+        mp_msg(log, lev, " - %s\n", mp_chmap_to_str(&s->chmaps[i]));
+    for (int i = 0; i < MP_SPEAKER_ID_COUNT; i++) {
+        if (!s->speakers[i])
+            continue;
+        struct mp_chmap l = {.num = 1, .speaker = { i }};
+        mp_msg(log, lev, " - #%s\n",
+                    i == MP_SPEAKER_ID_FC ? "fc" : mp_chmap_to_str_hr(&l));
+    }
+    if (s->allow_waveext)
+        mp_msg(log, lev, " - waveext\n");
+    if (s->allow_any)
+        mp_msg(log, lev, " - anything\n");
+}

audio/chmap_sel.h

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

audio/decode/ad_lavc.c

@@ -26,7 +26,7 @@
 #include <libavutil/common.h>
 #include <libavutil/intreadwrite.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 #include "config.h"
 #include "common/av_common.h"
@@ -79,12 +79,12 @@ static int init(struct dec_audio *da, const char *decoder)
     AVCodecContext *lavc_context;
     AVCodec *lavc_codec;
     struct sh_stream *sh = da->header;
-    struct sh_audio *sh_audio = sh->audio;
+    struct mp_codec_params *c = sh->codec;
 
     struct priv *ctx = talloc_zero(NULL, struct priv);
     da->priv = ctx;
 
-    ctx->force_channel_map = sh_audio->force_channels;
+    ctx->force_channel_map = c->force_channels;
 
     lavc_codec = avcodec_find_decoder_by_name(decoder);
     if (!lavc_codec) {
@@ -116,20 +116,20 @@ static int init(struct dec_audio *da, const char *decoder)
 
     mp_set_avopts(da->log, lavc_context, opts->avopts);
 
-    lavc_context->codec_tag = sh->codec_tag;
-    lavc_context->sample_rate = sh_audio->samplerate;
-    lavc_context->bit_rate = sh_audio->bitrate;
-    lavc_context->block_align = sh_audio->block_align;
-    lavc_context->bits_per_coded_sample = sh_audio->bits_per_coded_sample;
-    lavc_context->channels = sh_audio->channels.num;
-    if (!mp_chmap_is_unknown(&sh_audio->channels))
-        lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
+    lavc_context->codec_tag = c->codec_tag;
+    lavc_context->sample_rate = c->samplerate;
+    lavc_context->bit_rate = c->bitrate;
+    lavc_context->block_align = c->block_align;
+    lavc_context->bits_per_coded_sample = c->bits_per_coded_sample;
+    lavc_context->channels = c->channels.num;
+    if (!mp_chmap_is_unknown(&c->channels))
+        lavc_context->channel_layout = mp_chmap_to_lavc(&c->channels);
 
     // demux_mkv
-    mp_lavc_set_extradata(lavc_context, sh->extradata, sh->extradata_size);
+    mp_lavc_set_extradata(lavc_context, c->extradata, c->extradata_size);
 
-    if (sh->lav_headers)
-        mp_copy_lav_codec_headers(lavc_context, sh->lav_headers);
+    if (c->lav_headers)
+        mp_copy_lav_codec_headers(lavc_context, c->lav_headers);
 
     mp_set_avcodec_threads(da->log, lavc_context, opts->threads);
 
@@ -191,12 +191,6 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
     AVPacket pkt;
     mp_set_av_packet(&pkt, mpkt, NULL);
 
-    // If we don't have a PTS yet, use the first packet PTS we can get.
-    if (da->pts == MP_NOPTS_VALUE && mpkt && mpkt->pts != MP_NOPTS_VALUE) {
-        da->pts = mpkt->pts;
-        da->pts_offset = 0;
-    }
-
     int got_frame = 0;
     av_frame_unref(priv->avframe);
     int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
@@ -225,10 +219,6 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
         return mpkt ? AD_OK : AD_EOF;
 
     double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
-    if (out_pts != MP_NOPTS_VALUE) {
-        da->pts = out_pts;
-        da->pts_offset = 0;
-    }
 
     struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
     if (!mpframe)
@@ -238,12 +228,13 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
     if (lavc_chmap.num != avctx->channels)
         mp_chmap_from_channels(&lavc_chmap, avctx->channels);
     if (priv->force_channel_map) {
-        struct sh_audio *sh_audio = da->header->audio;
-        if (lavc_chmap.num == sh_audio->channels.num)
-            lavc_chmap = sh_audio->channels;
+        if (lavc_chmap.num == da->header->codec->channels.num)
+            lavc_chmap = da->header->codec->channels;
     }
     mp_audio_set_channels(mpframe, &lavc_chmap);
 
+    mpframe->pts = out_pts;
+
 #if HAVE_AVFRAME_SKIP_SAMPLES
     AVFrameSideData *sd =
         av_frame_get_side_data(priv->avframe, AV_FRAME_DATA_SKIP_SAMPLES);
@@ -254,7 +245,8 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
         uint32_t skip = MPMIN(priv->skip_samples, mpframe->samples);
         if (skip) {
             mp_audio_skip_samples(mpframe, skip);
-            da->pts_offset += skip;
+            if (mpframe->pts != MP_NOPTS_VALUE)
+                mpframe->pts += skip / (double)mpframe->rate;
             priv->skip_samples -= skip;
         }
         if (pad <= mpframe->samples)

audio/decode/ad_spdif.c

@@ -253,13 +253,11 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
     if (!mpkt)
         return AD_EOF;
 
+    double pts = mpkt->pts;
+
     AVPacket pkt;
     mp_set_av_packet(&pkt, mpkt, NULL);
     pkt.pts = pkt.dts = 0;
-    if (mpkt->pts != MP_NOPTS_VALUE) {
-        da->pts        = mpkt->pts;
-        da->pts_offset = 0;
-    }
     if (!spdif_ctx->lavf_ctx) {
         if (init_filter(da, &pkt) < 0)
             return AD_ERR;
@@ -276,6 +274,7 @@ static int decode_packet(struct dec_audio *da, struct mp_audio **out)
         return AD_ERR;
 
     memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
+    (*out)->pts = pts;
 
     return 0;
 }

audio/decode/dec_audio.c

@@ -18,6 +18,7 @@
 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
+#include <math.h>
 #include <assert.h>
 
 #include <libavutil/mem.h>
@@ -87,7 +88,7 @@ struct mp_decoder_list *audio_decoder_list(void)
 static struct mp_decoder_list *audio_select_decoders(struct dec_audio *d_audio)
 {
     struct MPOpts *opts = d_audio->opts;
-    const char *codec = d_audio->header->codec;
+    const char *codec = d_audio->header->codec->codec;
 
     struct mp_decoder_list *list = audio_decoder_list();
     struct mp_decoder_list *new =
@@ -145,7 +146,7 @@ int audio_init_best_codec(struct dec_audio *d_audio)
         MP_VERBOSE(d_audio, "Selected audio codec: %s\n", d_audio->decoder_desc);
     } else {
         MP_ERR(d_audio, "Failed to initialize an audio decoder for codec '%s'.\n",
-               d_audio->header->codec ? d_audio->header->codec : "<unknown>");
+               d_audio->header->codec->codec);
     }
 
     talloc_free(list);
@@ -170,14 +171,38 @@ static int decode_new_frame(struct dec_audio *da)
         if (ret < 0)
             return ret;
 
-        if (da->pts == MP_NOPTS_VALUE && da->header->missing_timestamps)
-            da->pts = 0;
-
         if (da->waiting) {
+            if (da->waiting->pts != MP_NOPTS_VALUE) {
+                if (da->pts != MP_NOPTS_VALUE) {
+                    da->pts += da->pts_offset / (double)da->waiting->rate;
+                    da->pts_offset = 0;
+                }
+                double newpts = da->waiting->pts;
+                // Keep the interpolated timestamp if it doesn't deviate more
+                // than 1 ms from the real one. (MKV rounded timestamps.)
+                if (da->pts == MP_NOPTS_VALUE || da->pts_offset != 0 ||
+                    fabs(da->pts - newpts) > 0.001)
+                {
+                    // Attempt to detect jumps in PTS. Even for the lowest
+                    // sample rates and with worst container rounded timestamp,
+                    // this should be a margin more than enough.
+                    if (da->pts != MP_NOPTS_VALUE && fabs(newpts - da->pts) > 0.1)
+                    {
+                        MP_WARN(da, "Invalid audio PTS: %f -> %f\n",
+                                da->pts, newpts);
+                        da->pts_reset = true;
+                    }
+                    da->pts = da->waiting->pts;
+                    da->pts_offset = 0;
+                }
+            }
             da->pts_offset += da->waiting->samples;
             da->decode_format = *da->waiting;
             mp_audio_set_null_data(&da->decode_format);
         }
+
+        if (da->pts == MP_NOPTS_VALUE && da->header->missing_timestamps)
+            da->pts = 0;
     }
     return mp_audio_config_valid(da->waiting) ? AD_OK : AD_ERR;
 }
@@ -259,6 +284,7 @@ void audio_reset_decoding(struct dec_audio *d_audio)
     af_seek_reset(d_audio->afilter);
     d_audio->pts = MP_NOPTS_VALUE;
     d_audio->pts_offset = 0;
+    d_audio->pts_reset = false;
     if (d_audio->waiting) {
         talloc_free(d_audio->waiting);
         d_audio->waiting = NULL;

audio/decode/dec_audio.h

@@ -30,7 +30,7 @@ struct dec_audio {
     struct mp_log *log;
     struct MPOpts *opts;
     struct mpv_global *global;
-    bool spdif_passthrough;
+    bool spdif_passthrough, spdif_failed;
     const struct ad_functions *ad_driver;
     struct sh_stream *header;
     struct af_stream *afilter;
@@ -45,6 +45,8 @@ struct dec_audio {
     double pts;
     // number of samples output by decoder after last known pts
     int pts_offset;
+    // set every time a jump in timestamps is encountered
+    bool pts_reset;
     // For free use by the ad_driver
     void *priv;
 };

audio/filter/af.c

@@ -461,7 +461,8 @@ static int af_reinit(struct af_stream *s)
                     struct af_instance *aft = af->prev;
                     af_remove(s, af);
                     af = aft->next;
-                    break;
+                    retry++;
+                    continue;
                 }
             }
             goto negotiate_error;

audio/filter/af_lavcac3enc.c

@@ -49,7 +49,6 @@ const uint16_t ac3_bitrate_tab[19] = {
 typedef struct af_ac3enc_s {
     struct AVCodec        *lavc_acodec;
     struct AVCodecContext *lavc_actx;
-    AVPacket pkt;
     int bit_rate;
     struct mp_audio *input;     // frame passed to libavcodec
     struct mp_audio *pending;   // unconsumed input data
@@ -146,7 +145,6 @@ static void uninit(struct af_instance* af)
     af_ac3enc_t *s = af->priv;
 
     if (s) {
-        av_free_packet(&s->pkt);
         if(s->lavc_actx) {
             avcodec_close(s->lavc_actx);
             av_free(s->lavc_actx);
@@ -212,6 +210,8 @@ static int filter_out(struct af_instance *af)
         MP_FATAL(af, "Could not allocate memory \n");
         return -1;
     }
+    int err = -1;
+
     frame->nb_samples = s->in_samples;
     frame->format = s->lavc_actx->sample_fmt;
     frame->channel_layout = s->lavc_actx->channel_layout;
@@ -221,30 +221,33 @@ static int filter_out(struct af_instance *af)
         frame->data[n] = s->input->planes[n];
     frame->linesize[0] = s->input->samples * s->input->sstride;
 
+    AVPacket pkt = {0};
+    av_init_packet(&pkt);
+
     int ok;
-    int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
+    int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &pkt, frame, &ok);
     av_frame_free(&frame);
     s->input->samples = 0;
     if (lavc_ret < 0 || !ok) {
         MP_FATAL(af, "Encode failed.\n");
-        return -1;
+        goto done;
     }
 
     MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
-            s->pkt.size, s->pending->samples);
+            pkt.size, s->pending->samples);
 
     struct mp_audio *out =
         mp_audio_pool_get(af->out_pool, af->data, s->out_samples);
     if (!out)
-        return -1;
+        goto done;
     mp_audio_copy_attributes(out, s->pending);
 
-    int frame_size = s->pkt.size;
+    int frame_size = pkt.size;
     int header_len = 0;
     char hdr[8];
 
-    if (s->cfg_add_iec61937_header && s->pkt.size > 5) {
-        int bsmod = s->pkt.data[5] & 0x7;
+    if (s->cfg_add_iec61937_header && pkt.size > 5) {
+        int bsmod = pkt.data[5] & 0x7;
         int len = frame_size;
 
         frame_size = AC3_FRAME_SIZE * 2 * 2;
@@ -262,14 +265,18 @@ static int filter_out(struct af_instance *af)
 
     char *buf = (char *)out->planes[0];
     memcpy(buf, hdr, header_len);
-    memcpy(buf + header_len, s->pkt.data, s->pkt.size);
-    memset(buf + header_len + s->pkt.size, 0,
-            frame_size - (header_len + s->pkt.size));
-    swap_16((uint16_t *)(buf + header_len), s->pkt.size / 2);
+    memcpy(buf + header_len, pkt.data, pkt.size);
+    memset(buf + header_len + pkt.size, 0,
+           frame_size - (header_len + pkt.size));
+    swap_16((uint16_t *)(buf + header_len), pkt.size / 2);
     out->samples = frame_size / out->sstride;
     af_add_output_frame(af, out);
     update_delay(af);
-    return 0;
+
+    err = 0;
+done:
+    av_packet_unref(&pkt);
+    return err;
 }
 
 static int af_open(struct af_instance* af){
@@ -307,8 +314,6 @@ static int af_open(struct af_instance* af){
     MP_VERBOSE(af, "[af_lavcac3enc]: in sample format: %s\n",
            af_fmt_to_str(s->in_sampleformat));
 
-    av_init_packet(&s->pkt);
-
     s->input = talloc_zero(s, struct mp_audio);
 
     if (s->cfg_bit_rate) {

audio/filter/af_lavfi.c

@@ -195,6 +195,10 @@ static int control(struct af_instance *af, int cmd, void *arg)
         if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
             mp_audio_set_format(in, AF_FORMAT_FLOAT);
 
+        // Removing this requires fixing AVFrame.data vs. AVFrame.extended_data
+        if (in->channels.num > AV_NUM_DATA_POINTERS)
+            return AF_ERROR;
+
         if (!mp_chmap_is_lavc(&in->channels))
             mp_chmap_reorder_to_lavc(&in->channels); // will always work
 
@@ -211,7 +215,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
         mp_chmap_from_lavc(&out_cm, l_out->channel_layout);
         mp_audio_set_channels(out, &out_cm);
 
-        if (!mp_audio_config_valid(out))
+        if (!mp_audio_config_valid(out) || out->channels.num > AV_NUM_DATA_POINTERS)
             return AF_ERROR;
 
         p->timebase_out = l_out->time_base;
@@ -349,8 +353,6 @@ static int af_open(struct af_instance *af)
     af->uninit = uninit;
     af->filter_frame = filter_frame;
     af->filter_out = filter_out;
-    // Removing this requires fixing AVFrame.data vs. AVFrame.extended_data
-    assert(MP_NUM_CHANNELS <= AV_NUM_DATA_POINTERS);
     return AF_OK;
 }
 

audio/filter/af_lavrresample.c

@@ -49,6 +49,7 @@
 #define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \
     swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples)
 #define avresample_set_channel_mapping swr_set_channel_mapping
+#define avresample_set_compensation swr_set_compensation
 #else
 #error "config.h broken or no resampler found"
 #endif
@@ -304,14 +305,15 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
     av_opt_set_int(s->avrctx, "in_sample_fmt",      in_samplefmt, 0);
     av_opt_set_int(s->avrctx, "out_sample_fmt",     out_samplefmtp, 0);
 
-    // Just needs the correct number of channels.
-    int fake_out_ch_layout = av_get_default_channel_layout(map_out.num);
+    // Just needs the correct number of channels for deplanarization.
+    struct mp_chmap fake_chmap;
+    mp_chmap_set_unknown(&fake_chmap, map_out.num);
+    uint64_t fake_out_ch_layout = mp_chmap_to_lavc_unchecked(&fake_chmap);
     if (!fake_out_ch_layout)
         goto error;
-
-    // Deplanarize if needed.
     av_opt_set_int(s->avrctx_out, "in_channel_layout",  fake_out_ch_layout, 0);
     av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
+
     av_opt_set_int(s->avrctx_out, "in_sample_fmt",      out_samplefmtp, 0);
     av_opt_set_int(s->avrctx_out, "out_sample_fmt",     out_samplefmt, 0);
     av_opt_set_int(s->avrctx_out, "in_sample_rate",     s->out_rate, 0);
@@ -386,14 +388,6 @@ static int control(struct af_instance *af, int cmd, void *arg)
         return AF_OK;
     case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
         s->playback_speed = *(double *)arg;
-        int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
-        if (new_rate != s->in_rate && s->avrctx && af->fmt_out.format) {
-            // Before reconfiguring, drain the audio that is still buffered
-            // in the resampler.
-            af->filter_frame(af, NULL);
-            // Reinitialize resampler.
-            configure_lavrr(af, &af->fmt_in, &af->fmt_out, false);
-        }
         return AF_OK;
     }
     case AF_CONTROL_RESET:
@@ -429,6 +423,18 @@ static void extra_output_conversion(struct af_instance *af, struct mp_audio *mpa
         }
         mp_audio_set_format(mpa, AF_FORMAT_S24);
     }
+
+    for (int p = 0; p < mpa->num_planes; p++) {
+        void *ptr = mpa->planes[p];
+        int total = mpa->samples * mpa->spf;
+        if (af_fmt_from_planar(mpa->format) == AF_FORMAT_FLOAT) {
+            for (int s = 0; s < total; s++)
+                ((float *)ptr)[s] = av_clipf(((float *)ptr)[s], -1.0f, 1.0f);
+        } else if (af_fmt_from_planar(mpa->format) == AF_FORMAT_DOUBLE) {
+            for (int s = 0; s < total; s++)
+                ((double *)ptr)[s] = MPCLAMP(((double *)ptr)[s], -1.0, 1.0);
+        }
+    }
 }
 
 // This relies on the tricky way mpa was allocated.
@@ -455,7 +461,7 @@ static void reorder_planes(struct mp_audio *mpa, int *reorder,
     }
 }
 
-static int filter(struct af_instance *af, struct mp_audio *in)
+static int filter_resample(struct af_instance *af, struct mp_audio *in)
 {
     struct af_resample *s = af->priv;
 
@@ -516,6 +522,39 @@ error:
     return -1;
 }
 
+static int filter(struct af_instance *af, struct mp_audio *in)
+{
+    struct af_resample *s = af->priv;
+
+    int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
+    bool need_reinit = fabs(new_rate / (double)s->in_rate - 1) > 0.01;
+
+    if (s->avrctx) {
+        AVRational r = av_d2q(s->playback_speed * s->in_rate_af / s->in_rate,
+                              INT_MAX / 2);
+        // Essentially, swr/avresample_set_compensation() does 2 things:
+        // - adjust output sample rate by sample_delta/compensation_distance
+        // - reset the adjustment after compensation_distance output samples
+        // Increase the compensation_distance to avoid undesired reset
+        // semantics - we want to keep the ratio for the whole frame we're
+        // feeding it, until the next filter() call.
+        int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1);
+        r = (AVRational){ r.num * mult, r.den * mult };
+        if (avresample_set_compensation(s->avrctx, r.den - r.num, r.den) < 0)
+            need_reinit = true;
+    }
+
+    if (need_reinit && new_rate != s->in_rate) {
+        // Before reconfiguring, drain the audio that is still buffered
+        // in the resampler.
+        filter_resample(af, NULL);
+        // Reinitialize resampler.
+        configure_lavrr(af, &af->fmt_in, &af->fmt_out, false);
+    }
+
+    return filter_resample(af, in);
+}
+
 static int af_open(struct af_instance *af)
 {
     struct af_resample *s = af->priv;

audio/filter/af_scaletempo.c

@@ -286,9 +286,6 @@ static void update_speed(struct af_instance *af, float speed)
 
     s->frames_stride_scaled = s->scale * s->frames_stride;
     s->frames_stride_error = MPMIN(s->frames_stride_error, s->frames_stride_scaled);
-
-    MP_VERBOSE(af, "%.3f speed * %.3f scale_nominal = %.3f\n",
-               s->speed, s->scale_nominal, s->scale);
 }
 
 // Initialization and runtime control

audio/format.c

@@ -51,7 +51,7 @@ int af_fmt_change_bytes(int format, int bytes)
         if (af_fmt_to_bytes(fmt) == bytes &&
             af_fmt_is_float(fmt) == af_fmt_is_float(format) &&
             af_fmt_is_planar(fmt) == af_fmt_is_planar(format) &&
-            af_fmt_is_spdif(fmt) == af_fmt_is_spdif(format))
+            (fmt == format || (!af_fmt_is_spdif(fmt) && !af_fmt_is_spdif(format))))
             return fmt;
     }
     return 0;

audio/mixer.c

@@ -27,7 +27,7 @@
 #include "audio/filter/af.h"
 #include "common/global.h"
 #include "common/msg.h"
-#include "talloc.h"
+#include "mpv_talloc.h"
 #include "mixer.h"
 
 struct mixer {
@@ -120,8 +120,11 @@ void mixer_getvolume(struct mixer *mixer, float *l, float *r)
     *r = mixer->vol_r;
 }
 
-static void setvolume_internal(struct mixer *mixer, float l, float r)
+static void setvolume_internal(struct mixer *mixer)
 {
+    float l = mixer->vol_l, r = mixer->vol_r;
+    if (mixer->emulate_mute && mixer->muted)
+        l = r = 0;
     if (!mixer->softvol) {
         MP_DBG(mixer, "Setting volume on AO.\n");
         struct ao_control_vol vol = {.left = l, .right = r};
@@ -147,8 +150,8 @@ void mixer_setvolume(struct mixer *mixer, float l, float r)
     float max = mixer_getmaxvolume(mixer);
     mixer->vol_l = MPCLAMP(l, 0, max);
     mixer->vol_r = MPCLAMP(r, 0, max);
-    if (mixer->ao && !(mixer->emulate_mute && mixer->muted))
-        setvolume_internal(mixer, mixer->vol_l, mixer->vol_r);
+    if (mixer->ao)
+        setvolume_internal(mixer);
 }
 
 void mixer_getbothvolume(struct mixer *mixer, float *b)
@@ -167,7 +170,7 @@ void mixer_setmute(struct mixer *mixer, bool mute)
         mixer->muted = mute;
         mixer->muted_by_us = mute;
         if (mixer->emulate_mute) {
-            setvolume_internal(mixer, mixer->vol_l*!mute, mixer->vol_r*!mute);
+            setvolume_internal(mixer);
         } else {
             ao_control(mixer->ao, AOCONTROL_SET_MUTE, &mute);
         }
@@ -293,6 +296,8 @@ static void restore_volume(struct mixer *mixer)
 
     const char *prev_driver = mixer->driver;
     mixer->driver = mixer->softvol ? "softvol" : ao_get_name(ao);
+    if (!prev_driver[0])
+        prev_driver = mixer->driver;
 
     // Restore old parameters if volume won't survive reinitialization.
     // But not if volume scale is possibly different.

audio/out/ao.c

@@ -20,7 +20,7 @@
 #include <string.h>
 #include <assert.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 #include "config.h"
 #include "ao.h"
@@ -45,7 +45,6 @@ extern const struct ao_driver audio_out_jack;
 extern const struct ao_driver audio_out_openal;
 extern const struct ao_driver audio_out_null;
 extern const struct ao_driver audio_out_alsa;
-extern const struct ao_driver audio_out_dsound;
 extern const struct ao_driver audio_out_wasapi;
 extern const struct ao_driver audio_out_pcm;
 extern const struct ao_driver audio_out_lavc;
@@ -65,9 +64,6 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #if HAVE_WASAPI
     &audio_out_wasapi,
 #endif
-#if HAVE_DSOUND
-    &audio_out_dsound,
-#endif
 #if HAVE_OSS_AUDIO
     &audio_out_oss,
 #endif
@@ -240,6 +236,7 @@ static void split_ao_device(void *tmp, char *opt, char **out_ao, char **out_dev)
 }
 
 struct ao *ao_init_best(struct mpv_global *global,
+                        bool ao_null_fallback,
                         struct input_ctx *input_ctx,
                         struct encode_lavc_context *encode_lavc_ctx,
                         int samplerate, int format, struct mp_chmap channels)
@@ -248,58 +245,67 @@ struct ao *ao_init_best(struct mpv_global *global,
     void *tmp = talloc_new(NULL);
     struct mp_log *log = mp_log_new(tmp, global->log, "ao");
     struct ao *ao = NULL;
-    struct m_obj_settings *ao_list = opts->audio_driver_list;
+    struct m_obj_settings *ao_list = NULL;
+    int ao_num = 0;
+
+    for (int n = 0; opts->audio_driver_list && opts->audio_driver_list[n].name; n++)
+        MP_TARRAY_APPEND(tmp, ao_list, ao_num, opts->audio_driver_list[n]);
 
     bool forced_dev = false;
     char *pref_ao, *pref_dev;
     split_ao_device(tmp, opts->audio_device, &pref_ao, &pref_dev);
-    if (!(ao_list && ao_list[0].name) && pref_ao) {
+    if (!ao_num && pref_ao) {
         // Reuse the autoselection code
-        ao_list = talloc_zero_array(tmp, struct m_obj_settings, 2);
-        ao_list[0].name = pref_ao;
+        MP_TARRAY_APPEND(tmp, ao_list, ao_num,
+            (struct m_obj_settings){.name = pref_ao});
         forced_dev = true;
     }
 
-    if (ao_list && ao_list[0].name) {
-        for (int n = 0; ao_list[n].name; n++) {
-            if (strlen(ao_list[n].name) == 0)
-                goto autoprobe;
-            mp_verbose(log, "Trying preferred audio driver '%s'\n",
-                       ao_list[n].name);
-            char *dev = NULL;
-            if (pref_ao && strcmp(ao_list[n].name, pref_ao) == 0)
-                dev = pref_dev;
-            if (dev)
-                mp_verbose(log, "Using preferred device '%s'\n", dev);
-            ao = ao_init(false, global, input_ctx, encode_lavc_ctx,
-                         samplerate, format, channels, dev,
-                         ao_list[n].name, ao_list[n].attribs);
-            if (ao)
-                goto done;
-            mp_err(log, "Failed to initialize audio driver '%s'\n",
-                   ao_list[n].name);
-            if (forced_dev) {
-                mp_err(log, "This audio driver/device was forced with the "
-                            "--audio-device option.\n"
-                            "Try unsetting it.\n");
-            }
+    bool autoprobe = ao_num == 0;
+
+    // Something like "--ao=a,b," means do autoprobing after a and b fail.
+    if (ao_num && strlen(ao_list[ao_num - 1].name) == 0) {
+        ao_num -= 1;
+        autoprobe = true;
+    }
+
+    if (autoprobe) {
+        for (int n = 0; audio_out_drivers[n]; n++) {
+            const struct ao_driver *driver = audio_out_drivers[n];
+            if (driver == &audio_out_null)
+                break;
+            MP_TARRAY_APPEND(tmp, ao_list, ao_num,
+                (struct m_obj_settings){.name = (char *)driver->name});
         }
-        goto done;
     }
 
-autoprobe: ;
-    // now try the rest...
-    for (int i = 0; audio_out_drivers[i]; i++) {
-        const struct ao_driver *driver = audio_out_drivers[i];
-        if (driver == &audio_out_null)
-            break;
-        ao = ao_init(true, global, input_ctx, encode_lavc_ctx, samplerate,
-                     format, channels, NULL, (char *)driver->name, NULL);
+    if (ao_null_fallback) {
+        MP_TARRAY_APPEND(tmp, ao_list, ao_num,
+            (struct m_obj_settings){.name = "null"});
+    }
+
+    for (int n = 0; n < ao_num; n++) {
+        struct m_obj_settings *entry = &ao_list[n];
+        bool probing = n + 1 != ao_num;
+        mp_verbose(log, "Trying audio driver '%s'\n", entry->name);
+        char *dev = NULL;
+        if (pref_ao && pref_dev && strcmp(entry->name, pref_ao) == 0) {
+            dev = pref_dev;
+            mp_verbose(log, "Using preferred device '%s'\n", dev);
+        }
+        ao = ao_init(probing, global, input_ctx, encode_lavc_ctx,
+                     samplerate, format, channels, dev,
+                     entry->name, entry->attribs);
         if (ao)
-            goto done;
+            break;
+        if (!probing)
+            mp_err(log, "Failed to initialize audio driver '%s'\n", entry->name);
+        if (dev && forced_dev) {
+            mp_err(log, "This audio driver/device was forced with the "
+                        "--audio-device option.\nTry unsetting it.\n");
+        }
     }
 
-done:
     talloc_free(tmp);
     return ao;
 }
@@ -411,18 +417,11 @@ void ao_hotplug_event(struct ao *ao)
 bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
                          struct mp_chmap *map)
 {
-    if (mp_msg_test(ao->log, MSGL_DEBUG)) {
-        for (int i = 0; i < s->num_chmaps; i++) {
-            struct mp_chmap c = s->chmaps[i];
-            MP_DBG(ao, "chmap_sel #%d: %s (%s)\n", i, mp_chmap_to_str(&c),
-                   mp_chmap_to_str_hr(&c));
-        }
-    }
+    MP_VERBOSE(ao, "Channel layouts:\n");
+    mp_chmal_sel_log(s, ao->log, MSGL_V);
     bool r = mp_chmap_sel_adjust(s, map);
-    if (r) {
-        MP_DBG(ao, "result: %s (%s)\n", mp_chmap_to_str(map),
-               mp_chmap_to_str_hr(map));
-    }
+    if (r)
+        MP_VERBOSE(ao, "result: %s\n", mp_chmap_to_str(map));
     return r;
 }
 
@@ -490,8 +489,12 @@ static void get_devices(struct ao *ao, struct ao_device_list *list)
     if (ao->driver->list_devs)
         ao->driver->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"});
+    if (list->num_devices == num) {
+        char name[80] = "Default";
+        if (num > 1)
+            mp_snprintf_cat(name, sizeof(name), " (%s)", ao->driver->name);
+        ao_device_list_add(list, ao, &(struct ao_device_desc){"", name});
+    }
 }
 
 bool ao_hotplug_check_update(struct ao_hotplug *hp)

audio/out/ao.h

@@ -72,6 +72,7 @@ struct encode_lavc_context;
 struct mp_audio;
 
 struct ao *ao_init_best(struct mpv_global *global,
+                        bool ao_null_fallback,
                         struct input_ctx *input_ctx,
                         struct encode_lavc_context *encode_lavc_ctx,
                         int samplerate, int format, struct mp_chmap channels);

audio/out/ao_alsa.c

@@ -50,12 +50,13 @@
 
 struct priv {
     snd_pcm_t *alsa;
+    bool device_lost;
     snd_pcm_format_t alsa_fmt;
-    int can_pause;
+    bool can_pause;
     snd_pcm_sframes_t prepause_frames;
     double delay_before_pause;
-    int buffersize; // in frames
-    int outburst; // in frames
+    snd_pcm_uframes_t buffersize;
+    snd_pcm_uframes_t outburst;
 
     char *cfg_device;
     char *cfg_mixer_device;
@@ -83,6 +84,21 @@ struct priv {
             MP_WARN(ao, "%s: %s\n", (message), snd_strerror(err)); \
     } while (0)
 
+// Common code for handling ENODEV, which happens if a device gets "lost", and
+// can't be used anymore. Returns true if alsa_err is not ENODEV.
+static bool check_device_present(struct ao *ao, int alsa_err)
+{
+    struct priv *p = ao->priv;
+    if (alsa_err != -ENODEV)
+        return true;
+    if (!p->device_lost) {
+        MP_WARN(ao, "Device lost, trying to recover...\n");
+        ao_request_reload(ao);
+        p->device_lost = true;
+    }
+    return false;
+}
+
 static int control(struct ao *ao, enum aocontrol cmd, void *arg)
 {
     struct priv *p = ao->priv;
@@ -211,24 +227,19 @@ static const int mp_to_alsa_format[][2] = {
             MP_SELECT_LE_BE(SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S24_3BE)},
     {AF_FORMAT_FLOAT,       SND_PCM_FORMAT_FLOAT},
     {AF_FORMAT_DOUBLE,      SND_PCM_FORMAT_FLOAT64},
+    {AF_FORMAT_S_MP3,       SND_PCM_FORMAT_MPEG},
     {AF_FORMAT_UNKNOWN,     SND_PCM_FORMAT_UNKNOWN},
 };
 
 static int find_alsa_format(int af_format)
 {
-    if (af_fmt_is_spdif(af_format)) {
-        if (af_format == AF_FORMAT_S_MP3) {
-            return SND_PCM_FORMAT_MPEG;
-        } else {
-            return SND_PCM_FORMAT_S16;
-        }
-    }
-
     af_format = af_fmt_from_planar(af_format);
     for (int n = 0; mp_to_alsa_format[n][0] != AF_FORMAT_UNKNOWN; n++) {
         if (mp_to_alsa_format[n][0] == af_format)
             return mp_to_alsa_format[n][1];
     }
+    if (af_fmt_is_spdif(af_format))
+        return SND_PCM_FORMAT_S16;
     return SND_PCM_FORMAT_UNKNOWN;
 }
 
@@ -259,6 +270,7 @@ static const int alsa_to_mp_channels[][2] = {
     {SND_CHMAP_RLC,     MP_SP(SDL)},
     {SND_CHMAP_MONO,    MP_SP(FC)},
     {SND_CHMAP_NA,      MP_SPEAKER_ID_NA},
+    {SND_CHMAP_UNKNOWN, MP_SPEAKER_ID_NA},
     {SND_CHMAP_LAST,    MP_SPEAKER_ID_COUNT}
 };
 
@@ -272,28 +284,48 @@ static int find_mp_channel(int alsa_channel)
     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];
-    }
+#define CHMAP(n, ...) &(struct mp_chmap) MP_CONCAT(MP_CHMAP, n) (__VA_ARGS__)
 
-    return SND_CHMAP_UNKNOWN;
+// Replace each channel in a with b (a->num == b->num)
+static void replace_submap(struct mp_chmap *dst, struct mp_chmap *a,
+                           struct mp_chmap *b)
+{
+    struct mp_chmap t = *dst;
+    if (!mp_chmap_is_valid(&t) || mp_chmap_diffn(a, &t) != 0)
+        return;
+    assert(a->num == b->num);
+    for (int n = 0; n < t.num; n++) {
+        for (int i = 0; i < a->num; i++) {
+            if (t.speaker[n] == a->speaker[i]) {
+                t.speaker[n] = b->speaker[i];
+                break;
+            }
+        }
+    }
+    if (mp_chmap_is_valid(&t))
+        *dst = t;
 }
 
-static int mp_chmap_from_alsa(struct mp_chmap *dst, snd_pcm_chmap_t *src)
+static bool mp_chmap_from_alsa(struct mp_chmap *dst, snd_pcm_chmap_t *src)
 {
     *dst = (struct mp_chmap) {0};
 
     if (src->channels > MP_NUM_CHANNELS)
-        return -1;
+        return false;
 
     dst->num = src->channels;
     for (int c = 0; c < dst->num; c++)
         dst->speaker[c] = find_mp_channel(src->pos[c]);
 
-    return 0;
+    // Assume anything with 1 channel is mono.
+    if (dst->num == 1)
+        dst->speaker[0] = MP_SP(FC);
+
+    // Remap weird Intel HDA HDMI 7.1 layouts correctly.
+    replace_submap(dst, CHMAP(6, FL, FR, BL, BR, SDL, SDR),
+                        CHMAP(6, FL, FR, SL, SR, BL,  BR));
+
+    return mp_chmap_is_valid(dst);
 }
 
 static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
@@ -306,20 +338,21 @@ static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
         return false;
 
     for (int i = 0; maps[i] != NULL; i++) {
-        struct mp_chmap entry;
-        mp_chmap_from_alsa(&entry, &maps[i]->map);
+        char aname[128];
+        if (snd_pcm_chmap_print(&maps[i]->map, sizeof(aname), aname) <= 0)
+            aname[0] = '\0';
 
-        if (mp_chmap_is_valid(&entry)) {
+        struct mp_chmap entry;
+        if (mp_chmap_from_alsa(&entry, &maps[i]->map)) {
+            struct mp_chmap reorder = entry;
             if (maps[i]->type == SND_CHMAP_TYPE_VAR)
-                mp_chmap_reorder_norm(&entry);
-            MP_VERBOSE(ao, "Got supported channel map: %s (type %s)\n",
-                       mp_chmap_to_str(&entry),
-                       snd_pcm_chmap_type_name(maps[i]->type));
-            mp_chmap_sel_add_map(&chmap_sel, &entry);
+                mp_chmap_reorder_norm(&reorder);
+            MP_DBG(ao, "Got supported channel map: %s (type %s) -> %s -> %s\n",
+                   aname, snd_pcm_chmap_type_name(maps[i]->type),
+                   mp_chmap_to_str(&entry), mp_chmap_to_str(&reorder));
+            mp_chmap_sel_add_map(&chmap_sel, &reorder);
         } 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);
+            MP_VERBOSE(ao, "skipping unknown ALSA channel map: %s\n", aname);
         }
     }
 
@@ -328,6 +361,108 @@ static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
     return ao_chmap_sel_adjust(ao, &chmap_sel, chmap);
 }
 
+// Map back our selected channel layout to an ALSA one. This is done this way so
+// that our ALSA->mp_chmap mapping function only has to go one way.
+// The return value is to be freed with free().
+static snd_pcm_chmap_t *map_back_chmap(struct ao *ao, struct mp_chmap *chmap)
+{
+    struct priv *p = ao->priv;
+    if (!mp_chmap_is_valid(chmap))
+        return NULL;
+
+    snd_pcm_chmap_query_t **maps = snd_pcm_query_chmaps(p->alsa);
+    if (!maps)
+        return NULL;
+
+    snd_pcm_chmap_t *alsa_chmap = NULL;
+
+    for (int i = 0; maps[i] != NULL; i++) {
+        struct mp_chmap entry;
+        if (!mp_chmap_from_alsa(&entry, &maps[i]->map))
+            continue;
+
+        if (mp_chmap_equals(chmap, &entry) ||
+            (mp_chmap_equals_reordered(chmap, &entry) &&
+                maps[i]->type == SND_CHMAP_TYPE_VAR))
+        {
+            alsa_chmap = calloc(1, sizeof(*alsa_chmap) +
+                                   sizeof(alsa_chmap->pos[0]) * entry.num);
+            if (!alsa_chmap)
+                break;
+            alsa_chmap->channels = entry.num;
+
+            // Undo if mp_chmap_reorder() was called on the result.
+            int reorder[MP_NUM_CHANNELS];
+            mp_chmap_get_reorder(reorder, chmap, &entry);
+            for (int n = 0; n < entry.num; n++)
+                alsa_chmap->pos[n] = maps[i]->map.pos[reorder[n]];
+            break;
+        }
+    }
+
+    snd_pcm_free_chmaps(maps);
+    return alsa_chmap;
+}
+
+
+static int set_chmap(struct ao *ao, struct mp_chmap *dev_chmap, int num_channels)
+{
+    struct priv *p = ao->priv;
+    int err;
+
+    snd_pcm_chmap_t *alsa_chmap = map_back_chmap(ao, dev_chmap);
+    if (alsa_chmap) {
+        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) {
+            // A device my not be able to set any channel map, even channel maps
+            // that were reported as supported. This is either because the ALSA
+            // device is broken (dmix), or because the driver has only 1
+            // channel map per channel count, and setting the map is not needed.
+            MP_VERBOSE(ao, "device returned ENXIO when setting channel map %s\n",
+                       mp_chmap_to_str(dev_chmap));
+        } else {
+            CHECK_ALSA_WARN("Channel map setup failed");
+        }
+
+        free(alsa_chmap);
+    }
+
+    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;
+        mp_chmap_from_alsa(&chmap, alsa_chmap);
+
+        MP_VERBOSE(ao, "which we understand as: %s\n", mp_chmap_to_str(&chmap));
+
+        if (p->cfg_ignore_chmap) {
+            MP_VERBOSE(ao, "user set ignore-chmap; ignoring the channel map.\n");
+        } else if (af_fmt_is_spdif(ao->format)) {
+            MP_VERBOSE(ao, "using spdif passthrough; ignoring the channel map.\n");
+        } else if (!mp_chmap_is_valid(&chmap)) {
+            MP_WARN(ao, "Got unknown channel map from ALSA.\n");
+        } else if (chmap.num != num_channels) {
+            MP_WARN(ao, "ALSA channel map conflicts with channel count!\n");
+        } else {
+            MP_VERBOSE(ao, "using the ALSA channel map.\n");
+            if (mp_chmap_equals(&chmap, &ao->channels))
+                MP_VERBOSE(ao, "which is what we requested.\n");
+            ao->channels = chmap;
+        }
+
+        free(alsa_chmap);
+    }
+
+    return 0;
+}
+
 #else /* HAVE_CHMAP_API */
 
 static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
@@ -335,6 +470,11 @@ static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
     return false;
 }
 
+static int set_chmap(struct ao *ao, struct mp_chmap *dev_chmap, int num_channels)
+{
+    return 0;
+}
+
 #endif /* else HAVE_CHMAP_API */
 
 static int map_iec958_srate(int srate)
@@ -355,8 +495,8 @@ 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.
+// device name. 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])
@@ -427,16 +567,14 @@ static void uninit(struct ao *ao)
         int err;
 
         err = snd_pcm_close(p->alsa);
+        p->alsa = NULL;
         CHECK_ALSA_ERROR("pcm close error");
     }
 
 alsa_error: ;
 }
 
-#define INIT_OK 0
-#define INIT_ERROR -1
-#define INIT_BRAINDEATH -2
-static int init_device(struct ao *ao, bool second_try)
+static int init_device(struct ao *ao)
 {
     struct priv *p = ao->priv;
     int err;
@@ -454,31 +592,18 @@ static int init_device(struct ao *ao, bool second_try)
     CHECK_ALSA_WARN("Unable to set blocking mode");
 
     snd_pcm_hw_params_t *alsa_hwparams;
-    snd_pcm_sw_params_t *alsa_swparams;
-
     snd_pcm_hw_params_alloca(&alsa_hwparams);
-    snd_pcm_sw_params_alloca(&alsa_swparams);
 
     err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
     CHECK_ALSA_ERROR("Unable to get initial parameters");
 
-    int try_formats[AF_FORMAT_COUNT];
-    af_get_best_sample_formats(ao->format, try_formats);
-    for (int n = 0; try_formats[n]; n++) {
-        ao->format = try_formats[n];
-        p->alsa_fmt = find_alsa_format(ao->format);
-        if (snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt) >= 0)
-            break;
+    // Some ALSA drivers have broken delay reporting, so disable the ALSA
+    // resampling plugin by default.
+    if (!p->cfg_resample) {
+        err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
+        CHECK_ALSA_ERROR("Unable to disable resampling");
     }
 
-    if (!ao->format) {
-        MP_ERR(ao, "Can't find appropriate sample format.\n");
-        goto alsa_error;
-    }
-
-    err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
-    CHECK_ALSA_ERROR("Unable to set format");
-
     snd_pcm_access_t access = af_fmt_is_planar(ao->format)
                                     ? SND_PCM_ACCESS_RW_NONINTERLEAVED
                                     : SND_PCM_ACCESS_RW_INTERLEAVED;
@@ -490,6 +615,29 @@ static int init_device(struct ao *ao, bool second_try)
     }
     CHECK_ALSA_ERROR("Unable to set access type");
 
+    bool found_format = false;
+    int try_formats[AF_FORMAT_COUNT];
+    af_get_best_sample_formats(ao->format, try_formats);
+    for (int n = 0; try_formats[n]; n++) {
+        if (af_fmt_is_planar(ao->format) != af_fmt_is_planar(try_formats[n]))
+            continue; // implied SND_PCM_ACCESS mismatches
+        p->alsa_fmt = find_alsa_format(try_formats[n]);
+        MP_VERBOSE(ao, "trying format %s\n", af_fmt_to_str(try_formats[n]));
+        if (snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt) >= 0) {
+            ao->format = try_formats[n];
+            found_format = true;
+            break;
+        }
+    }
+
+    if (!found_format) {
+        MP_ERR(ao, "Can't find appropriate sample format.\n");
+        goto alsa_error;
+    }
+
+    err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
+    CHECK_ALSA_ERROR("Unable to set format");
+
     struct mp_chmap dev_chmap = ao->channels;
     if (af_fmt_is_spdif(ao->format) || p->cfg_ignore_chmap) {
         dev_chmap.num = 0; // disable chmap API
@@ -514,157 +662,51 @@ static int init_device(struct ao *ao, bool second_try)
         goto alsa_error;
     }
 
-    if (num_channels != ao->channels.num) {
-        int req = ao->channels.num;
-        mp_chmap_from_channels_alsa(&ao->channels, num_channels);
-        if (!mp_chmap_is_valid(&ao->channels))
-            mp_chmap_from_channels(&ao->channels, 2);
-        MP_ERR(ao, "Asked for %d channels, got %d - fallback to %s.\n", req,
-               num_channels, mp_chmap_to_str(&ao->channels));
-    }
-
-    // Some ALSA drivers have broken delay reporting, so disable the ALSA
-    // resampling plugin by default.
-    if (!p->cfg_resample) {
-        err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
-        CHECK_ALSA_ERROR("Unable to disable resampling");
-    }
-
     err = snd_pcm_hw_params_set_rate_near
             (p->alsa, alsa_hwparams, &ao->samplerate, NULL);
     CHECK_ALSA_ERROR("Unable to set samplerate-2");
 
+    snd_pcm_hw_params_t *hwparams_backup;
+    snd_pcm_hw_params_alloca(&hwparams_backup);
+    snd_pcm_hw_params_copy(hwparams_backup, alsa_hwparams);
+
+    // Cargo-culted buffer settings; might still be useful for PulseAudio.
     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");
-
-    err = snd_pcm_hw_params_set_periods_near
-            (p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
-    CHECK_ALSA_WARN("Unable to set periods");
+    if (err >= 0) {
+        err = snd_pcm_hw_params_set_periods_near
+                    (p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
+        CHECK_ALSA_WARN("Unable to set periods");
+    }
+    if (err < 0)
+        snd_pcm_hw_params_copy(alsa_hwparams, hwparams_backup);
 
     /* 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 */
+    if (set_chmap(ao, &dev_chmap, num_channels) < 0)
+        goto alsa_error;
 
-#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]);
-
-        // mpv and ALSA use different conventions for mono
-        if (dev_chmap.num == 1 && dev_chmap.speaker[0] == MP_SP(FC))
-            alsa_chmap->pos[0] = SND_CHMAP_MONO;
-
-        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) {
-            // A device my not be able to set any channel map, even channel maps
-            // that were reported as supported. This is either because the ALSA
-            // device is broken (dmix), or because the driver has only 1
-            // channel map per channel count, and setting the map is not needed.
-            MP_VERBOSE(ao, "device returned ENXIO when setting channel map %s\n",
-                       mp_chmap_to_str(&dev_chmap));
-        } else {
-            CHECK_ALSA_WARN("Channel map setup failed");
-        }
-
-        free(alsa_chmap);
+    if (num_channels != ao->channels.num) {
+        int req = ao->channels.num;
+        mp_chmap_from_channels(&ao->channels, MPMIN(2, num_channels));
+        MP_ERR(ao, "Asked for %d channels, got %d - fallback to %s.\n", req,
+               num_channels, mp_chmap_to_str(&ao->channels));
     }
 
-    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;
-        mp_chmap_from_alsa(&chmap, alsa_chmap);
-
-        MP_VERBOSE(ao, "which we understand as: %s\n", mp_chmap_to_str(&chmap));
-
-        if (p->cfg_ignore_chmap) {
-            MP_VERBOSE(ao, "user set ignore-chmap; ignoring the channel map.\n");
-        } else if (af_fmt_is_spdif(ao->format)) {
-            MP_VERBOSE(ao, "using spdif passthrough; ignoring the channel map.\n");
-        } else if (mp_chmap_is_valid(&chmap)) {
-            // Is it one that contains NA channels?
-            struct mp_chmap without_na = chmap;
-            mp_chmap_remove_na(&without_na);
-
-            if (mp_chmap_is_valid(&without_na) &&
-                !mp_chmap_equals(&without_na, &chmap) &&
-                !mp_chmap_equals(&chmap, &ao->channels) &&
-                !second_try)
-            {
-                // Sometimes, ALSA will advertise certain chmaps, but it's not
-                // possible to set them. This can happen with dmix: as of
-                // alsa 1.0.28, dmix can do stereo only, but advertises the
-                // surround chmaps of the underlying device. In this case,
-                // e.g. setting 6 channels will succeed, but requesting  5.1
-                // afterwards will fail. Then it will return something like
-                // "FL FR NA NA NA NA" as channel map. This means we would
-                // have to pad stereo output to 6 channels with silence, which
-                // would require lots of extra processing. You can't change
-                // the number of channels to 2 either, because the hw params
-                // are already set! So just fuck it and reopen the device with
-                // the chmap "cleaned out" of NA entries.
-                MP_VERBOSE(ao, "Working around braindead ALSA behavior.\n");
-                err = snd_pcm_close(p->alsa);
-                p->alsa = NULL;
-                CHECK_ALSA_ERROR("pcm close error");
-                ao->channels = without_na;
-                return INIT_BRAINDEATH;
-            }
-
-            if (mp_chmap_equals(&chmap, &ao->channels)) {
-                MP_VERBOSE(ao, "which is what we requested.\n");
-            } else if (chmap.num == ao->channels.num) {
-                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");
-        }
-
-        // mpv and ALSA use different conventions for mono
-        if (ao->channels.num == 1) {
-            MP_VERBOSE(ao, "assuming we actually got MONO from ALSA.\n");
-            ao->channels.speaker[0] = MP_SP(FC);
-        }
-
-        free(alsa_chmap);
-    }
-#endif
-
-    snd_pcm_uframes_t bufsize;
-    err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize);
+    err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &p->buffersize);
     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);
+    err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &p->outburst, NULL);
     CHECK_ALSA_ERROR("Unable to get period size");
 
-    MP_VERBOSE(ao, "got period size %li\n", chunk_size);
-    p->outburst = chunk_size;
+    p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
+
+    snd_pcm_sw_params_t *alsa_swparams;
+    snd_pcm_sw_params_alloca(&alsa_swparams);
 
-    /* setting software parameters */
     err = snd_pcm_sw_params_current(p->alsa, alsa_swparams);
     CHECK_ALSA_ERROR("Unable to get sw-parameters");
 
@@ -674,7 +716,7 @@ static int init_device(struct ao *ao, bool second_try)
 
     /* start playing when one period has been written */
     err = snd_pcm_sw_params_set_start_threshold
-            (p->alsa, alsa_swparams, chunk_size);
+            (p->alsa, alsa_swparams, p->outburst);
     CHECK_ALSA_ERROR("Unable to set start threshold");
 
     /* disable underrun reporting */
@@ -690,15 +732,15 @@ static int init_device(struct ao *ao, bool second_try)
     err = snd_pcm_sw_params(p->alsa, alsa_swparams);
     CHECK_ALSA_ERROR("Unable to set sw-parameters");
 
-    /* end setting sw-params */
+    MP_VERBOSE(ao, "hw pausing supported: %s\n", p->can_pause ? "yes" : "no");
+    MP_VERBOSE(ao, "buffersize: %d samples\n", (int)p->buffersize);
+    MP_VERBOSE(ao, "period size: %d samples\n", (int)p->outburst);
 
-    p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
-
-    return INIT_OK;
+    return 0;
 
 alsa_error:
     uninit(ao);
-    return INIT_ERROR;
+    return -1;
 }
 
 static int init(struct ao *ao)
@@ -709,10 +751,33 @@ static int init(struct ao *ao)
 
     MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());
 
-    int r = init_device(ao, false);
-    if (r == INIT_BRAINDEATH)
-        r = init_device(ao, true); // retry with normalized channel layout
-    return r == INIT_OK ? 0 : -1;
+    int r = init_device(ao);
+
+    // Sometimes, ALSA will advertise certain chmaps, but it's not possible to
+    // set them. This can happen with dmix: as of alsa 1.0.29, dmix can do
+    // stereo only, but advertises the surround chmaps of the underlying device.
+    // In this case, e.g. setting 6 channels will succeed, but requesting  5.1
+    // afterwards will fail. Then it will return something like "FL FR NA NA NA NA"
+    // as channel map. This means we would have to pad stereo output to 6
+    // channels with silence, which would require lots of extra processing. You
+    // can't change the number of channels to 2 either, because the hw params
+    // are already set! So just fuck it and reopen the device with the chmap
+    // "cleaned out" of NA entries.
+    if (r >= 0) {
+        struct mp_chmap without_na = ao->channels;
+        mp_chmap_remove_na(&without_na);
+
+        if (mp_chmap_is_valid(&without_na) && without_na.num <= 2 &&
+            ao->channels.num > 2)
+        {
+            MP_VERBOSE(ao, "Working around braindead dmix multichannel behavior.\n");
+            uninit(ao);
+            ao->channels = without_na;
+            r = init_device(ao);
+        }
+    }
+
+    return r;
 }
 
 static void drain(struct ao *ao)
@@ -730,6 +795,8 @@ static int get_space(struct ao *ao)
     snd_pcm_status_alloca(&status);
 
     err = snd_pcm_status(p->alsa, status);
+    if (!check_device_present(ao, err))
+        goto alsa_error;
     CHECK_ALSA_ERROR("cannot get pcm status");
 
     unsigned space = snd_pcm_status_get_avail(status);
@@ -773,7 +840,6 @@ static void audio_pause(struct ao *ao)
             CHECK_ALSA_ERROR("pcm pause error");
         }
     } else {
-        MP_VERBOSE(ao, "pause not supported by hardware\n");
         if (snd_pcm_delay(p->alsa, &p->prepause_frames) < 0
             || p->prepause_frames < 0)
             p->prepause_frames = 0;
@@ -856,9 +922,8 @@ static int play(struct ao *ao, void **data, int samples, int flags)
 
         if (res == -EINTR || res == -EAGAIN) { /* retry */
             res = 0;
-        } else if (res == -ENODEV) {
-            MP_WARN(ao, "Device lost, trying to recover...\n");
-            ao_request_reload(ao);
+        } else if (!check_device_present(ao, res)) {
+            goto alsa_error;
         } else if (res < 0) {
             if (res == -ESTRPIPE) {  /* suspend */
                 resume_device(ao);
@@ -912,6 +977,20 @@ alsa_error:
     return -1;
 }
 
+static bool is_useless_device(char *name)
+{
+    char *crap[] = {"front", "rear", "center_lfe", "side", "surround21",
+        "surround40", "surround41", "surround50", "surround51", "surround71",
+        "sysdefault", "pulse", "null", "dsnoop", "dmix", "hw", "iec958"};
+    for (int i = 0; i < MP_ARRAY_SIZE(crap); i++) {
+        int l = strlen(crap[i]);
+        if (name && strncmp(name, crap[i], l) == 0 &&
+            (!name[l] || name[l] == ':'))
+            return true;
+    }
+    return false;
+}
+
 static void list_devs(struct ao *ao, struct ao_device_list *list)
 {
     void **hints;
@@ -922,7 +1001,7 @@ static void list_devs(struct ao *ao, struct ao_device_list *list)
         char *name = snd_device_name_get_hint(hints[n], "NAME");
         char *desc = snd_device_name_get_hint(hints[n], "DESC");
         char *io = snd_device_name_get_hint(hints[n], "IOID");
-        if (io && strcmp(io, "Output") == 0) {
+        if (!is_useless_device(name) && (!io || strcmp(io, "Output") == 0)) {
             char desc2[1024];
             snprintf(desc2, sizeof(desc2), "%s", desc ? desc : "");
             for (int i = 0; desc2[i]; i++) {

audio/out/ao_coreaudio.c

@@ -180,6 +180,8 @@ static void init_physical_format(struct ao *ao)
     struct priv *p = ao->priv;
     OSErr err;
 
+    void *tmp = talloc_new(NULL);
+
     AudioStreamBasicDescription asbd;
     ca_fill_asbd(ao, &asbd);
 
@@ -190,6 +192,8 @@ static void init_physical_format(struct ao *ao)
                        &streams, &n_streams);
     CHECK_CA_ERROR("could not get number of streams");
 
+    talloc_steal(tmp, streams);
+
     MP_VERBOSE(ao, "Found %zd substream(s).\n", n_streams);
 
     for (int i = 0; i < n_streams; i++) {
@@ -204,6 +208,7 @@ static void init_physical_format(struct ao *ao)
         if (!CHECK_CA_WARN("could not get number of stream formats"))
             continue; // try next one
 
+        talloc_steal(tmp, formats);
 
         uint32_t direction;
         err = CA_GET(streams[i], kAudioStreamPropertyDirection, &direction);
@@ -232,6 +237,12 @@ static void init_physical_format(struct ao *ao)
                          &p->original_asbd);
             CHECK_CA_WARN("could not get current physical stream format");
 
+            if (ca_asbd_equals(&p->original_asbd, &best_asbd)) {
+                MP_VERBOSE(ao, "Requested format already set, not changing.\n");
+                p->original_asbd.mFormatID = 0;
+                break;
+            }
+
             if (!ca_change_physical_format_sync(ao, streams[i], best_asbd))
                 p->original_asbd = (AudioStreamBasicDescription){0};
             break;
@@ -239,6 +250,7 @@ static void init_physical_format(struct ao *ao)
     }
 
 coreaudio_error:
+    talloc_free(tmp);
     return;
 }
 
@@ -372,8 +384,8 @@ static int hotplug_init(struct ao *ao)
         err = AudioObjectAddPropertyListener(
             kAudioObjectSystemObject, &addr, hotplug_cb, (void *)ao);
         if (err != noErr) {
-            char *c1 = fourcc_repr(hotplug_properties[i]);
-            char *c2 = fourcc_repr(err);
+            char *c1 = mp_tag_str(hotplug_properties[i]);
+            char *c2 = mp_tag_str(err);
             MP_ERR(ao, "failed to set device listener %s (%s)", c1, c2);
             goto coreaudio_error;
         }
@@ -397,8 +409,8 @@ static void hotplug_uninit(struct ao *ao)
         err = AudioObjectRemovePropertyListener(
             kAudioObjectSystemObject, &addr, hotplug_cb, (void *)ao);
         if (err != noErr) {
-            char *c1 = fourcc_repr(hotplug_properties[i]);
-            char *c2 = fourcc_repr(err);
+            char *c1 = mp_tag_str(hotplug_properties[i]);
+            char *c2 = mp_tag_str(err);
             MP_ERR(ao, "failed to set device listener %s (%s)", c1, c2);
         }
     }

audio/out/ao_coreaudio_chmap.c

@@ -98,6 +98,9 @@ static AudioChannelLayout *ca_layout_to_custom_layout(struct ao *ao,
     AudioChannelLayout *r;
     OSStatus err;
 
+    if (tag == kAudioChannelLayoutTag_UseChannelDescriptions)
+        return l;
+
     if (tag == kAudioChannelLayoutTag_UseChannelBitmap) {
         uint32_t psize;
         err = AudioFormatGetPropertyInfo(
@@ -109,7 +112,7 @@ static AudioChannelLayout *ca_layout_to_custom_layout(struct ao *ao,
             kAudioFormatProperty_ChannelLayoutForBitmap,
             sizeof(uint32_t), &l->mChannelBitmap, &psize, r);
         CHECK_CA_ERROR("failed to convert channel bitmap to descriptions (get)");
-    } else if (tag != kAudioChannelLayoutTag_UseChannelDescriptions) {
+    } else {
         uint32_t psize;
         err = AudioFormatGetPropertyInfo(
             kAudioFormatProperty_ChannelLayoutForTag,
@@ -120,10 +123,11 @@ static AudioChannelLayout *ca_layout_to_custom_layout(struct ao *ao,
             kAudioFormatProperty_ChannelLayoutForTag,
             sizeof(AudioChannelLayoutTag), &l->mChannelLayoutTag, &psize, r);
         CHECK_CA_ERROR("failed to convert channel tag to descriptions (get)");
-    } else {
-        r = l;
     }
 
+    MP_VERBOSE(ao, "converted input channel layout:\n");
+    ca_log_layout(ao, MSGL_V, l);
+
     return r;
 coreaudio_error:
     return NULL;
@@ -134,22 +138,20 @@ static bool ca_layout_to_mp_chmap(struct ao *ao, AudioChannelLayout *layout,
 {
     void *talloc_ctx = talloc_new(NULL);
 
-    MP_DBG(ao, "input channel layout:\n");
-    ca_log_layout(ao, MSGL_DEBUG, layout);
+    MP_VERBOSE(ao, "input channel layout:\n");
+    ca_log_layout(ao, MSGL_V, layout);
 
     AudioChannelLayout *l = ca_layout_to_custom_layout(ao, talloc_ctx, layout);
     if (!l)
         goto coreaudio_error;
 
-    MP_VERBOSE(ao, "converted input channel layout:\n");
-    ca_log_layout(ao, MSGL_V, l);
-
     if (l->mNumberChannelDescriptions > MP_NUM_CHANNELS) {
         MP_VERBOSE(ao, "layout has too many descriptions (%u, max: %d)\n",
                    (unsigned) l->mNumberChannelDescriptions, MP_NUM_CHANNELS);
         return false;
     }
 
+    chmap->num = l->mNumberChannelDescriptions;
     for (int n = 0; n < l->mNumberChannelDescriptions; n++) {
         AudioChannelLabel label = l->mChannelDescriptions[n].mChannelLabel;
         int speaker = ca_label_to_mp_speaker_id(label);
@@ -157,14 +159,13 @@ static bool ca_layout_to_mp_chmap(struct ao *ao, AudioChannelLayout *layout,
             MP_VERBOSE(ao, "channel label=%u unusable to build channel "
                            "bitmap, skipping layout\n", (unsigned) label);
             goto coreaudio_error;
-        } else {
-            chmap->speaker[n] = speaker;
-            chmap->num = n + 1;
         }
+        chmap->speaker[n] = speaker;
     }
 
     talloc_free(talloc_ctx);
-    return chmap->num > 0;
+    MP_VERBOSE(ao, "mp chmap: %s\n", mp_chmap_to_str(chmap));
+    return mp_chmap_is_valid(chmap) && !mp_chmap_is_unknown(chmap);
 coreaudio_error:
     MP_VERBOSE(ao, "converted input channel layout (failed):\n");
     ca_log_layout(ao, MSGL_V, layout);
@@ -234,34 +235,68 @@ coreaudio_error:
     return r;
 }
 
-bool ca_init_chmap(struct ao *ao, AudioDeviceID device)
+static void ca_retrieve_layouts(struct ao *ao, struct mp_chmap_sel *s,
+                                AudioDeviceID device)
 {
     void *ta_ctx = talloc_new(NULL);
-
-    struct mp_chmap_sel chmap_sel = {.tmp = ta_ctx};
-    struct mp_chmap chmap = {0};
-
-    mp_chmap_sel_add_map(&chmap_sel, &(struct mp_chmap)MP_CHMAP_INIT_MONO);
+    struct mp_chmap chmap;
 
     AudioChannelLayout *ml = ca_query_layout(ao, device, ta_ctx);
     if (ml && ca_layout_to_mp_chmap(ao, ml, &chmap))
-        mp_chmap_sel_add_map(&chmap_sel, &chmap);
+        mp_chmap_sel_add_map(s, &chmap);
 
     AudioChannelLayout *sl = ca_query_stereo_layout(ao, device, ta_ctx);
     if (sl && ca_layout_to_mp_chmap(ao, sl, &chmap))
-        mp_chmap_sel_add_map(&chmap_sel, &chmap);
+        mp_chmap_sel_add_map(s, &chmap);
+
+    talloc_free(ta_ctx);
+}
+
+bool ca_init_chmap(struct ao *ao, AudioDeviceID device)
+{
+    struct mp_chmap_sel chmap_sel = {0};
+    ca_retrieve_layouts(ao, &chmap_sel, device);
+
+    if (!chmap_sel.num_chmaps)
+        mp_chmap_sel_add_map(&chmap_sel, &(struct mp_chmap)MP_CHMAP_INIT_STEREO);
+
+    mp_chmap_sel_add_map(&chmap_sel, &(struct mp_chmap)MP_CHMAP_INIT_MONO);
 
     if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels)) {
         MP_ERR(ao, "could not select a suitable channel map among the "
                    "hardware supported ones. Make sure to configure your "
                    "output device correctly in 'Audio MIDI Setup.app'\n");
-        goto coreaudio_error;
+        return false;
+    }
+    return true;
+}
+
+void ca_get_active_chmap(struct ao *ao, AudioDeviceID device, int channel_count,
+                         struct mp_chmap *out_map)
+{
+    // Apparently, we have to guess by looking back at the supported layouts,
+    // and I haven't found a property that retrieves the actual currently
+    // active channel layout.
+
+    struct mp_chmap_sel chmap_sel = {0};
+    ca_retrieve_layouts(ao, &chmap_sel, device);
+
+    // Use any exact match.
+    for (int n = 0; n < chmap_sel.num_chmaps; n++) {
+        if (chmap_sel.chmaps[n].num == channel_count) {
+            MP_VERBOSE(ao, "mismatching channels - fallback #%d\n", n);
+            *out_map = chmap_sel.chmaps[n];
+            return;
+        }
     }
 
-    talloc_free(ta_ctx);
-    return true;
-
-coreaudio_error:
-    talloc_free(ta_ctx);
-    return false;
+    // Fall back to stereo or mono, and fill the rest with silence. (We don't
+    // know what the device expects. We could use a larger default layout here,
+    // but let's not.)
+    mp_chmap_from_channels(out_map, MPMIN(2, channel_count));
+    out_map->num = channel_count;
+    for (int n = 2; n < out_map->num; n++)
+        out_map->speaker[n] = MP_SPEAKER_ID_NA;
+    MP_WARN(ao, "mismatching channels - falling back to %s\n",
+            mp_chmap_to_str(out_map));
 }

audio/out/ao_coreaudio_chmap.h

@@ -20,6 +20,10 @@
 
 #include <AudioToolbox/AudioToolbox.h>
 
+struct mp_chmap;
+
 bool ca_init_chmap(struct ao *ao, AudioDeviceID device);
+void ca_get_active_chmap(struct ao *ao, AudioDeviceID device, int channel_count,
+                         struct mp_chmap *out_map);
 
 #endif

audio/out/ao_coreaudio_exclusive.c

@@ -300,6 +300,12 @@ static int init(struct ao *ao)
 
     ca_print_asbd(ao, "virtual format", &p->stream_asbd);
 
+    if (p->stream_asbd.mChannelsPerFrame > MP_NUM_CHANNELS) {
+        MP_ERR(ao, "unsupported number of channels: %d > %d.\n",
+               p->stream_asbd.mChannelsPerFrame, MP_NUM_CHANNELS);
+        goto coreaudio_error;
+    }
+
     int new_format = ca_asbd_to_mp_format(&p->stream_asbd);
 
     // If both old and new formats are spdif, avoid changing it due to the
@@ -315,7 +321,10 @@ static int init(struct ao *ao)
     ao->samplerate = p->stream_asbd.mSampleRate;
 
     if (ao->channels.num != p->stream_asbd.mChannelsPerFrame) {
-        // We really expect that ca_init_chmap() fixes the layout to the HW's.
+        ca_get_active_chmap(ao, p->device, p->stream_asbd.mChannelsPerFrame,
+                            &ao->channels);
+    }
+    if (!ao->channels.num) {
         MP_ERR(ao, "number of channels changed, and unknown channel layout!\n");
         goto coreaudio_error;
     }

audio/out/ao_coreaudio_properties.c

@@ -21,7 +21,7 @@
 
 #include "audio/out/ao_coreaudio_properties.h"
 #include "audio/out/ao_coreaudio_utils.h"
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 OSStatus ca_get(AudioObjectID id, ca_scope scope, ca_sel selector,
                 uint32_t size, void *data)

audio/out/ao_coreaudio_utils.c

@@ -51,7 +51,9 @@ static bool ca_is_output_device(struct ao *ao, AudioDeviceID dev)
     size_t n_buffers;
     AudioBufferList *buffers;
     const ca_scope scope = kAudioDevicePropertyStreamConfiguration;
-    CA_GET_ARY_O(dev, scope, &buffers, &n_buffers);
+    OSStatus err = CA_GET_ARY_O(dev, scope, &buffers, &n_buffers);
+    if (err != noErr)
+        return false;
     talloc_free(buffers);
     return n_buffers > 0;
 }
@@ -71,11 +73,16 @@ void ca_get_device_list(struct ao *ao, struct ao_device_list *list)
         char *name;
         char *desc;
         err = CA_GET_STR(devs[i], kAudioDevicePropertyDeviceUID, &name);
+        if (err != noErr) {
+            MP_VERBOSE(ao, "skipping device %d, which has no UID\n", i);
+            talloc_free(ta_ctx);
+            continue;
+        }
         talloc_steal(ta_ctx, name);
         err = CA_GET_STR(devs[i], kAudioObjectPropertyName, &desc);
-        talloc_steal(ta_ctx, desc);
         if (err != noErr)
-            desc = "Unknown";
+            desc = talloc_strdup(NULL, "Unknown");
+        talloc_steal(ta_ctx, desc);
         ao_device_list_add(list, ao, &(struct ao_device_desc){name, desc});
         talloc_free(ta_ctx);
     }
@@ -129,36 +136,11 @@ coreaudio_error:
     return err;
 }
 
-char *fourcc_repr_buf(char *buf, size_t buf_size, uint32_t code)
-{
-    // Extract FourCC letters from the uint32_t and finde out if it's a valid
-    // code that is made of letters.
-    unsigned char fcc[4] = {
-        (code >> 24) & 0xFF,
-        (code >> 16) & 0xFF,
-        (code >> 8)  & 0xFF,
-        code         & 0xFF,
-    };
-
-    bool valid_fourcc = true;
-    for (int i = 0; i < 4; i++) {
-        if (fcc[i] < 32 || fcc[i] >= 128)
-            valid_fourcc = false;
-    }
-
-    if (valid_fourcc)
-        snprintf(buf, buf_size, "'%c%c%c%c'", fcc[0], fcc[1], fcc[2], fcc[3]);
-    else
-        snprintf(buf, buf_size, "%u", (unsigned int)code);
-
-    return buf;
-}
-
 bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message)
 {
     if (code == noErr) return true;
 
-    mp_msg(ao->log, level, "%s (%s)\n", message, fourcc_repr(code));
+    mp_msg(ao->log, level, "%s (%s)\n", message, mp_tag_str(code));
 
     return false;
 }
@@ -251,7 +233,7 @@ void ca_print_asbd(struct ao *ao, const char *description,
                    const AudioStreamBasicDescription *asbd)
 {
     uint32_t flags  = asbd->mFormatFlags;
-    char *format    = fourcc_repr(asbd->mFormatID);
+    char *format    = mp_tag_str(asbd->mFormatID);
     int mpfmt       = ca_asbd_to_mp_format(asbd);
 
     MP_VERBOSE(ao,
@@ -463,7 +445,7 @@ int64_t ca_get_device_latency_us(struct ao *ao, AudioDeviceID device)
         if (err == noErr) {
             latency_frames += temp;
             MP_VERBOSE(ao, "Latency property %s: %d frames\n",
-                       fourcc_repr(latency_properties[n]), (int)temp);
+                       mp_tag_str(latency_properties[n]), (int)temp);
         }
     }
 
@@ -498,6 +480,8 @@ bool ca_change_physical_format_sync(struct ao *ao, AudioStreamID stream,
     err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &prev_format);
     CHECK_CA_ERROR("can't get current physical format");
 
+    ca_print_asbd(ao, "format in use before switching:", &prev_format);
+
     /* Install the callback. */
     AudioObjectPropertyAddress p_addr = {
         .mSelector = kAudioStreamPropertyPhysicalFormat,
@@ -516,7 +500,7 @@ bool ca_change_physical_format_sync(struct ao *ao, AudioStreamID stream,
 
     /* The AudioStreamSetProperty is not only asynchronous,
      * it is also not Atomic, in its behaviour. */
-    struct timespec timeout = mp_rel_time_to_timespec(0.5);
+    struct timespec timeout = mp_rel_time_to_timespec(2.0);
     AudioStreamBasicDescription actual_format = {0};
     while (1) {
         err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format);

audio/out/ao_coreaudio_utils.h

@@ -31,9 +31,6 @@
 CFStringRef cfstr_from_cstr(char *str);
 char *cfstr_get_cstr(CFStringRef cfstr);
 
-char *fourcc_repr_buf(char *buf, size_t buf_size, uint32_t code);
-#define fourcc_repr(code) fourcc_repr_buf((char[40]){0}, 40, code)
-
 bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message);
 
 #define CHECK_CA_ERROR_L(label, message) \

audio/out/ao_dsound.c

@@ -1,707 +0,0 @@
-/*
- * Windows DirectSound interface
- *
- * Copyright (c) 2004 Gabor Szecsi <deje@miki.hu>
- *
- * 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/>.
- */
-
-/**
-\todo verify/extend multichannel support
-*/
-
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <windows.h>
-#define DIRECTSOUND_VERSION 0x0600
-#include <dsound.h>
-#include <math.h>
-
-#include <libavutil/avutil.h>
-#include <libavutil/common.h>
-
-#include "config.h"
-#include "audio/format.h"
-#include "ao.h"
-#include "internal.h"
-#include "common/msg.h"
-#include "osdep/timer.h"
-#include "osdep/io.h"
-#include "options/m_option.h"
-
-/**
-\todo use the definitions from the win32 api headers when they define these
-*/
-#define WAVE_FORMAT_IEEE_FLOAT 0x0003
-#define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
-#define WAVE_FORMAT_EXTENSIBLE 0xFFFE
-
-static const GUID KSDATAFORMAT_SUBTYPE_PCM = {
-    0x1, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}
-};
-
-#if 0
-#define DSSPEAKER_HEADPHONE         0x00000001
-#define DSSPEAKER_MONO              0x00000002
-#define DSSPEAKER_QUAD              0x00000003
-#define DSSPEAKER_STEREO            0x00000004
-#define DSSPEAKER_SURROUND          0x00000005
-#define DSSPEAKER_5POINT1           0x00000006
-#endif
-
-#ifndef _WAVEFORMATEXTENSIBLE_
-typedef struct {
-    WAVEFORMATEX Format;
-    union {
-        WORD wValidBitsPerSample;       /* bits of precision  */
-        WORD wSamplesPerBlock;          /* valid if wBitsPerSample==0 */
-        WORD wReserved;                 /* If neither applies, set to zero. */
-    } Samples;
-    DWORD dwChannelMask;                /* which channels are */
-                                        /* present in stream  */
-    GUID SubFormat;
-} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
-#endif
-
-struct priv {
-    HINSTANCE hdsound_dll;          ///handle to the dll
-    LPDIRECTSOUND hds;              ///direct sound object
-    LPDIRECTSOUNDBUFFER hdspribuf;  ///primary direct sound buffer
-    LPDIRECTSOUNDBUFFER hdsbuf;     ///secondary direct sound buffer (stream buffer)
-    int buffer_size;                ///size in bytes of the direct sound buffer
-    int write_offset;               ///offset of the write cursor in the direct sound buffer
-    int min_free_space;             ///if the free space is below this value get_space() will return 0
-                                    ///there will always be at least this amout of free space to prevent
-                                    ///get_space() from returning wrong values when buffer is 100% full.
-                                    ///will be replaced with nBlockAlign in init()
-    int underrun_check;             ///0 or last reported free space (underrun detection)
-    int device_num;                 ///wanted device number
-    GUID device;                    ///guid of the device
-    int audio_volume;
-
-    int device_index;
-
-    int outburst;                   ///play in multiple of chunks of this size
-
-    int cfg_device;
-    int cfg_buffersize;
-
-    struct ao_device_list *listing; ///temporary during list_devs()
-};
-
-/***************************************************************************************/
-
-/**
-\brief output error message
-\param err error code
-\return string with the error message
-*/
-static char * dserr2str(int err)
-{
-    switch (err) {
-    case DS_OK:                         return "DS_OK";
-    case DS_NO_VIRTUALIZATION:          return "DS_NO_VIRTUALIZATION";
-    case DSERR_ALLOCATED:               return "DS_NO_VIRTUALIZATION";
-    case DSERR_CONTROLUNAVAIL:          return "DSERR_CONTROLUNAVAIL";
-    case DSERR_INVALIDPARAM:            return "DSERR_INVALIDPARAM";
-    case DSERR_INVALIDCALL:             return "DSERR_INVALIDCALL";
-    case DSERR_GENERIC:                 return "DSERR_GENERIC";
-    case DSERR_PRIOLEVELNEEDED:         return "DSERR_PRIOLEVELNEEDED";
-    case DSERR_OUTOFMEMORY:             return "DSERR_OUTOFMEMORY";
-    case DSERR_BADFORMAT:               return "DSERR_BADFORMAT";
-    case DSERR_UNSUPPORTED:             return "DSERR_UNSUPPORTED";
-    case DSERR_NODRIVER:                return "DSERR_NODRIVER";
-    case DSERR_ALREADYINITIALIZED:      return "DSERR_ALREADYINITIALIZED";
-    case DSERR_NOAGGREGATION:           return "DSERR_NOAGGREGATION";
-    case DSERR_BUFFERLOST:              return "DSERR_BUFFERLOST";
-    case DSERR_OTHERAPPHASPRIO:         return "DSERR_OTHERAPPHASPRIO";
-    case DSERR_UNINITIALIZED:           return "DSERR_UNINITIALIZED";
-    case DSERR_NOINTERFACE:             return "DSERR_NOINTERFACE";
-    case DSERR_ACCESSDENIED:            return "DSERR_ACCESSDENIED";
-    }
-    return "unknown";
-}
-
-/**
-\brief uninitialize direct sound
-*/
-static void UninitDirectSound(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    // finally release the DirectSound object
-    if (p->hds) {
-        IDirectSound_Release(p->hds);
-        p->hds = NULL;
-    }
-    // free DSOUND.DLL
-    if (p->hdsound_dll) {
-        FreeLibrary(p->hdsound_dll);
-        p->hdsound_dll = NULL;
-    }
-    MP_VERBOSE(ao, "DirectSound uninitialized\n");
-}
-
-/**
-\brief enumerate direct sound devices
-\return TRUE to continue with the enumeration
-*/
-static BOOL CALLBACK DirectSoundEnum(LPGUID guid, LPCSTR desc, LPCSTR module,
-                                     LPVOID context)
-{
-    struct ao *ao = context;
-    struct priv *p = ao->priv;
-
-    MP_VERBOSE(ao, "%i %s ", p->device_index, desc);
-    if (p->device_num == p->device_index) {
-        MP_VERBOSE(ao, "<--");
-        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(L"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
-\return 0 if error, 1 if ok
-*/
-static int InitDirectSound(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    DSCAPS dscaps;
-
-    if (!LoadDirectSound(ao))
-        return 0;
-
-    HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
-    OurDirectSoundCreate =
-        (void *)GetProcAddress(p->hdsound_dll, "DirectSoundCreate");
-
-    if (OurDirectSoundCreate == NULL) {
-        MP_ERR(ao, "GetProcAddress FAILED\n");
-        FreeLibrary(p->hdsound_dll);
-        return 0;
-    }
-
-    EnumDevs(ao);
-
-    // Create the direct sound object
-    if (FAILED(OurDirectSoundCreate((p->device_num > 0) ? &p->device : NULL,
-                                    &p->hds, NULL)))
-    {
-        MP_ERR(ao, "cannot create a DirectSound device\n");
-        FreeLibrary(p->hdsound_dll);
-        return 0;
-    }
-
-    /* Set DirectSound Cooperative level, ie what control we want over Windows
-     * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
-     * settings of the primary buffer, but also that only the sound of our
-     * application will be audible when it will have the focus.
-     * !!! (this is not really working as intended yet because to set the
-     * cooperative level you need the window handle of your application, and
-     * I don't know of any easy way to get it. Especially since we might play
-     * sound without any video, and so what window handle should we use ???
-     * The hack for now is to use the Desktop window handle - it seems to be
-     * working */
-    if (IDirectSound_SetCooperativeLevel(p->hds, GetDesktopWindow(),
-                                         DSSCL_EXCLUSIVE))
-    {
-        MP_ERR(ao, "cannot set direct sound cooperative level\n");
-        IDirectSound_Release(p->hds);
-        FreeLibrary(p->hdsound_dll);
-        return 0;
-    }
-    MP_VERBOSE(ao, "DirectSound initialized\n");
-
-    memset(&dscaps, 0, sizeof(DSCAPS));
-    dscaps.dwSize = sizeof(DSCAPS);
-    if (DS_OK == IDirectSound_GetCaps(p->hds, &dscaps)) {
-        if (dscaps.dwFlags & DSCAPS_EMULDRIVER)
-            MP_VERBOSE(ao, "DirectSound is emulated\n");
-    } else {
-        MP_VERBOSE(ao, "cannot get device capabilities\n");
-    }
-
-    return 1;
-}
-
-/**
-\brief destroy the direct sound buffer
-*/
-static void DestroyBuffer(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    if (p->hdsbuf) {
-        IDirectSoundBuffer_Release(p->hdsbuf);
-        p->hdsbuf = NULL;
-    }
-    if (p->hdspribuf) {
-        IDirectSoundBuffer_Release(p->hdspribuf);
-        p->hdspribuf = NULL;
-    }
-}
-
-/**
-\brief fill sound buffer
-\param data pointer to the sound data to copy
-\param len length of the data to copy in bytes
-\return number of copyed bytes
-*/
-static int write_buffer(struct ao *ao, unsigned char *data, int len)
-{
-    struct priv *p = ao->priv;
-    HRESULT res;
-    LPVOID lpvPtr1;
-    DWORD dwBytes1;
-    LPVOID lpvPtr2;
-    DWORD dwBytes2;
-
-    p->underrun_check = 0;
-
-    // Lock the buffer
-    res = IDirectSoundBuffer_Lock(p->hdsbuf, p->write_offset, len, &lpvPtr1,
-                                  &dwBytes1, &lpvPtr2, &dwBytes2, 0);
-    // If the buffer was lost, restore and retry lock.
-    if (DSERR_BUFFERLOST == res) {
-        IDirectSoundBuffer_Restore(p->hdsbuf);
-        res = IDirectSoundBuffer_Lock(p->hdsbuf, p->write_offset, len, &lpvPtr1,
-                                      &dwBytes1, &lpvPtr2, &dwBytes2, 0);
-    }
-
-
-    if (SUCCEEDED(res)) {
-        memcpy(lpvPtr1, data, dwBytes1);
-        if (NULL != lpvPtr2)
-            memcpy(lpvPtr2, data + dwBytes1, dwBytes2);
-        p->write_offset += dwBytes1 + dwBytes2;
-        if (p->write_offset >= p->buffer_size)
-            p->write_offset = dwBytes2;
-
-        // Release the data back to DirectSound.
-        res = IDirectSoundBuffer_Unlock(p->hdsbuf, lpvPtr1, dwBytes1, lpvPtr2,
-                                        dwBytes2);
-        if (SUCCEEDED(res)) {
-            // Success.
-            DWORD status;
-            IDirectSoundBuffer_GetStatus(p->hdsbuf, &status);
-            if (!(status & DSBSTATUS_PLAYING))
-                res = IDirectSoundBuffer_Play(p->hdsbuf, 0, 0, DSBPLAY_LOOPING);
-            return dwBytes1 + dwBytes2;
-        }
-    }
-    // Lock, Unlock, or Restore failed.
-    return 0;
-}
-
-/***************************************************************************************/
-
-/**
-\brief handle control commands
-\param cmd command
-\param arg argument
-\return CONTROL_OK or CONTROL_UNKNOWN in case the command is not supported
-*/
-static int control(struct ao *ao, enum aocontrol cmd, void *arg)
-{
-    struct priv *p = ao->priv;
-    DWORD volume;
-    switch (cmd) {
-    case AOCONTROL_GET_VOLUME: {
-        ao_control_vol_t *vol = (ao_control_vol_t *)arg;
-        vol->left = vol->right = p->audio_volume;
-        return CONTROL_OK;
-    }
-    case AOCONTROL_SET_VOLUME: {
-        ao_control_vol_t *vol = (ao_control_vol_t *)arg;
-        volume = p->audio_volume = vol->right;
-        if (volume < 1)
-            volume = 1;
-        volume = (DWORD)(log10(volume) * 5000.0) - 10000;
-        IDirectSoundBuffer_SetVolume(p->hdsbuf, volume);
-        return CONTROL_OK;
-    }
-    case AOCONTROL_HAS_SOFT_VOLUME:
-        return CONTROL_TRUE;
-    }
-    return CONTROL_UNKNOWN;
-}
-
-/**
-\brief setup sound device
-\param rate samplerate
-\param channels number of channels
-\param format format
-\param flags unused
-\return 0=success -1=fail
-*/
-static int init(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-    int res;
-
-    if (!InitDirectSound(ao))
-        return -1;
-
-    p->audio_volume = 100;
-
-    // ok, now create the buffers
-    WAVEFORMATEXTENSIBLE wformat;
-    DSBUFFERDESC dsbpridesc;
-    DSBUFFERDESC dsbdesc;
-    int format = af_fmt_from_planar(ao->format);
-    int rate = ao->samplerate;
-
-    if (!af_fmt_is_spdif(format)) {
-        struct mp_chmap_sel sel = {0};
-        mp_chmap_sel_add_waveext(&sel);
-        if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
-            return -1;
-    }
-    switch (format) {
-    case AF_FORMAT_S24:
-    case AF_FORMAT_S16:
-    case AF_FORMAT_U8:
-        break;
-    default:
-        if (af_fmt_is_spdif(format))
-            break;
-        MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
-                   af_fmt_to_str(format));
-        format = AF_FORMAT_S16;
-    }
-    //set our audio parameters
-    ao->samplerate = rate;
-    ao->format = format;
-    ao->bps = ao->channels.num * rate * af_fmt_to_bytes(format);
-    int buffersize = ao->bps * p->cfg_buffersize / 1000;
-    MP_VERBOSE(ao, "Samplerate:%iHz Channels:%i Format:%s\n", rate,
-               ao->channels.num, af_fmt_to_str(format));
-    MP_VERBOSE(ao, "Buffersize:%d bytes (%f msec)\n",
-               buffersize, buffersize * 1000.0 / ao->bps);
-
-    //fill waveformatex
-    ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
-    wformat.Format.cbSize = (ao->channels.num > 2)
-                    ? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
-    wformat.Format.nChannels = ao->channels.num;
-    wformat.Format.nSamplesPerSec = rate;
-    if (af_fmt_is_spdif(format)) {
-        // Whether it also works with e.g. DTS is unknown, but probably does.
-        wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
-        wformat.Format.wBitsPerSample = 16;
-        wformat.Format.nBlockAlign = 4;
-    } else {
-        wformat.Format.wFormatTag = (ao->channels.num > 2)
-                                    ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
-        int bps = af_fmt_to_bytes(format);
-        wformat.Format.wBitsPerSample = bps * 8;
-        wformat.Format.nBlockAlign = wformat.Format.nChannels * bps;
-    }
-
-    // fill in primary sound buffer descriptor
-    memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC));
-    dsbpridesc.dwSize = sizeof(DSBUFFERDESC);
-    dsbpridesc.dwFlags       = DSBCAPS_PRIMARYBUFFER;
-    dsbpridesc.dwBufferBytes = 0;
-    dsbpridesc.lpwfxFormat   = NULL;
-
-    // fill in the secondary sound buffer (=stream buffer) descriptor
-    memset(&dsbdesc, 0, sizeof(DSBUFFERDESC));
-    dsbdesc.dwSize = sizeof(DSBUFFERDESC);
-    dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */
-                      | DSBCAPS_GLOBALFOCUS       /** Allows background playing */
-                      | DSBCAPS_CTRLVOLUME;       /** volume control enabled */
-
-    if (ao->channels.num > 2) {
-        wformat.dwChannelMask = mp_chmap_to_waveext(&ao->channels);
-        wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
-        wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
-        // Needed for 5.1 on emu101k - shit soundblaster
-        dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE;
-    }
-    wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec *
-                                     wformat.Format.nBlockAlign;
-
-    dsbdesc.dwBufferBytes = buffersize;
-    dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
-    p->buffer_size = dsbdesc.dwBufferBytes;
-    p->write_offset = 0;
-    p->min_free_space = wformat.Format.nBlockAlign;
-    p->outburst = wformat.Format.nBlockAlign * 512;
-
-    // create primary buffer and set its format
-
-    res = IDirectSound_CreateSoundBuffer(p->hds, &dsbpridesc, &p->hdspribuf, NULL);
-    if (res != DS_OK) {
-        UninitDirectSound(ao);
-        MP_ERR(ao, "cannot create primary buffer (%s)\n", dserr2str(res));
-        return -1;
-    }
-    res = IDirectSoundBuffer_SetFormat(p->hdspribuf, (WAVEFORMATEX *)&wformat);
-    if (res != DS_OK) {
-        MP_WARN(ao, "cannot set primary buffer format (%s), using "
-                "standard setting (bad quality)", dserr2str(res));
-    }
-
-    MP_VERBOSE(ao, "primary buffer created\n");
-
-    // now create the stream buffer
-
-    res = IDirectSound_CreateSoundBuffer(p->hds, &dsbdesc, &p->hdsbuf, NULL);
-    if (res != DS_OK) {
-        if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) {
-            // Try without DSBCAPS_LOCHARDWARE
-            dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE;
-            res = IDirectSound_CreateSoundBuffer(p->hds, &dsbdesc, &p->hdsbuf, NULL);
-        }
-        if (res != DS_OK) {
-            UninitDirectSound(ao);
-            MP_ERR(ao, "cannot create secondary (stream)buffer (%s)\n",
-                   dserr2str(res));
-            return -1;
-        }
-    }
-    MP_VERBOSE(ao, "secondary (stream)buffer created\n");
-    return 0;
-}
-
-
-
-/**
-\brief stop playing and empty buffers (for seeking/pause)
-*/
-static void reset(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    IDirectSoundBuffer_Stop(p->hdsbuf);
-    // reset directsound buffer
-    IDirectSoundBuffer_SetCurrentPosition(p->hdsbuf, 0);
-    p->write_offset = 0;
-    p->underrun_check = 0;
-}
-
-/**
-\brief stop playing, keep buffers (for pause)
-*/
-static void audio_pause(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    IDirectSoundBuffer_Stop(p->hdsbuf);
-}
-
-/**
-\brief resume playing, after audio_pause()
-*/
-static void audio_resume(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    IDirectSoundBuffer_Play(p->hdsbuf, 0, 0, DSBPLAY_LOOPING);
-}
-
-/**
-\brief close audio device
-\param immed stop playback immediately
-*/
-static void uninit(struct ao *ao)
-{
-    reset(ao);
-
-    DestroyBuffer(ao);
-    UninitDirectSound(ao);
-}
-
-// return exact number of free (safe to write) bytes
-static int check_free_buffer_size(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-    int space;
-    DWORD play_offset;
-    IDirectSoundBuffer_GetCurrentPosition(p->hdsbuf, &play_offset, NULL);
-    space = p->buffer_size - (p->write_offset - play_offset);
-    // |              | <-- const --> |                |                 |
-    // buffer start   play_cursor     write_cursor     p->write_offset   buffer end
-    // play_cursor is the actual position of the play cursor
-    // write_cursor is the position after which it is assumed to be save to write data
-    // p->write_offset is the position where we actually write the data to
-    if (space > p->buffer_size)
-        space -= p->buffer_size;                        // p->write_offset < play_offset
-    // Check for buffer underruns. An underrun happens if DirectSound
-    // started to play old data beyond the current p->write_offset. Detect this
-    // by checking whether the free space shrinks, even though no data was
-    // written (i.e. no write_buffer). Doesn't always work, but the only
-    // reason we need this is to deal with the situation when playback ends,
-    // and the buffer is only half-filled.
-    if (space < p->underrun_check) {
-        // there's no useful data in the buffers
-        space = p->buffer_size;
-        reset(ao);
-    }
-    p->underrun_check = space;
-    return space;
-}
-
-/**
-   \brief find out how many bytes can be written into the audio buffer without
-   \return free space in bytes, has to return 0 if the buffer is almost full
- */
-static int get_space(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-
-    int space = check_free_buffer_size(ao);
-    if (space < p->min_free_space)
-        return 0;
-    return (space - p->min_free_space) / p->outburst * p->outburst / ao->sstride;
-}
-
-/**
-\brief play 'len' bytes of 'data'
-\param data pointer to the data to play
-\param len size in bytes of the data buffer, gets rounded down to outburst*n
-\param flags currently unused
-\return number of played bytes
-*/
-static int play(struct ao *ao, void **data, int samples, int flags)
-{
-    struct priv *p = ao->priv;
-    int len = samples * ao->sstride;
-
-    int space = check_free_buffer_size(ao);
-    if (space < len)
-        len = space;
-
-    if (!(flags & AOPLAY_FINAL_CHUNK))
-        len = (len / p->outburst) * p->outburst;
-    return write_buffer(ao, data[0], len) / ao->sstride;
-}
-
-/**
-\brief get the delay between the first and last sample in the buffer
-\return delay in seconds
-*/
-static double 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;
-}
-
-#define OPT_BASE_STRUCT struct priv
-
-const struct ao_driver audio_out_dsound = {
-    .description = "Windows DirectSound audio output",
-    .name      = "dsound",
-    .init      = init,
-    .uninit    = uninit,
-    .control   = control,
-    .get_space = get_space,
-    .play      = play,
-    .get_delay = get_delay,
-    .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_INTRANGE("buffersize", cfg_buffersize, 0, 1, 10000, OPTDEF_INT(200)),
-        {0}
-    },
-};

audio/out/ao_jack.c

@@ -162,10 +162,6 @@ static int init(struct ao *ao)
         mp_chmap_sel_add_waveext(&sel);
         break;
 
-    case 1:
-        mp_chmap_sel_add_alsa_def(&sel);
-        break;
-
     default:
         mp_chmap_sel_add_any(&sel);
     }
@@ -236,7 +232,7 @@ const struct ao_driver audio_out_jack = {
         OPT_FLAG("autostart", autostart, 0),
         OPT_FLAG("connect", connect, 0),
         OPT_CHOICE("std-channel-layout", stdlayout, 0,
-                ({"waveext", 0}, {"alsa", 1}, {"any", 2})),
+                ({"waveext", 0}, {"any", 1})),
         {0}
     },
 };

audio/out/ao_lavc.c

@@ -31,7 +31,7 @@
 #include "common/common.h"
 #include "audio/format.h"
 #include "audio/fmt-conversion.h"
-#include "talloc.h"
+#include "mpv_talloc.h"
 #include "ao.h"
 #include "internal.h"
 #include "common/msg.h"
@@ -164,6 +164,9 @@ static int init(struct ao *ao)
 
     ao->untimed = true;
 
+    if (ao->channels.num > AV_NUM_DATA_POINTERS)
+        goto fail;
+
     pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
     return 0;
 

audio/out/ao_null.c

@@ -26,7 +26,7 @@
 #include <stdlib.h>
 #include <math.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 #include "config.h"
 #include "osdep/timer.h"

audio/out/ao_openal.c

@@ -44,8 +44,7 @@
 
 #define MAX_CHANS MP_NUM_CHANNELS
 #define NUM_BUF 128
-#define CHUNK_SIZE 512
-#define CHUNK_SAMPLES (CHUNK_SIZE / 2)
+#define CHUNK_SAMPLES 256
 static ALuint buffers[MAX_CHANS][NUM_BUF];
 static ALuint sources[MAX_CHANS];
 
@@ -56,6 +55,8 @@ static struct ao *ao_data;
 
 struct priv {
     char *cfg_device;
+    ALenum al_format;
+    int chunk_size;
 };
 
 static void reset(struct ao *ao);
@@ -117,22 +118,51 @@ struct speaker {
 };
 
 static const struct speaker speaker_pos[] = {
-    {MP_SPEAKER_ID_FL,   {-1, 0, 0.5}},
-    {MP_SPEAKER_ID_FR,   { 1, 0, 0.5}},
-    {MP_SPEAKER_ID_FC,   { 0, 0,   1}},
-    {MP_SPEAKER_ID_LFE,  { 0, 0, 0.1}},
-    {MP_SPEAKER_ID_BL,   {-1, 0,  -1}},
-    {MP_SPEAKER_ID_BR,   { 1, 0,  -1}},
-    {MP_SPEAKER_ID_BC,   { 0, 0,  -1}},
-    {MP_SPEAKER_ID_SL,   {-1, 0,   0}},
-    {MP_SPEAKER_ID_SR,   { 1, 0,   0}},
+    {MP_SPEAKER_ID_FL,   {-0.500,  0, -0.866}}, // -30 deg
+    {MP_SPEAKER_ID_FR,   { 0.500,  0, -0.866}}, //  30 deg
+    {MP_SPEAKER_ID_FC,   {     0,  0,     -1}}, //   0 deg
+    {MP_SPEAKER_ID_LFE,  {     0, -1,      0}}, //   below
+    {MP_SPEAKER_ID_BL,   {-0.609,  0,  0.793}}, // -142.5 deg
+    {MP_SPEAKER_ID_BR,   { 0.609,  0,  0.793}}, //  142.5 deg
+    {MP_SPEAKER_ID_BC,   {     0,  0,      1}}, //  180 deg
+    {MP_SPEAKER_ID_SL,   {-0.985,  0,  0.174}}, // -100 deg
+    {MP_SPEAKER_ID_SR,   { 0.985,  0,  0.174}}, //  100 deg
     {-1},
 };
 
+static ALenum get_al_format(int format)
+{
+    switch (format) {
+    case AF_FORMAT_U8P: return AL_FORMAT_MONO8;
+    case AF_FORMAT_S16P: return  AL_FORMAT_MONO16;
+    case AF_FORMAT_FLOATP:
+        if (alIsExtensionPresent((ALchar*)"AL_EXT_float32") == AL_TRUE)
+            return AL_FORMAT_MONO_FLOAT32;
+        break;
+    case AF_FORMAT_DOUBLEP:
+        if (alIsExtensionPresent((ALchar*)"AL_EXT_double") == AL_TRUE)
+            return AL_FORMAT_MONO_DOUBLE_EXT;
+        break;
+    }
+    return AL_FALSE;
+}
+
+// close audio device
+static void uninit(struct ao *ao)
+{
+    ALCcontext *ctx = alcGetCurrentContext();
+    ALCdevice *dev = alcGetContextsDevice(ctx);
+    reset(ao);
+    alcMakeContextCurrent(NULL);
+    alcDestroyContext(ctx);
+    alcCloseDevice(dev);
+    ao_data = NULL;
+}
+
 static int init(struct ao *ao)
 {
     float position[3] = {0, 0, 0};
-    float direction[6] = {0, 0, 1, 0, -1, 0};
+    float direction[6] = {0, 0, -1, 0, 1, 0};
     ALCdevice *dev = NULL;
     ALCcontext *ctx = NULL;
     ALCint freq = 0;
@@ -184,25 +214,31 @@ static int init(struct ao *ao)
     alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
     if (alcGetError(dev) == ALC_NO_ERROR && freq)
         ao->samplerate = freq;
-    ao->format = AF_FORMAT_S16P;
+
+    p->al_format = AL_FALSE;
+    int try_formats[AF_FORMAT_COUNT];
+    af_get_best_sample_formats(ao->format, try_formats);
+    for (int n = 0; try_formats[n]; n++) {
+        p->al_format = get_al_format(try_formats[n]);
+        if (p->al_format != AL_FALSE) {
+            ao->format = try_formats[n];
+            break;
+        }
+    }
+
+    if (p->al_format == AL_FALSE) {
+        MP_FATAL(ao, "Can't find appropriate sample format.\n");
+        uninit(ao);
+        goto err_out;
+    }
+
+    p->chunk_size = CHUNK_SAMPLES * af_fmt_to_bytes(ao->format);
     return 0;
 
 err_out:
     return -1;
 }
 
-// close audio device
-static void uninit(struct ao *ao)
-{
-    ALCcontext *ctx = alcGetCurrentContext();
-    ALCdevice *dev = alcGetContextsDevice(ctx);
-    reset(ao);
-    alcMakeContextCurrent(NULL);
-    alcDestroyContext(ctx);
-    alcCloseDevice(dev);
-    ao_data = NULL;
-}
-
 static void drain(struct ao *ao)
 {
     ALint state;
@@ -274,14 +310,15 @@ static int get_space(struct ao *ao)
  */
 static int play(struct ao *ao, void **data, int samples, int flags)
 {
+    struct priv *p = ao->priv;
     ALint state;
     int num = samples / CHUNK_SAMPLES;
     for (int i = 0; i < num; i++) {
         for (int ch = 0; ch < ao->channels.num; ch++) {
-            int16_t *d = data[ch];
-            d += i * CHUNK_SAMPLES;
-            alBufferData(buffers[ch][cur_buf[ch]], AL_FORMAT_MONO16, d,
-                         CHUNK_SIZE, ao->samplerate);
+            char *d = data[ch];
+            d += i * p->chunk_size;
+            alBufferData(buffers[ch][cur_buf[ch]], p->al_format, d,
+                         p->chunk_size, ao->samplerate);
             alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
             cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
         }

audio/out/ao_pcm.c

@@ -27,7 +27,7 @@
 
 #include <libavutil/common.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 #include "options/m_option.h"
 #include "audio/format.h"

audio/out/ao_pulse.c

@@ -376,22 +376,8 @@ fail:
     return -1;
 }
 
-static int init(struct ao *ao)
+static bool set_format(struct ao *ao, pa_format_info *format)
 {
-    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 (!(format = pa_format_info_new()))
-        goto unlock_and_fail;
-
     ao->format = af_fmt_from_planar(ao->format);
 
     format->encoding = map_digital_format(ao->format);
@@ -411,8 +397,29 @@ static int init(struct ao *ao)
         pa_format_info_set_sample_format(format, fmt_map->pa_format);
     }
 
+    struct pa_channel_map map;
+
     if (!select_chmap(ao, &map))
-        goto unlock_and_fail;
+        return false;
+
+    pa_format_info_set_rate(format, ao->samplerate);
+    pa_format_info_set_channels(format, ao->channels.num);
+    pa_format_info_set_channel_map(format, &map);
+
+    return ao->samplerate < PA_RATE_MAX && pa_format_info_valid(format);
+}
+
+static int init(struct ao *ao)
+{
+    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 (!(proplist = pa_proplist_new())) {
         MP_ERR(ao, "Failed to allocate proplist\n");
@@ -420,13 +427,17 @@ static int init(struct ao *ao)
     }
     (void)pa_proplist_sets(proplist, PA_PROP_MEDIA_ICON_NAME, ao->client_name);
 
-    pa_format_info_set_rate(format, ao->samplerate);
-    pa_format_info_set_channels(format, ao->channels.num);
-    pa_format_info_set_channel_map(format, &map);
-
-    if (!pa_format_info_valid(format)) {
-        MP_ERR(ao, "Invalid audio format\n");
+    if (!(format = pa_format_info_new()))
         goto unlock_and_fail;
+
+    if (!set_format(ao, format)) {
+        ao->channels = (struct mp_chmap) MP_CHMAP_INIT_STEREO;
+        ao->samplerate = 48000;
+        ao->format = AF_FORMAT_FLOAT;
+        if (!set_format(ao, format)) {
+            MP_ERR(ao, "Invalid audio format\n");
+            goto unlock_and_fail;
+        }
     }
 
     if (!(priv->stream = pa_stream_new_extended(priv->context, "audio stream",

audio/out/ao_rsound.c

@@ -26,7 +26,7 @@
 #include <unistd.h>
 #include <rsound.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 #include "options/m_option.h"
 #include "osdep/timer.h"

audio/out/ao_sdl.c

@@ -20,7 +20,7 @@
 
 #include "config.h"
 #include "audio/format.h"
-#include "talloc.h"
+#include "mpv_talloc.h"
 #include "ao.h"
 #include "internal.h"
 #include "common/common.h"

audio/out/ao_wasapi.c

@@ -17,49 +17,61 @@
  * with mpv.  If not, see <http://www.gnu.org/licenses/>.
  */
 
-#include <stdlib.h>
+#include <math.h>
 #include <inttypes.h>
-#include <process.h>
-#include <initguid.h>
-#include <audioclient.h>
-#include <endpointvolume.h>
-#include <mmdeviceapi.h>
-#include <avrt.h>
+#include <libavutil/mathematics.h>
 
-#include "audio/out/ao_wasapi.h"
-#include "audio/out/ao_wasapi_utils.h"
-
-#include "audio/format.h"
+#include "options/m_option.h"
 #include "osdep/timer.h"
 #include "osdep/io.h"
+#include "ao_wasapi.h"
 
-static HRESULT get_device_delay(struct wasapi_state *state, double *delay) {
+// naive av_rescale for unsigned
+static UINT64 uint64_scale(UINT64 x, UINT64 num, UINT64 den)
+{
+    return (x / den) * num
+        + ((x % den) * (num / den))
+        + ((x % den) * (num % den)) / den;
+}
+
+static HRESULT get_device_delay(struct wasapi_state *state, double *delay_us) {
     UINT64 sample_count = atomic_load(&state->sample_count);
     UINT64 position, qpc_position;
     HRESULT hr;
 
     hr = IAudioClock_GetPosition(state->pAudioClock, &position, &qpc_position);
-    /* GetPosition succeeded, but the result may be inaccurate due to the length of the call */
-    /* http://msdn.microsoft.com/en-us/library/windows/desktop/dd370889%28v=vs.85%29.aspx */
+    // GetPosition succeeded, but the result may be
+    // inaccurate due to the length of the call
+    // http://msdn.microsoft.com/en-us/library/windows/desktop/dd370889%28v=vs.85%29.aspx
     if (hr == S_FALSE) {
-        MP_DBG(state, "Possibly inaccurate device position.\n");
+        MP_VERBOSE(state, "Possibly inaccurate device position.\n");
         hr = S_OK;
     }
     EXIT_ON_ERROR(hr);
 
-    LARGE_INTEGER qpc_count;
-    QueryPerformanceCounter(&qpc_count);
-    double qpc_diff = (qpc_count.QuadPart * 1e7 / state->qpc_frequency.QuadPart) - qpc_position;
+    // convert position to number of samples careful to avoid overflow
+    UINT64 sample_position = uint64_scale(position,
+                                          state->format.Format.nSamplesPerSec,
+                                          state->clock_frequency);
+    INT64 diff = sample_count - sample_position;
+    *delay_us = diff * 1e6 / state->format.Format.nSamplesPerSec;
 
-    position += state->clock_frequency * (uint64_t) (qpc_diff / 1e7);
+    // Correct for any delay in IAudioClock_GetPosition above.
+    // This should normally be very small (<1 us), but just in case. . .
+    LARGE_INTEGER qpc;
+    QueryPerformanceCounter(&qpc);
+    INT64 qpc_diff = av_rescale(qpc.QuadPart, 10000000, state->qpc_frequency.QuadPart)
+                     - qpc_position;
+    // ignore the above calculation if it yeilds more than 10 seconds (due to
+    // possible overflow inside IAudioClock_GetPosition)
+    if (qpc_diff < 10 * 10000000) {
+        *delay_us -= qpc_diff / 10.0; // convert to us
+    } else {
+        MP_VERBOSE(state, "Insane qpc delay correction of %g seconds. "
+                   "Ignoring it.\n", qpc_diff / 10000000.0);
+    }
 
-    /* convert position to the same base as sample_count */
-    position = position * state->format.Format.nSamplesPerSec / state->clock_frequency;
-
-    double diff = sample_count - position;
-    *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 us\n", *delay_us);
 
     return S_OK;
 exit_label:
@@ -80,11 +92,14 @@ 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);
+        MP_TRACE(ao, "Frame to fill: %"PRIu32". Padding: %"PRIu32"\n",
+                 frame_count, padding);
     }
-    double delay;
-    hr = get_device_delay(state, &delay);
+    double delay_us;
+    hr = get_device_delay(state, &delay_us);
     EXIT_ON_ERROR(hr);
+    // add the buffer delay
+    delay_us += frame_count * 1e6 / state->format.Format.nSamplesPerSec;
 
     BYTE *pData;
     hr = IAudioRenderClient_GetBuffer(state->pRenderClient,
@@ -93,10 +108,11 @@ static void thread_feed(struct ao *ao)
 
     BYTE *data[1] = {pData};
 
-    ao_read_data(ao, (void**)data, frame_count, (int64_t) (
-                     mp_time_us() + delay * 1e6 +
-                     frame_count * 1e6 / state->format.Format.nSamplesPerSec));
+    ao_read_data(ao, (void **)data, frame_count,
+                 mp_time_us() + (int64_t)llrint(delay_us));
 
+    // note, we can't use ao_read_data return value here since we already
+    // commited to frame_count above in the GetBuffer call
     hr = IAudioRenderClient_ReleaseBuffer(state->pRenderClient,
                                           frame_count, 0);
     EXIT_ON_ERROR(hr);
@@ -124,17 +140,21 @@ static void thread_resume(struct ao *ao)
                mp_HRESULT_to_str(hr));
     }
 
-    /* Fill the buffer before starting, but only if there is no audio queued to play. */
-    /* This prevents overfilling the buffer, which leads to problems in exclusive mode */
+    // Fill the buffer before starting, but only if there is no audio queued to
+    // play.  This prevents overfilling the buffer, which leads to problems in
+    // exclusive mode
     if (padding < (UINT32) state->bufferFrameCount)
         thread_feed(ao);
 
     // start feeding next wakeup if something else hasn't been requested
     int expected = WASAPI_THREAD_RESUME;
-    atomic_compare_exchange_strong(&state->thread_state, &expected, WASAPI_THREAD_FEED);
+    atomic_compare_exchange_strong(&state->thread_state, &expected,
+                                   WASAPI_THREAD_FEED);
     hr = IAudioClient_Start(state->pAudioClient);
-    if (hr != S_OK)
-        MP_ERR(state, "IAudioClient_Start returned %s\n", mp_HRESULT_to_str(hr));
+    if (hr != S_OK) {
+        MP_ERR(state, "IAudioClient_Start returned %s\n",
+               mp_HRESULT_to_str(hr));
+    }
 
     return;
 }
@@ -145,11 +165,11 @@ static void thread_reset(struct ao *ao)
     HRESULT hr;
     MP_DBG(state, "Thread Reset\n");
     hr = IAudioClient_Stop(state->pAudioClient);
-    /* we may get S_FALSE if the stream is already stopped */
+    // we may get S_FALSE if the stream is already stopped
     if (hr != S_OK && hr != S_FALSE)
         MP_ERR(state, "IAudioClient_Stop returned: %s\n", mp_HRESULT_to_str(hr));
 
-    /* we may get S_FALSE if the stream is already reset */
+    // we may get S_FALSE if the stream is already reset
     hr = IAudioClient_Reset(state->pAudioClient);
     if (hr != S_OK && hr != S_FALSE)
         MP_ERR(state, "IAudioClient_Reset returned: %s\n", mp_HRESULT_to_str(hr));
@@ -157,7 +177,8 @@ static void thread_reset(struct ao *ao)
     atomic_store(&state->sample_count, 0);
     // start feeding next wakeup if something else hasn't been requested
     int expected = WASAPI_THREAD_RESET;
-    atomic_compare_exchange_strong(&state->thread_state, &expected, WASAPI_THREAD_FEED);
+    atomic_compare_exchange_strong(&state->thread_state, &expected,
+                                   WASAPI_THREAD_FEED);
     return;
 }
 
@@ -173,11 +194,12 @@ static DWORD __stdcall AudioThread(void *lpParameter)
         goto exit_label;
 
     MP_DBG(ao, "Entering dispatch loop\n");
-    while (true) { /* watch events */
+    while (true) { // watch events
         HANDLE events[] = {state->hWake};
-        switch (MsgWaitForMultipleObjects(MP_ARRAY_SIZE(events), events, FALSE, INFINITE,
+        switch (MsgWaitForMultipleObjects(MP_ARRAY_SIZE(events), events,
+                                          FALSE, INFINITE,
                                           QS_POSTMESSAGE | QS_SENDMESSAGE)) {
-        /* AudioThread wakeup */
+        // AudioThread wakeup
         case WAIT_OBJECT_0:
             switch (atomic_load(&state->thread_state)) {
             case WASAPI_THREAD_FEED:
@@ -198,7 +220,7 @@ static DWORD __stdcall AudioThread(void *lpParameter)
                 goto exit_label;
             }
             break;
-        /* messages to dispatch (COM marshalling) */
+        // messages to dispatch (COM marshalling)
         case (WAIT_OBJECT_0 + MP_ARRAY_SIZE(events)):
             wasapi_dispatch(ao);
             break;
@@ -215,7 +237,8 @@ exit_label:
     return 0;
 }
 
-static void set_thread_state(struct ao *ao, enum wasapi_thread_state thread_state)
+static void set_thread_state(struct ao *ao,
+                             enum wasapi_thread_state thread_state)
 {
     struct wasapi_state *state = ao->priv;
     atomic_store(&state->thread_state, thread_state);
@@ -230,20 +253,22 @@ static void uninit(struct ao *ao)
     if (state->hWake)
         set_thread_state(ao, WASAPI_THREAD_SHUTDOWN);
 
-    /* wait up to 10 seconds */
+    // wait up to 10 seconds
     if (state->hAudioThread &&
         WaitForSingleObject(state->hAudioThread, 10000) == WAIT_TIMEOUT)
     {
         MP_ERR(ao, "Audio loop thread refuses to abort\n");
         return;
     }
-    if (state->VistaBlob.hAvrt)
-        FreeLibrary(state->VistaBlob.hAvrt);
 
     SAFE_RELEASE(state->hInitDone,   CloseHandle(state->hInitDone));
     SAFE_RELEASE(state->hWake,       CloseHandle(state->hWake));
     SAFE_RELEASE(state->hAudioThread,CloseHandle(state->hAudioThread));
 
+    wasapi_change_uninit(ao);
+
+    talloc_free(state->deviceID);
+
     CoUninitialize();
     MP_DBG(ao, "Uninit wasapi done\n");
 }
@@ -255,8 +280,14 @@ static int init(struct ao *ao)
 
     struct wasapi_state *state = ao->priv;
     state->log = ao->log;
-    if(!wasapi_fill_VistaBlob(state))
-        MP_WARN(ao, "Error loading thread priority functions\n");
+
+    state->deviceID = find_deviceID(ao);
+    if (!state->deviceID) {
+        uninit(ao);
+        return -1;
+    }
+
+    wasapi_change_init(ao, false);
 
     state->hInitDone = CreateEventW(NULL, FALSE, FALSE, NULL);
     state->hWake     = CreateEventW(NULL, FALSE, FALSE, NULL);
@@ -274,7 +305,7 @@ static int init(struct ao *ao)
         return -1;
     }
 
-    WaitForSingleObject(state->hInitDone, INFINITE); /* wait on init complete */
+    WaitForSingleObject(state->hInitDone, INFINITE); // wait on init complete
     SAFE_RELEASE(state->hInitDone,CloseHandle(state->hInitDone));
     if (state->init_ret != S_OK) {
         if (!ao->probing)
@@ -283,88 +314,135 @@ static int init(struct ao *ao)
         return -1;
     }
 
-    wasapi_setup_proxies(state);
+    wasapi_receive_proxies(state);
     MP_DBG(ao, "Init wasapi done\n");
     return 0;
 }
 
+static int control_exclusive(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+    struct wasapi_state *state = ao->priv;
+
+    switch (cmd) {
+    case AOCONTROL_GET_VOLUME:
+    case AOCONTROL_SET_VOLUME:
+        if (!state->pEndpointVolumeProxy ||
+            !(state->vol_hw_support & ENDPOINT_HARDWARE_SUPPORT_VOLUME)) {
+            return CONTROL_FALSE;
+        }
+
+        float volume;
+        switch (cmd) {
+        case AOCONTROL_GET_VOLUME:
+            IAudioEndpointVolume_GetMasterVolumeLevelScalar(
+                state->pEndpointVolumeProxy,
+                &volume);
+            *(ao_control_vol_t *)arg = (ao_control_vol_t){
+                .left  = 100.0f * volume,
+                .right = 100.0f * volume,
+            };
+            return CONTROL_OK;
+        case AOCONTROL_SET_VOLUME:
+            volume = ((ao_control_vol_t *)arg)->left / 100.f;
+            IAudioEndpointVolume_SetMasterVolumeLevelScalar(
+                state->pEndpointVolumeProxy,
+                volume, NULL);
+            return CONTROL_OK;
+        }
+    case AOCONTROL_GET_MUTE:
+    case AOCONTROL_SET_MUTE:
+        if (!state->pEndpointVolumeProxy ||
+            !(state->vol_hw_support & ENDPOINT_HARDWARE_SUPPORT_MUTE)) {
+            return CONTROL_FALSE;
+        }
+
+        BOOL mute;
+        switch (cmd) {
+        case AOCONTROL_GET_MUTE:
+            IAudioEndpointVolume_GetMute(state->pEndpointVolumeProxy,
+                                         &mute);
+            *(bool *)arg = mute;
+            return CONTROL_OK;
+        case AOCONTROL_SET_MUTE:
+            mute = *(bool *)arg;
+            IAudioEndpointVolume_SetMute(state->pEndpointVolumeProxy,
+                                         mute, NULL);
+            return CONTROL_OK;
+        }
+    case AOCONTROL_HAS_PER_APP_VOLUME:
+        return CONTROL_FALSE;
+    default:
+        return CONTROL_UNKNOWN;
+    }
+}
+
+static int control_shared(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+    struct wasapi_state *state = ao->priv;
+    if (!state->pAudioVolumeProxy)
+        return CONTROL_UNKNOWN;
+
+    float volume;
+    BOOL mute;
+    switch(cmd) {
+    case AOCONTROL_GET_VOLUME:
+        ISimpleAudioVolume_GetMasterVolume(state->pAudioVolumeProxy,
+                                           &volume);
+        *(ao_control_vol_t *)arg = (ao_control_vol_t){
+            .left  = 100.0f * volume,
+            .right = 100.0f * volume,
+        };
+        return CONTROL_OK;
+    case AOCONTROL_SET_VOLUME:
+        volume = ((ao_control_vol_t *)arg)->left / 100.f;
+        ISimpleAudioVolume_SetMasterVolume(state->pAudioVolumeProxy,
+                                           volume, NULL);
+        return CONTROL_OK;
+    case AOCONTROL_GET_MUTE:
+        ISimpleAudioVolume_GetMute(state->pAudioVolumeProxy, &mute);
+        *(bool *)arg = mute;
+        return CONTROL_OK;
+    case AOCONTROL_SET_MUTE:
+        mute = *(bool *)arg;
+        ISimpleAudioVolume_SetMute(state->pAudioVolumeProxy, mute, NULL);
+        return CONTROL_OK;
+    case AOCONTROL_HAS_PER_APP_VOLUME:
+        return CONTROL_TRUE;
+    default:
+        return CONTROL_UNKNOWN;
+    }
+}
+
 static int control(struct ao *ao, enum aocontrol cmd, void *arg)
 {
     struct wasapi_state *state = ao->priv;
-    ao_control_vol_t *vol = arg;
-    BOOL mute;
 
+    // common to exclusive and shared
     switch (cmd) {
-    case AOCONTROL_GET_VOLUME:
-        if (state->opt_exclusive)
-            IAudioEndpointVolume_GetMasterVolumeLevelScalar(state->pEndpointVolumeProxy,
-                                                            &state->audio_volume);
-        else
-            ISimpleAudioVolume_GetMasterVolume(state->pAudioVolumeProxy,
-                                               &state->audio_volume);
+    case AOCONTROL_UPDATE_STREAM_TITLE:
+        if (!state->pSessionControlProxy)
+            return CONTROL_FALSE;
 
-        /* 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",
-                       state->audio_volume, state->previous_volume);
-            state->initial_volume = state->audio_volume;
-        }
-
-        vol->left = vol->right = 100.0f * state->audio_volume;
-        return CONTROL_OK;
-    case AOCONTROL_SET_VOLUME:
-        state->audio_volume = vol->left / 100.f;
-        if (state->opt_exclusive)
-            IAudioEndpointVolume_SetMasterVolumeLevelScalar(state->pEndpointVolumeProxy,
-                                                            state->audio_volume, NULL);
-        else
-            ISimpleAudioVolume_SetMasterVolume(state->pAudioVolumeProxy,
-                                               state->audio_volume, NULL);
-
-        state->previous_volume = state->audio_volume;
-        return CONTROL_OK;
-    case AOCONTROL_GET_MUTE:
-        if (state->opt_exclusive)
-            IAudioEndpointVolume_GetMute(state->pEndpointVolumeProxy, &mute);
-        else
-            ISimpleAudioVolume_GetMute(state->pAudioVolumeProxy, &mute);
-        *(bool*)arg = mute;
-
-        return CONTROL_OK;
-    case AOCONTROL_SET_MUTE:
-        mute = *(bool*)arg;
-        if (state->opt_exclusive)
-            IAudioEndpointVolume_SetMute(state->pEndpointVolumeProxy, mute, NULL);
-        else
-            ISimpleAudioVolume_SetMute(state->pAudioVolumeProxy, mute, NULL);
-
-        return CONTROL_OK;
-    case AOCONTROL_HAS_PER_APP_VOLUME:
-        return CONTROL_TRUE;
-    case AOCONTROL_UPDATE_STREAM_TITLE: {
-        MP_VERBOSE(state, "Updating stream title to \"%s\"\n", (char*)arg);
         wchar_t *title = mp_from_utf8(NULL, (char*)arg);
-
         wchar_t *tmp = NULL;
-
-        /* There is a weird race condition in the IAudioSessionControl itself --
-           it seems that *sometimes* the SetDisplayName does not take effect and it still shows
-           the old title. Use this loop to insist until it works. */
+        // There is a weird race condition in the IAudioSessionControl itself --
+        // it seems that *sometimes* the SetDisplayName does not take effect and
+        // it still shows the old title. Use this loop to insist until it works.
         do {
-            IAudioSessionControl_SetDisplayName(state->pSessionControlProxy, title, NULL);
+            IAudioSessionControl_SetDisplayName(state->pSessionControlProxy,
+                                                title, NULL);
 
             SAFE_RELEASE(tmp, CoTaskMemFree(tmp));
-            IAudioSessionControl_GetDisplayName(state->pSessionControlProxy, &tmp);
+            IAudioSessionControl_GetDisplayName(state->pSessionControlProxy,
+                                                &tmp);
         } while (lstrcmpW(title, tmp));
         SAFE_RELEASE(tmp, CoTaskMemFree(tmp));
         talloc_free(title);
-
         return CONTROL_OK;
     }
-    default:
-        return CONTROL_UNKNOWN;
-    }
+
+    return state->opt_exclusive ?
+        control_exclusive(ao, cmd, arg) : control_shared(ao, cmd, arg);
 }
 
 static void audio_reset(struct ao *ao)
@@ -380,9 +458,7 @@ static void audio_resume(struct ao *ao)
 static void hotplug_uninit(struct ao *ao)
 {
     MP_DBG(ao, "Hotplug uninit\n");
-    struct wasapi_state *state = ao->priv;
     wasapi_change_uninit(ao);
-    SAFE_RELEASE(state->pEnumerator, IMMDeviceEnumerator_Release(state->pEnumerator));
     CoUninitialize();
 }
 
@@ -392,10 +468,7 @@ static int hotplug_init(struct ao *ao)
     struct wasapi_state *state = ao->priv;
     state->log = ao->log;
     CoInitializeEx(NULL, COINIT_APARTMENTTHREADED);
-    HRESULT hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
-                                  &IID_IMMDeviceEnumerator, (void **)&state->pEnumerator);
-    EXIT_ON_ERROR(hr);
-    hr = wasapi_change_init(ao, true);
+    HRESULT hr = wasapi_change_init(ao, true);
     EXIT_ON_ERROR(hr);
 
     return 0;

audio/out/ao_wasapi.h

@@ -20,17 +20,24 @@
 #ifndef MP_AO_WASAPI_H_
 #define MP_AO_WASAPI_H_
 
+#include <stdlib.h>
 #include <stdbool.h>
+#include <windows.h>
+#include <mmdeviceapi.h>
 #include <audioclient.h>
 #include <audiopolicy.h>
-#include <mmdeviceapi.h>
-#include <avrt.h>
+#include <endpointvolume.h>
 
+#include "common/msg.h"
 #include "osdep/atomics.h"
+#include "osdep/windows_utils.h"
+#include "internal.h"
+#include "ao.h"
 
 typedef struct change_notify {
-    IMMNotificationClient client; /* this must be first in the structure! */
-    LPWSTR monitored; /* Monitored device */
+    IMMNotificationClient client; // this must be first in the structure!
+    IMMDeviceEnumerator *pEnumerator; // object where client is registered
+    LPWSTR monitored; // Monitored device
     bool is_hotplug;
     struct ao *ao;
 } change_notify;
@@ -52,75 +59,73 @@ enum wasapi_thread_state {
 
 typedef struct wasapi_state {
     struct mp_log *log;
-    /* Init phase */
-    HRESULT init_ret;
-    HANDLE hInitDone;
-    int share_mode;
 
-    /* volume control */
-    DWORD vol_hw_support;
-    float audio_volume;
-    float previous_volume;
-    float initial_volume;
+    // Thread handles
+    HRESULT init_ret;        // status of init phase
+    HANDLE hInitDone;        // set when init is complete in audio thread
+    HANDLE hAudioThread;     // the audio thread itself
+    HANDLE hWake;            // thread wakeup event
+    atomic_int thread_state; // enum wasapi_thread_state (what to do on wakeup)
 
-    /* WASAPI handles, owned by audio thread */
+    // for setting the audio thread priority
+    HANDLE hTask;
+
+    // ID of the device to use
+    LPWSTR deviceID;
+    // WASAPI object handles owned and used by audio thread
     IMMDevice *pDevice;
     IAudioClient *pAudioClient;
     IAudioRenderClient *pRenderClient;
-    ISimpleAudioVolume *pAudioVolume;
-    IAudioEndpointVolume *pEndpointVolume;
-    IAudioSessionControl *pSessionControl;
-    IMMDeviceEnumerator *pEnumerator;
 
-    /* thread handles */
-    HANDLE hAudioThread; /* the thread itself */
-    HANDLE hWake; /* thread wakeup event */
-    atomic_int thread_state; /* enum wasapi_thread_state */
-
-    /* for setting the audio thread priority */
-    HANDLE hTask; /* AV thread */
-    DWORD taskIndex; /* AV task ID */
-
-    /* WASAPI proxy handles, for Single-Threaded Apartment communication.
-       One is needed for each audio thread object that's accessed from the main thread. */
-    ISimpleAudioVolume *pAudioVolumeProxy;
-    IAudioEndpointVolume *pEndpointVolumeProxy;
-    IAudioSessionControl *pSessionControlProxy;
-
-    /* Streams used to marshal the proxy objects. The thread owning the actual objects
-       needs to marshal proxy objects into these streams, and the thread that wants the
-       proxies unmarshals them from here. */
-    IStream *sAudioVolume;
-    IStream *sEndpointVolume;
-    IStream *sSessionControl;
-
-    /* WASAPI internal clock information, for estimating delay */
+    // WASAPI internal clock information, for estimating delay
     IAudioClock *pAudioClock;
-    UINT64 clock_frequency; /* scale for the "samples" returned by the clock */
-    atomic_ullong sample_count; /* the amount of samples per channel written to a GetBuffer buffer */
-    LARGE_INTEGER qpc_frequency; /* frequency of windows' high resolution timer */
+    atomic_ullong sample_count;  // samples per channel written by GetBuffer
+    UINT64 clock_frequency;      // scale for position returned by GetPosition
+    LARGE_INTEGER qpc_frequency; // frequency of Windows' high resolution timer
 
-    /* ao options */
+    // WASAPI control (handles owned by audio thread but used by main thread)
+    IAudioSessionControl *pSessionControl; // setting the stream title
+    IAudioEndpointVolume *pEndpointVolume; // exclusive mode volume/mute
+    ISimpleAudioVolume *pAudioVolume;      // shared mode volume/mute
+    DWORD vol_hw_support; // is hardware volume supported for exclusive-mode?
+
+    // Streams used to marshal the proxy objects. The thread owning the actual
+    // objects needs to marshal proxy objects into these streams, and the thread
+    // that wants the proxies unmarshals them from here.
+    IStream *sSessionControl;
+    IStream *sEndpointVolume;
+    IStream *sAudioVolume;
+
+    // WASAPI proxy handles, for Single-Threaded Apartment communication. One is
+    // needed for each audio thread object that's accessed from the main thread.
+    IAudioSessionControl *pSessionControlProxy;
+    IAudioEndpointVolume *pEndpointVolumeProxy;
+    ISimpleAudioVolume *pAudioVolumeProxy;
+
+    // ao options
     int opt_exclusive;
     int opt_list;
     char *opt_device;
 
-    /* format info */
+    // format info
     WAVEFORMATEXTENSIBLE format;
-    size_t buffer_block_size; /* Size of each block in bytes */
-    UINT32 bufferFrameCount; /* wasapi buffer block size, number of frames, frame size at format.nBlockAlign */
-
-    /* Don't use these functions directly in case
-       they are unimplemented for some reason.
-       (XP shouldn't be an issue since it doesn't support wasapi, maybe wine?)
-       Blob is owned by the main thread */
-    struct {
-        HMODULE hAvrt;
-        HANDLE (WINAPI *pAvSetMmThreadCharacteristicsW)(LPCWSTR, LPDWORD);
-        WINBOOL (WINAPI *pAvRevertMmThreadCharacteristics)(HANDLE);
-    } VistaBlob;
+    AUDCLNT_SHAREMODE share_mode; // AUDCLNT_SHAREMODE_EXCLUSIVE / SHARED
+    UINT32 bufferFrameCount;      // number of frames in buffer
 
     change_notify change;
 } wasapi_state;
 
+char *mp_PKEY_to_str_buf(char *buf, size_t buf_size, const PROPERTYKEY *pkey);
+#define mp_PKEY_to_str(pkey) mp_PKEY_to_str_buf((char[42]){0}, 42, (pkey))
+
+void wasapi_list_devs(struct ao *ao, struct ao_device_list *list);
+LPWSTR find_deviceID(struct ao *ao);
+
+void wasapi_dispatch(struct ao *ao);
+HRESULT wasapi_thread_init(struct ao *ao);
+void wasapi_thread_uninit(struct ao *ao);
+
+void wasapi_receive_proxies(wasapi_state *state);
+void wasapi_release_proxies(wasapi_state *state);
+
 #endif

audio/out/ao_wasapi_changenotify.c

@@ -17,20 +17,14 @@
  * with mpv.  If not, see <http://www.gnu.org/licenses/>.
  */
 
-#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 HRESULT STDMETHODCALLTYPE sIMMNotificationClient_QueryInterface(
     IMMNotificationClient* This, REFIID riid, void **ppvObject)
 {
-    /* Compatible with IMMNotificationClient and IUnknown */
+    // Compatible with IMMNotificationClient and IUnknown
     if (IsEqualGUID(&IID_IMMNotificationClient, riid) ||
         IsEqualGUID(&IID_IUnknown, riid))
     {
@@ -42,14 +36,14 @@ static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_QueryInterface(
     }
 }
 
-/* these are required, but not actually used */
+// 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 */
+// MSDN says it should free itself, but we're static
 static ULONG STDMETHODCALLTYPE sIMMNotificationClient_Release(
     IMMNotificationClient *This)
 {
@@ -65,7 +59,8 @@ static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnDeviceStateChanged(
     struct ao *ao = change->ao;
 
     if (change->is_hotplug) {
-        MP_VERBOSE(ao, "OnDeviceStateChanged triggered: sending hotplug event\n");
+        MP_VERBOSE(ao,
+                   "OnDeviceStateChanged triggered: sending hotplug event\n");
         ao_hotplug_event(ao);
     } else if (pwstrDeviceId && !wcscmp(pwstrDeviceId, change->monitored)) {
         switch (dwNewState) {
@@ -99,7 +94,7 @@ static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnDeviceAdded(
     return S_OK;
 }
 
-/* maybe MPV can go over to the prefered device once it is plugged in? */
+// maybe MPV can go over to the prefered device once it is plugged in?
 static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnDeviceRemoved(
     IMMNotificationClient *This,
     LPCWSTR pwstrDeviceId)
@@ -129,29 +124,31 @@ static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnDefaultDeviceChanged(
     struct ao *ao = change->ao;
     struct wasapi_state *state = ao->priv;
 
-    /* don't care about "eCapture" or non-"eMultimedia" roles  */
+    // don't care about "eCapture" or non-"eMultimedia" roles
     if (flow == eCapture || role != eMultimedia) return S_OK;
 
     if (change->is_hotplug) {
-        MP_VERBOSE(ao, "OnDefaultDeviceChanged triggered: sending hotplug event\n");
+        MP_VERBOSE(ao,
+                   "OnDefaultDeviceChanged triggered: sending hotplug event\n");
         ao_hotplug_event(ao);
     } else {
-        /* stay on the device the user specified */
+        // stay on the device the user specified
         if (state->opt_device) {
             MP_VERBOSE(ao, "OnDefaultDeviceChanged triggered: "
                        "staying on specified device %s\n", state->opt_device);
             return S_OK;
         }
 
-        /* don't reload if already on the new default */
+        // don't reload if already on the new default
         if (pwstrDeviceId && !wcscmp(pwstrDeviceId, change->monitored)) {
             MP_VERBOSE(ao, "OnDefaultDeviceChanged triggered: "
                        "already using default device, no reload required\n");
             return S_OK;
         }
 
-        /* if we got here, we need to reload */
-        MP_VERBOSE(ao, "OnDefaultDeviceChanged triggered: requesting ao reload\n");
+        // if we got here, we need to reload
+        MP_VERBOSE(ao,
+                   "OnDefaultDeviceChanged triggered: requesting ao reload\n");
         ao_request_reload(ao);
     }
 
@@ -184,7 +181,7 @@ static HRESULT STDMETHODCALLTYPE sIMMNotificationClient_OnPropertyValueChanged(
     return S_OK;
 }
 
-static CONST_VTBL IMMNotificationClientVtbl sIMMDeviceEnumeratorVtbl_vtbl = {
+static CONST_VTBL IMMNotificationClientVtbl sIMMNotificationClientVtbl = {
     .QueryInterface = sIMMNotificationClient_QueryInterface,
     .AddRef = sIMMNotificationClient_AddRef,
     .Release = sIMMNotificationClient_Release,
@@ -200,27 +197,30 @@ HRESULT wasapi_change_init(struct ao *ao, bool is_hotplug)
 {
     struct wasapi_state *state = ao->priv;
     struct change_notify *change = &state->change;
-    HRESULT hr;
-    /* COM voodoo to emulate c++ class */
-    change->client.lpVtbl = &sIMMDeviceEnumeratorVtbl_vtbl;
-
-    /* register the change notification client */
-    hr = IMMDeviceEnumerator_RegisterEndpointNotificationCallback(
-        state->pEnumerator, (IMMNotificationClient *)change);
+    HRESULT hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
+                                  &IID_IMMDeviceEnumerator,
+                                  (void **)&change->pEnumerator);
     EXIT_ON_ERROR(hr);
 
-    /* so the callbacks can access the ao */
+    // COM voodoo to emulate c++ class
+    change->client.lpVtbl = &sIMMNotificationClientVtbl;
+
+    // register the change notification client
+    hr = IMMDeviceEnumerator_RegisterEndpointNotificationCallback(
+        change->pEnumerator, (IMMNotificationClient *)change);
+    EXIT_ON_ERROR(hr);
+
+    // so the callbacks can access the ao
     change->ao = ao;
 
-    /* whether or not this is the hotplug instance */
+    // whether or not this is the hotplug instance
     change->is_hotplug = is_hotplug;
 
     if (is_hotplug) {
         MP_DBG(ao, "Monitoring for hotplug events\n");
     } else {
-        /* Get the device string to compare with the pwstrDeviceId */
-        hr = IMMDevice_GetId(state->pDevice, &change->monitored);
-        EXIT_ON_ERROR(hr);
+        // Get the device string to compare with the pwstrDeviceId
+        change->monitored = state->deviceID;
         MP_VERBOSE(ao, "Monitoring changes in device %S\n", change->monitored);
     }
 
@@ -237,10 +237,10 @@ void wasapi_change_uninit(struct ao *ao)
     struct wasapi_state *state = ao->priv;
     struct change_notify *change = &state->change;
 
-    if (state->pEnumerator && change->client.lpVtbl) {
+    if (change->pEnumerator && change->client.lpVtbl) {
         IMMDeviceEnumerator_UnregisterEndpointNotificationCallback(
-            state->pEnumerator, (IMMNotificationClient *)change);
+            change->pEnumerator, (IMMNotificationClient *)change);
     }
 
-    if (change->monitored) CoTaskMemFree(change->monitored);
+    SAFE_RELEASE(change->pEnumerator, IMMDeviceEnumerator_Release(change->pEnumerator));
 }

audio/out/ao_wasapi_utils.h

@@ -1,48 +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/>.
- */
-
-#ifndef MP_AO_WASAPI_UTILS_H_
-#define MP_AO_WASAPI_UTILS_H_
-
-#include "audio/out/ao_wasapi.h"
-
-#include "options/m_option.h"
-#include "common/msg.h"
-#include "ao.h"
-#include "internal.h"
-
-char *mp_GUID_to_str_buf(char *buf, size_t buf_size, const GUID *guid);
-char *mp_PKEY_to_str_buf(char *buf, size_t buf_size, const PROPERTYKEY *pkey);
-char *mp_HRESULT_to_str_buf(char *buf, size_t buf_size, HRESULT hr);
-#define mp_GUID_to_str(guid) mp_GUID_to_str_buf((char[40]){0}, 40, (guid))
-#define mp_PKEY_to_str(pkey) mp_PKEY_to_str_buf((char[42]){0}, 42, (pkey))
-#define mp_HRESULT_to_str(hr) mp_HRESULT_to_str_buf((char[60]){0}, 60, (hr))
-
-bool wasapi_fill_VistaBlob(wasapi_state *state);
-
-void wasapi_list_devs(struct ao *ao, struct ao_device_list *list);
-
-void wasapi_dispatch(struct ao *ao);
-HRESULT wasapi_thread_init(struct ao *ao);
-void wasapi_thread_uninit(struct ao *ao);
-
-HRESULT wasapi_setup_proxies(wasapi_state *state);
-void wasapi_release_proxies(wasapi_state *state);
-
-#endif

audio/out/pull.c

@@ -113,7 +113,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
 // rest of the user-provided buffer with silence.
 // This basically assumes that the audio device doesn't care about underruns.
 // If this is called in paused mode, it will always return 0.
-// The caller should set out_time_us to the expected delay the last sample
+// The caller should set out_time_us to the expected delay until the last sample
 // reaches the speakers, in microseconds, using mp_time_us() as reference.
 int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us)
 {

common/av_common.c

@@ -65,6 +65,7 @@ void mp_copy_lav_codec_headers(AVCodecContext *avctx, AVCodecContext *st)
     avctx->block_align              = st->block_align;
     avctx->channel_layout           = st->channel_layout;
     avctx->bits_per_coded_sample    = st->bits_per_coded_sample;
+    avctx->has_b_frames             = st->has_b_frames;
 }
 
 // We merely pass-through our PTS/DTS as an int64_t; libavcodec won't use it.

common/codecs.c

@@ -16,7 +16,7 @@
  */
 
 #include <assert.h>
-#include "talloc.h"
+#include "mpv_talloc.h"
 #include "misc/bstr.h"
 #include "common/msg.h"
 #include "codecs.h"

common/common.c

@@ -21,8 +21,9 @@
 #include <libavutil/common.h>
 #include <libavutil/error.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 #include "misc/bstr.h"
+#include "misc/ctype.h"
 #include "common/common.h"
 
 #define appendf(ptr, ...) \
@@ -148,7 +149,7 @@ void mp_append_utf8_bstr(void *talloc_ctx, struct bstr *buf, uint32_t codepoint)
     bstr_xappend(talloc_ctx, buf, (bstr){data, output - data});
 }
 
-// Parse a C-style escape beginning at code, and append the result to *str
+// Parse a C/JSON-style escape beginning at code, and append the result to *str
 // using talloc. The input string (*code) must point to the first character
 // after the initial '\', and after parsing *code is set to the first character
 // after the current escape.
@@ -161,6 +162,7 @@ static bool mp_parse_escape(void *talloc_ctx, bstr *dst, bstr *code)
     switch (code->start[0]) {
     case '"':  replace = '"';  break;
     case '\\': replace = '\\'; break;
+    case '/':  replace = '/'; break;
     case 'b':  replace = '\b'; break;
     case 'f':  replace = '\f'; break;
     case 'n':  replace = '\n'; break;
@@ -257,3 +259,19 @@ char *mp_strerror_buf(char *buf, size_t buf_size, int errnum)
     av_strerror(AVERROR(errnum), buf, buf_size);
     return buf;
 }
+
+char *mp_tag_str_buf(char *buf, size_t buf_size, uint32_t tag)
+{
+    if (buf_size < 1)
+        return buf;
+    buf[0] = '\0';
+    for (int n = 0; n < 4; n++) {
+        uint8_t val = (tag >> (n * 8)) & 0xFF;
+        if (mp_isalnum(val) || val == '_' || val == ' ') {
+            mp_snprintf_cat(buf, buf_size, "%c", val);
+        } else {
+            mp_snprintf_cat(buf, buf_size, "[%d]", val);
+        }
+    }
+    return buf;
+}

common/common.h

@@ -24,7 +24,7 @@
 #include <stdint.h>
 
 #include "osdep/compiler.h"
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 // double should be able to represent this exactly
 #define MP_NOPTS_VALUE (-0x1p+63)
@@ -32,8 +32,6 @@
 #define MP_CONCAT_(a, b) a ## b
 #define MP_CONCAT(a, b) MP_CONCAT_(a, b)
 
-#define ROUND(x) ((int)((x) < 0 ? (x) - 0.5 : (x) + 0.5))
-
 #define MPMAX(a, b) ((a) > (b) ? (a) : (b))
 #define MPMIN(a, b) ((a) > (b) ? (b) : (a))
 #define MPCLAMP(a, min, max) (((a) < (min)) ? (min) : (((a) > (max)) ? (max) : (a)))
@@ -92,4 +90,7 @@ bool mp_append_escaped_string(void *talloc_ctx, struct bstr *dst,
 char *mp_strerror_buf(char *buf, size_t buf_size, int errnum);
 #define mp_strerror(e) mp_strerror_buf((char[80]){0}, 80, e)
 
+char *mp_tag_str_buf(char *buf, size_t buf_size, uint32_t tag);
+#define mp_tag_str(t) mp_tag_str_buf((char[22]){0}, 22, t)
+
 #endif /* MPLAYER_MPCOMMON_H */

common/encode_lavc.c

@@ -29,7 +29,7 @@
 #include "options/options.h"
 #include "osdep/timer.h"
 #include "video/out/vo.h"
-#include "talloc.h"
+#include "mpv_talloc.h"
 #include "stream/stream.h"
 
 #define OPT_BASE_STRUCT struct encode_opts

common/global.h

@@ -7,6 +7,7 @@
 struct mpv_global {
     struct MPOpts *opts;
     struct mp_log *log;
+    struct mp_client_api *client_api;
 };
 
 #endif

common/msg.c

@@ -24,7 +24,7 @@
 #include <pthread.h>
 #include <stdint.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 #include "misc/bstr.h"
 #include "osdep/atomics.h"
@@ -326,10 +326,8 @@ static void write_msg_to_buffers(struct mp_log *log, int lev, char *text)
 static void dump_stats(struct mp_log *log, int lev, char *text)
 {
     struct mp_log_root *root = log->root;
-    if (lev == MSGL_STATS && root->stats_file) {
-        fprintf(root->stats_file, "%"PRId64" %s #%s\n", mp_time_us(), text,
-                log->verbose_prefix);
-    }
+    if (lev == MSGL_STATS && root->stats_file)
+        fprintf(root->stats_file, "%"PRId64" %s\n", mp_time_us(), text);
 }
 
 void mp_msg_va(struct mp_log *log, int lev, const char *format, va_list va)
@@ -634,7 +632,7 @@ const int mp_mpv_log_levels[MSGL_MAX + 1] = {
 int mp_msg_find_level(const char *s)
 {
     for (int n = 0; n < MP_ARRAY_SIZE(mp_log_levels); n++) {
-        if (mp_log_levels[n] && mp_log_levels[n] && !strcmp(s, mp_log_levels[n]))
+        if (mp_log_levels[n] && !strcmp(s, mp_log_levels[n]))
             return n;
     }
     return -1;

common/playlist.c

@@ -21,7 +21,7 @@
 #include "common/common.h"
 #include "common/global.h"
 #include "common/msg.h"
-#include "talloc.h"
+#include "mpv_talloc.h"
 #include "options/path.h"
 
 #include "demux/demux.h"
@@ -206,6 +206,18 @@ void playlist_add_base_path(struct playlist *pl, bstr base_path)
     }
 }
 
+// Add redirected_from as new redirect entry to each item in pl.
+void playlist_add_redirect(struct playlist *pl, const char *redirected_from)
+{
+    for (struct playlist_entry *e = pl->first; e; e = e->next) {
+        if (e->num_redirects >= 10) // arbitrary limit for sanity
+            break;
+        char *s = talloc_strdup(e, redirected_from);
+        if (s)
+            MP_TARRAY_APPEND(e, e->redirects, e->num_redirects, s);
+    }
+}
+
 // Move all entries from source_pl to pl, appending them after the current entry
 // of pl. source_pl will be empty, and all entries have changed ownership to pl.
 void playlist_transfer_entries(struct playlist *pl, struct playlist *source_pl)

common/playlist.h

@@ -36,6 +36,11 @@ struct playlist_entry {
 
     char *title;
 
+    // If the user plays a playlist, then the playlist's URL will be appended
+    // as redirect to each entry. (Same for directories etc.)
+    char **redirects;
+    int num_redirects;
+
     // Set to true if playback didn't seem to work, or if the file could be
     // played only for a very short time. This is used to make playlist
     // navigation just work in case the user has unplayable files in the
@@ -88,6 +93,7 @@ void playlist_add_file(struct playlist *pl, const char *filename);
 void playlist_shuffle(struct playlist *pl);
 struct playlist_entry *playlist_get_next(struct playlist *pl, int direction);
 void playlist_add_base_path(struct playlist *pl, bstr base_path);
+void playlist_add_redirect(struct playlist *pl, const char *redirected_from);
 void playlist_transfer_entries(struct playlist *pl, struct playlist *source_pl);
 void playlist_append_entries(struct playlist *pl, struct playlist *source_pl);
 

demux/codec_tags.c

@@ -70,20 +70,20 @@ static const char *map_audio_pcm_tag(uint32_t tag, int bits)
     }
 }
 
-void mp_set_codec_from_tag(struct sh_stream *sh)
+void mp_set_codec_from_tag(struct mp_codec_params *c)
 {
-    sh->codec = lookup_tag(sh->type, sh->codec_tag);
+    c->codec = lookup_tag(c->type, c->codec_tag);
 
-    if (sh->audio && sh->audio->bits_per_coded_sample) {
+    if (c->type == STREAM_AUDIO && c->bits_per_coded_sample) {
         const char *codec =
-            map_audio_pcm_tag(sh->codec_tag, sh->audio->bits_per_coded_sample);
+            map_audio_pcm_tag(c->codec_tag, c->bits_per_coded_sample);
         if (codec)
-            sh->codec = codec;
+            c->codec = codec;
     }
 }
 
-void mp_set_pcm_codec(struct sh_stream *sh, bool sign, bool is_float, int bits,
-                      bool is_be)
+void mp_set_pcm_codec(struct mp_codec_params *c, bool sign, bool is_float,
+                      int bits, bool is_be)
 {
     // This uses libavcodec pcm codec names, e.g. "pcm_u16le".
     char codec[64] = "pcm_";
@@ -95,7 +95,7 @@ void mp_set_pcm_codec(struct sh_stream *sh, bool sign, bool is_float, int bits,
     mp_snprintf_cat(codec, sizeof(codec), "%d", bits);
     if (bits != 8)
         mp_snprintf_cat(codec, sizeof(codec), is_be ? "be" : "le");
-    sh->codec = talloc_strdup(sh->audio, codec);
+    c->codec = talloc_strdup(c, codec);
 }
 
 static const char *const mimetype_to_codec[][2] = {

demux/codec_tags.h

@@ -21,12 +21,12 @@
 #include <stdint.h>
 #include <stdbool.h>
 
-struct sh_stream;
+struct mp_codec_params;
 
-void mp_set_codec_from_tag(struct sh_stream *sh);
+void mp_set_codec_from_tag(struct mp_codec_params *c);
 
-void mp_set_pcm_codec(struct sh_stream *sh, bool sign, bool is_float, int bits,
-                      bool is_be);
+void mp_set_pcm_codec(struct mp_codec_params *c, bool sign, bool is_float,
+                      int bits, bool is_be);
 
 const char *mp_map_mimetype_to_video_codec(const char *mimetype);
 

demux/cue.c

@@ -20,10 +20,11 @@
 #include <string.h>
 #include <inttypes.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 #include "misc/bstr.h"
 #include "common/common.h"
+#include "common/tags.h"
 
 #include "cue.h"
 
@@ -37,6 +38,7 @@ enum cue_command {
     CUE_TRACK,
     CUE_INDEX,
     CUE_TITLE,
+    CUE_PERFORMER,
 };
 
 static const struct {
@@ -51,7 +53,7 @@ static const struct {
     { CUE_UNUSED, "CDTEXTFILE" },
     { CUE_UNUSED, "FLAGS" },
     { CUE_UNUSED, "ISRC" },
-    { CUE_UNUSED, "PERFORMER" },
+    { CUE_PERFORMER, "PERFORMER" },
     { CUE_UNUSED, "POSTGAP" },
     { CUE_UNUSED, "PREGAP" },
     { CUE_UNUSED, "REM" },
@@ -160,17 +162,19 @@ bool mp_probe_cue(struct bstr data)
 struct cue_file *mp_parse_cue(struct bstr data)
 {
     struct cue_file *f = talloc_zero(NULL, struct cue_file);
+    f->tags = talloc_zero(f, struct mp_tags);
 
     data = skip_utf8_bom(data);
 
     char *filename = NULL;
     // Global metadata, and copied into new tracks.
     struct cue_track proto_track = {0};
-    struct cue_track *cur_track = &proto_track;
+    struct cue_track *cur_track = NULL;
 
     while (data.len) {
         struct bstr param;
-        switch (read_cmd(&data, &param)) {
+        int cmd = read_cmd(&data, &param);
+        switch (cmd) {
         case CUE_ERROR:
             talloc_free(f);
             return NULL;
@@ -179,19 +183,29 @@ struct cue_file *mp_parse_cue(struct bstr data)
             f->num_tracks += 1;
             cur_track = &f->tracks[f->num_tracks - 1];
             *cur_track = proto_track;
+            cur_track->tags = talloc_zero(f, struct mp_tags);
             break;
         }
         case CUE_TITLE:
-            cur_track->title = read_quoted(f, &param);
+        case CUE_PERFORMER: {
+            static const char *metanames[] = {
+                [CUE_TITLE] = "title",
+                [CUE_PERFORMER] = "performer",
+            };
+            struct mp_tags *tags = cur_track ? cur_track->tags : f->tags;
+            mp_tags_set_bstr(tags, bstr0(metanames[cmd]), param);
             break;
+        }
         case CUE_INDEX: {
             int type = read_int_2(&param);
             double time = read_time(&param);
-            if (type == 1) {
-                cur_track->start = time;
-                cur_track->filename = filename;
-            } else if (type == 0) {
-                cur_track->pregap_start = time;
+            if (cur_track) {
+                if (type == 1) {
+                    cur_track->start = time;
+                    cur_track->filename = filename;
+                } else if (type == 0) {
+                    cur_track->pregap_start = time;
+                }
             }
             break;
         }
@@ -204,3 +218,18 @@ struct cue_file *mp_parse_cue(struct bstr data)
 
     return f;
 }
+
+int mp_check_embedded_cue(struct cue_file *f)
+{
+    char *fn0 = f->tracks[0].filename;
+    for (int n = 1; n < f->num_tracks; n++) {
+        char *fn = f->tracks[n].filename;
+        // both filenames have the same address (including NULL)
+        if (fn0 == fn)
+            continue;
+        // only one filename is NULL, or the strings don't match
+        if (!fn0 || !fn || strcmp(fn0, fn) != 0)
+            return -1;
+    }
+    return 0;
+}

demux/cue.h

@@ -25,6 +25,7 @@
 struct cue_file {
     struct cue_track *tracks;
     int num_tracks;
+    struct mp_tags *tags;
 };
 
 struct cue_track {
@@ -32,10 +33,11 @@ struct cue_track {
     double start;               // corresponds to INDEX 01
     char *filename;
     int source;
-    char *title;
+    struct mp_tags *tags;
 };
 
 bool mp_probe_cue(struct bstr data);
 struct cue_file *mp_parse_cue(struct bstr data);
+int mp_check_embedded_cue(struct cue_file *f);
 
 #endif

demux/demux.c

@@ -29,7 +29,7 @@
 
 #include "config.h"
 #include "options/options.h"
-#include "talloc.h"
+#include "mpv_talloc.h"
 #include "common/msg.h"
 #include "common/global.h"
 #include "osdep/threads.h"
@@ -48,8 +48,6 @@ extern const demuxer_desc_t demuxer_desc_tv;
 extern const demuxer_desc_t demuxer_desc_mf;
 extern const demuxer_desc_t demuxer_desc_matroska;
 extern const demuxer_desc_t demuxer_desc_lavf;
-extern const demuxer_desc_t demuxer_desc_libass;
-extern const demuxer_desc_t demuxer_desc_subreader;
 extern const demuxer_desc_t demuxer_desc_playlist;
 extern const demuxer_desc_t demuxer_desc_disc;
 extern const demuxer_desc_t demuxer_desc_rar;
@@ -67,9 +65,6 @@ const demuxer_desc_t *const demuxer_list[] = {
     &demuxer_desc_rawvideo,
 #if HAVE_TV
     &demuxer_desc_tv,
-#endif
-#if HAVE_LIBASS
-    &demuxer_desc_libass,
 #endif
     &demuxer_desc_matroska,
 #if HAVE_LIBARCHIVE
@@ -79,8 +74,6 @@ const demuxer_desc_t *const demuxer_list[] = {
     &demuxer_desc_lavf,
     &demuxer_desc_mf,
     &demuxer_desc_playlist,
-    // Pretty aggressive, so should be last.
-    &demuxer_desc_subreader,
     NULL
 };
 
@@ -110,6 +103,11 @@ struct demux_internal {
     void (*wakeup_cb)(void *ctx);
     void *wakeup_cb_ctx;
 
+    struct sh_stream **streams;
+    int num_streams;
+
+    int events;
+
     bool warned_queue_overflow;
     bool last_eof;              // last actual global EOF status
     bool eof;                   // whether we're in EOF state (reset for retry)
@@ -128,6 +126,8 @@ struct demux_internal {
     bool refresh_seeks_enabled;
     bool start_refresh_seek;
 
+    double ts_offset;           // timestamp offset to apply everything
+
     // Cached state.
     bool force_cache_update;
     double time_length;
@@ -168,6 +168,8 @@ struct demux_stream {
 #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_ADD_PTS(a, b) ((a) == MP_NOPTS_VALUE ? (a) : ((a) + (b)))
+
 static void demuxer_sort_chapters(demuxer_t *demuxer);
 static void *demux_thread(void *pctx);
 static void update_cache(struct demux_internal *in);
@@ -193,42 +195,92 @@ static void ds_flush(struct demux_stream *ds)
     ds->last_pos = -1;
 }
 
-struct sh_stream *new_sh_stream(demuxer_t *demuxer, enum stream_type type)
+void demux_set_ts_offset(struct demuxer *demuxer, double offset)
 {
-    assert(demuxer == demuxer->in->d_thread);
+    struct demux_internal *in = demuxer->in;
+    pthread_mutex_lock(&in->lock);
+    in->ts_offset = offset;
+    pthread_mutex_unlock(&in->lock);
+}
 
-    if (demuxer->num_streams > MAX_SH_STREAMS) {
-        MP_WARN(demuxer, "Too many streams.\n");
-        return NULL;
-    }
-
-    int demuxer_id = 0;
-    for (int n = 0; n < demuxer->num_streams; n++) {
-        if (demuxer->streams[n]->type == type)
-            demuxer_id++;
-    }
-
-    struct sh_stream *sh = talloc_ptrtype(demuxer, sh);
+// Allocate a new sh_stream of the given type. It either has to be released
+// with talloc_free(), or added to a demuxer with demux_add_sh_stream(). You
+// cannot add or read packets from the stream before it has been added.
+struct sh_stream *demux_alloc_sh_stream(enum stream_type type)
+{
+    struct sh_stream *sh = talloc_ptrtype(NULL, sh);
     *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),
+        .index = -1,
+        .ff_index = -1,     // may be overwritten by demuxer
+        .demuxer_id = -1,   // ... same
+        .codec = talloc_zero(sh, struct mp_codec_params),
     };
+    sh->codec->type = type;
+    return sh;
+}
+
+// Add a new sh_stream to the demuxer. Note that as soon as the stream has been
+// added, it must be immutable, and must not be released (this will happen when
+// the demuxer is destroyed).
+void demux_add_sh_stream(struct demuxer *demuxer, struct sh_stream *sh)
+{
+    struct demux_internal *in = demuxer->in;
+    pthread_mutex_lock(&in->lock);
+
+    assert(!sh->ds); // must not be added yet
+
+    sh->ds = talloc(sh, struct demux_stream);
     *sh->ds = (struct demux_stream) {
-        .in = demuxer->in,
+        .in = in,
         .type = sh->type,
-        .selected = demuxer->in->autoselect,
+        .selected = in->autoselect,
     };
-    MP_TARRAY_APPEND(demuxer, demuxer->streams, demuxer->num_streams, sh);
-    switch (sh->type) {
-    case STREAM_VIDEO: sh->video = talloc_zero(demuxer, struct sh_video); break;
-    case STREAM_AUDIO: sh->audio = talloc_zero(demuxer, struct sh_audio); break;
-    case STREAM_SUB:   sh->sub = talloc_zero(demuxer, struct sh_sub); break;
+
+    if (!sh->codec->codec)
+        sh->codec->codec = "";
+
+    sh->index = in->num_streams;
+    if (sh->ff_index < 0)
+        sh->ff_index = sh->index;
+    if (sh->demuxer_id < 0) {
+        sh->demuxer_id = 0;
+        for (int n = 0; n < in->num_streams; n++) {
+            if (in->streams[n]->type == sh->type)
+                sh->demuxer_id += 1;
+        }
     }
 
-    return sh;
+    MP_TARRAY_APPEND(in, in->streams, in->num_streams, sh);
+
+    in->events |= DEMUX_EVENT_STREAMS;
+    if (in->wakeup_cb)
+        in->wakeup_cb(in->wakeup_cb_ctx);
+    pthread_mutex_unlock(&in->lock);
+}
+
+// Return a stream with the given index. Since streams can only be added during
+// the lifetime of the demuxer, it is guaranteed that an index within the valid
+// range [0, demux_get_num_stream()) always returns a valid sh_stream pointer,
+// which will be valid until the demuxer is destroyed.
+struct sh_stream *demux_get_stream(struct demuxer *demuxer, int index)
+{
+    struct demux_internal *in = demuxer->in;
+    pthread_mutex_lock(&in->lock);
+    assert(index >= 0 && index < in->num_streams);
+    struct sh_stream *r = in->streams[index];
+    pthread_mutex_unlock(&in->lock);
+    return r;
+}
+
+// See demux_get_stream().
+int demux_get_num_stream(struct demuxer *demuxer)
+{
+    struct demux_internal *in = demuxer->in;
+    pthread_mutex_lock(&in->lock);
+    int r = in->num_streams;
+    pthread_mutex_unlock(&in->lock);
+    return r;
 }
 
 void free_demuxer(demuxer_t *demuxer)
@@ -242,8 +294,10 @@ void free_demuxer(demuxer_t *demuxer)
 
     if (demuxer->desc->close)
         demuxer->desc->close(in->d_thread);
-    for (int n = 0; n < demuxer->num_streams; n++)
-        ds_flush(demuxer->streams[n]->ds);
+    for (int n = in->num_streams - 1; n >= 0; n--) {
+        ds_flush(in->streams[n]->ds);
+        talloc_free(in->streams[n]);
+    }
     pthread_mutex_destroy(&in->lock);
     pthread_cond_destroy(&in->wakeup);
     talloc_free(demuxer);
@@ -307,13 +361,12 @@ const char *stream_type_name(enum stream_type type)
     }
 }
 
-// Returns the same value as demuxer->fill_buffer: 1 ok, 0 EOF/not selected.
-int demux_add_packet(struct sh_stream *stream, demux_packet_t *dp)
+void demux_add_packet(struct sh_stream *stream, demux_packet_t *dp)
 {
     struct demux_stream *ds = stream ? stream->ds : NULL;
     if (!dp || !ds) {
         talloc_free(dp);
-        return 0;
+        return;
     }
     struct demux_internal *in = ds->in;
     pthread_mutex_lock(&in->lock);
@@ -330,7 +383,7 @@ int demux_add_packet(struct sh_stream *stream, demux_packet_t *dp)
     if (!ds->selected || in->seeking || drop) {
         pthread_mutex_unlock(&in->lock);
         talloc_free(dp);
-        return 0;
+        return;
     }
 
     dp->stream = stream->index;
@@ -371,7 +424,6 @@ int demux_add_packet(struct sh_stream *stream, demux_packet_t *dp)
         ds->in->wakeup_cb(ds->in->wakeup_cb_ctx);
     pthread_cond_signal(&in->wakeup);
     pthread_mutex_unlock(&in->lock);
-    return 1;
 }
 
 // Returns true if there was "progress" (lock was released temporarily).
@@ -385,8 +437,8 @@ static bool read_packet(struct demux_internal *in)
     // safe-guards against packet queue overflow.
     bool active = false, read_more = false;
     size_t packs = 0, bytes = 0;
-    for (int n = 0; n < in->d_buffer->num_streams; n++) {
-        struct demux_stream *ds = in->d_buffer->streams[n]->ds;
+    for (int n = 0; n < in->num_streams; n++) {
+        struct demux_stream *ds = in->streams[n]->ds;
         active |= ds->active;
         read_more |= ds->active && !ds->head;
         packs += ds->packs;
@@ -401,16 +453,16 @@ static bool read_packet(struct demux_internal *in)
         if (!in->warned_queue_overflow) {
             in->warned_queue_overflow = true;
             MP_ERR(in, "Too many packets in the demuxer packet queues:\n");
-            for (int n = 0; n < in->d_buffer->num_streams; n++) {
-                struct demux_stream *ds = in->d_buffer->streams[n]->ds;
+            for (int n = 0; n < in->num_streams; n++) {
+                struct demux_stream *ds = in->streams[n]->ds;
                 if (ds->selected) {
                     MP_ERR(in, "  %s/%d: %zd packets, %zd bytes\n",
                            stream_type_name(ds->type), n, ds->packs, ds->bytes);
                 }
             }
         }
-        for (int n = 0; n < in->d_buffer->num_streams; n++) {
-            struct demux_stream *ds = in->d_buffer->streams[n]->ds;
+        for (int n = 0; n < in->num_streams; n++) {
+            struct demux_stream *ds = in->streams[n]->ds;
             ds->eof |= !ds->head;
         }
         pthread_cond_signal(&in->wakeup);
@@ -430,10 +482,8 @@ static bool read_packet(struct demux_internal *in)
     pthread_mutex_lock(&in->lock);
 
     if (eof) {
-        for (int n = 0; n < in->d_buffer->num_streams; n++) {
-            struct demux_stream *ds = in->d_buffer->streams[n]->ds;
-            ds->eof = true;
-        }
+        for (int n = 0; n < in->num_streams; n++)
+            in->streams[n]->ds->eof = true;
         // If we had EOF previously, then don't wakeup (avoids wakeup loop)
         if (!in->last_eof) {
             if (in->wakeup_cb)
@@ -477,8 +527,8 @@ static void start_refreshing(struct demux_internal *in)
     in->start_refresh_seek = false;
 
     double start_ts = MP_NOPTS_VALUE;
-    for (int n = 0; n < demux->num_streams; n++) {
-        struct demux_stream *ds = demux->streams[n]->ds;
+    for (int n = 0; n < in->num_streams; n++) {
+        struct demux_stream *ds = in->streams[n]->ds;
         if (ds->type == STREAM_VIDEO || ds->type == STREAM_AUDIO)
             start_ts = MP_PTS_MIN(start_ts, ds->base_ts);
     }
@@ -487,8 +537,8 @@ static void start_refreshing(struct demux_internal *in)
         demux->partially_seekable || !demux->allow_refresh_seeks)
         return;
 
-    for (int n = 0; n < demux->num_streams; n++) {
-        struct demux_stream *ds = demux->streams[n]->ds;
+    for (int n = 0; n < in->num_streams; n++) {
+        struct demux_stream *ds = in->streams[n]->ds;
         // Streams which didn't read any packets yet can return all packets,
         // or they'd be stuck forever; affects newly selected streams too.
         if (ds->last_pos != -1)
@@ -611,6 +661,9 @@ static struct demux_packet *dequeue_packet(struct demux_stream *ds)
     if (pkt->pos >= ds->in->d_user->filepos)
         ds->in->d_user->filepos = pkt->pos;
 
+    pkt->pts = MP_ADD_PTS(pkt->pts, ds->in->ts_offset);
+    pkt->dts = MP_ADD_PTS(pkt->dts, ds->in->ts_offset);
+
     return pkt;
 }
 
@@ -631,15 +684,33 @@ struct demux_packet *demux_read_packet(struct sh_stream *sh)
     return pkt;
 }
 
+// Sparse packets (Subtitles) interleaved with other non-sparse packets (video,
+// audio) should never be read actively, meaning the demuxer thread does not
+// try to exceed default readahead in order to find a new packet.
+static bool use_lazy_subtitle_reading(struct demux_stream *ds)
+{
+    if (ds->type != STREAM_SUB)
+        return false;
+    for (int n = 0; n < ds->in->num_streams; n++) {
+        struct demux_stream *s = ds->in->streams[n]->ds;
+        if (s->type != STREAM_SUB && s->selected && !s->eof)
+            return true;
+    }
+    return false;
+}
+
 // Poll the demuxer queue, and if there's a packet, return it. Otherwise, just
 // make the demuxer thread read packets for this stream, and if there's at
 // least one packet, call the wakeup callback.
-// Unlike demux_read_packet(), this always enables readahead (which means you
-// must not use it on interleaved subtitle streams).
+// Unlike demux_read_packet(), this always enables readahead (except for
+// interleaved subtitles).
 // Returns:
 //   < 0: EOF was reached, *out_pkt=NULL
 //  == 0: no new packet yet, but maybe later, *out_pkt=NULL
 //   > 0: new packet read, *out_pkt is set
+// Note: when reading interleaved subtitles, the demuxer won't try to forcibly
+// read ahead to get the next subtitle packet (as the next packet could be
+// minutes away). In this situation, this function will just return -1.
 int demux_read_packet_async(struct sh_stream *sh, struct demux_packet **out_pkt)
 {
     struct demux_stream *ds = sh ? sh->ds : NULL;
@@ -649,10 +720,14 @@ int demux_read_packet_async(struct sh_stream *sh, struct demux_packet **out_pkt)
         if (ds->in->threading) {
             pthread_mutex_lock(&ds->in->lock);
             *out_pkt = dequeue_packet(ds);
-            r = *out_pkt ? 1 : ((ds->eof || !ds->selected) ? -1 : 0);
-            ds->active = ds->selected; // enable readahead
-            ds->in->eof = false; // force retry
-            pthread_cond_signal(&ds->in->wakeup); // possibly read more
+            if (use_lazy_subtitle_reading(ds)) {
+                r = *out_pkt ? 1 : -1;
+            } else {
+                r = *out_pkt ? 1 : ((ds->eof || !ds->selected) ? -1 : 0);
+                ds->active = ds->selected; // enable readahead
+                ds->in->eof = false; // force retry
+                pthread_cond_signal(&ds->in->wakeup); // possibly read more
+            }
             pthread_mutex_unlock(&ds->in->lock);
         } else {
             *out_pkt = demux_read_packet(sh);
@@ -662,22 +737,6 @@ int demux_read_packet_async(struct sh_stream *sh, struct demux_packet **out_pkt)
     return r;
 }
 
-// Return the pts of the next packet that demux_read_packet() would return.
-// Might block. Sometimes used to force a packet read, without removing any
-// packets from the queue.
-double demux_get_next_pts(struct sh_stream *sh)
-{
-    double res = MP_NOPTS_VALUE;
-    if (sh) {
-        pthread_mutex_lock(&sh->ds->in->lock);
-        ds_get_packets(sh->ds);
-        if (sh->ds->head)
-            res = sh->ds->head->pts;
-        pthread_mutex_unlock(&sh->ds->in->lock);
-    }
-    return res;
-}
-
 // Return whether a packet is queued. Never blocks, never forces any reads.
 bool demux_has_packet(struct sh_stream *sh)
 {
@@ -693,21 +752,22 @@ bool demux_has_packet(struct sh_stream *sh)
 // Read and return any packet we find.
 struct demux_packet *demux_read_any_packet(struct demuxer *demuxer)
 {
-    assert(!demuxer->in->threading); // doesn't work with threading
+    struct demux_internal *in = demuxer->in;
+    assert(!in->threading); // doesn't work with threading
     bool read_more = true;
     while (read_more) {
-        for (int n = 0; n < demuxer->num_streams; n++) {
-            struct sh_stream *sh = demuxer->streams[n];
+        for (int n = 0; n < in->num_streams; n++) {
+            struct sh_stream *sh = in->streams[n];
             sh->ds->active = sh->ds->selected; // force read_packet() to read
             struct demux_packet *pkt = dequeue_packet(sh->ds);
             if (pkt)
                 return pkt;
         }
         // retry after calling this
-        pthread_mutex_lock(&demuxer->in->lock);
-        read_more = read_packet(demuxer->in);
-        read_more &= !demuxer->in->eof;
-        pthread_mutex_unlock(&demuxer->in->lock);
+        pthread_mutex_lock(&in->lock); // lock only because read_packet unlocks
+        read_more = read_packet(in);
+        read_more &= !in->eof;
+        pthread_mutex_unlock(&in->lock);
     }
     return NULL;
 }
@@ -791,13 +851,14 @@ static int decode_peak(demuxer_t *demuxer, const char *tag, float *out)
 
 static void apply_replaygain(demuxer_t *demuxer, struct replaygain_data *rg)
 {
-    for (int n = 0; n < demuxer->num_streams; n++) {
-        struct sh_stream *sh = demuxer->streams[n];
-        if (sh->audio && !sh->audio->replaygain_data) {
+    struct demux_internal *in = demuxer->in;
+    for (int n = 0; n < in->num_streams; n++) {
+        struct sh_stream *sh = in->streams[n];
+        if (sh->type == STREAM_AUDIO && !sh->codec->replaygain_data) {
             MP_VERBOSE(demuxer, "Replaygain: Track=%f/%f Album=%f/%f\n",
                        rg->track_gain, rg->track_peak,
                        rg->album_gain, rg->album_peak);
-            sh->audio->replaygain_data = talloc_memdup(demuxer, rg, sizeof(*rg));
+            sh->codec->replaygain_data = talloc_memdup(in, rg, sizeof(*rg));
         }
     }
 }
@@ -849,11 +910,6 @@ static void demux_copy(struct demuxer *dst, struct demuxer *src)
         dst->start_time = src->start_time;
         dst->priv = src->priv;
     }
-    if (src->events & DEMUX_EVENT_STREAMS) {
-        // The stream structs themselves are immutable.
-        for (int n = dst->num_streams; n < src->num_streams; n++)
-            MP_TARRAY_APPEND(dst, dst->streams, dst->num_streams, src->streams[n]);
-    }
     if (src->events & DEMUX_EVENT_METADATA) {
         talloc_free(dst->metadata);
         dst->metadata = mp_tags_dup(dst, src->metadata);
@@ -901,6 +957,8 @@ void demux_update(demuxer_t *demuxer)
 
     pthread_mutex_lock(&in->lock);
     demux_copy(demuxer, in->d_buffer);
+    demuxer->events |= in->events;
+    in->events = 0;
     if (in->stream_metadata && (demuxer->events & DEMUX_EVENT_METADATA))
         mp_tags_merge(demuxer->metadata, in->stream_metadata);
     pthread_mutex_unlock(&in->lock);
@@ -922,9 +980,15 @@ static void demux_init_cuesheet(struct demuxer *demuxer)
     if (cue && !demuxer->num_chapters) {
         struct cue_file *f = mp_parse_cue(bstr0(cue));
         if (f) {
-            for (int n = 0; n < f->num_tracks; n++) {
-                struct cue_track *t = &f->tracks[n];
-                demuxer_add_chapter(demuxer, t->title, t->start, -1);
+            if (mp_check_embedded_cue(f) < 0) {
+                MP_WARN(demuxer, "Embedded cue sheet references more than one file. "
+                        "Ignoring it.\n");
+            } else {
+                for (int n = 0; n < f->num_tracks; n++) {
+                    struct cue_track *t = &f->tracks[n];
+                    int idx = demuxer_add_chapter(demuxer, "", t->start, -1);
+                    mp_tags_merge(demuxer->chapters[idx].metadata, t->tags);
+                }
             }
         }
         talloc_free(f);
@@ -938,6 +1002,9 @@ static struct demuxer *open_given_type(struct mpv_global *global,
                                        struct demuxer_params *params,
                                        enum demux_check check)
 {
+    if (mp_cancel_test(stream->cancel))
+        return NULL;
+
     struct demuxer *demuxer = talloc_ptrtype(NULL, demuxer);
     *demuxer = (struct demuxer) {
         .desc = desc,
@@ -1100,8 +1167,8 @@ struct demuxer *demux_open_url(const char *url,
 
 static void flush_locked(demuxer_t *demuxer)
 {
-    for (int n = 0; n < demuxer->num_streams; n++)
-        ds_flush(demuxer->streams[n]->ds);
+    for (int n = 0; n < demuxer->in->num_streams; n++)
+        ds_flush(demuxer->in->streams[n]->ds);
     demuxer->in->warned_queue_overflow = false;
     demuxer->in->eof = false;
     demuxer->in->last_eof = false;
@@ -1152,6 +1219,8 @@ int demux_seek(demuxer_t *demuxer, double rel_seek_secs, int flags)
     in->seeking = true;
     in->seek_flags = flags;
     in->seek_pts = rel_seek_secs;
+    if ((flags & SEEK_ABSOLUTE) && !(flags & SEEK_FACTOR))
+        in->seek_pts = MP_ADD_PTS(in->seek_pts, -in->ts_offset);
 
     if (!in->threading)
         execute_seek(in);
@@ -1177,10 +1246,11 @@ void demux_set_enable_refresh_seeks(struct demuxer *demuxer, bool enabled)
 struct sh_stream *demuxer_stream_by_demuxer_id(struct demuxer *d,
                                                enum stream_type t, int id)
 {
-    for (int n = 0; n < d->num_streams; n++) {
-        struct sh_stream *s = d->streams[n];
+    int num = demux_get_num_stream(d);
+    for (int n = 0; n < num; n++) {
+        struct sh_stream *s = demux_get_stream(d, n);
         if (s->type == t && s->demuxer_id == id)
-            return d->streams[n];
+            return s;
     }
     return NULL;
 }
@@ -1190,8 +1260,9 @@ void demuxer_switch_track(struct demuxer *demuxer, enum stream_type type,
 {
     assert(!stream || stream->type == type);
 
-    for (int n = 0; n < demuxer->num_streams; n++) {
-        struct sh_stream *cur = demuxer->streams[n];
+    int num = demux_get_num_stream(demuxer);
+    for (int n = 0; n < num; n++) {
+        struct sh_stream *cur = demux_get_stream(demuxer, n);
         if (cur->type == type)
             demuxer_select_track(demuxer, cur, cur == stream);
     }
@@ -1392,10 +1463,11 @@ static int cached_demux_control(struct demux_internal *in, int cmd, void *arg)
     case DEMUXER_CTRL_GET_BITRATE_STATS: {
         double *rates = arg;
         for (int n = 0; n < STREAM_TYPE_COUNT; n++)
-            rates[n] = 0;
-        for (int n = 0; n < in->d_user->num_streams; n++) {
-            struct demux_stream *ds = in->d_user->streams[n]->ds;
-            rates[ds->type] += MPMAX(0, ds->bitrate);
+            rates[n] = -1;
+        for (int n = 0; n < in->num_streams; n++) {
+            struct demux_stream *ds = in->streams[n]->ds;
+            if (ds->selected && ds->bitrate >= 0)
+                rates[ds->type] = MPMAX(0, rates[ds->type]) + ds->bitrate;
         }
         return DEMUXER_CTRL_OK;
     }
@@ -1407,8 +1479,8 @@ static int cached_demux_control(struct demux_internal *in, int cmd, void *arg)
             .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;
+        for (int n = 0; n < in->num_streams; n++) {
+            struct demux_stream *ds = in->streams[n]->ds;
             if (ds->active) {
                 r->underrun |= !ds->head && !ds->eof;
                 r->ts_range[0] = MP_PTS_MAX(r->ts_range[0], ds->base_ts);
@@ -1422,6 +1494,8 @@ static int cached_demux_control(struct demux_internal *in, int cmd, void *arg)
             r->ts_duration = MPMAX(0, r->ts_range[1] - r->ts_range[0]);
         if (!num_packets || in->seeking)
             r->ts_duration = 0;
+        r->ts_range[0] = MP_ADD_PTS(r->ts_range[0], in->ts_offset);
+        r->ts_range[1] = MP_ADD_PTS(r->ts_range[1], in->ts_offset);
         return DEMUXER_CTRL_OK;
     }
     }

demux/demux.h

@@ -86,8 +86,6 @@ enum demux_event {
     DEMUX_EVENT_ALL = 0xFFFF,
 };
 
-#define MAX_SH_STREAMS 256
-
 struct demuxer;
 struct timeline;
 
@@ -199,9 +197,6 @@ typedef struct demuxer {
     // Bitmask of DEMUX_EVENT_*
     int events;
 
-    struct sh_stream **streams;
-    int num_streams;
-
     struct demux_edition *editions;
     int num_editions;
     int edition;
@@ -242,16 +237,19 @@ typedef struct {
 void free_demuxer(struct demuxer *demuxer);
 void free_demuxer_and_stream(struct demuxer *demuxer);
 
-int demux_add_packet(struct sh_stream *stream, demux_packet_t *dp);
+void demux_add_packet(struct sh_stream *stream, demux_packet_t *dp);
 
 struct demux_packet *demux_read_packet(struct sh_stream *sh);
 int demux_read_packet_async(struct sh_stream *sh, struct demux_packet **out_pkt);
 bool demux_stream_is_selected(struct sh_stream *stream);
-double demux_get_next_pts(struct sh_stream *sh);
 bool demux_has_packet(struct sh_stream *sh);
 struct demux_packet *demux_read_any_packet(struct demuxer *demuxer);
 
-struct sh_stream *new_sh_stream(struct demuxer *demuxer, enum stream_type type);
+struct sh_stream *demux_get_stream(struct demuxer *demuxer, int index);
+int demux_get_num_stream(struct demuxer *demuxer);
+
+struct sh_stream *demux_alloc_sh_stream(enum stream_type type);
+void demux_add_sh_stream(struct demuxer *demuxer, struct sh_stream *sh);
 
 struct demuxer *demux_open(struct stream *stream, struct demuxer_params *params,
                            struct mpv_global *global);
@@ -271,6 +269,7 @@ bool demux_cancel_test(struct demuxer *demuxer);
 void demux_flush(struct demuxer *demuxer);
 int demux_seek(struct demuxer *demuxer, double rel_seek_secs, int flags);
 void demux_set_enable_refresh_seeks(struct demuxer *demuxer, bool enabled);
+void demux_set_ts_offset(struct demuxer *demuxer, double offset);
 
 int demux_control(struct demuxer *demuxer, int cmd, void *arg);
 

demux/demux_cue.c

@@ -23,7 +23,7 @@
 #include <dirent.h>
 #include <inttypes.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 #include "misc/bstr.h"
 #include "common/msg.h"
@@ -38,7 +38,7 @@
 #define PROBE_SIZE 512
 
 struct priv {
-    bstr data;
+    struct cue_file *f;
 };
 
 static void add_source(struct timeline *tl, struct demuxer *d)
@@ -150,15 +150,18 @@ static void build_timeline(struct timeline *tl)
 
     add_source(tl, tl->demuxer);
 
-    struct cue_file *f = mp_parse_cue(p->data);
-    if (!f) {
-        MP_ERR(tl, "CUE: error parsing input file!\n");
-        goto out;
-    }
-    talloc_steal(ctx, f);
+    struct cue_track *tracks = NULL;
+    size_t track_count = 0;
 
-    struct cue_track *tracks = f->tracks;
-    size_t track_count = f->num_tracks;
+    for (size_t n = 0; n < p->f->num_tracks; n++) {
+        struct cue_track *track = &p->f->tracks[n];
+        if (track->filename) {
+            MP_TARRAY_APPEND(ctx, tracks, track_count, *track);
+        } else {
+            MP_WARN(tl->demuxer, "No file specified for track entry %zd. "
+                    "It will be removed\n", n + 1);
+        }
+    }
 
     if (track_count == 0) {
         MP_ERR(tl, "CUE: no tracks found!\n");
@@ -224,10 +227,8 @@ static void build_timeline(struct timeline *tl)
         };
         chapters[i] = (struct demux_chapter) {
             .pts = timeline[i].start,
-            .metadata = talloc_zero(tl, struct mp_tags),
+            .metadata = mp_tags_dup(tl, tracks[i].tags),
         };
-        // might want to include other metadata here
-        mp_tags_set_str(chapters[i].metadata, "title", tracks[i].title);
         starttime += duration;
     }
 
@@ -261,9 +262,18 @@ static int try_open_file(struct demuxer *demuxer, enum demux_check check)
     struct priv *p = talloc_zero(demuxer, struct priv);
     demuxer->priv = p;
     demuxer->fully_read = true;
-    p->data = stream_read_complete(s, demuxer, 1000000);
-    if (p->data.start == NULL)
+
+    bstr data = stream_read_complete(s, p, 1000000);
+    if (data.start == NULL)
         return -1;
+    p->f = mp_parse_cue(data);
+    talloc_steal(p, p->f);
+    if (!p->f) {
+        MP_ERR(demuxer, "error parsing input file!\n");
+        return -1;
+    }
+
+    mp_tags_merge(demuxer->metadata, p->f->tags);
     return 0;
 }
 

demux/demux_disc.c

@@ -23,6 +23,7 @@
 #include "common/msg.h"
 
 #include "stream/stream.h"
+#include "video/mp_image.h"
 #include "demux.h"
 #include "stheader.h"
 
@@ -53,9 +54,10 @@ struct priv {
 static void reselect_streams(demuxer_t *demuxer)
 {
     struct priv *p = demuxer->priv;
-    for (int n = 0; n < MPMIN(p->slave->num_streams, p->num_streams); n++) {
+    int num_slave = demux_get_num_stream(p->slave);
+    for (int n = 0; n < MPMIN(num_slave, p->num_streams); n++) {
         if (p->streams[n]) {
-            demuxer_select_track(p->slave, p->slave->streams[n],
+            demuxer_select_track(p->slave, demux_get_stream(p->slave, n),
                 demux_stream_is_selected(p->streams[n]));
         }
     }
@@ -80,11 +82,9 @@ static void add_dvd_streams(demuxer_t *demuxer)
     struct stream_dvd_info_req info;
     if (stream_control(stream, STREAM_CTRL_GET_DVD_INFO, &info) > 0) {
         for (int n = 0; n < MPMIN(32, info.num_subs); n++) {
-            struct sh_stream *sh = new_sh_stream(demuxer, STREAM_SUB);
-            if (!sh)
-                break;
+            struct sh_stream *sh = demux_alloc_sh_stream(STREAM_SUB);
             sh->demuxer_id = n + 0x20;
-            sh->codec = "dvd_subtitle";
+            sh->codec->codec = "dvd_subtitle";
             get_disc_lang(stream, sh);
             // p->streams _must_ match with p->slave->streams, so we can't add
             // it yet - it has to be done when the real stream appears, which
@@ -93,17 +93,16 @@ static void add_dvd_streams(demuxer_t *demuxer)
 
             // emulate the extradata
             struct mp_csp_params csp = MP_CSP_PARAMS_DEFAULTS;
-            csp.int_bits_in = 8;
-            csp.int_bits_out = 8;
             struct mp_cmat cmatrix;
-            mp_get_yuv2rgb_coeffs(&csp, &cmatrix);
+            mp_get_csp_matrix(&csp, &cmatrix);
 
             char *s = talloc_strdup(sh, "");
             s = talloc_asprintf_append(s, "palette: ");
             for (int i = 0; i < 16; i++) {
                 int color = info.palette[i];
-                int c[3] = {(color >> 16) & 0xff, (color >> 8) & 0xff, color & 0xff};
-                mp_map_int_color(&cmatrix, 8, c);
+                int y[3] = {(color >> 16) & 0xff, (color >> 8) & 0xff, color & 0xff};
+                int c[3];
+                mp_map_fixp_color(&cmatrix, 8, y, 8, c);
                 color = (c[2] << 16) | (c[1] << 8) | c[0];
 
                 if (i != 0)
@@ -112,8 +111,10 @@ static void add_dvd_streams(demuxer_t *demuxer)
             }
             s = talloc_asprintf_append(s, "\n");
 
-            sh->extradata = s;
-            sh->extradata_size = strlen(s);
+            sh->codec->extradata = s;
+            sh->codec->extradata_size = strlen(s);
+
+            demux_add_sh_stream(demuxer, sh);
         }
     }
 }
@@ -122,9 +123,9 @@ static void add_streams(demuxer_t *demuxer)
 {
     struct priv *p = demuxer->priv;
 
-    for (int n = p->num_streams; n < p->slave->num_streams; n++) {
-        struct sh_stream *src = p->slave->streams[n];
-        if (src->sub) {
+    for (int n = p->num_streams; n < demux_get_num_stream(p->slave); n++) {
+        struct sh_stream *src = demux_get_stream(p->slave, n);
+        if (src->type == STREAM_SUB) {
             struct sh_stream *sub = NULL;
             if (src->demuxer_id >= 0x20 && src->demuxer_id <= 0x3F)
                 sub = p->dvd_subs[src->demuxer_id - 0x20];
@@ -134,25 +135,26 @@ static void add_streams(demuxer_t *demuxer)
                 continue;
             }
         }
-        struct sh_stream *sh = new_sh_stream(demuxer, src->type);
-        if (!sh)
-            break;
+        struct sh_stream *sh = demux_alloc_sh_stream(src->type);
         assert(p->num_streams == n); // directly mapped
         MP_TARRAY_APPEND(p, p->streams, p->num_streams, sh);
         // Copy all stream fields that might be relevant
-        sh->codec = talloc_strdup(sh, src->codec);
-        sh->codec_tag = src->codec_tag;
-        sh->lav_headers = src->lav_headers;
+        *sh->codec = *src->codec;
         sh->demuxer_id = src->demuxer_id;
-        if (src->video) {
+        if (src->type == STREAM_VIDEO) {
             double ar;
             if (stream_control(demuxer->stream, STREAM_CTRL_GET_ASPECT_RATIO, &ar)
                                 == STREAM_OK)
-                sh->video->aspect = ar;
+            {
+                struct mp_image_params f = {.w = src->codec->disp_w,
+                                            .h = src->codec->disp_h};
+                mp_image_params_set_dsize(&f, 1728 * ar, 1728);
+                sh->codec->par_w = f.p_w;
+                sh->codec->par_h = f.p_h;
+            }
         }
-        if (src->audio)
-            sh->audio = src->audio;
         get_disc_lang(demuxer->stream, sh);
+        demux_add_sh_stream(demuxer, sh);
     }
     reselect_streams(demuxer);
 }

demux/demux_edl.c

@@ -23,7 +23,7 @@
 #include <inttypes.h>
 #include <math.h>
 
-#include "talloc.h"
+#include "mpv_talloc.h"
 
 #include "demux.h"
 #include "timeline.h"

demux/demux_lavf.c

@@ -40,6 +40,7 @@
 #include "common/tags.h"
 #include "common/av_common.h"
 #include "misc/bstr.h"
+#include "misc/charset_conv.h"
 
 #include "stream/stream.h"
 #include "demux.h"
@@ -108,6 +109,7 @@ struct format_hack {
     bool no_stream : 1;         // do not wrap struct stream as AVIOContext
     bool use_stream_ids : 1;    // export the native stream IDs
     bool fully_read : 1;        // set demuxer.fully_read flag
+    bool detect_charset : 1;    // format is a small text file, possibly not UTF8
     bool image_format : 1;      // expected to contain exactly 1 frame
     // Do not confuse player's position estimation (position is into external
     // segment, with e.g. HLS, player knows about the playlist main file only).
@@ -115,8 +117,8 @@ struct format_hack {
 };
 
 #define BLACKLIST(fmt) {fmt, .ignore = true}
-#define TEXTSUB(fmt) {fmt, .fully_read = true}
-#define IMAGEFMT(fmt) {fmt, .image_format = true}
+#define TEXTSUB(fmt) {fmt, .fully_read = true, .detect_charset = true}
+#define TEXTSUB_UTF8(fmt) {fmt, .fully_read = true}
 
 static const struct format_hack format_hacks[] = {
     // for webradios
@@ -135,10 +137,12 @@ static const struct format_hack format_hacks[] = {
     {"h264", .if_flags = AVFMT_NOTIMESTAMPS },
     {"hevc", .if_flags = AVFMT_NOTIMESTAMPS },
 
-    TEXTSUB("aqtitle"), TEXTSUB("ass"), TEXTSUB("jacosub"), TEXTSUB("microdvd"),
+    TEXTSUB("aqtitle"), TEXTSUB("jacosub"), TEXTSUB("microdvd"),
     TEXTSUB("mpl2"), TEXTSUB("mpsub"), TEXTSUB("pjs"), TEXTSUB("realtext"),
     TEXTSUB("sami"), TEXTSUB("srt"), TEXTSUB("stl"), TEXTSUB("subviewer"),
-    TEXTSUB("subviewer1"), TEXTSUB("vplayer"), TEXTSUB("webvtt"),
+    TEXTSUB("subviewer1"), TEXTSUB("vplayer"), TEXTSUB("ass"),
+
+    TEXTSUB_UTF8("webvtt"),
 
     // Useless non-sense, sometimes breaks MLP2 subreader.c fallback
     BLACKLIST("tty"),
@@ -148,11 +152,12 @@ static const struct format_hack format_hacks[] = {
     // Useless, does not work with custom streams.
     BLACKLIST("image2"),
     // Image demuxers ("<name>_pipe" is detected explicitly)
-    IMAGEFMT("image2pipe"),
+    {"image2pipe", .image_format = true},
     {0}
 };
 
 typedef struct lavf_priv {
+    struct stream *stream;
     char *filename;
     struct format_hack format_hack;
     AVInputFormat *avif;
@@ -187,7 +192,8 @@ static bool matches_avinputformat_name(struct lavf_priv *priv,
 static int mp_read(void *opaque, uint8_t *buf, int size)
 {
     struct demuxer *demuxer = opaque;
-    struct stream *stream = demuxer->stream;
+    lavf_priv_t *priv = demuxer->priv;
+    struct stream *stream = priv->stream;
     int ret;
 
     ret = stream_read(stream, buf, size);
@@ -200,9 +206,13 @@ static int mp_read(void *opaque, uint8_t *buf, int size)
 static int64_t mp_seek(void *opaque, int64_t pos, int whence)
 {
     struct demuxer *demuxer = opaque;
-    struct stream *stream = demuxer->stream;
+    lavf_priv_t *priv = demuxer->priv;
+    struct stream *stream = priv->stream;
     int64_t current_pos;
-    MP_TRACE(demuxer, "mp_seek(%p, %"PRId64", %d)\n", stream, pos, whence);
+    MP_TRACE(demuxer, "mp_seek(%p, %"PRId64", %s)\n", stream, pos,
+             whence == SEEK_END ? "end" :
+             whence == SEEK_CUR ? "cur" :
+             whence == SEEK_SET ? "set" : "size");
     if (whence == SEEK_END || whence == AVSEEK_SIZE) {
         int64_t end = stream_get_size(stream);
         if (end < 0)
@@ -230,7 +240,8 @@ static int64_t mp_seek(void *opaque, int64_t pos, int whence)
 static int64_t mp_read_seek(void *opaque, int stream_idx, int64_t ts, int flags)
 {
     struct demuxer *demuxer = opaque;
-    struct stream *stream = demuxer->stream;
+    lavf_priv_t *priv = demuxer->priv;
+    struct stream *stream = priv->stream;
 
     struct stream_avseek cmd = {
         .stream_index = stream_idx,
@@ -253,6 +264,32 @@ static void list_formats(struct demuxer *demuxer)
         MP_INFO(demuxer, "%15s : %s\n", fmt->name, fmt->long_name);
 }
 
+static void convert_charset(struct demuxer *demuxer)
+{
+    lavf_priv_t *priv = demuxer->priv;
+    char *cp = demuxer->opts->sub_cp;
+    if (!cp || mp_charset_is_utf8(cp))
+        return;
+    bstr data = stream_read_complete(priv->stream, NULL, 128 * 1024 * 1024);
+    if (!data.start) {
+        MP_WARN(demuxer, "File too big (or error reading) - skip charset probing.\n");
+        return;
+    }
+    void *alloc = data.start;
+    cp = (char *)mp_charset_guess(priv, demuxer->log, data, cp, 0);
+    if (cp && !mp_charset_is_utf8(cp))
+        MP_INFO(demuxer, "Using subtitle charset: %s\n", cp);
+    // libavformat transparently converts UTF-16 to UTF-8
+    if (!mp_charset_is_utf16(cp) && !mp_charset_is_utf8(cp)) {
+        bstr conv = mp_iconv_to_utf8(demuxer->log, data, cp, MP_ICONV_VERBOSE);
+        if (conv.start)
+            data = conv;
+    }
+    if (data.start)
+        priv->stream = open_memory_stream(data.start, data.len);
+    talloc_free(alloc);
+}
+
 static char *remove_prefix(char *s, const char *const *prefixes)
 {
     for (int n = 0; prefixes[n]; n++) {
@@ -270,12 +307,8 @@ static int lavf_check_file(demuxer_t *demuxer, enum demux_check check)
 {
     struct MPOpts *opts = demuxer->opts;
     struct demux_lavf_opts *lavfdopts = opts->demux_lavf;
-    struct stream *s = demuxer->stream;
-    lavf_priv_t *priv;
-
-    assert(!demuxer->priv);
-    demuxer->priv = talloc_zero(NULL, lavf_priv_t);
-    priv = demuxer->priv;
+    lavf_priv_t *priv = demuxer->priv;
+    struct stream *s = priv->stream;
 
     priv->filename = remove_prefix(s->url, prefixes);
 
@@ -292,7 +325,7 @@ static int lavf_check_file(demuxer_t *demuxer, enum demux_check check)
         priv->filename = sep + 1;
     }
 
-    char *mime_type = demuxer->stream->mime_type;
+    char *mime_type = s->mime_type;
     if (!lavfdopts->allow_mimetype || !mime_type)
         mime_type = "";
 
@@ -393,6 +426,9 @@ static int lavf_check_file(demuxer_t *demuxer, enum demux_check check)
 
     demuxer->filetype = priv->avif->name;
 
+    if (priv->format_hack.detect_charset)
+        convert_charset(demuxer);
+
     return 0;
 }
 
@@ -464,7 +500,8 @@ static void select_tracks(struct demuxer *demuxer, int start)
     }
 }
 
-static void export_replaygain(demuxer_t *demuxer, sh_audio_t *sh, AVStream *st)
+static void export_replaygain(demuxer_t *demuxer, struct mp_codec_params *c,
+                              AVStream *st)
 {
     for (int i = 0; i < st->nb_side_data; i++) {
         AVReplayGain *av_rgain;
@@ -489,7 +526,7 @@ static void export_replaygain(demuxer_t *demuxer, sh_audio_t *sh, AVStream *st)
         rgain->album_peak = (av_rgain->album_peak != 0.0) ?
             av_rgain->album_peak / 100000.0f : 1.0;
 
-        sh->replaygain_data = rgain;
+        c->replaygain_data = rgain;
     }
 }
 
@@ -506,7 +543,7 @@ static int dict_get_decimal(AVDictionary *dict, const char *entry, int def)
     return def;
 }
 
-static void handle_stream(demuxer_t *demuxer, int i)
+static void handle_new_stream(demuxer_t *demuxer, int i)
 {
     lavf_priv_t *priv = demuxer->priv;
     AVFormatContext *avfc = priv->avfc;
@@ -516,27 +553,21 @@ static void handle_stream(demuxer_t *demuxer, int i)
 
     switch (codec->codec_type) {
     case AVMEDIA_TYPE_AUDIO: {
-        sh = new_sh_stream(demuxer, STREAM_AUDIO);
-        if (!sh)
-            break;
-        sh_audio_t *sh_audio = sh->audio;
+        sh = demux_alloc_sh_stream(STREAM_AUDIO);
 
         // probably unneeded
-        mp_chmap_set_unknown(&sh_audio->channels, codec->channels);
+        mp_chmap_set_unknown(&sh->codec->channels, codec->channels);
         if (codec->channel_layout)
-            mp_chmap_from_lavc(&sh_audio->channels, codec->channel_layout);
-        sh_audio->samplerate = codec->sample_rate;
-        sh_audio->bitrate = codec->bit_rate;
+            mp_chmap_from_lavc(&sh->codec->channels, codec->channel_layout);
+        sh->codec->samplerate = codec->sample_rate;
+        sh->codec->bitrate = codec->bit_rate;
 
-        export_replaygain(demuxer, sh_audio, st);
+        export_replaygain(demuxer, sh->codec, st);
 
         break;
     }
     case AVMEDIA_TYPE_VIDEO: {
-        sh = new_sh_stream(demuxer, STREAM_VIDEO);
-        if (!sh)
-            break;
-        sh_video_t *sh_video = sh->video;
+        sh = demux_alloc_sh_stream(STREAM_VIDEO);
 
         if (st->disposition & AV_DISPOSITION_ATTACHED_PIC) {
             sh->attached_picture = new_demux_packet_from(st->attached_pic.data,
@@ -548,8 +579,8 @@ static void handle_stream(demuxer_t *demuxer, int i)
             }
         }
 
-        sh_video->disp_w = codec->width;
-        sh_video->disp_h = codec->height;
+        sh->codec->disp_w = codec->width;
+        sh->codec->disp_h = codec->height;
         /* Try to make up some frame rate value, even if it's not reliable.
          * FPS information is needed to support subtitle formats which base
          * timing on frame numbers.
@@ -564,39 +595,31 @@ static void handle_stream(demuxer_t *demuxer, int i)
             fps = 1.0 / FFMAX(av_q2d(st->time_base),
                               av_q2d(st->codec->time_base) *
                               st->codec->ticks_per_frame);
-        sh_video->fps = fps;
+        sh->codec->fps = fps;
         if (priv->format_hack.image_format)
-            sh_video->fps = demuxer->opts->mf_fps;
-        if (st->sample_aspect_ratio.num)
-            sh_video->aspect = codec->width  * st->sample_aspect_ratio.num
-                    / (float)(codec->height * st->sample_aspect_ratio.den);
-        else
-            sh_video->aspect = codec->width  * codec->sample_aspect_ratio.num
-                    / (float)(codec->height * codec->sample_aspect_ratio.den);
+            sh->codec->fps = demuxer->opts->mf_fps;
+        sh->codec->par_w = st->sample_aspect_ratio.num;
+        sh->codec->par_h = st->sample_aspect_ratio.den;
 
         uint8_t *sd = av_stream_get_side_data(st, AV_PKT_DATA_DISPLAYMATRIX, NULL);
         if (sd) {
             double r = av_display_rotation_get((uint32_t *)sd);
             if (!isnan(r))
-                sh_video->rotate = (((int)(-r) % 360) + 360) % 360;
+                sh->codec->rotate = (((int)(-r) % 360) + 360) % 360;
         }
 
         // This also applies to vfw-muxed mkv, but we can't detect these easily.
-        sh_video->avi_dts = matches_avinputformat_name(priv, "avi");
+        sh->codec->avi_dts = matches_avinputformat_name(priv, "avi");
 
         break;
     }
     case AVMEDIA_TYPE_SUBTITLE: {
-        sh_sub_t *sh_sub;
-        sh = new_sh_stream(demuxer, STREAM_SUB);
-        if (!sh)
-            break;
-        sh_sub = sh->sub;
+        sh = demux_alloc_sh_stream(STREAM_SUB);
 
         if (codec->extradata_size) {
-            sh->extradata = talloc_size(sh, codec->extradata_size);
-            memcpy(sh->extradata, codec->extradata, codec->extradata_size);
-            sh->extradata_size = codec->extradata_size;
+            sh->codec->extradata = talloc_size(sh, codec->extradata_size);
+            memcpy(sh->codec->extradata, codec->extradata, codec->extradata_size);
+            sh->codec->extradata_size = codec->extradata_size;
         }
 
         if (matches_avinputformat_name(priv, "microdvd")) {
@@ -604,12 +627,12 @@ static void handle_stream(demuxer_t *demuxer, int i)
             if (av_opt_get_q(avfc, "subfps", AV_OPT_SEARCH_CHILDREN, &r) >= 0) {
                 // File headers don't have a FPS set.
                 if (r.num < 1 || r.den < 1)
-                    sh_sub->frame_based = av_q2d(av_inv_q(codec->time_base));
+                    sh->codec->frame_based = av_q2d(av_inv_q(codec->time_base));
             } else {
                 // Older libavformat versions. If the FPS matches the microdvd
                 // reader's default, assume it uses frame based timing.
                 if (codec->time_base.num == 125 && codec->time_base.den == 2997)
-                    sh_sub->frame_based = 23.976;
+                    sh->codec->frame_based = 23.976;
             }
         }
         break;
@@ -633,9 +656,11 @@ static void handle_stream(demuxer_t *demuxer, int i)
 
     if (sh) {
         sh->ff_index = st->index;
-        sh->codec = mp_codec_from_av_codec_id(codec->codec_id);
-        sh->codec_tag = codec->codec_tag;
-        sh->lav_headers = codec;
+        sh->codec->codec = mp_codec_from_av_codec_id(codec->codec_id);
+        sh->codec->codec_tag = codec->codec_tag;
+        sh->codec->lav_headers = avcodec_alloc_context3(NULL);
+        if (sh->codec->lav_headers)
+            mp_copy_lav_codec_headers(sh->codec->lav_headers, codec);
 
         if (st->disposition & AV_DISPOSITION_DEFAULT)
             sh->default_track = true;
@@ -653,10 +678,10 @@ static void handle_stream(demuxer_t *demuxer, int i)
         if (!sh->title && sh->hls_bitrate > 0)
             sh->title = talloc_asprintf(sh, "bitrate %d", sh->hls_bitrate);
         sh->missing_timestamps = !!(priv->avif_flags & AVFMT_NOTIMESTAMPS);
+        demux_add_sh_stream(demuxer, sh);
     }
 
     select_tracks(demuxer, i);
-    demux_changed(demuxer, DEMUX_EVENT_STREAMS);
 }
 
 // Add any new streams that might have been added
@@ -664,7 +689,7 @@ static void add_new_streams(demuxer_t *demuxer)
 {
     lavf_priv_t *priv = demuxer->priv;
     while (priv->num_streams < priv->avfc->nb_streams)
-        handle_stream(demuxer, priv->num_streams);
+        handle_new_stream(demuxer, priv->num_streams);
 }
 
 static void update_metadata(demuxer_t *demuxer, AVPacket *pkt)
@@ -690,7 +715,8 @@ static void update_metadata(demuxer_t *demuxer, AVPacket *pkt)
 static int interrupt_cb(void *ctx)
 {
     struct demuxer *demuxer = ctx;
-    return mp_cancel_test(demuxer->stream->cancel);
+    lavf_priv_t *priv = demuxer->priv;
+    return mp_cancel_test(priv->stream->cancel);
 }
 
 static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
@@ -700,15 +726,13 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
     AVFormatContext *avfc;
     AVDictionaryEntry *t = NULL;
     float analyze_duration = 0;
-    int i;
+    lavf_priv_t *priv = talloc_zero(NULL, lavf_priv_t);
+    demuxer->priv = priv;
+    priv->stream = demuxer->stream;
 
     if (lavf_check_file(demuxer, check) < 0)
         return -1;
 
-    lavf_priv_t *priv = demuxer->priv;
-    if (!priv)
-        return -1;
-
     avfc = avformat_alloc_context();
     if (!avfc)
         return -1;
@@ -747,7 +771,7 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
     AVDictionary *dopts = NULL;
 
     if ((priv->avif_flags & AVFMT_NOFILE) ||
-        demuxer->stream->type == STREAMTYPE_AVDEVICE ||
+        priv->stream->type == STREAMTYPE_AVDEVICE ||
         priv->format_hack.no_stream)
     {
         mp_setup_av_network_options(&dopts, demuxer->global, demuxer->log, opts);
@@ -766,7 +790,7 @@ 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)
+        if (stream_control(priv->stream, STREAM_CTRL_HAS_AVSEEK, NULL) > 0)
             demuxer->seekable = true;
     }
 
@@ -806,9 +830,9 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
     }
 
     MP_VERBOSE(demuxer, "avformat_find_stream_info() finished after %"PRId64
-               " bytes.\n", stream_tell(demuxer->stream));
+               " bytes.\n", stream_tell(priv->stream));
 
-    for (i = 0; i < avfc->nb_chapters; i++) {
+    for (int i = 0; i < avfc->nb_chapters; i++) {
         AVChapter *c = avfc->chapters[i];
         t = av_dict_get(c->metadata, "title", NULL, 0);
         int index = demuxer_add_chapter(demuxer, t ? t->value : "",
@@ -820,7 +844,7 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
 
     // Often useful with OGG audio-only files, which have metadata in the audio
     // track metadata instead of the main metadata.
-    if (demuxer->num_streams == 1) {
+    if (demux_get_num_stream(demuxer) == 1) {
         priv->merge_track_metadata = true;
         for (int n = 0; n < priv->num_streams; n++) {
             if (priv->streams[n])
@@ -850,7 +874,7 @@ static int demux_lavf_fill_buffer(demuxer_t *demux)
     AVPacket *pkt = &(AVPacket){0};
     int r = av_read_frame(priv->avfc, pkt);
     if (r < 0) {
-        av_free_packet(pkt);
+        av_packet_unref(pkt);
         if (r == AVERROR(EAGAIN))
             return 1;
         if (r == AVERROR_EOF)
@@ -867,13 +891,13 @@ static int demux_lavf_fill_buffer(demuxer_t *demux)
     AVStream *st = priv->avfc->streams[pkt->stream_index];
 
     if (!demux_stream_is_selected(stream)) {
-        av_free_packet(pkt);
+        av_packet_unref(pkt);
         return 1; // don't signal EOF if skipping a packet
     }
 
     struct demux_packet *dp = new_demux_packet_from_avpacket(pkt);
     if (!dp) {
-        av_free_packet(pkt);
+        av_packet_unref(pkt);
         return 1;
     }
 
@@ -882,8 +906,10 @@ static int demux_lavf_fill_buffer(demuxer_t *demux)
     if (pkt->dts != AV_NOPTS_VALUE)
         dp->dts = pkt->dts * av_q2d(st->time_base);
     dp->duration = pkt->duration * av_q2d(st->time_base);
+#if !HAVE_AV_AVPACKET_INT64_DURATION
     if (pkt->convergence_duration > 0)
         dp->duration = pkt->convergence_duration * av_q2d(st->time_base);
+#endif
     dp->pos = pkt->pos;
     dp->keyframe = pkt->flags & AV_PKT_FLAG_KEY;
     if (dp->pts != MP_NOPTS_VALUE) {
@@ -891,7 +917,7 @@ static int demux_lavf_fill_buffer(demuxer_t *demux)
     } else if (dp->dts != MP_NOPTS_VALUE) {
         priv->last_pts = dp->dts * AV_TIME_BASE;
     }
-    av_free_packet(pkt);
+    av_packet_unref(pkt);
 
     if (priv->format_hack.clear_filepos)
         dp->pos = -1;
@@ -916,7 +942,7 @@ static void demux_seek_lavf(demuxer_t *demuxer, double rel_seek_secs, int flags)
         avsflags = AVSEEK_FLAG_BACKWARD;
 
     if (flags & SEEK_FACTOR) {
-        struct stream *s = demuxer->stream;
+        struct stream *s = priv->stream;
         int64_t end = stream_get_size(s);
         if (end > 0 && demuxer->ts_resets_possible &&
             !(priv->avif_flags & AVFMT_NO_BYTE_SEEK))
@@ -1036,6 +1062,12 @@ redo:
             goto redo;
         }
         priv->cur_program = prog->progid = program->id;
+
+        mp_tags_copy_from_av_dictionary(demuxer->metadata, priv->avfc->programs[p]->metadata);
+        update_metadata(demuxer, NULL);
+        // Enforce metadata update even if no explicit METADATA_UPDATED since we switched program.
+        demux_changed(demuxer, DEMUX_EVENT_METADATA);
+
         return DEMUXER_CTRL_OK;
     }
     case DEMUXER_CTRL_RESYNC:
@@ -1052,10 +1084,10 @@ redo:
          */
         // avio_flush() is designed for write-only streams, and does the wrong
         // thing when reading. Flush it manually instead.
-        stream_drop_buffers(demuxer->stream);
+        stream_drop_buffers(priv->stream);
         priv->avfc->pb->buf_ptr = priv->avfc->pb->buf_end = priv->avfc->pb->buffer;
-        priv->avfc->pb->pos = stream_tell(demuxer->stream);
-        av_seek_frame(priv->avfc, 0, stream_tell(demuxer->stream),
+        priv->avfc->pb->pos = stream_tell(priv->stream);
+        av_seek_frame(priv->avfc, 0, stream_tell(priv->stream),
                       AVSEEK_FLAG_BYTE);
         return DEMUXER_CTRL_OK;
     default:
@@ -1074,6 +1106,12 @@ static void demux_close_lavf(demuxer_t *demuxer)
         if (priv->pb)
             av_freep(&priv->pb->buffer);
         av_freep(&priv->pb);
+        for (int n = 0; n < priv->num_streams; n++) {
+            if (priv->streams[n])
+                avcodec_free_context(&priv->streams[n]->codec->lav_headers);
+        }
+        if (priv->stream != demuxer->stream)
+            free_stream(priv->stream);
         talloc_free(priv);
         demuxer->priv = NULL;
     }

demux/demux_libarchive.c

@@ -71,7 +71,7 @@ static int open_file(struct demuxer *demuxer, enum demux_check check)
         if (r == ARCHIVE_EOF)
             break;
         if (r < ARCHIVE_OK)
-            MP_ERR(demuxer, "libarchive: %s\n", archive_error_string(mpa->arch));
+            MP_ERR(demuxer, "%s\n", archive_error_string(mpa->arch));
         if (r < ARCHIVE_WARN)
             break;
         if (archive_entry_filetype(entry) != AE_IFREG)