Release 0.29.1

cocoa-cb: fix side by side Split View
when entering a Split View a windowDidEnterFullScreen event happens without a previous toggleFullScreen call. in that case it tries to stop an animation that was never initiated by us and basically breaks the system initiated fullscreen, or in this case the Split View. immediately after entering the fullscreen it tries top stop the animation and resizes the window, which causes the window to exit fullscreen. only try to stop an animation that was initiated by us and is safe to stop. (cherry picked from commit 2b0b9bb6a1)
2018-10-02 22:03:41 +03:00 · 2018-10-02 20:36:55 +03:00 · 2018-10-02 00:42:58 +03:00 · 2018-10-02 00:42:51 +03:00 · 2018-10-01 20:29:51 +03:00 · 2018-10-01 18:55:06 +03:00
344 changed files with 26001 additions and 17698 deletions
@@ -12,7 +12,7 @@ Releases are listed here: https://github.com/mpv-player/mpv/releases
 ### Log file

 Make a log file made with -v or --log-file=output.txt, paste it to
-http://sprunge.us or a similar site, and replace this text with a link to it.
+https://0x0.st/ or a similar site, and replace this text with a link to it.

 Providing a log file is strongly encouraged. It is very helpful for reproducing
 bugs or getting important technical context. Without a log it might not be
@@ -1,43 +1,39 @@
-language: c
+language: generic

 os:
  - linux
-  #- osx
+
+services:
+  - docker
+
+sudo: required
+
 env:
  matrix:
-    - LIBAV=ffmpeg-git
-    - LIBAV=libav-git
+    - CONTAINER=registry.cirno.systems/kiwi/containers/mpv-ci:stable-deps CI_SCRIPT=ci/build-tumbleweed.sh CC=gcc
+    - CONTAINER=registry.cirno.systems/kiwi/containers/mpv-ci:stable-deps CI_SCRIPT=ci/build-tumbleweed.sh CC=clang
+    - CONTAINER=registry.cirno.systems/kiwi/containers/mpv-ci:snapshot-deps CI_SCRIPT=ci/build-tumbleweed.sh CC=gcc
+    - CONTAINER=registry.cirno.systems/kiwi/containers/mpv-ci:snapshot-deps CI_SCRIPT=ci/build-tumbleweed.sh CC=clang
+    - CONTAINER=registry.cirno.systems/kiwi/containers/mpv-ci-mingw64:i686 CI_SCRIPT=ci/build-mingw64.sh TARGET=i686-w64-mingw32
+    - CONTAINER=registry.cirno.systems/kiwi/containers/mpv-ci-mingw64:x86_64 CI_SCRIPT=ci/build-mingw64.sh TARGET=x86_64-w64-mingw32
  global:
    # Coverity token
    - secure: "H21mSRlMhk4BKS0xHZvCFGJxteCP0hRVUxTuNfM2Z9HBsyutuLEYMtViLO86VtM+Tqla3xXPzUdS4ozLwI72Ax/5ZUDXACROj73yW6QhFB5D6rLut12+FjqC7M33Qv2hl0xwgNBmR5dsm1ToP37+Wn+ecJQNvN8fkTXF+HVzOEw="
    # Travis token for mpv.io
    - secure: "nlTVLJK6kRhtXvhKCoJ3YdFGHuKaq/eHowfPw25hqRWuBOZd+HjHY5KIYjV7SxuKFDpJE4GpNcvA3Q31nsqomxpkLYgrwjg6TSazN7ZP+x85ZgV1QGFebrPfGm2n5UR5CAPAwFoeF3pZheLi4bajVzwq1fWW+x3grS188P9OZso="
-compiler:
-  - clang
-  - gcc

 branches:
  only:
    - master
    - ci
    - coverity_scan
+    - /release\/.*$/

-matrix:
-  exclude:
-    - os: osx
-      env: LIBAV=libav-git
-    - os: osx
-      env: LIBAV=ffmpeg-git
-    - os: osx
-      compiler: gcc
-    - os: linux
-      compiler: clang
-
-before_install: TOOLS/travis-deps libass-stable $LIBAV
+before_install:
+  - docker pull $CONTAINER
 script:
  - ./bootstrap.py
-  - ./waf configure
-  - ./waf build
+  - docker run --env CC --env TARGET -v $TRAVIS_BUILD_DIR:/build $CONTAINER /bin/sh -c "cd /build && $CI_SCRIPT"
 after_failure: cat ./build/config.log
 after_script: TOOLS/travis-rebuild-website

@@ -16,7 +16,7 @@ licensed as LGPLv2.1+. Also see the list of files with specific licenses below
 All new contributions must be LGPLv2.1+ licensed. Using a more liberal license
 compatible to LGPLv2.1+ is also ok.

-If changes are done on GPL code, must come with the implicit agreement that the
+Changes done to GPL code must come with the implicit/explicit agreement that the
 project can relicense the changes to LGPLv2.1+ at a later point without asking
 the contributor. This is a safeguard for making potential relicensing of
 remaining GPL code to LGPLv2.1+ easier.
@@ -34,8 +34,6 @@ a LGPL mode to mpv, MPlayer code had to be relicensed from GPLv2+ to LGPLv2.1+
 by asking the MPlayer authors for permission. Since permission could not be
 obtained from everyone, LGPL mode disables the following features, some of
 them quite central:
- no audio filtering, which breaks: --af, pitch correction, fine control over
-  downmix/upmix/resampling behavior
 - Linux X11 video output
 - BSD audio output via OSS
 - NVIDIA/Linux hardware decoding (vdpau, although nvdec usually works)
@@ -47,15 +45,8 @@ at all.

 The following files are still GPL only (--enable-lgpl disables them):

-    audio/filter/*                  will be replaced with new filter chain
-    audio/filter/af_format.c        mostly LGPL (except af glue code)
-    audio/filter/af_lavc3enc.c          as above
-    audio/filter/af_lavfi.c             as above
-    audio/filter/af_scaletempo.c        as above
-    audio/filter/af_rubberband.c        as above
    audio/out/ao_jack.c             will stay GPL
    audio/out/ao_oss.c              will stay GPL
-    audio/audio.*                   needed by af code only
    demux/demux_tv.c                will stay GPL
    stream/ai_*                     will stay GPL (TV code)
    stream/audio_in.*               will stay GPL (TV code)
@@ -88,7 +79,6 @@ The following files are still GPL only (--enable-lgpl disables them):
 The following files contain some optional GPL code (--enable-lgpl disables it):

    options/parse_commandline.c     dvd:// expansion
-    player/audio.c                  libaf glue code

 None of the exceptions listed above affect the final binary if it's built as
 LGPL. Linked libraries still can affect the final license (for example if
@@ -33,6 +33,61 @@ API changes
 ::

 --- mpv 0.29.0 ---
+ 1.101  - add MPV_RENDER_PARAM_ADVANCED_CONTROL and related API
+        - add MPV_RENDER_PARAM_NEXT_FRAME_INFO and related symbols
+        - add MPV_RENDER_PARAM_BLOCK_FOR_TARGET_TIME
+        - add MPV_RENDER_PARAM_SKIP_RENDERING
+        - add mpv_render_context_get_info()
+ 1.100  - bump API number to avoid confusion with mpv release versions
+        - actually apply the GL_MP_MPGetNativeDisplay change for the new render
+          API. This also means compatibility for anything but x11 and wayland
+          through the old opengl-cb GL_MP_MPGetNativeDisplay method is now
+          unsupported.
+        - deprecate mpv_get_wakeup_pipe(). It's complex, but easy to replace
+          using normal API (just set a wakeup callback to a function which
+          writes to a pipe).
+        - add a 1st class hook API, which replaces the hacky mpv_command()
+          based one. The old API is deprecated and will be removed soon. The
+          old API was never meant to be stable, while the new API is.
+ 1.29   - the behavior of mpv_terminate_destroy() and mpv_detach_destroy()
+          changes subtly (see documentation in the header file). In particular,
+          mpv_detach_destroy() will not leave the player running in all
+          situations anymore (it gets closer to refcounting).
+        - rename mpv_detach_destroy() to mpv_destroy() (the old function will
+          remain valid as deprecated alias)
+        - add mpv_create_weak_client(), which makes use of above changes
+        - MPV_EVENT_SHUTDOWN is now returned exactly once if a mpv_handle
+          should terminate, instead of spamming the event queue with this event
+ 1.28   - deprecate the render opengl_cb API, and replace it with render.h
+          and render_gl.h. The goal is allowing support for APIs other than
+          OpenGL. The old API is emulated with the new API.
+          Likewise, the "opengl-cb" VO is renamed to "libmpv".
+          mpv_get_sub_api() is deprecated along the opengl_cb API.
+          The new API is relatively similar, but not the same. The rough
+          equivalents are:
+            mpv_opengl_cb_init_gl => mpv_render_context_create
+            mpv_opengl_cb_set_update_callback => mpv_render_context_set_update_callback
+            mpv_opengl_cb_draw => mpv_render_context_render
+            mpv_opengl_cb_report_flip => mpv_render_context_report_swap
+            mpv_opengl_cb_uninit_gl => mpv_render_context_free
+          The VO opengl-cb is also renamed to "libmpv".
+          Also, the GL_MP_MPGetNativeDisplay pseudo extension is not used by the
+          render API anymore, and the old opengl-cb API only handles the "x11"
+          and "wl" names anymore. Support for everything else has been removed.
+          The new render API uses proper API parameters, e.g. for X11 you pass
+          MPV_RENDER_PARAM_X11_DISPLAY directly.
+        - deprecate the qthelper.hpp header file. This provided some C++ helper
+          utility functions for Qt with use of libmpv. There is no reason to
+          keep this in the mpv git repository, nor to make it part of the libmpv
+          API. If you're using this header, you can safely copy it into your
+          project - it uses only libmpv public API. Alternatively, it could be
+          maintained in a separate repository by interested parties.
+ 1.27   - make opengl-cb the default VO. This causes a subtle behavior change
+          if the API user called mpv_opengl_cb_init_gl(), but does not set
+          the "vo" option. Before, it would still have used another VO (like
+          on the CLI, e.g. vo=gpu). Now it'll behave as if vo=opengl-cb was
+          used.
+ --- mpv 0.28.0 ---
 1.26   - remove glMPGetNativeDisplay("drm") support
        - add mpv_opengl_cb_window_pos and mpv_opengl_cb_drm_params and
          support via glMPGetNativeDisplay() for using it
@@ -3,8 +3,7 @@ General usage

 ::

-  mpv infile -o outfile [-of outfileformat] [-ofopts formatoptions] \
-    [-ofps outfps | -oautofps] [-oharddup] [-ocopyts | -orawts] [-oneverdrop] \
+  mpv infile -o outfile [-of outfileformat] [-ofopts formatoptions] [-orawts] \
    [(any other mpv options)] \
    -ovc outvideocodec [-ovcopts outvideocodecoptions] \
    -oac outaudiocodec [-oacopts outaudiocodecoptions]
@@ -60,13 +59,13 @@ for.
 Typical MPEG-4 Part 2 ("ASP", "DivX") encoding, AVI container::

  mpv infile -o outfile.avi \
-    -ofps 25 \
+    --vf=fps=25 \
    -ovc mpeg4 -ovcopts qscale=4 \
    -oac libmp3lame -oacopts ab=128k

-Note: AVI does not support variable frame rate, so -ofps must be used. The
-frame rate should ideally match the input (25 for PAL, 24000/1001 or 30000/1001
-for NTSC)
+Note: AVI does not support variable frame rate, so the fps filter must be used.
+The frame rate should ideally match the input (25 for PAL, 24000/1001 or
+30000/1001 for NTSC)

 Typical MPEG-4 Part 10 ("AVC", "H.264") encoding, Matroska (MKV) container::

@@ -129,7 +128,7 @@ What works
 ==========

 * Encoding at variable frame rate (default)
-* Encoding at constant frame rate using -ofps framerate -oharddup
+* Encoding at constant frame rate using --vf=fps=RATE
 * 2-pass encoding (specify flags=+pass1 in the first pass's -ovcopts, specify
  flags=+pass2 in the second pass)
 * Hardcoding subtitles using vobsub, ass or srt subtitle rendering (just
@@ -19,6 +19,96 @@ Interface changes

 ::

+ --- mpv 0.29.1 ---
+    - add --cocoa-cb-sw-renderer to control the usage of Apple Software Renderer
+ --- mpv 0.29.0 ---
+    - drop --opensles-sample-rate, as --audio-samplerate should be used if desired
+    - drop deprecated --videotoolbox-format, --ff-aid, --ff-vid, --ff-sid,
+      --ad-spdif-dtshd, --softvol options
+    - fix --external-files: strictly never select any tracks from them, unless
+      explicitly selected (this may or may not be expected)
+    - --ytdl is now always enabled, even for libmpv
+    - add a number of --audio-resample-* options, which should from now on be
+      used instead of --af-defaults=lavrresample:...
+    - deprecate --vf-defaults and --af-defaults. These didn't work with the
+      lavfi bridge, so they have very little use left. The only potential use
+      is with af_lavrresample (going to be deprecated, --audio-resample-... set
+      its defaults), and various hw deinterlacing filters (like vf_vavpp), for
+      which you will have to stop using --deinterlace=yes, and instead use the
+      vf toggle commands and the filter enable/disable flag to customize it.
+    - deprecate --af=lavrresample. Use the ``--audio-resample-...`` options to
+      customize resampling, or the libavfilter ``--af=aresample`` filter.
+    - add --osd-on-seek
+    - remove outfmt sub-parameter from "format" video filter (no replacement)
+    - some behavior changes in the video filter chain, including:
+        - before, using an incompatible filter with hwdec would disable hwdec;
+          now it disables the filter at runtime instead
+        - inserting an incompatible filter with hwdec at runtime would refuse
+          to insert the filter; now it will add it successfully, but disables
+          the filter slightly later
+    - some behavior changes in the audio filter chain, including:
+        - a manually inserted lavrresample filter is not necessarily used for
+          sample format conversion anymore, so it's pretty useless
+        - changing playback speed will not respect --af-defaults anymore
+        - having libavfilter based filters after the scaletempo or rubberband
+          filters is not supported anymore, and may desync if playback speed is
+          changed (libavfilter does not support the metadata for playback speed)
+        - the lavcac3enc filter does not auto detach itself anymore; instead it
+          passes through the data after converting it to the sample rate and
+          channel configuration the ac3 encoder expects; also, if the audio
+          format changes midstream in a way that causes the filter to switch
+          between PCM and AC3 output, the audio output won't be reconfigured,
+          and audio playback will fail due to libswresample being unable to
+          convert between PCM and AC3 (Note: the responsible developer didn't
+          give a shit. Later changes might have improved or worsened this.)
+        - inserting a filter that changes the output sample format will not
+          reconfigure the AO - you need to run an additional "ao-reload"
+          command to force this if you want that
+        - using "strong" gapless audio (--gapless-audio=yes) can fail if the
+          audio formats are not convertible (such as switching between PCM and
+          AC3 passthrough)
+        - if filters do not pass through PTS values correctly, A/V sync can
+          result over time. Some libavfilter filters are known to be affected by
+          this, such as af_loudnorm, which can desync over time, depending on
+          how the audio track was muxed (af_lavfi's fix-pts suboption can help).
+    - remove out-format sub-parameter from "format" audio filter (no replacement)
+    - --lavfi-complex now requires uniquely named filter pads. In addition,
+      unconnected filter pads are not allowed anymore (that means every filter
+      pad must be connected either to another filter, or to a video/audio track
+      or video/audio output). If they are disconnected at runtime, the stream
+      will probably stall.
+    - rename --vo=opengl-cb to --vo=libmpv (goes in hand with the opengl-cb
+      API deprecation, see client-api-changes.rst)
+    - deprecate the OpenGL cocoa backend, option choice --gpu-context=cocoa
+      when used with --gpu-api=opengl (use --vo=libmpv)
+    - make --deinterlace=yes always deinterlace, instead of trying to check
+      certain unreliable video metadata. Also flip the defaults of all builtin
+      HW deinterlace filters to always deinterlace.
+    - change vf_vavpp default to use the best deinterlace algorithm by default
+    - remove a compatibility hack that allowed CLI aliases to be set as property
+      (such as "sub-file"), deprecated in mpv 0.26.0
+    - deprecate the old command based hook API, and introduce a proper C API
+      (the high level Lua API for this does not change)
+    - rename the the lua-settings/ config directory to script-opts/
+    - the way the player waits for scripts getting loaded changes slightly. Now
+      scripts are loaded in parallel, and block the player from continuing
+      playback only in the player initialization phase. It could change again in
+      the future. (This kind of waiting was always a feature to prevent that
+      playback is started while scripts are only half-loaded.)
+    - deprecate --ovoffset, --oaoffset, --ovfirst, --oafirst
+    - remove the following encoding options: --ocopyts (now the default, old
+      timestamp handling is gone), --oneverdrop (now default), --oharddup (you
+      need to use --vf=fps=VALUE), --ofps, --oautofps, --omaxfps
+    - remove --video-stereo-mode. This option was broken out of laziness, and
+      nobody wants to fix it. Automatic 3D down-conversion to 2D is also broken,
+      although you can just insert the stereo3d filter manually. The obscurity
+      of 3D content doesn't justify such an option anyway.
+    - change cycle-values command to use the current value, instead of an
+      internal counter that remembered the current position.
+    - edition and disc title switching will now fully reload playback (may have
+      consequences for scripts, client API, or when using file-local options)
+    - remove deprecated ao/vo auto profiles. Consider using scripts like
+      auto-profiles.lua instead.
 --- mpv 0.28.0 ---
    - rename --hwdec=mediacodec option to mediacodec-copy, to reflect
      conventions followed by other hardware video decoding APIs
@@ -59,11 +149,12 @@ Interface changes
      now.
    - change the --hwdec option from a choice to a plain string (affects
      introspection of the option/property), also affects some properties
-    - rename --hwdec=rpi to --hwdec=mmal, sane for the -copy variant (no
+    - rename --hwdec=rpi to --hwdec=mmal, same for the -copy variant (no
      backwards compatibility)
-    - deprecate the --ff-aid, --ff-vid, -ff-sid options and properties (there is
+    - deprecate the --ff-aid, --ff-vid, --ff-sid options and properties (there is
      no replacement, but you can manually query the track property and use the
      "ff-index" field to find the mpv track ID to imitate this behavior)
+    - rename --no-ometadata to --no-ocopy-metadata
 --- mpv 0.27.0 ---
    - drop previously deprecated --field-dominance option
    - drop previously deprecated "osd" command
@@ -5,7 +5,8 @@ Audio filters allow you to modify the audio stream and its properties. The
 syntax is:

 ``--af=...``
-    Setup a chain of audio filters. See ``--vf`` for the syntax.
+    Setup a chain of audio filters. See ``--vf`` (`VIDEO FILTERS`_) for the
+    full syntax.

 .. note::

@@ -29,6 +30,12 @@ Available filters are:
    This filter is automatically enabled if the audio output does not support
    the audio configuration of the file being played.

+    .. warning::
+
+        Deprecated. Either use the ``--audio-resample-...`` options to customize
+        resampling, or the libavfilter ``--af=aresample`` filter, which has its
+        own options.
+
    It supports only the following sample formats: u8, s16, s32, float.

    ``filter-size=<length>``
@@ -46,7 +53,8 @@ Available filters are:
        Do not detach if input and output audio format/rate/channels match.
        (If you just want to set defaults for this filter that will be used
        even by automatically inserted lavrresample instances, you should
-        prefer setting them with ``--af-defaults=lavrresample:...``.)
+        prefer setting them with the ``--audio-resample-...`` options.) This
+        does not do anything anymore and the filter will never detach.
    ``normalize=<yes|no|auto>``
        Whether to normalize when remixing channel layouts (default: auto).
        ``auto`` uses the value set by ``--audio-normalize-downmix``.
@@ -91,7 +99,7 @@ Available filters are:
        Select the libavcodec encoder used. Currently, this should be an AC-3
        encoder, and using another codec will fail horribly.

-``format=format:srate:channels:out-format:out-srate:out-channels``
+``format=format:srate:channels:out-srate:out-channels``
    Does not do any format conversion itself. Rather, it may cause the
    filter system to insert necessary conversion filters before or after this
    filter if needed. It is primarily useful for controlling the audio format
@@ -120,8 +128,6 @@ Available filters are:
        Force mixing to a specific channel layout. See ``--audio-channels`` option
        for possible values.

-    ``<out-format>``
-
    ``<out-srate>``

    ``<out-channels>``
@@ -219,6 +225,12 @@ Available filters are:
        change the playback pitch at runtime. Note that speed is controlled
        using the standard ``speed`` property, not ``af-command``.

+    ``multiply-pitch <factor>``
+        Multiply the current value of ``<pitch-scale>`` dynamically.  For
+        example: 0.5 to go down by an octave, 1.5 to go up by a perfect fifth.
+        If you want to go up or down by semi-tones, use 1.059463094352953 and
+        0.9438743126816935
+
 ``lavfi=graph``
    Filter audio using FFmpeg's libavfilter.

@@ -233,3 +245,16 @@ Available filters are:

    ``o=<string>``
        AVOptions.
+
+    ``fix-pts=<yes|no>``
+        Determine PTS based on sample count (default: no). If this is enabled,
+        the player won't rely on libavfilter passing through PTS accurately.
+        Instead, it pass a sample count as PTS to libavfilter, and compute the
+        PTS used by mpv based on that and the input PTS. This helps with filters
+        which output a recomputed PTS instead of the original PTS (including
+        filters which require the PTS to start at 0). mpv normally expects
+        filters to not touch the PTS (or only to the extent of changing frame
+        boundaries), so this is not the default, but it will be needed to use
+        broken filters. In practice, these broken filters will either cause slow
+        A/V desync over time (with some files), or break playback completely if
+        you seek or start playback from the middle of a file.
@@ -27,7 +27,7 @@ Available audio output drivers are:
    .. warning::

        To get multichannel/surround audio, use ``--audio-channels=auto``. The
-        default for this option is ``auto-safe``, which makes this audio otuput
+        default for this option is ``auto-safe``, which makes this audio output
        explicitly reject multichannel output, as there is no way to detect
        whether a certain channel layout is actually supported.

@@ -110,10 +110,21 @@ Available audio output drivers are:
    exclusive mode (bypasses the sound server).

 ``openal``
-    Experimental OpenAL audio output driver
+    OpenAL audio output driver

-    .. note:: This driver is not very useful. Playing multi-channel audio with
-              it is slow.
+    ``--openal-num-buffers=<2-128>``
+        Specify the number of audio buffers to use. Lower values are better for
+        lower CPU usage. Default: 4.
+
+    ``--openal-num-samples=<256-32768>``
+        Specify the number of complete samples to use for each buffer. Higher
+        values are better for lower CPU usage. Default: 8192.
+
+    ``--openal-direct-channels=<yes|no>``
+        Enable OpenAL Soft's direct channel extension when available to avoid
+        tinting the sound with ambisonics or HRTF.
+        Channels are dropped when when they are not available as downmixing
+        will be disabled. Default: no.

 ``pulse``
    PulseAudio audio output driver
@@ -128,7 +139,7 @@ Available audio output drivers are:
        Set the audio buffer size in milliseconds. A higher value buffers
        more data, and has a lower probability of buffer underruns. A smaller
        value makes the audio stream react faster, e.g. to playback speed
-        changes. Default: 250.
+        changes.

    ``--pulse-latency-hacks=<yes|no>``
        Enable hacks to workaround PulseAudio timing bugs (default: no). If
@@ -198,6 +209,9 @@ Available audio output drivers are:
        If not empty, this is a ``,`` separated list of channel layouts the
        AO allows. This can be used to test channel layout selection.

+    ``--ao-null-format``
+        Force the audio output format the AO will accept. If unset accepts any.
+
 ``pcm``
    Raw PCM/WAVE file writer audio output

@@ -8,9 +8,8 @@ You can encode files from one format/codec to another using this facility.

 ``--of=<format>``
    Specifies the output format (overrides autodetection by the file name
-    extension of the file specified by ``-o``). This can be a comma separated
-    list of possible formats to try. See ``--of=help`` for a full list of
-    supported formats.
+    extension of the file specified by ``-o``). See ``--of=help`` for a full
+    list of supported formats.

 ``--ofopts=<options>``
    Specifies the output format options for libavformat.
@@ -22,50 +21,16 @@ You can encode files from one format/codec to another using this facility.
    ``--ofopts-add=<options1[,options2,...]>``
        Appends the options given as arguments to the options list.

-    ``--ofopts-pre=<options1[,options2,...]>``
-        Prepends the options given as arguments to the options list.
-
-    ``--ofopts-del=<index1[,index2,...]>``
-        Deletes the options at the given indexes. Index numbers start at 0,
-        negative numbers address the end of the list (-1 is the last).
-
-    ``--ofopts-clr``
+    ``--ofopts=""``
        Completely empties the options list.

-``--ofps=<float value>``
-    Specifies the output format time base (default: 24000). Low values like 25
-    limit video fps by dropping frames.
-
-``--oautofps``
-    Sets the output format time base to the guessed frame rate of the input
-    video (simulates MEncoder behavior, useful for AVI; may cause frame drops).
-    Note that not all codecs and not all formats support VFR encoding, and some
-    which do have bugs when a target bitrate is specified - use ``--ofps`` or
-    ``--oautofps`` to force CFR encoding in these cases.
-
-``--omaxfps=<float value>``
-    Specifies the minimum distance of adjacent frames (default: 0, which means
-    unset). Content of lower frame rate is not readjusted to this frame rate;
-    content of higher frame rate is decimated to this frame rate.
-
-``--oharddup``
-    If set, the frame rate given by ``--ofps`` is attained not by skipping time
-    codes, but by duplicating frames (constant frame rate mode).
-
-``--oneverdrop``
-    If set, frames are never dropped. Instead, time codes of video are
-    readjusted to always increase. This may cause AV desync, though; to work
-    around this, use a high-fps time base using ``--ofps`` and absolutely
-    avoid ``--oautofps``.
-
 ``--oac=<codec>``
-    Specifies the output audio codec. This can be a comma separated list of
-    possible codecs to try. See ``--oac=help`` for a full list of supported
-    codecs.
+    Specifies the output audio codec. See ``--oac=help`` for a full list of
+    supported codecs.

 ``--oaoffset=<value>``
    Shifts audio data by the given time (in seconds) by adding/removing
-    samples at the start.
+    samples at the start. Deprecated.

 ``--oacopts=<options>``
    Specifies the output audio codec options for libavcodec.
@@ -82,30 +47,22 @@ You can encode files from one format/codec to another using this facility.
    ``--oacopts-add=<options1[,options2,...]>``
        Appends the options given as arguments to the options list.

-    ``--oacopts-pre=<options1[,options2,...]>``
-        Prepends the options given as arguments to the options list.
-
-    ``--oacopts-del=<index1[,index2,...]>``
-        Deletes the options at the given indexes. Index numbers start at 0,
-        negative numbers address the end of the list (-1 is the last).
-
-    ``--oacopts-clr``
+    ``--oacopts=""``
        Completely empties the options list.

 ``--oafirst``
    Force the audio stream to become the first stream in the output.
-    By default, the order is unspecified.
+    By default, the order is unspecified. Deprecated.

 ``--ovc=<codec>``
-    Specifies the output video codec. This can be a comma separated list of
-    possible codecs to try. See ``--ovc=help`` for a full list of supported
-    codecs.
+    Specifies the output video codec. See ``--ovc=help`` for a full list of
+    supported codecs.

 ``--ovoffset=<value>``
    Shifts video data by the given time (in seconds) by shifting the pts
-    values.
+    values. Deprecated.

-``--ovcopts <options>``
+``--ovcopts=<options>``
    Specifies the output video codec options for libavcodec.
    See --ovcopts=help for a full list of supported options.

@@ -123,26 +80,12 @@ You can encode files from one format/codec to another using this facility.
    ``--ovcopts-add=<options1[,options2,...]>``
        Appends the options given as arguments to the options list.

-    ``--ovcopts-pre=<options1[,options2,...]>``
-        Prepends the options given as arguments to the options list.
-
-    ``--ovcopts-del=<index1[,index2,...]>``
-        Deletes the options at the given indexes. Index numbers start at 0,
-        negative numbers address the end of the list (-1 is the last).
-
-    ``--ovcopts-clr``
+    ``--ovcopts=""``
        Completely empties the options list.

 ``--ovfirst``
    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
-    container formats, e.g. AVI). Discontinuities are still fixed.
-    By default, audio pts are set to playback time and video pts are
-    synchronized to match audio pts, as some output formats do not support
-    anything else.
+    By default, the order is unspecified. Deprecated.

 ``--orawts``
    Copies input pts to the output video (not supported by some output
@@ -150,6 +93,27 @@ You can encode files from one format/codec to another using this facility.
    and all pts are passed through as-is. Never seek backwards or use multiple
    input files in this mode!

-``--no-ometadata``
+``--no-ocopy-metadata``
    Turns off copying of metadata from input files to output files when
    encoding (which is enabled by default).
+
+``--oset-metadata=<metadata-tag[,metadata-tag,...]>``
+    Specifies metadata to include in the output file.
+    Supported keys vary between output formats. For example, Matroska (MKV) and
+    FLAC allow almost arbitrary keys, while support in MP4 and MP3 is more
+    limited.
+
+    .. admonition:: Example
+
+        "``--oset-metadata=title="Output title",comment="Another tag"``"
+            adds a title and a comment to the output file.
+
+``--oremove-metadata=<metadata-tag[,metadata-tag,...]>``
+    Specifies metadata to exclude from the output file when copying from the
+    input file.
+
+    .. admonition:: Example
+
+        "``--oremove-metadata=comment,genre``"
+            excludes copying of the the comment and genre tags to the output
+            file.
@@ -350,7 +350,7 @@ List of Input Commands
    Print text to stdout. The string can contain properties (see
    `Property Expansion`_).

-``show-text "<string>" [<duration>|- [<level>]]``
+``show-text "<string>" [<duration>|-1 [<level>]]``
    Show text on the OSD. The string can contain properties, which are expanded
    as described in `Property Expansion`_. This can be used to show playback
    time, filename, and so on.
@@ -499,35 +499,29 @@ Input Commands that are Possibly Subject to Change

        - ``a vf set flip`` turn video upside-down on the ``a`` key
        - ``b vf set ""`` remove all video filters on ``b``
-        - ``c vf toggle lavfi=gradfun`` toggle debanding on ``c``
+        - ``c vf toggle gradfun`` toggle debanding on ``c``

    .. admonition:: Example how to toggle disabled filters at runtime

-        - Add something ``vf-add=@deband:!lavfi=[gradfun]`` to ``mpv.conf``. The
-          ``@deband:`` is the label, and ``deband`` is an arbitrary, user-given
-          name for this filter entry. The ``!`` before the filter name disables
-          the filter by default. Everything after this is the normal filter name
-          and the filter parameters.
+        - Add something like ``vf-add=@deband:!gradfun`` to ``mpv.conf``.
+          The ``@deband:`` is the label, an arbitrary, user-given name for this
+          filter entry. The ``!`` before the filter name disables the filter by
+          default. Everything after this is the normal filter name and possibly
+          filter parameters, like in the normal ``--vf`` syntax.
        - Add ``a vf toggle @deband`` to ``input.conf``. This toggles the
-          "disabled" flag for the filter identified with ``deband``.
+          "disabled" flag for the filter with the label ``deband`` when the
+          ``a`` key is hit.

 ``cycle-values ["!reverse"] <property> "<value1>" "<value2>" ...``
    Cycle through a list of values. Each invocation of the command will set the
-    given property to the next value in the list. The command maintains an
-    internal counter which value to pick next, and which is initially 0. It is
-    reset to 0 once the last value is reached.
-
-    The internal counter is associated using the property name and the value
-    list. If multiple commands (bound to different keys) use the same name
-    and value list, they will share the internal counter.
+    given property to the next value in the list. The command will use the
+    current value of the property/option, and use it to determine the current
+    position in the list of values. Once it has found it, it will set the
+    next value in the list (wrapping around to the first item if needed).

    The special argument ``!reverse`` can be used to cycle the value list in
-    reverse. Compared with a command that just lists the value in reverse, this
-    command will actually share the internal counter with the forward-cycling
-    key binding (as long as the rest of the arguments are the same).
-
-    Note that there is a static limit of (as of this writing) 10 arguments
-    (this limit could be raised on demand).
+    reverse. The only advantage is that you don't need to reverse the value
+    list yourself when adding a second key binding for cycling backwards.

 ``enable-section "<section>" [flags]``
    Enable all key bindings in the named input section.
@@ -729,7 +723,26 @@ Input Commands that are Possibly Subject to Change
    merely sets all option values listed within the profile.

 ``load-script "<path>"``
-    Load a script, similar to the ``--script`` option.
+    Load a script, similar to the ``--script`` option. Whether this waits for
+    the script to finish initialization or not changed multiple times, and the
+    future behavior is left undefined.
+
+``change-list "<option>" "<operation>" "<value>"``
+    This command changes list options as described in `List Options`_. The
+    ``<option>`` parameter is the normal option name, while ``<operation>`` is
+    the suffix or action used on the option.
+
+    Some operations take no value, but the command still requires the value
+    parameter. In these cases, the value must be an empty string.
+
+    .. admonition:: Example
+
+        ``change-list glsl-shaders append file.glsl``
+
+        Add a filename to the ``glsl-shaders`` list. The command line
+        equivalent is ``--glsl-shaders-append=file.glsl`` or alternatively
+        ``--glsl-shader=file.glsl``.
+

 Undocumented commands: ``tv-last-channel`` (TV/DVB only),
 ``ao-reload`` (experimental/internal).
@@ -744,6 +757,49 @@ and obscure way to handle events that require stricter coordination. There are
 no API stability guarantees made. Not following the protocol exactly can make
 the player freeze randomly. Basically, nobody should use this API.

+The C API is described in the header files. The Lua API is described in the
+Lua section.
+
+The following hooks are currently defined:
+
+``on_load``
+    Called when a file is to be opened, before anything is actually done.
+    For example, you could read and write the ``stream-open-filename``
+    property to redirect an URL to something else (consider support for
+    streaming sites which rarely give the user a direct media URL), or
+    you could set per-file options with by setting the property
+    ``file-local-options/<option name>``. The player will wait until all
+    hooks are run.
+
+``on_load_fail``
+    Called after after a file has been opened, but failed to. This can be
+    used to provide a fallback in case native demuxers failed to recognize
+    the file, instead of always running before the native demuxers like
+    ``on_load``. Demux will only be retried if ``stream-open-filename``
+    was changed.
+
+``on_preloaded``
+    Called after a file has been opened, and before tracks are selected and
+    decoders are created. This has some usefulness if an API users wants
+    to select tracks manually, based on the set of available tracks. It's
+    also useful to initialize ``--lavfi-complex`` in a specific way by API,
+    without having to "probe" the available streams at first.
+
+    Note that this does not yet apply default track selection. Which operations
+    exactly can be done and not be done, and what information is available and
+    what is not yet available yet, is all subject to change.
+
+``on_unload``
+    Run before closing a file, and before actually uninitializing
+    everything. It's not possible to resume playback in this state.
+
+Legacy hook API
+~~~~~~~~~~~~~~~
+
+.. warning::
+
+    The legacy API is deprecated and will be removed soon.
+
 There are two special commands involved. Also, the client must listen for
 client messages (``MPV_EVENT_CLIENT_MESSAGE`` in the C API).

@@ -765,7 +821,6 @@ client messages (``MPV_EVENT_CLIENT_MESSAGE`` in the C API).
       used to correctly handle multiple hooks registered by the same client,
       as long as the ``id`` argument is unique in the client)
    3. something undefined, used by the hook mechanism to track hook execution
-       (currently, it's the hook-name, but this might change without warning)

    Upon receiving this message, the client can handle the event. While doing
    this, the player core will still react to requests, but playback will
@@ -779,32 +834,6 @@ client messages (``MPV_EVENT_CLIENT_MESSAGE`` in the C API).
    argument of the client message that starts hook execution for the
    current client.

-The following hooks are currently defined:
-
-``on_load``
-    Called when a file is to be opened, before anything is actually done.
-    For example, you could read and write the ``stream-open-filename``
-    property to redirect an URL to something else (consider support for
-    streaming sites which rarely give the user a direct media URL), or
-    you could set per-file options with by setting the property
-    ``file-local-options/<option name>``. The player will wait until all
-    hooks are run.
-
-``on_preloaded``
-    Called after a file has been opened, and before tracks are selected and
-    decoders are created. This has some usefulness if an API users wants
-    to select tracks manually, based on the set of available tracks. It's
-    also useful to initialize ``--lavfi-complex`` in a specific way by API,
-    without having to "probe" the available streams at first.
-
-    Note that this does not yet apply default track selection. Which operations
-    exactly can be done and not be done, and what information is available and
-    what is not yet available yet, is all subject to change.
-
-``on_unload``
-    Run before closing a file, and before actually uninitializing
-    everything. It's not possible to resume playback in this state.
-
 Input Command Prefixes
 ----------------------

@@ -1435,7 +1464,7 @@ Property list
 ``hwdec-interop``
    This returns the currently loaded hardware decoding/output interop driver.
    This is known only once the VO has opened (and possibly later). With some
-    VOs (like ``opengl``), this might be never known in advance, but only when
+    VOs (like ``gpu``), this might be never known in advance, but only when
    the decoder attempted to create the hw decoder successfully. (Using
    ``--gpu-hwdec-interop`` can load it eagerly.) If there are multiple
    drivers loaded, they will be separated by ``,``.
@@ -1565,12 +1594,12 @@ Property list
    redrawing and frame display being somewhat disconnected, and you might
    have to pause and force a redraw.

-    Sub-properties:
+    Sub-properties::

-    ``video-frame-info/picture-type``
-    ``video-frame-info/interlaced``
-    ``video-frame-info/tff``
-    ``video-frame-info/repeat``
+        video-frame-info/picture-type
+        video-frame-info/interlaced
+        video-frame-info/tff
+        video-frame-info/repeat

 ``container-fps``
    Container FPS. This can easily contain bogus values. For videos that use
@@ -2079,9 +2108,6 @@ Property list
    List of decoders supported. This lists decoders which can be passed to
    ``--vd`` and ``--ad``.

-    ``family``
-        Decoder driver. Usually ``lavc`` for libavcodec.
-
    ``codec``
        Canonical codec name, which identifies the format the decoder can
        handle.
@@ -2102,7 +2128,6 @@ Property list

        MPV_FORMAT_NODE_ARRAY
            MPV_FORMAT_NODE_MAP (for each decoder entry)
-                "family"        MPV_FORMAT_STRING
                "codec"         MPV_FORMAT_STRING
                "driver"        MPV_FORMAT_STRING
                "description"   MPV_FORMAT_STRING
@@ -2112,6 +2137,10 @@ Property list
    The encoder names (``driver`` entries) can be passed to ``--ovc`` and
    ``--oac`` (without the ``lavc:`` prefix required by ``--vd`` and ``--ad``).

+``demuxer-lavf-list``
+    List of available libavformat demuxers' names. This can be used to check
+    for support for a specific format or use with ``--demuxer-lavf-format``.
+
 ``mpv-version``
    Return the mpv version/copyright string. Depending on how the binary was
    built, it might contain either a release version, or just a git hash.
@@ -180,14 +180,7 @@ extra commands can also be used as part of the protocol:
        { "error": "success" }

 ``set_property_string``
-    Like ``set_property``, but the argument value must be passed as string.
-
-    Example:
-
-    ::
-
-        { "command": ["set_property_string", "pause", "yes"] }
-        { "error": "success" }
+    Alias for ``set_property``. Both commands accept native values and strings.

 ``observe_property``
    Watch a property for changes. If the given property is changed, then an
@@ -27,16 +27,16 @@ otherwise, the documented Lua options, script directories, loading, etc apply to
 JavaScript files too.

 Script initialization and lifecycle is the same as with Lua, and most of the Lua
-functions at the modules ``mp``, ``mp.utils`` and ``mp.msg`` are available to
-JavaScript with identical APIs - including running commands, getting/setting
-properties, registering events/key-bindings/property-changes/hooks, etc.
+functions at the modules ``mp``, ``mp.utils``, ``mp.msg`` and ``mp.options`` are
+available to JavaScript with identical APIs - including running commands,
+getting/setting properties, registering events/key-bindings/hooks, etc.

 Differences from Lua
 --------------------

-No need to load modules. ``mp``, ``mp.utils`` and ``mp.msg`` are preloaded, and
-you can use e.g. ``var cwd = mp.utils.getcwd();`` without prior setup.
-``mp.options`` is currently not implemented, but ``mp.get_opt(...)`` is.
+No need to load modules. ``mp``, ``mp.utils``,  ``mp.msg`` and ``mp.options``
+are preloaded, and you can use e.g. ``var cwd = mp.utils.getcwd();`` without
+prior setup.

 Errors are slightly different. Where the Lua APIs return ``nil`` for error,
 the JavaScript ones return ``undefined``. Where Lua returns ``something, error``
@@ -67,10 +67,6 @@ Unsupported Lua APIs and their JS alternatives

 ``mp.add_periodic_timer(seconds, fn)``  JS: ``id = setInterval(fn, ms)``

-``mp.register_idle(fn)``  JS: ``id = setTimeout(fn)``
-
-``mp.unregister_idle(fn)``  JS:  ``clearTimeout(id)``
-
 ``utils.parse_json(str [, trail])``  JS: ``JSON.parse(str)``

 ``utils.format_json(v)``  JS: ``JSON.stringify(v)``
@@ -87,8 +83,6 @@ Unsupported Lua APIs and their JS alternatives

 ``mp.dispatch_events([allow_wait])`` see event loop below.

-``mp.options`` module is not implemented currently for JS.
-
 Scripting APIs - identical to Lua
 ---------------------------------

@@ -144,6 +138,10 @@ Otherwise, where the Lua APIs return ``nil`` on error, JS returns ``undefined``.

 ``mp.get_wakeup_pipe()``

+``mp.register_idle(fn)``
+
+``mp.unregister_idle(fn)``
+
 ``mp.enable_messages(level)``

 ``mp.register_script_message(name, fn)``
@@ -180,8 +178,12 @@ Otherwise, where the Lua APIs return ``nil`` on error, JS returns ``undefined``.

 ``mp.utils.subprocess_detached(t)``

+``mp.utils.getpid()`` (LE)
+
 ``mp.add_hook(type, priority, fn)``

+``mp.options.read_options(obj [, identifier])`` (types: string/boolean/number)
+
 Additional utilities
 --------------------

@@ -265,7 +267,7 @@ event loop iteration or at a later event loop iteration. This is true also for
 intervals - which also never call back twice at the same event loop iteration.

 Additionally, timers are processed after the event queue is empty, so it's valid
-to use ``setTimeout(fn)`` instead of Lua's ``mp.register_idle(fn)``.
+to use ``setTimeout(fn)`` as a one-time idle observer.

 CommonJS modules and ``require(id)``
 ------------------------------------
@@ -318,6 +320,10 @@ also print every event which mpv sends to your script:
                wait = 0;
            } else {
                wait = mp.process_timers() / 1000;
+                if (wait != 0) {
+                    mp.notify_idle_observers();
+                    wait = mp.peek_timers_wait() / 1000;
+                }
            }
        } while (mp.keep_running);
    }
@@ -337,5 +343,12 @@ if there are such (event handlers, property observers, script messages, etc).
 and returns the duration in ms till the next due timer (possibly 0), or -1 if
 there are no pending timers. Must not be called recursively.

+``mp.notify_idle_observers()`` calls back the idle observers, which we do when
+we're about to sleep (wait != 0), but the observers may add timers or take
+non-negligible duration to complete, so we re-calculate ``wait`` afterwards.
+
+``mp.peek_timers_wait()`` returns the same values as ``mp.process_timers()``
+but without doing anything. Invalid result if called from a timer callback.
+
 Note: ``exit()`` is also registered for the ``shutdown`` event, and its
 implementation is a simple ``mp.keep_running = false``.
@@ -63,7 +63,10 @@ the player will more or less hang until the script returns from the main chunk
 (and ``mp_event_loop`` is called), or the script calls ``mp_event_loop`` or
 ``mp.dispatch_events`` directly. This is done to make it possible for a script
 to fully setup event handlers etc. before playback actually starts. In older
-mpv versions, this happened asynchronously.
+mpv versions, this happened asynchronously. With mpv 0.29.0, this changes
+slightly, and it merely waits for scripts to be loaded in this manner before
+starting playback as part of the player initialization phase. Scripts run though
+initialization in parallel. This might change again.

 mp functions
 ------------
@@ -427,7 +430,7 @@ are useful only in special situations.

 ``mp.get_wakeup_pipe()``
    Calls ``mpv_get_wakeup_pipe()`` and returns the read end of the wakeup
-    pipe. (See ``client.h`` for details.)
+    pipe. This is deprecated, but still works. (See ``client.h`` for details.)

 ``mp.get_next_timeout()``
    Return the relative time in seconds when the next timer (``mp.add_timeout``
@@ -531,7 +534,7 @@ Example implementation::
    print(options.optionA)


-The config file will be stored in ``lua-settings/identifier.conf`` in mpv's user
+The config file will be stored in ``script-opts/identifier.conf`` in mpv's user
 folder. Comment lines can be started with # and stray spaces are not removed.
 Boolean values will be represented with yes/no.

@@ -682,6 +685,10 @@ strictly part of the guaranteed API.

    The function returns ``nil``.

+``utils.getpid()``
+    Returns the process ID of the running mpv process. This can be used to identify
+    the calling mpv when launching (detached) subprocesses.
+
 ``utils.parse_json(str [, trail])``
    Parses the given string argument as JSON, and returns it as a Lua table. On
    error, returns ``nil, error``. (Currently, ``error`` is just a string
@@ -58,6 +58,10 @@ Ctrl+LEFT and Ctrl+RIGHT
    Seek to the previous/next subtitle. Subject to some restrictions and
    might not always work; see ``sub-seek`` command.

+Ctrl+Shift+Left and Ctrl+Shift+Right
+    Adjust subtitle delay so that the next or previous subtitle is displayed
+    now. This is especially useful to sync subtitles to audio.
+
 [ and ]
    Decrease/increase current playback speed by 10%.

@@ -67,6 +71,17 @@ Ctrl+LEFT and Ctrl+RIGHT
 BACKSPACE
    Reset playback speed to normal.

+Shift+BACKSPACE
+    Undo the last seek. This works only if the playlist entry was not changed.
+    Hitting it a second time will go back to the original position.
+    See ``revert-seek`` command for details.
+
+Shift+Ctrl+BACKSPACE
+    Mark the current position. This will then be used by ``Shift+BACKSPACE``
+    as revert position (once you seek back, the marker will be reset). You can
+    use this to seek around in the file and then return to the exact position
+    where you left off.
+
 < and >
    Go backward/forward in the playlist.

@@ -115,8 +130,9 @@ ESC
 T
    Toggle stay-on-top (see also ``--ontop``).

-w and e
-    Decrease/increase pan-and-scan range.
+w and W
+    Decrease/increase pan-and-scan range. The ``e`` key does the same as
+    ``W`` currently, but use is discouraged.

 o (also P)
    Show progression bar, elapsed time and total duration on the OSD.
@@ -130,8 +146,9 @@ v
 j and J
    Cycle through the available subtitles.

-x and z
-    Adjust subtitle delay by +/- 0.1 seconds.
+z and Z
+    Adjust subtitle delay by +/- 0.1 seconds. The ``x`` key does the same as
+    ``Z`` currently, but use is discouraged.

 l
    Set/clear A-B loop points. See ``ab-loop`` command for details.
@@ -151,8 +168,9 @@ V
    Toggle subtitle VSFilter aspect compatibility mode. See
    ``--sub-ass-vsfilter-aspect-compat`` for more info.

-r and t
-    Move subtitles up/down.
+r and R
+    Move subtitles up/down. The ``t`` key does the same as ``R`` currently, but
+    use is discouraged.

 s
    Take a screenshot.
@@ -180,8 +198,28 @@ d
 A
    Cycle aspect ratio override.

+Ctrl h
+    Toggle hardware video decoding on/off.
+
+Alt+LEFT, Alt+RIGHT, Alt+UP, Alt+DOWN
+    Move the video rectangle (panning).
+
+Alt + and Alt -
+    Combining ``Alt`` with the ``+`` or ``-`` keys changes video zoom.
+
+Alt+BACKSPACE
+    Reset the pan/zoom settings.
+
+F8
+    Show the playlist and the current position in it (useful only if a UI window
+    is used, broken on the terminal).
+
+F9
+    Show the list of audio and subtitle streams (useful only if a UI window  is
+    used, broken on the terminal).
+
 (The following keys are valid only when using a video output that supports the
-corresponding adjustment, or the software equalizer (``--vf=eq``).)
+corresponding adjustment.)

 1 and 2
    Adjust contrast.
@@ -435,6 +473,7 @@ Suffix        Meaning
 -del          Delete an existing item by integer index
 -pre          Prepend 1 or more items
 -set          Set a list of items
+-toggle       Append an item, or remove if if it already exists
 ============= ===============================================

 Although some operations allow specifying multiple ``,``-separated items, using
@@ -446,6 +485,11 @@ Some options (like ``--sub-file``, ``--audio-file``, ``--glsl-shader``) are
 aliases for the proper option with ``-append`` action. For example,
 ``--sub-file`` is an alias for ``--sub-files-append``.

+Some options only support a subset of the above.
+
+Options of this type can be changed at runtime using the ``change-list``
+command, which takes the suffix as separate operation parameter.
+
 Playing DVDs
 ------------

@@ -567,6 +611,11 @@ 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.

+You can list profiles with ``--profile=help``, and show the contents of a
+profile with ``--show-profile=<name>`` (replace ``<name>`` with the profile
+name). You can apply profiles on start with the ``--profile=<name>`` option,
+or at runtime with the ``apply-profile <name>`` command.
+
 .. admonition:: Example mpv config file with profiles

    ::
@@ -689,6 +738,45 @@ listed.
  properties.)


+LOW LATENCY PLAYBACK
+====================
+
+mpv is optimized for normal video playback, meaning it actually tries to buffer
+as much data as it seems to make sense. This will increase latency. Reducing
+latency is possible only by specifically disabling features which increase
+latency.
+
+The builtin ``low-latency`` profile tries to apply some of the options which can
+reduce latency. You can use  ``--profile=low-latency`` to apply all of them. You
+can list the contents with ``--show-profile=low-latency`` (some of the options
+are quite obscure, and may change every mpv release).
+
+Be aware that some of the options can reduce playback quality.
+
+Most latency is actually caused by inconvenient timing behavior. You can disable
+this with ``--untimed``, but it will likely break, unless the stream has no
+audio, and the input feeds data to the player at a constant rate.
+
+Another common problem is with MJPEG streams. These do not signal the correct
+framerate. Using ``--untimed`` or ``--no-correct-pts --fps=60`` might help.
+
+For livestreams, data can build up due to pausing the stream, due to slightly
+lower playback rate, or "buffering" pauses. If the demuxer cache is enabled,
+these can be skipped manually. The experimental ``drop-buffers`` command can
+be used to discard any buffered data, though it's very disruptive.
+
+In some cases, manually tuning TCP buffer sizes and such can help to reduce
+latency.
+
+Additional options that can be tried:
+
+- ``--opengl-glfinish=yes``, can reduce buffering in the graphics driver
+- ``--opengl-swapinterval=0``, same
+- ``--vo=xv``, same
+- without audio ``--framedrop=no --speed=1.01`` may help for live sources
+  (results can be mixed)
+
+
 PROTOCOLS
 =========

@@ -726,14 +814,13 @@ PROTOCOLS

 ``bd://[title][/device]`` ``--bluray-device=PATH``

-    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.
+    Play a Blu-ray disc. Since libbluray 1.0.1, you can read from ISO files
+    by passing them to ``--bluray-device``.

    ``title`` can be: ``longest`` or ``first`` (selects the default
    playlist); ``mpls/<number>`` (selects <number>.mpls playlist);
-    ``<number>`` (select playlist with the same index). You can list
-    the available playlists with ``--msg-level=bd=v``.
+    ``<number>`` (select playlist with the same index). mpv will list
+    the available playlists on loading.

    ``bluray://`` is an alias.

@@ -748,7 +835,7 @@ PROTOCOLS
 ``dvdread://...:``

    Play a DVD using the old libdvdread code. This is what MPlayer and
-    older mpv versions use for ``dvd://``. Use is discouraged. It's
+    older mpv versions used for ``dvd://``. Use is discouraged. It's
    provided only for compatibility and for transition, and to work
    around outstanding dvdnav bugs (see "DVD library choices" above).

@@ -795,10 +882,25 @@ PROTOCOLS
    ``PATH`` itself should start with a third ``/`` to make the path an
    absolute path.

+``appending://PATH``
+
+    Play a local file, but assume it's being appended to. This is useful for
+    example for files that are currently being downloaded to disk. This will
+    block playback, and stop playback only if no new data was appended after
+    a timeout of about 2 seconds.
+
+    Using this is still a bit of a bad idea, because there is no way to detect
+    if a file is actually being appended, or if it's still written. If you're
+    trying to play the  output of some program, consider using a pipe
+    (``something | mpv -``). If it really has to be a file on disk, use tail to
+    make it wait forever, e.g. ``tail -f -c +0 file.mkv | mpv -``.
+
 ``fd://123``

    Read data from the given file descriptor (for example 123). This is similar
    to piping data to stdin via ``-``, but can use an arbitrary file descriptor.
+    mpv may modify some file descriptor properties when the stream layer "opens"
+    it.

 ``fdclose://123``

@@ -1072,7 +1174,7 @@ For Windows-specifics, see `FILES ON WINDOWS`_ section.
    ``--write-filename-in-watch-later-config`` option, and the player will
    add the media filename to the contents of the resume config file.

-``~/.config/mpv/lua-settings/osc.conf``
+``~/.config/mpv/script-opts/osc.conf``
    This is loaded by the OSC script. See the `ON SCREEN CONTROLLER`_ docs
    for details.

@@ -64,17 +64,6 @@ Track Selection
    streamed with youtube-dl, because it saves bandwidth. This is done by
    setting the ytdl_format to "bestaudio/best" in the ytdl_hook.lua script.

-``--ff-aid=<ID|auto|no>``, ``--ff-sid=<ID|auto|no>``, ``--ff-vid=<ID|auto|no>``
-    Select audio/subtitle/video streams by the FFmpeg stream index. The FFmpeg
-    stream index is relatively arbitrary, but useful when interacting with
-    other software using FFmpeg (consider ``ffprobe``).
-
-    Note that with external tracks (added with ``--sub-files`` and similar
-    options), there will be streams with duplicate IDs. In this case, the
-    first stream in order is selected.
-
-    Deprecated.
-
 ``--edition=<ID|auto>``
    (Matroska files only)
    Specify the edition (set of chapters) to use, where 0 is the first. If set
@@ -129,8 +118,8 @@ Playback Control
        ``--start='#2' --end='#4'``
            Plays chapters 2 and 3, and exits.

-``--end=<time>``
-    Stop at given absolute time. Use ``--length`` if the time should be relative
+``--end=<relative time>``
+    Stop at given time. Use ``--length`` if the time should be relative
    to ``--start``. See ``--start`` for valid option values and examples.

 ``--length=<relative time>``
@@ -547,12 +536,16 @@ Program Behavior
    Enable the youtube-dl hook-script. It will look at the input URL, and will
    play the video located on the website. This works with many streaming sites,
    not just the one that the script is named after. This requires a recent
-    version of youtube-dl to be installed on the system. (Enabled by default,
-    except when the client API / libmpv is used.)
+    version of youtube-dl to be installed on the system. (Enabled by default.)

    If the script can't do anything with an URL, it will do nothing.

-    The `exclude` script option accepts a ``|``-separated list of URL patterns
+    The ``try_ytdl_first`` script option accepts a boolean 'yes' or 'no', and if
+    'yes' will try parsing the URL with youtube-dl first, instead of the default
+    where it's only after mpv failed to open it. This mostly depends on whether
+    most of your URLs need youtube-dl parsing.
+
+    The ``exclude`` script option accepts a ``|``-separated list of URL patterns
    which mpv should not use with youtube-dl. The patterns are matched after
    the ``http(s)://`` part of the URL.

@@ -570,6 +563,9 @@ Program Behavior

    See more lua patterns here: https://www.lua.org/manual/5.1/manual.html#5.4.1

+    The ``use_manifests`` script option makes mpv use the master manifest URL for
+    formats like HLS and DASH, if available, allowing for video/audio selection
+    in runtime. It's disabled ("no") by default for performance reasons.

 ``--ytdl-format=<best|worst|mp4|webm|...>``
    Video format/quality that is directly passed to youtube-dl. The possible
@@ -586,10 +582,18 @@ Program Behavior
    There is no sanity checking so it's possible to break things (i.e.
    passing invalid parameters to youtube-dl).

+    A proxy URL can be passed for youtube-dl to use it in parsing the website.
+    This is useful for geo-restricted URLs. After youtube-dl parsing, some
+    URLs also require a proxy for playback, so this can pass that proxy
+    information to mpv. Take note that SOCKS proxies aren't supported and
+    https URLs also bypass the proxy. This is a limitation in FFmpeg.
+
    .. admonition:: Example

        - ``--ytdl-raw-options=username=user,password=pass``
        - ``--ytdl-raw-options=force-ipv6=``
+        - ``--ytdl-raw-options=proxy=[http://127.0.0.1:3128]``
+        - ``--ytdl-raw-options-append=proxy=http://127.0.0.1:3128``

 ``--load-stats-overlay=<yes|no>``
    Enable the builtin script that shows useful playback information on a key
@@ -663,6 +667,26 @@ Video
        ``--vo=vdpau`` has its own code for the ``vo`` framedrop mode. Slight
        differences to other VOs are possible.

+``--video-latency-hacks=<yes|no>``
+    Enable some things which tend to reduce video latency by 1 or 2 frames
+    (default: no). Note that this option might be removed without notice once
+    the player's timing code does not inherently need to do these things
+    anymore.
+
+    This does:
+
+    - Use the demuxer reported FPS for frame dropping. This avoids that the
+      player needs to decode 1 frame in advance, lowering total latency in
+      effect. This also means that if the demuxer reported FPS is wrong, or
+      the video filter chain changes FPS (e.g. deinterlacing), then it could
+      drop too many or not enough frames.
+    - Disable waiting for the first video frame. Normally the player waits for
+      the first video frame to be fully rendered before starting playback
+      properly. Some VOs will lazily initialize stuff when rendering the first
+      frame, so if this is not done, there is some likeliness that the VO has
+      to drop some frames if rendering the first frame takes longer than needed.
+
+
 ``--display-fps=<fps>``
    Set the display FPS used with the ``--video-sync=display-*`` modes. By
    default, a detected value is used. Keep in mind that setting an incorrect
@@ -691,8 +715,9 @@ Video
    :videotoolbox: requires ``--vo=gpu`` (OS X 10.8 and up),
                   or ``--vo=opengl-cb`` (iOS 9.0 and up)
    :videotoolbox-copy: copies video back into system RAM (OS X 10.8 or iOS 9.0 and up)
-    :dxva2:     requires ``--vo=gpu`` with ``--gpu-context=angle`` or
-                ``--gpu-context=dxinterop`` (Windows only)
+    :dxva2:     requires ``--vo=gpu`` with ``--gpu-context=d3d11``,
+                ``--gpu-context=angle`` or ``--gpu-context=dxinterop``
+                (Windows only)
    :dxva2-copy: copies video back to system RAM (Windows only)
    :d3d11va:   requires ``--vo=gpu`` with ``--gpu-context=d3d11`` or
                ``--gpu-context=angle`` (Windows 8+ only)
@@ -862,18 +887,19 @@ Video
    older hardware. d3d11va can always use ``yuv420p``, which uses an opaque
    format, with likely no advantages.

-``--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. ``yuv420p`` also
-    works.
-    Since mpv 0.25.0, ``no`` is an accepted value, which lets the decoder pick
-    the format on newer FFmpeg versions (will use ``nv12`` on older versions).
+``--cuda-decode-device=<auto|0..>``
+    Choose the GPU device used for decoding when using the ``cuda`` hwdec.

-    Deprecated. Use ``--hwdec-image-format`` if you really need this. If both
-    are specified, ``--hwdec-image-format`` wins.
+    By default, the device that is being used to provide OpenGL output will
+    also be used for decoding (and in the vast majority of cases, only one
+    GPU will be present).
+
+    Note that when using the ``cuda-copy`` hwdec, a different option must be
+    passed: ``--vd-lavc-o=gpu=<0..>``.
+
+``--vaapi-device=<device file>``
+    Choose the DRM device for ``vaapi-copy``. This should be the path to a
+    DRM device file. (Default: ``/dev/dri/renderD128``)

 ``--panscan=<0.0-1.0>``
    Enables pan-and-scan functionality (cropping the sides of e.g. a 16:9
@@ -993,8 +1019,8 @@ Video
    determined using a fixed framerate value (either using the ``--fps``
    option, or using file information). Sometimes, files with very broken
    timestamps can be played somewhat well in this mode. Note that video
-    filters, subtitle rendering and audio synchronization can be completely
-    broken in this mode.
+    filters, subtitle rendering, seeking (including hr-seeks and backstepping),
+    and audio synchronization can be completely broken in this mode.

 ``--fps=<float>``
    Override video framerate. Useful if the original value is wrong or missing.
@@ -1019,6 +1045,8 @@ Video
    ``--deinterlace=auto`` was removed, which used to mean that the default
    interlacing option of possibly inserted video filters was used.)

+    Note that this will make video look worse if it's not actually interlaced.
+
 ``--frames=<number>``
    Play/convert only first ``<number>`` video frames, then quit.

@@ -1081,22 +1109,19 @@ Video
    N frames fail to decode in a row. 1 is equivalent to ``yes``.

 ``--vd-lavc-dr=<yes|no>``
-    Enable direct rendering (default: no). If this is set to ``yes``, the
+    Enable direct rendering (default: yes). If this is set to ``yes``, the
    video will be decoded directly to GPU video memory (or staging buffers).
    This can speed up video upload, and may help with large resolutions or
    slow hardware. This works only with the following VOs:

-        - ``gpu``: requires at least OpenGL 4.4.
+        - ``gpu``: requires at least OpenGL 4.4 or Vulkan.

-    (In particular, this can't be made work with ``opengl-cb``.)
+    (In particular, this can't be made work with ``opengl-cb``, but the libmpv
+    render API has optional support.)

    Using video filters of any kind that write to the image data (or output
    newly allocated frames) will silently disable the DR code path.

-    There are some corner cases that will result in undefined behavior (crashes
-    and other strange behavior) if this option is enabled. These are pending
-    towards being fixed properly at a later point.
-
 ``--vd-lavc-bitexact``
    Only use bit-exact algorithms in all decoding steps (for codec testing).

@@ -1159,6 +1184,18 @@ Video
    on the machine and use that, up to the maximum of 16. You can set more than
    16 threads manually.

+``--vd-lavc-assume-old-x264=<yes|no>``
+    Assume the video was encoded by an old, buggy x264 version (default: no).
+    Normally, this is autodetected by libavcodec. But if the bitstream contains
+    no x264 version info (or it was somehow skipped), and the stream was in fact
+    encoded by an old x264 version (build 150 or earlier), and if the stream
+    uses ``4:4:4`` chroma, then libavcodec will by default show corrupted video.
+    This option sets the libavcodec ``x264_build`` option to ``150``, which
+    means that if the stream contains no version info, or was not encoded by
+    x264 at all, it assumes it was encoded by the old version. Enabling this
+    option is pretty safe if you want your broken files to work, but in theory
+    this can break on streams not encoded by x264, or if a stream encoded by a
+    newer x264 version contains no version info.


 Audio
@@ -1183,7 +1220,9 @@ Audio
    While the option normally takes one of the strings as indicated by the
    methods above, you can also force the device for most AOs by building it
    manually. For example ``name/foobar`` forces the AO ``name`` to use the
-    device ``foobar``.
+    device ``foobar``. However, the ``--ao`` option will strictly force a
+    specific AO. To avoid confusion, don't use ``--ao`` and ``--audio-device``
+    together.

    .. admonition:: Example for ALSA

@@ -1424,16 +1463,6 @@ Audio
        want. For example, most A/V receivers connected via HDMI and that can
        do 7.1 would  be served by: ``--audio-channels=7.1,5.1,stereo``

-``--audio-normalize-downmix=<yes|no>``
-    Enable/disable normalization if surround audio is downmixed to stereo
-    (default: no). If this is disabled, downmix can cause clipping. If it's
-    enabled, the output might be too silent. It depends on the source audio.
-
-    Technically, this changes the ``normalize`` suboption of the
-    ``lavrresample`` audio filter, which performs the downmixing.
-
-    If downmix happens outside of mpv for some reason, this has no effect.
-
 ``--audio-display=<no|attachment>``
    Setting this option to ``attachment`` (default) will display image
    attachments (e.g. album cover art) when playing audio files. It will
@@ -1484,9 +1513,10 @@ Audio
            changes, the audio device is closed and reopened. This means that
            you will normally get gapless audio with files that were encoded
            using the same settings, but might not be gapless in other cases.
-            (Unlike with ``yes``, you don't have to worry about corner cases
-            like the first file setting a very low quality output format, and
-            ruining the playback of higher quality files that follow.)
+            The exact conditions under which the audio device is kept open is
+            an implementation detail, and can change from version to version.
+            Currently, the device is kept even if the sample format changes,
+            but the sample formats are convertible.

    .. note::

@@ -2013,7 +2043,8 @@ Subtitles
    Default: 55.

 ``--sub-back-color=<color>``
-    See ``--sub-color``. Color used for sub text background.
+    See ``--sub-color``. Color used for sub text background. You can use
+    ``--sub-shadow-offset`` to change its size relative to the text.

 ``--sub-blur=<0..20.0>``
    Gaussian blur factor. 0 means no blur applied (default).
@@ -2193,7 +2224,7 @@ Window

    .. admonition:: Note (X11)

-        This option does works properly only with window managers which
+        This option works properly only with window managers which
        understand the EWMH ``_NET_WM_FULLSCREEN_MONITORS`` hint.

    .. admonition:: Note (OS X)
@@ -2514,8 +2545,7 @@ 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 (known to happen with GNOME). You might be
-    able to work around this using ``--heartbeat-cmd`` instead.
+    implemented, but does not work (especially with Linux "desktops").

 ``--wid=<ID>``
    This tells mpv to attach to an existing window. If a VO is selected that
@@ -2696,7 +2726,7 @@ Demuxer
 ``--demuxer-lavf-analyzeduration=<value>``
    Maximum length in seconds to analyze the stream properties.

-``--demuxer-lavf-probe-info=<yes|no|auto>``
+``--demuxer-lavf-probe-info=<yes|no|auto|nostreams>``
    Whether to probe stream information (default: auto). Technically, this
    controls whether libavformat's ``avformat_find_stream_info()`` function
    is called. Usually it's safer to call it, but it can also make startup
@@ -2705,6 +2735,10 @@ Demuxer
    The ``auto`` choice (the default) tries to skip this for a few know-safe
    whitelisted formats, while calling it for everything else.

+    The ``nostreams`` choice only calls it if and only if the file seems to
+    contain no streams after opening (helpful in cases when calling the function
+    is needed to detect streams at all, such as with FLV files).
+
 ``--demuxer-lavf-probescore=<1-100>``
    Minimum required libavformat probe score. Lower values will require
    less data to be loaded (makes streams start faster), but makes file
@@ -2728,12 +2762,6 @@ Demuxer
    imperfect behavior from libavformat demuxers. Passing ``no`` disables
    these. For debugging and testing only.

-``--demuxer-lavf-genpts-mode=<no|lavf>``
-    Mode for deriving missing packet PTS values from packet DTS. ``lavf``
-    enables libavformat's ``genpts`` option. ``no`` disables it. This used
-    to be enabled by default, but then it was deemed as not needed anymore.
-    Enabling this might help with timestamp problems, or make them worse.
-
 ``--demuxer-lavf-o=<key>=<value>[,<key>=<value>[,...]]``
    Pass AVOptions to libavformat demuxer.

@@ -2860,7 +2888,7 @@ Demuxer
 ``--demuxer-rawvideo-size=<value>``
    Frame size in bytes when using ``--demuxer=rawvideo``.

-``--demuxer-max-bytes=<bytes>``
+``--demuxer-max-bytes=<bytesize>``
    This controls how much the demuxer is allowed to buffer ahead. The demuxer
    will normally try to read ahead as much as necessary, or as much is
    requested with ``--demuxer-readahead-secs``. The option can be used to
@@ -2872,9 +2900,10 @@ Demuxer
    Set these limits higher if you get a packet queue overflow warning, and
    you think normal playback would be possible with a larger packet queue.

-    See ``--list-options`` for defaults and value range.
+    See ``--list-options`` for defaults and value range. ``<bytesize>`` options
+    accept suffixes such as ``KiB`` and ``MiB``.

-``--demuxer-max-back-bytes=<value>``
+``--demuxer-max-back-bytes=<bytesize>``
    This controls how much past data the demuxer is allowed to preserve. This
    is useful only if the ``--demuxer-seekable-cache`` option is enabled.
    Unlike the forward cache, there is no control how many seconds are actually
@@ -2900,8 +2929,7 @@ Demuxer

    Keep in mind that some events can flush the cache or force a low level
    seek anyway, such as switching tracks, or attempting to seek before the
-    start or after the end of the file. This option is experimental - thus
-    disabled, and bugs are to be expected.
+    start or after the end of the file.

    The special value ``auto`` means ``yes`` in the same situation as
    ``--cache-secs`` is used (i.e. when the stream appears to be a network
@@ -2909,9 +2937,12 @@ Demuxer

 ``--demuxer-thread=<yes|no>``
    Run the demuxer in a separate thread, and let it prefetch a certain amount
-    of packets (default: yes). Having this enabled may lead to smoother
-    playback, but on the other hand can add delays to seeking or track
-    switching.
+    of packets (default: yes). Having this enabled leads to smoother playback,
+    enables features like prefetching, and prevents that stuck network freezes
+    the player. On the other hand, it can add overhead, or the background
+    prefetching can hog CPU resources.
+
+    Disabling this option is not recommended. Use it for debugging only.

 ``--demuxer-readahead-secs=<seconds>``
    If ``--demuxer-thread`` is enabled, this controls how much the demuxer
@@ -3084,13 +3115,18 @@ OSD
    Whether to load the on-screen-controller (default: yes).

 ``--no-osd-bar``, ``--osd-bar``
-    Disable display of the OSD bar. This will make some things (like seeking)
-    use OSD text messages instead of the bar.
+    Disable display of the OSD bar.

    You can configure this on a per-command basis in input.conf using ``osd-``
-    prefixes, see ``Input command prefixes``. If you want to disable the OSD
+    prefixes, see ``Input Command Prefixes``. If you want to disable the OSD
    completely, use ``--osd-level=0``.

+``--osd-on-seek=<no,bar,msg,msg-bar>``
+    Set what is displayed on the OSD during seeks. The default is ``bar``.
+
+    You can configure this on a per-command basis in input.conf using ``osd-``
+    prefixes, see ``Input Command Prefixes``.
+
 ``--osd-duration=<time>``
    Set the duration of the OSD messages in ms (default: 1000).

@@ -3109,30 +3145,31 @@ OSD

 ``--osd-msg1=<string>``
    Show this string as message on OSD with OSD level 1 (visible by default).
-    The message will be visible by default, and as long no other message
+    The message will be visible by default, and as long as no other message
    covers it, and the OSD level isn't changed (see ``--osd-level``).
    Expands properties; see `Property Expansion`_.

 ``--osd-msg2=<string>``
-    Similar as ``--osd-msg1``, but for OSD level 2. If this is an empty string
+    Similar to ``--osd-msg1``, but for OSD level 2. If this is an empty string
    (default), then the playback time is shown.

 ``--osd-msg3=<string>``
-    Similar as ``--osd-msg1``, but for OSD level 3. If this is an empty string
+    Similar to ``--osd-msg1``, but for OSD level 3. If this is an empty string
    (default), then the playback time, duration, and some more information is
    shown.

-    This is also used for the ``show-progress`` command (by default mapped to
-    ``P``), or in some non-default cases when seeking.
+    This is used for the ``show-progress`` command (by default mapped to ``P``),
+    and when seeking if enabled with ``--osd-on-seek`` or by ``osd-`` prefixes
+    in input.conf (see ``Input Command Prefixes``).

    ``--osd-status-msg`` is a legacy equivalent (but with a minor difference).

 ``--osd-status-msg=<string>``
    Show a custom string during playback instead of the standard status text.
    This overrides the status text used for ``--osd-level=3``, when using the
-    ``show-progress`` command (by default mapped to ``P``), or in some
-    non-default cases when seeking. Expands properties. See
-    `Property Expansion`_.
+    ``show-progress`` command (by default mapped to ``P``), and when seeking if
+    enabled with ``--osd-on-seek`` or ``osd-`` prefixes in input.conf (see
+    ``Input Command Prefixes``). Expands properties. See `Property Expansion`_.

    This option has been replaced with ``--osd-msg3``. The only difference is
    that this option implicitly includes ``${osd-sym-cc}``. This option is
@@ -3286,7 +3323,8 @@ Screenshot
 ``--screenshot-high-bit-depth=<yes|no>``
    If possible, write screenshots with a bit depth similar to the source
    video (default: yes). This is interesting in particular for PNG, as this
-    sometimes triggers writing 16 bit PNGs with huge file sizes.
+    sometimes triggers writing 16 bit PNGs with huge file sizes. This will also
+    include an unused alpha channel in the resulting files if 16 bit is used.

 ``--screenshot-template=<template>``
    Specify the filename template used to save screenshots. The template
@@ -3438,6 +3476,53 @@ Software Scaler
 ``--sws-cvs=<v>``
    Software scaler chroma vertical shifting. See ``--sws-scaler``.

+Audio Resampler
+---------------
+
+This controls the default options of any resampling done by mpv (but not within
+libavfilter, within the system audio API resampler, or any other places).
+
+It also sets the defaults for the ``lavrresample`` audio filter.
+
+``--audio-resample-filter-size=<length>``
+    Length of the filter with respect to the lower sampling rate. (default:
+    16)
+
+``--audio-resample-phase-shift=<count>``
+    Log2 of the number of polyphase entries. (..., 10->1024, 11->2048,
+    12->4096, ...) (default: 10->1024)
+
+``--audio-resample-cutoff=<cutoff>``
+    Cutoff frequency (0.0-1.0), default set depending upon filter length.
+
+``--audio-resample-linear=<yes|no>``
+    If set then filters will be linearly interpolated between polyphase
+    entries. (default: no)
+
+``--audio-normalize-downmix=<yes|no>``
+    Enable/disable normalization if surround audio is downmixed to stereo
+    (default: no). If this is disabled, downmix can cause clipping. If it's
+    enabled, the output might be too quiet. It depends on the source audio.
+
+    Technically, this changes the ``normalize`` suboption of the
+    ``lavrresample`` audio filter, which performs the downmixing.
+
+    If downmix happens outside of mpv for some reason, or in the decoder
+    (decoder downmixing), or in the audio output (system mixer), this has no
+    effect.
+
+``--audio-resample-max-output-size=<length>``
+    Limit maximum size of audio frames filtered at once, in ms (default: 40).
+    The output size size is limited in order to make resample speed changes
+    react faster. This is necessary especially if decoders or filters output
+    very large frame sizes (like some lossless codecs or some DRC filters).
+    This option does not affect the resampling algorithm in any way.
+
+    For testing/debugging only. Can be removed or changed any time.
+
+``--audio-swresample-o=<string>``
+    Set AVOptions on the SwrContext or AVAudioResampleContext. These should
+    be documented by FFmpeg or Libav.

 Terminal
 --------
@@ -3839,9 +3924,33 @@ Cache
    very high, so the actually achieved readahead will usually be limited by
    the value of the ``--demuxer-max-bytes`` option.

-``--cache-pause``, ``--no-cache-pause``
-    Whether the player should automatically pause when the cache runs low,
-    and unpause once more data is available ("buffering").
+``--cache-pause=<yes|no>``
+    Whether the player should automatically pause when the cache runs out of
+    data and stalls decoding/playback (default: yes). If enabled, it will
+    pause and unpause once more data is available, aka "buffering".
+
+``--cache-pause-wait=<seconds>``
+    Number of seconds the packet cache should have buffered before starting
+    playback again if "buffering" was entered (default: 1). This can be used
+    to control how long the player rebuffers if ``--cache-pause`` is enabled,
+    and the demuxer underruns. If the given time is higher than the maximum
+    set with ``--cache-secs`` or  ``--demuxer-readahead-secs``, or prefetching
+    ends before that for some other reason (like file end), playback resumes
+    earlier.
+
+``--cache-pause-initial=<yes|no>``
+    Enter "buffering" mode before starting playback (default: no). This can be
+    used to ensure playback starts smoothly, in exchange for waiting some time
+    to prefetch network data (as controlled by ``--cache-pause-wait``). For
+    example, some common behavior is that playback starts, but network caches
+    immediately underrun when trying to decode more data as playback progresses.
+
+    Another thing that can happen is that the network prefetching is so CPU
+    demanding (due to demuxing in the background) that playback drops frames
+    at first. In these cases, it helps enabling this option, and setting
+    ``--cache-secs`` and ``--cache-pause-wait`` to roughly the same value.
+
+    This option also triggers when playback is restarted after seeking.


 Network
@@ -3877,6 +3986,13 @@ Network
            Field2: value2
            Connection: close

+``--http-proxy=<proxy>``
+    URL of the HTTP/HTTPS proxy. If this is set, the ``http_proxy`` environment
+    is ignored. The ``no_proxy`` environment variable is still respected. This
+    option is silently ignored if it does not start with ``http://``. Proxies
+    are not used for https URLs. Setting this option does not try to make the
+    ytdl script use the proxy.
+
 ``--tls-ca-file=<filename>``
    Certificate authority database file for use with TLS. (Silently fails with
    older FFmpeg or Libav versions.)
@@ -3900,6 +4016,16 @@ Network
    special value 0 (default) uses the FFmpeg/Libav defaults. If a protocol
    is used which does not support timeouts, this option is silently ignored.

+    .. warning::
+
+        This breaks the RTSP protocol, because of inconsistent FFmpeg API
+        regarding its internal timeout option. Not only does the RTSP timeout
+        option accept different units (seconds instead of microseconds, causing
+        mpv to pass it huge values), it will also overflow FFmpeg internal
+        calculations. The worst is that merely setting the option will put RTSP
+        into listening mode, which breaks any client uses. Do not use this
+        option with RTSP URLs.
+
 ``--rtsp-transport=<lavf|udp|tcp|http>``
    Select RTSP transport method (default: tcp). This selects the underlying
    network transport when playing ``rtsp://...`` URLs. The value ``lavf``
@@ -3996,6 +4122,20 @@ ALSA audio output options
    or it will work only for files which use the layout implicit to your
    ALSA device).

+``--alsa-buffer-time=<microseconds>``
+    Set the requested buffer time in microseconds. A value of 0 skips requesting
+    anything from the ALSA API. This and the ``--alsa-periods`` option uses the
+    ALSA ``near`` functions to set the requested parameters. If doing so results
+    in an empty configuration set, setting these parameters is skipped.
+
+    Both options control the buffer size. A low buffer size can lead to higher
+    CPU usage and audio dropouts, while a high buffer size can lead to higher
+    latency in volume changes and other filtering.
+
+``--alsa-periods=<number>``
+    Number of periods requested from the ALSA API. See ``--alsa-buffer-time``
+    for further remarks.
+

 GPU renderer options
 -----------------------
@@ -4295,11 +4435,25 @@ The following video options are currently all specific to ``--vo=gpu`` and
    Controls the number of VkQueues used for rendering (limited by how many
    your device supports). In theory, using more queues could enable some
    parallelism between frames (when using a ``--swapchain-depth`` higher than
-    1). (Default: 1)
+    1), but it can also slow things down on hardware where there's no true
+    parallelism between queues. (Default: 1)

-    NOTE: Setting this to a value higher than 1 may cause graphical corruption,
-    as mpv's vulkan implementation currently does not try and protect textures
-    against concurrent access.
+``--vulkan-async-transfer``
+    Enables the use of async transfer queues on supported vulkan devices. Using
+    them allows transfer operations like texture uploads and blits to happen
+    concurrently with the actual rendering, thus improving overall throughput
+    and power consumption. Enabled by default, and should be relatively safe.
+
+``--vulkan-async-compute``
+    Enables the use of async compute queues on supported vulkan devices. Using
+    this, in theory, allows out-of-order scheduling of compute shaders with
+    graphics shaders, thus enabling the hardware to do more effective work while
+    waiting for pipeline bubbles and memory operations. Not beneficial on all
+    GPUs. It's worth noting that if async compute is enabled, and the device
+    supports more compute queues than graphics queues (bound by the restrictions
+    set by ``--vulkan-queue-count``), mpv will internally try and prefer the
+    use of compute shaders over fragment shaders wherever possible. Not enabled
+    by default, since it seems to cause issues with some drivers.

 ``--d3d11-warp=<yes|no|auto>``
    Use WARP (Windows Advanced Rasterization Platform) with the D3D11 GPU
@@ -4589,7 +4743,7 @@ The following video options are currently all specific to ``--vo=gpu`` and

 ``--deband``
    Enable the debanding algorithm. This greatly reduces the amount of visible
-    banding, blocking and other quantization artifacts, at the expensive of
+    banding, blocking and other quantization artifacts, at the expense of
    very slightly blurring some of the finest details. In practice, it's
    virtually always an improvement - the only reason to disable it would be
    for performance.
@@ -4727,6 +4881,46 @@ The following video options are currently all specific to ``--vo=gpu`` and

    OS X only.

+``--cocoa-cb-sw-renderer=<yes|no|auto>``
+    Use the Apple Software Renderer when using cocoa-cb (default: auto). If set
+    to ``no`` the software renderer is never used and instead fails when a the
+    usual pixel format could not be created, ``yes`` will always only use the
+    software renderer, and ``auto`` only falls back to the software renderer
+    when the usual pixel format couldn't be created.
+
+    OS X only.
+
+``--macos-title-bar-style=<dark|ultradark|light|mediumlight|auto>``
+    Sets the styling of the title bar (default: dark).
+    OS X and cocoa-cb only
+
+    :dark:        Dark title bar with vibrancy, a subtle blurring effect that
+                  dynamically blends the background (Video) into the title bar.
+    :ultradark:   Darker title bar with vibrancy (like QuickTime Player).
+    :light:       Bright title bar with vibrancy.
+    :mediumlight: Less bright title bar with vibrancy.
+    :auto:        Detects the system settings and sets the title bar styling
+                  appropriately, either ultradark or mediumlight.
+
+``--macos-fs-animation-duration=<default|0-1000>``
+    Sets the fullscreen resize animation duration in ms (default: default).
+    The default value is slightly less than the system's animation duration
+    (500ms) to prevent some problems when the end of an async animation happens
+    at the same time as the end of the system wide fullscreen animation. Setting
+    anything higher than 500ms will only prematurely cancel the resize animation
+    after the system wide animation ended. The upper limit is still set at
+    1000ms since it's possible that Apple or the user changes the system
+    defaults. Anything higher than 1000ms though seems too long and shouldn't be
+    set anyway.
+    OS X and cocoa-cb only
+
+``--android-surface-size=<WxH>``
+    Set dimensions of the rendering surface used by the Android gpu context.
+    Needs to be set by the embedding application if the dimensions change during
+    runtime (i.e. if the device is rotated), via the surfaceChanged callback.
+
+    Android with ``--gpu-context=android`` only.
+
 ``--swapchain-depth=<N>``
    Allow up to N in-flight frames. This essentially controls the frame
    latency. Increasing the swapchain depth can improve pipelining and prevent
@@ -4746,7 +4940,7 @@ The following video options are currently all specific to ``--vo=gpu`` and
    auto
        auto-select (default)
    cocoa
-        Cocoa/OS X
+        Cocoa/OS X (deprecated, use --vo=opengl-cb instead)
    win
        Win32/WGL
    winvk
@@ -4819,9 +5013,12 @@ The following video options are currently all specific to ``--vo=gpu`` and
    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: rgb8, rgb10, rgb10_a2, rgb16, rgb16f, rgb32f, rgba12, rgba16, rgba16f,
-    rgba16hf, rgba32f. Default: ``auto``, which maps to rgba16 on desktop GL,
-    and rgba16f or rgb10_a2 on GLES (e.g. ANGLE), unless GL_EXT_texture_norm16
-    is available.
+    rgba16hf, rgba32f.
+
+    Default: ``auto``, which first attempts to utilize 16bit float
+    (rgba16f, rgba16hf), and falls back to rgba16 if those are not available.
+    Finally, attempts to utilize rgb10_a2 or rgba8 if all of the previous formats
+    are not available.

 ``--gamma-factor=<0.1..2.0>``
    Set an additional raw gamma factor (default: 1.0). If gamma is adjusted in
@@ -4856,7 +5053,9 @@ The following video options are currently all specific to ``--vo=gpu`` and
    are:

    auto
-        Disable any adaptation (default)
+        Disable any adaptation, except for atypical color spaces. Specifically,
+        wide/unusual gamuts get automatically adapted to BT.709, while standard
+        gamut (i.e. BT.601 and BT.709) content is not touched. (default)
    bt.470m
        ITU-R BT.470 M
    bt.601-525
@@ -4888,7 +5087,9 @@ The following video options are currently all specific to ``--vo=gpu`` and
    Valid values are:

    auto
-        Disable any adaptation (default)
+        Disable any adaptation, except for atypical transfers. Specifically,
+        HDR or linear light source material gets automatically converted to
+        gamma 2.2, while SDR content is not touched. (default)
    bt.1886
        ITU-R BT.1886 curve (assuming infinite contrast)
    srgb
@@ -4922,6 +5123,39 @@ The following video options are currently all specific to ``--vo=gpu`` and
        The user should independently guarantee this before using these signal
        formats for display.

+``--target-peak=<nits>``
+    Specifies the measured peak brightness of the output display, in cd/m^2
+    (AKA nits). The interpretation of this brightness depends on the configured
+    ``--target-trc``. In all cases, it imposes a limit on the signal values
+    that will be sent to the display. If the source exceeds this brightness
+    level, a tone mapping filter will be inserted. For HLG, it has the
+    additional effect of parametrizing the inverse OOTF, in order to get
+    colorimetrically consistent results with the mastering display. For SDR, or
+    when using an ICC (profile (``--icc-profile``), setting this to a value
+    above 100 essentially causes the display to be treated as if it were an HDR
+    display in disguise. (See the note below)
+
+    By default, the chosen peak defaults to an appropriate value based on the
+    TRC in use. For SDR curves, it defaults to 100. For HDR curves, it
+    defaults to 100 * the transfer function's nominal peak.
+
+    .. note::
+
+        When using an SDR transfer function, this is normally not needed, and
+        setting it may lead to very unexpected results. The one time it *is*
+        useful is if you want to calibrate a HDR display using traditional
+        transfer functions and calibration equipment. In such cases, you can
+        set your HDR display to a high brightness such as 800 cd/m^2, and then
+        calibrate it to a standard curve like gamma2.8. Setting this value to
+        800 would then instruct mpv to essentially treat it as an HDR display
+        with the given peak. This may be a good alternative in environments
+        where PQ or HLG input to the display is not possible, and makes it
+        possible to use HDR displays with mpv regardless of operating system
+        support for HDMI HDR metadata.
+
+        In such a configuration, we highly recommend setting ``--tone-mapping``
+        to ``mobius`` or even ``clip``.
+
 ``--tone-mapping=<value>``
    Specifies the algorithm used for tone-mapping images onto the target
    display. This is relevant for both HDR->SDR conversion as well as gamut
@@ -4938,7 +5172,7 @@ The following video options are currently all specific to ``--vo=gpu`` and
        for in-range material as much as possible. Use this when you care about
        color accuracy more than detail preservation. This is somewhere in
        between ``clip`` and ``reinhard``, depending on the value of
-        ``--tone-mapping-param``. (default)
+        ``--tone-mapping-param``.
    reinhard
        Reinhard tone mapping algorithm. Very simple continuous curve.
        Preserves overall image brightness but uses nonlinear contrast, which
@@ -4949,7 +5183,9 @@ The following video options are currently all specific to ``--vo=gpu`` and
        desaturating everything. Developed by John Hable for use in video
        games. Use this when you care about detail preservation more than
        color/brightness accuracy. This is roughly equivalent to
-        ``--hdr-tone-mapping=reinhard --tone-mapping-param=0.24``.
+        ``--tone-mapping=reinhard --tone-mapping-param=0.24``. If possible,
+        you should also enable ``--hdr-compute-peak`` for the best results.
+        (Default)
    gamma
        Fits a logarithmic transfer between the tone curves.
    linear
@@ -4978,13 +5214,15 @@ The following video options are currently all specific to ``--vo=gpu`` and
    linear
        Specifies the scale factor to use while stretching. Defaults to 1.0.

-``--hdr-compute-peak``
-    Compute the HDR peak per-frame of relying on tagged metadata. These values
-    are averaged over local regions as well as over several frames to prevent
-    the value from jittering around too much. This option basically gives you
-    dynamic, per-scene tone mapping. Requires compute shaders, which is a
-    fairly recent OpenGL feature, and will probably also perform horribly on
-    some drivers, so enable at your own risk.
+``--hdr-compute-peak=<auto|yes|no>``
+    Compute the HDR peak and frame average brightness per-frame instead of
+    relying on tagged metadata. These values are averaged over local regions as
+    well as over several frames to prevent the value from jittering around too
+    much. This option basically gives you dynamic, per-scene tone mapping.
+    Requires compute shaders, which is a fairly recent OpenGL feature, and will
+    probably also perform horribly on some drivers, so enable at your own risk.
+    The special value ``auto`` (default) will enable HDR peak computation
+    automatically if compute shaders and SSBOs are supported.

 ``--tone-mapping-desaturate=<value>``
    Apply desaturation for highlights. The parameter essentially controls the
@@ -4994,8 +5232,9 @@ The following video options are currently all specific to ``--vo=gpu`` and
    into white instead. This makes images feel more natural, at the cost of
    reducing information about out-of-range colors.

-    The default of 1.0 provides a good balance that roughly matches the look
-    and feel of the ACES ODT curves. A setting of 0.0 disables this option.
+    The default of 0.5 provides a good balance. This value is weaker than the
+    ACES ODT curves' recommendation, but works better for most content in
+    practice. A setting of 0.0 disables this option.

 ``--gamut-warning``
    If enabled, mpv will mark all clipped/out-of-gamut pixels that exceed a
@@ -5027,6 +5266,9 @@ The following video options are currently all specific to ``--vo=gpu`` and
    NOTE: On Windows, the default profile must be an ICC profile. WCS profiles
    are not supported.

+    Applications using libmpv with the render API need to provide the ICC
+    profile via ``MPV_RENDER_PARAM_ICC_PROFILE``.
+
 ``--icc-cache-dir=<dirname>``
    Store and load the 3D LUTs created from the ICC profile in this directory.
    This can be used to speed up loading, since LittleCMS 2 can take a while to
@@ -5053,7 +5295,7 @@ The following video options are currently all specific to ``--vo=gpu`` and
    Size of the 3D LUT generated from the ICC profile in each dimension.
    Default is 64x64x64. Sizes may range from 2 to 512.

-``--icc-contrast=<0-100000>``
+``--icc-contrast=<0-1000000>``
    Specifies an upper limit on the target device's contrast ratio. This is
    detected automatically from the profile if possible, but for some profiles
    it might be missing, causing the contrast to be assumed as infinite. As a
@@ -5123,6 +5365,9 @@ The following video options are currently all specific to ``--vo=gpu`` and
    flipping GL front and backbuffers immediately (i.e. it doesn't call it
    in display-sync mode).

+    On OSX this is always deactivated because it only causes performance
+    problems and other regressions.
+
 ``--gpu-dumb-mode=<yes|no|auto>``
    This mode is extremely restricted, and will disable most extended
    features. That includes high quality scalers and custom shaders!
@@ -5147,16 +5392,6 @@ The following video options are currently all specific to ``--vo=gpu`` and
    NOTE: This is not cleaned automatically, so old, unused cache files may
    stick around indefinitely.

-``--cuda-decode-device=<auto|0..>``
-    Choose the GPU device used for decoding when using the ``cuda`` hwdec.
-
-    By default, the device that is being used to provide OpenGL output will
-    also be used for decoding (and in the vast majority of cases, only one
-    GPU will be present).
-
-    Note that when using the ``cuda-copy`` hwdec, a different option must be
-    passed: ``--vd-lavc-o=gpu=<0..>``.
-
 Miscellaneous
 -------------

@@ -5186,6 +5421,26 @@ Miscellaneous
    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.

+``--video-timing-offset=<seconds>``
+    Control how long before video display target time the frame should be
+    rendered (default: 0.050). If a video frame should be displayed at a
+    certain time, the VO will start rendering the frame earlier, and then will
+    perform a blocking wait until the display time, and only then "swap" the
+    frame to display. The rendering cannot start before the previous frame is
+    displayed, so this value is implicitly limited by the video framerate. With
+    normal video frame rates, the default value will ensure that rendering is
+    always immediately started after the previous frame was displayed. On the
+    other hand, setting a too high value can reduce responsiveness with low
+    FPS value.
+
+    For client API users using the render API (or the deprecated ``opengl-cb``
+    API), this option is interesting, because you can stop the render API
+    from limiting your FPS (see ``mpv_render_context_render()`` documentation).
+
+    This applies only to audio timing modes (e.g. ``--video-sync=audio``). In
+    other modes (``--video-sync=display-...``), video timing relies on vsync
+    blocking, and this option is not used.
+
 ``--video-sync=<audio|...>``
    How the player synchronizes audio and video.

@@ -5318,7 +5573,8 @@ Miscellaneous

    Unlike ``--sub-files`` and ``--audio-files``, this includes all tracks, and
    does not cause default stream selection over the "proper" file. This makes
-    it slightly less intrusive.
+    it slightly less intrusive. (In mpv 0.28.0 and before, this was not quite
+    strictly enforced.)

    This is a list option. See `List Options`_ for details.

@@ -5399,6 +5655,8 @@ Miscellaneous
          both tracks need to have the same width, or filter initialization
          will fail (you can add ``scale`` filters before the ``vstack`` filter
          to fix the size).
+          To load a video track from another file, you can use
+          ``--external-file=other.mkv``.
        - ``--lavfi-complex='[aid1] asplit [t1] [ao] ; [t1] showvolume [t2] ; [vid1] [t2] overlay [vo]'``
          Play audio track 1, and overlay the measured volume for each speaker
          over video track 1.
@@ -128,7 +128,7 @@ Configuration
 -------------

 The OSC offers limited configuration through a config file
-``lua-settings/osc.conf`` placed in mpv's user dir and through the
+``script-opts/osc.conf`` placed in mpv's user dir and through the
 ``--script-opts`` command-line option. Options provided through the command-line
 will override those from the config file.

@@ -171,6 +171,15 @@ Configurable Options
    marker with guide), or bar (fill).
    Default pre-0.21.0 was 'slider'.

+``seekbarkeyframes``
+    Default: yes
+
+    Controls the mode used to seek when dragging the seekbar. By default,
+    keyframes are used. If set to false, exact seeking on mouse drags
+    will be used instead. Keyframes are preferred, but exact seeks may be
+    useful in cases where keyframes cannot be found. Note that using exact
+    seeks can potentially make mouse dragging much slower.
+
 ``deadzonesize``
    Default: 0.5

@@ -272,6 +281,11 @@ Configurable Options

    Display timecodes with milliseconds

+``seekranges``
+    Default: yes
+
+    Display seekable ranges on the seekbar
+
 ``visibility``
    Default: auto (auto hide/show on mouse move)

@@ -35,7 +35,7 @@ monospaced digits is recommended. By default, the open source font
 Configuration
 -------------

-This script can be customized through a config file ``lua-settings/stats.conf``
+This script can be customized through a config file ``script-opts/stats.conf``
 placed in mpv's user directory and through the ``--script-opts`` command-line
 option. The configuration syntax is described in `ON SCREEN CONTROLLER`_.

@@ -1,8 +1,11 @@
 VIDEO FILTERS
 =============

-Video filters allow you to modify the video stream and its properties. The
-syntax is:
+Video filters allow you to modify the video stream and its properties. All of
+the information described in this section applies to audio filters as well
+(generally using the prefix ``--af`` instead of ``--vf``).
+
+The exact syntax is:

 ``--vf=<filter1[=parameter1:parameter2:...],filter2,...>``
    Setup a chain of video filters. This consists on the filter name, and an
@@ -13,7 +16,7 @@ syntax is:
    name is an arbitrary user-given name, which identifies the filter. This
    is only needed if you want to toggle the filter at runtime.

-    A ``!`` before the filter name means the filter is enabled by default. It
+    A ``!`` before the filter name means the filter is disabled by default. It
    will be skipped on filter creation. This is also useful for runtime filter
    toggling.

@@ -41,11 +44,18 @@ syntax is:
    the ``lavfi`` filter, which uses a very similar syntax as mpv (MPlayer
    historically) to specify filters and their parameters.

+Filters can be manipulated at run time. You can use ``@`` labels as described
+above in combination with the ``vf`` command (see `COMMAND INTERFACE`_) to get
+more control over this. Initially disabled filters with ``!`` are useful for
+this as well.
+
 You can also set defaults for each filter. The defaults are applied before the
-normal filter parameters.
+normal filter parameters. This is deprecated and never worked for the
+libavfilter bridge.

 ``--vf-defaults=<filter1[=parameter1:parameter2:...],filter2,...>``
-    Set defaults for each filter.
+    Set defaults for each filter. (Deprecated. ``--af-defaults`` is deprecated
+    as well.)

 .. note::

@@ -75,15 +85,20 @@ normal filter parameters.
 Video filters are managed in lists. There are a few commands to manage the
 filter list.

-``--vf-add=<filter1[,filter2,...]>``
-    Appends the filters given as arguments to the filter list.
+``--vf-add=filter``
+    Appends the filter given as arguments to the filter list. (Passing multiple
+    filters is currently still possible, but deprecated.)

-``--vf-pre=<filter1[,filter2,...]>``
-    Prepends the filters given as arguments to the filter list.
+``--vf-pre=filter``
+    Prepends the filters given as arguments to the filter list. (Passing
+    multiple filters is currently still possible, but deprecated.)

-``--vf-del=<index1[,index2,...]>``
-    Deletes the filters at the given indexes. Index numbers start at 0,
-    negative numbers address the end of the list (-1 is the last).
+``--vf-del=filter``
+    Deletes the filter. The filter can even given the way it was added (filter
+    name and its full argument list), by label (prefixed with ``@``), or as
+    index number. Index numbers start at 0, negative numbers address the end of
+    the list (-1 is the last). (Passing multiple filters is currently still
+    possible, but deprecated.)

 ``--vf-clr``
    Completely empties the filter list.
@@ -94,10 +109,6 @@ With filters that support it, you can access parameters by their name.
    Prints the parameter names and parameter value ranges for a particular
    filter.

-``--vf=<filter=named_parameter1=value1[:named_parameter2=value2:...]>``
-    Sets a named parameter to the given value. Use on and off or yes and no to
-    set flag parameters.
-
 Available mpv-only filters are:

 ``format=fmt=<value>:colormatrix=<value>:...``
@@ -110,9 +121,7 @@ Available mpv-only filters are:

    ``<fmt>``
        Format name, e.g. rgb15, bgr24, 420p, etc. (default: don't change).
-    ``<outfmt>``
-        Format name that should be substituted for the output. If they do not
-        have the same bytes per pixel and chroma subsampling, it will fail.
+
    ``<colormatrix>``
        Controls the YUV to RGB color space conversion when playing video. There
        are various standards. Normally, BT.601 should be used for SD video, and
@@ -120,7 +129,7 @@ Available mpv-only filters are:
        results in slightly under or over saturated and shifted colors.

        These options are not always supported. Different video outputs provide
-        varying degrees of support. The ``opengl`` and ``vdpau`` video output
+        varying degrees of support. The ``gpu`` and ``vdpau`` video output
        drivers usually offer full support. The ``xv`` output can set the color
        space if the system video driver supports it, but not input and output
        levels. The ``scale`` video filter can configure color space and input
@@ -163,7 +172,7 @@ Available mpv-only filters are:
        used to override the setting.

        This option only affects video output drivers that perform color
-        management, for example ``opengl`` with the ``target-prim`` or
+        management, for example ``gpu`` with the ``target-prim`` or
        ``icc-profile`` suboptions set.

        If this option is set to ``auto`` (which is the default), the video's
@@ -459,7 +468,7 @@ Available mpv-only filters are:
    other userdata type will result in hard crashes.

 ``vavpp``
-    VA-AP-API video post processing. Works with ``--vo=vaapi`` and ``--vo=opengl``
+    VA-AP-API video post processing. Works with ``--vo=vaapi`` and ``--vo=gpu``
    only. Currently deinterlaces. This filter is automatically inserted if
    deinterlacing is requested (either using the ``d`` key, by default mapped to
    the command ``cycle deinterlace``, or the ``--deinterlace`` option).
@@ -469,18 +478,22 @@ Available mpv-only filters are:

        no
            Don't perform deinterlacing.
+        auto
+             Select the best quality deinterlacing algorithm (default). This
+             goes by the order of the options as documented, with
+             ``motion-compensated`` being considered best quality.
        first-field
            Show only first field.
        bob
-            bob deinterlacing (default).
+            bob deinterlacing.
        weave, motion-adaptive, motion-compensated
            Advanced deinterlacing algorithms. Whether these actually work
            depends on the GPU hardware, the GPU drivers, driver bugs, and
            mpv bugs.

    ``<interlaced-only>``
-        :no:  Deinterlace all frames.
-        :yes: Only deinterlace frames marked as interlaced (default).
+        :no:  Deinterlace all frames (default).
+        :yes: Only deinterlace frames marked as interlaced.

    ``reversal-bug=<yes|no>``
        :no:  Use the API as it was interpreted by older Mesa drivers. While
@@ -492,12 +505,12 @@ Available mpv-only filters are:
              algorithms.

 ``vdpaupp``
-    VDPAU video post processing. Works with ``--vo=vdpau`` and ``--vo=opengl``
+    VDPAU video post processing. Works with ``--vo=vdpau`` and ``--vo=gpu``
    only. This filter is automatically inserted if deinterlacing is requested
    (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
+    if the ``vdpau`` VO is used, and also if ``gpu`` is used and hardware
    decoding was activated at least once (i.e. vdpau was loaded).

    ``sharpen=<-1-1>``
@@ -535,7 +548,7 @@ Available mpv-only filters are:
        Try to apply inverse telecine, needs motion adaptive temporal
        deinterlacing.
    ``interlaced-only=<yes|no>``
-        If ``yes`` (default), only deinterlace frames marked as interlaced.
+        If ``yes``, only deinterlace frames marked as interlaced (default: no).
    ``hqscaling=<0-9>``
        0
            Use default VDPAU scaling (default).
@@ -549,7 +562,7 @@ Available mpv-only filters are:
    ``deint=<yes|no>``
        Whether deinterlacing is enabled (default: no).
    ``interlaced-only=<yes|no>``
-        If ``yes`` (default), only deinterlace frames marked as interlaced.
+        If ``yes``, only deinterlace frames marked as interlaced (default: no).
    ``mode=<blend|bob|adaptive|mocomp|ivctc|none>``
        Tries to select a video processor with the given processing capability.
        If a video processor supports multiple capabilities, it is not clear
@@ -324,7 +324,7 @@ Available video output drivers are:
    This is low quality, and has issues with OSD.

    .. note:: This driver is for compatibility with crappy systems. You can
-              use vaapi hardware decoding with ``--vo=opengl`` too.
+              use vaapi hardware decoding with ``--vo=gpu`` too.

    The following global options are supported by this video output:

@@ -428,37 +428,21 @@ Available video output drivers are:
    ``--vo-image-outdir=<dirname>``
        Specify the directory to save the image files to (default: ``./``).

-``wayland`` (Wayland only)
-    Wayland shared memory video output as fallback for ``opengl``.
+``libmpv``
+    For use with libmpv direct embedding. As a special case, on OS X it
+    is used like a normal VO within mpv (cocoa-cb). Otherwise useless in any
+    other contexts.
+    (See ``<mpv/render.h>``.)

-    .. note:: This driver is for compatibility with systems that don't provide
-              working OpenGL drivers.
-
-    The following global options are supported by this video output:
-
-    ``--vo-wayland-alpha``
-        Use a buffer format that supports videos and images with alpha
-        information
-    ``--vo-wayland-rgb565``
-        Use RGB565 as buffer format. This format is implemented on most
-        platforms, especially on embedded where it is far more efficient then
-        RGB8888.
-    ``--vo-wayland-triple-buffering``
-        Use 3 buffers instead of 2. This can lead to more fluid playback, but
-        uses more memory.
-
-``opengl-cb``
-    For use with libmpv direct OpenGL embedding; useless in any other contexts.
-    (See ``<mpv/opengl_cb.h>``.)
-
-    This also supports many of the options the ``opengl`` VO has.
+    This also supports many of the options the ``gpu`` VO has, depending on the
+    backend.

 ``rpi`` (Raspberry Pi)
    Native video output on the Raspberry Pi using the MMAL API.

-    This is deprecated. Use ``--vo=opengl`` instead, which is the default and
+    This is deprecated. Use ``--vo=gpu`` instead, which is the default and
    provides the same functionality. The ``rpi`` VO will be removed in
-    mpv 0.23.0. Its functionality was folded into --vo=opengl, which now uses
+    mpv 0.23.0. Its functionality was folded into --vo=gpu, which now uses
    RPI hardware decoding by treating it as a hardware overlay (without applying
    GL filtering). Also to be changed in 0.23.0: the --fs flag will be reset to
    "no" by default (like on the other platforms).
@@ -488,7 +472,7 @@ Available video output drivers are:
    Video output driver using Kernel Mode Setting / Direct Rendering Manager.
    Should be used when one doesn't want to install full-blown graphical
    environment (e.g. no X). Does not support hardware acceleration (if you
-    need this, check the ``drm`` backend for ``opengl`` VO).
+    need this, check the ``drm`` backend for ``gpu`` VO).

    The following global options are supported by this video output:

@@ -501,13 +485,42 @@ Available video output drivers are:
        (default: empty)

    ``--drm-mode=<number>``
-        Mode ID to use (resolution, bit depth and frame rate).
+        Mode ID to use (resolution and frame rate).
        (default: 0)

-    ``--drm-overlay=<number>``
-        Select the DRM overlay index to use.
-        Overlay index is zero based, and related to crtc.
-        (default: 0)
+    ``--drm-osd-plane-id=<number>``
+        Select the DRM plane index to use for OSD (or OSD and video).
+        Index is zero based, and related to crtc.
+        When using this option with the drm_prime renderer, it will only affect
+        the OSD contents. Otherwise it will set OSD & video plane.
+        (default: primary plane)
+
+    ``--drm-video-plane-id=<number>``
+        Select the DRM plane index to use for video layer.
+        Index is zero based, and related to crtc.
+        This option only has effect when using the drm_prime renderer (which
+        supports several layers) together with ``vo=gpu`` and ``gpu-context=drm``.
+        (default: first overlay plane)
+
+    ``--drm-format=<xrgb8888|xrgb2101010>``
+        Select the DRM format to use (default: xrgb8888). This allows you to
+        choose the bit depth of the DRM mode. xrgb8888 is your usual 24 bit per
+        pixel/8 bits per channel packed RGB format with 8 bits of padding.
+        xrgb2101010 is a packed 30 bits per pixel/10 bits per channel packed RGB
+        format with 2 bits of padding.
+
+        Unless you have an intel graphics card, a recent kernel and a recent
+        version of mesa (>=18) xrgb2101010 is unlikely to work for you.
+
+        This currently only has an effect when used together with the ``drm``
+        backend for the ``gpu`` VO. The ``drm`` VO always uses xrgb8888.
+
+    ``--drm-osd-size=<[WxH]>``
+        Sets the OSD OpenGL size to the specified size. OSD will then be upscaled
+        to the current screen resolution. This option can be useful when using
+        several layers in high resolutions with a GPU which cannot handle it.
+        Note : this option is only available with DRM atomic support.
+        (default: display resolution)

 ``mediacodec_embed`` (Android)
    Renders ``IMGFMT_MEDIACODEC`` frames directly to an ``android.view.Surface``.
@@ -177,13 +177,13 @@ video/out/:
    Video output. They also create GUI windows and handle user input. In most
    cases, the windowing code is shared among VOs, like x11_common.c for X11 and
    w32_common.c for Windows. The VOs stand between frontend and windowing code.
-    vo_opengl can pick a windowing system at runtime, e.g. the same binary can
+    vo_gpu can pick a windowing system at runtime, e.g. the same binary can
    provide both X11 and Cocoa support on OSX.

    VOs can be reconfigured at runtime. A vo_reconfig() call can change the video
    resolution and format, without destroying the window.

-    vo_opengl should be taken as reference.
+    vo_gpu should be taken as reference.

 audio/:
    format.h/format.c define the uncompressed audio formats. (As well as some
@@ -132,7 +132,11 @@ FFmpeg dependencies:
 - OpenSSL or GnuTLS (have to be explicitly enabled when compiling FFmpeg)
 - libx264/libmp3lame/libfdk-aac if you want to use encoding (have to be
  explicitly enabled when compiling FFmpeg)
- Libav also works, but some features will not work. (See section below.)
+- For native DASH playback, FFmpeg needs to be built with --enable-libxml2
+  (although there are security implications).
+- For good nvidia support on Linux, make sure nv-codec-headers is installed
+  and can be found by configure.
+- Libav support is broken. (See section below.)

 Most of the above libraries are available in suitable versions on normal
 Linux distributions. For ease of compiling the latest git master of everything,
@@ -149,8 +153,8 @@ or cross-compile from Linux with MinGW. See


 Generally, mpv should work with the latest release as well as the git version
-of both FFmpeg and Libav. But FFmpeg is preferred, and some mpv features work
-with FFmpeg only (subtitle formats in particular).
+of FFmpeg. Libav support is currently broken, because they did not add certain
+FFmpeg API changes which mpv relies on.


 ## FFmpeg ABI compatibility
@@ -1,18 +1,49 @@
-Release 0.28.0
+Release 0.29.1
 ==============

-This release needs recent FFmpeg (newer than 3.4) due to major refactoring.
-Required library versions:
+A bug fix release for the 0.29 release branch.

-  * libavutil >= 56.6.100
-  * libavcodec >= 58.7.100
-  * libavformat >= 58.0.102
-  * libswscale >= 5.0.101
-  * libavfilter >= 7.0.101
-  * libswresample >= 3.0.100
+Fixes and Minor Enhancements
+----------------------------
+
+- af_rubberband: Fix for A-V drift on seeking.
+- ao_alsa: Fixes to remove spam during pausing and end-of-file.
+- ao_jack: Fix for mpv attempting to connect to non-standard devices such as MIDI.
+- ao_openal: Drop support for Apple's bundled OpenAL due to lack of features to successfully build.
+- ao_pulse: Fix for audio buffer length calculation.
+- demux_lavf: Fix EDL playback of certain DASH streams.
+- demux_lavf: Mark v4l streams as not seekable.
+- demux_raw: Fix for gapless playback from raw audio input.
+- drm_atomic: Fix to enable video output with devices without a video overlay plane.
+- encode: Fix for crashes in case of failure to write to output.
+- encode: Fix for encoding when utilizing lavfi-complex.
+- Fix to strip HDR peak metadata if attached to SDR video (#6111)
+- macOS: Capability to fall back on SW rendering for testing purposes.
+- macOS: Fix for issues with the main menu.
+- macOS: Fix for side by side split view.
+- macOS: Fixes for crashes, most notably those happening on 10.14 (#5908, #6097, #6041, #6062).
+- macOS: No longer let older, unsupported versions of the Swift build tools through configure.
+- manpage: Misc fixups.
+- stream_{smb,file}: Fix for partial writes to buffers.
+- stream_libarchive: Fix hang when an out-of-bound seek happens.
+- stream_smb: Fix for crashes due to libsmbclient and threading (#5936).
+- vo_gpu: Fix for corruption when multiple compute shaders override each others' block sizes (#6083).
+- ytdl_hook: Always load "ytdl://" URLs with ytdl_hook first.
+- ytdl_hook: Fix audio streams not being picked up for some sites.
+
+Options and Commands
+--------------------
+
+Added
+~~~~~
+
+- `--cocoa-cb-sw-renderer` to control whether or not to fall back on software rendering on macOS.
+
+Release 0.29.0
+==============
+
+This release requires FFmpeg 4.0 or newer.

-The LGPL 2.1+ relicensing process is now mostly complete. The remaining GPL-only
-code (see Copyright file) can be disabled at build time by using --enable-lgpl.

 Features
 --------
@@ -20,33 +51,40 @@ Features
 Added
 ~~~~~

- Add DRM_PRIME Format Handling and Display for RockChip MPP decoders
- csputils: Add support for Display P3 primaries
- demux: support multiple seekable cached ranges, display cache ranges on OSC
- demux_playlist: support .url files (#5107)
- dvb: Add multiple frontends support (up to 8)
- dvb: implement parsing of modulation for VDR-style channels config
- hwdec: add mediacodec hardware decoder for IMGFMT_MEDIACODEC frames,
-  rename mediacodec to mediacodec-copy
- lua: integrate stats.lua script (bound to i/I by default)
- vd_lavc: add support for nvdec hwaccel
- vo_gpu: add android opengl backend
- vo_gpu: initial d3d11 support
- vo_gpu: vulkan support
+- ao/openal: Add better sample format and channel layout selection
+- ao/openal: Add support for direct channels output
+- cocoa-cb: initial implementation via opengl-cb API
+- context_drm_egl: Introduce 30bpp support (requires Mesa 18.0.0_rc4 or later)
+- osx: add some more menu bar items as suggested by Apples's HIG
+- vo_gpu: make screenshots use the GL renderer (#5498, #5240)
+- x11: support Shift+TAB as an input key bind (#5849)
+- ytdl_hook: support native dash demuxer, if present
+
+
+Changed
+~~~~~~~
+
+- ao_alsa, ao_pulse: reduce requested buffer size
+- audio: change format negotiation, remove channel remix fudging
+- client API: deprecate opengl-cb API and introduce a replacement API
+- demux: lower demuxer cache default sizes from 400MB/400MB to 50MB/150MB
+- osx: always deactivate the early opengl flush on macOS
+- scripting: change when/how player waits for scripts being loaded
+- vo_gpu: various improvements to HDR peak detection
+- vo_gpu: various improvements to HDR tone mapping
+- wayland_common: update to stable xdg-shell
+- ytdl_hook: parse youtube playlist urls to set start index if `--ytdl-raw-option=yes-playlist=` is specified
+
+Deprecated
+~~~~~~~~~~
+
+- af_lavrresample: deprecate this filter


 Removed
 ~~~~~~~

- af: remove deprecated audio filters (channels, equalizer, pan, volume;
-  replacements in lavfi)
- vf: remove most GPL video filters (crop, dsize, expand, flip, gradfun, mirror,
-  noformat, pullup, rotate, scale, stereo3d, yadif; replacements in lavfi)
- vf_buffer: remove this filter
- video: remove automatic stereo3d filter insertion
- vo_gpu: remove hwdec_vaglx interop
- vo_opengl: refactor into vo_gpu
- vo_wayland: remove
+- build: drop support for SDL1 (only applied to ao_sdl)


 Options and Commands
@@ -55,106 +93,88 @@ Options and Commands
 Added
 ~~~~~

- demux: add option to create CC tracks eagerly (--sub-create-cc-track)
- options: add --start=none to reset previously set start time
- options: add --vlang switch
+- add a number of --audio-resample-* options, which should from now on be used instead of --af-defaults=lavrresample:
+- ao/openal: --openal-direct-channels
+- command: add --osd-on-seek option defaulting to bar
+- command: add a change-list command (#5435)
+- options: add --http-proxy
+- options: add a builtin low-latency profile
+- vaapi: add option to select a non-default device path with --vaapi-device (#4320)
+- video: add option to reduce latency by 1 or 2 frames
+- vo_gpu: introduce --target-peak (#5521)


 Changed
 ~~~~~~~

- cache: lower default size to 2*10MB
- demux: bump the demuxer cache readahead duration to 10 hours
- demux: use seekable cache for network by default, bump prefetch limit
- msg: make --msg-level affect --log-file too
- player/misc.c: allow both --length and --end to control play endpoint
- player: match subtitles with language tags with --sub-auto=exact
- rename --opengl-hwdec-interop to --gpu-hwdec-interop (now mostly useless)
- vd_lavc: prefer nvdec over vdpau with --hwdec=auto (better codec and surface
-  format support)
- vd_lavc: rename --hwdec=rpi to --hwdec=mmal
+- command: change cycle-value command behavior (#5772)
+- config: replace config dir lua-settings/ with dir script-opts/
+- options: --lavfi-complex now requires uniquely named filter pads.
+- options: --ytdl is now always enabled, even for libmpv
+- options: make --deinterlace=yes always deinterlace (#5219)
+- options: slightly improve filter help output for lavfi bridge
+- vo: rename --vo=opengl-cb to --vo=libmpv
+
+
+Deprecated
+~~~~~~~~~~
+
+- encoding: deprecate a bunch of obscure options (--ovoffset, --oaoffset, --ovfirst, --oafirst)
+- options: deprecate --vf-defaults and --af-defaults
+- osx: --gpu-context=cocoa with --gpu-api=opengl is now deprecated in favour of --vo=libmpv


 Removed
 ~~~~~~~

- options: remove --heartbeat-cmd and --heartbeat-interval
-  (incidentally fixes #4888)
+- command: remove an old compatibility hack that allowed CLI aliases to be set as property, deprecated in 0.26.0
+- input: remove legacy command handling for MPlayer commands that were deprecated in 2013
+- options: drop --opensles-sample-rate, as --audio-samplerate should be used if desired
+- options: drop deprecated --videotoolbox-format, --ff-aid, --ff-vid, --ff-sid, --ad-spdif-dtshd, --softvol options
+- options: remove --video-stereo-mode
+- options: remove deprecated ao/vo auto profiles
+- options: remove the following encoding options: --ocopyts (now the default, old timestamp handling is gone), --oneverdrop (now default), --oharddup (you need to use --vf=fps=VALUE), --ofps, --oautofps, --omaxfps


 Fixes and Minor Enhancements
 ----------------------------

- TOOLS/autoload.lua: add ogm, ogg and opus extensions
- Use /dev/tty instead of stdin for terminal input (#4190)
- audio: add audio softvol processing to AO (replaces previously GPL’d code)
- audio: fix channel conversion with NA channels (e.g. with ALSA)
- audio: fix missing volume update on init and reinit
- csputils: Fix DCI P3 primaries white point
- demux: don't allow subtitles to mess up buffered time display
- demux: fix .cue files with audio files that contain attached pictures
- demux: fix accounting for seekable ranges on track switches (fixes missing
-  audio when cycling through audio tracks with e.g. EDL, --merge-files, ordered
-  chapters and youtube-dl pseudo DASH)
- demux: fix crash with cue/ordered chapter files (#5027)
- demux: speed up cache seeking with a coarse index
- demux_lavf: always give libavformat the filename when probing
-  (helps with mp3 files)
- demux_mkv: add V_SNOW tag to mkv_video_tags
- dvb: Fix long channel switching: next/prev channel
- dvb: fixes for ATSC tuning
- lavc_conv: clamp timestamps to positive (#5047)
- macOS: fix bundle on macOS High Sierra (10.13) (#4926, #4866)
- mp_image: always copy color attributes on hw download (#4804)
- mp_image: select an explicit fallback for chroma location (#4804)
- msg: bump log level of --log-file to -v -v
- msg: reinterpret a bunch of message levels
- osc: fix rare stack overflow when changing visibility mode
- osdep/io: add android-related bullshit to fix files >2 GiB
- player/playloop.c: respect playback start time when using --loop-file
- player: allow seeking in cached parts of unseekable streams
- player: make track language matching case insensitive (#5272)
- player: use start timestamp for ab-looping if --ab-loop-a is absent
- player: when loading external file, always add all track types (#5132)
- restore-old-bindings.conf: add old macOS/Wayland AXIS bindings
- screenshot: create directories from template
- scripting: report dlerror() output
- sd_ass: accept RFC8081 font media types
- sd_ass: accept otc as fallback OpenType collection file extension
- stream_libarchive: work around various types of locale braindeath
-  (https://git.io/vbiFJ)
- subprocess-win: don't change the mouse cursor when creating processes
- video: add a hack to avoid missing subtitles with vf_sub (#5194)
- video: fix alpha handling (#4983)
- video: fix memory leaks (roughly 1 KB per decoded frame) with hwdec copy modes
- video: fix rotation and deinterlace auto filters
- video: properly pass through ICC data
- vo: add support for externally driven renderloop and make wayland use it
-  (partially fixes display-sync under wayland; disables rendering when window
-  is invisible)
- vo_gpu: change --tone-mapping-desaturate algorithm
- vo_gpu: enable 3DLUTs in dumb mode
- vo_gpu: fix gamma scale
- vo_gpu: fix mobius tone mapping compatibility to GLSL 120 (#5069)
- vo_gpu: fix video sometimes not being rerendered on equalizer change
- vo_gpu: kill off FBOTEX_FUZZY (#1814)
- vo_gpu: opengl: fix possible screenshot window crash (#4905)
- vo_gpu: opengl: use GLX_MESA_swap_control where available
- vo_gpu: reduce the --alpha=blend-tiles checkerboard intensity
- vo_gpu: win: remove exclusive-fullscreen detection hack
- vo_lavc: remove messy delayed subtitle rendering logic (#4689)
- wayland_common: implement output tracking, many cleanups and bugfixes
- Windows: skip window snapping if Windows handled it
- Windows: add more-POSIXy versions of open() and fstat() (#4711)
- ytdl_hook: don't prepend ytdl:// to non-youtube links in playlists (#5003)
+- HIDRemote: fix volume buttons on macOS 10.13 (#5721)
+- TOOLS/autoload: Fix broken "disabled" option
+- TOOLS/autoload: be more robust with slow directory listings (#5618)
+- ao_sdl: fix default buffer size
+- build: add static libraries to libmpv.pc
+- build: bump waf to 2.0.9 (fixes Python 3.7 compatibility)
+- build: manually add standard library search paths for linking (#5791)
+- common: round all integer times to milliseconds
+- demux, player: fix playback of sparse video streams (w/ still images)
+- demux: support for some kinds of timed metadata (e.g. ICY)
+- demux_mkv: add V_AV1 identifier for AV1
+- demux_mkv: support Meridian Lossless Packing in Matroska (#5923)
+- display additional metadata tags during video playback
+- drm_atomic: Fix memory leaks in drm_atomic_create
+- enable cache for FUSE filesystems on OpenBSD and FreeBSD
+- gpu: prefer 16bit floating point FBO formats to 16bit integer ones
+- hwdec_vaegl: Fix VAAPI EGL interop used with gpu-context=drm
+- input.conf: use exact value for [ binding, which does the inverse of ]
+- input: add a keybinding to toggle hardware decoding
+- input: minor additions to default key bindings (#973)
+- osc: fix accidentally skipping files when seeking with slider
+- player: fix strange behavior on edition switching
+- player: fix track autoselection of external files yet again
+- terminal-unix: stop trying to read when terminal disappears (#5842)
+- video: actually wait for last frame being rendered on EOF
+- video: fix --video-rotate in some cases
+- video: fix crash with vdpau when reinitializing rendering (#5447)
+- vo_gpu: fix anamorphic screenshots (#5619)
+- vo_vdpau: fix resizing and rotation problems


 This listing is not complete. Check DOCS/client-api-changes.rst for a history
 of changes to the client API, and DOCS/interface-changes.rst for a history
 of changes to other user-visible interfaces.

-A complete changelog can be seen by running `git log v0.27.0..v0.28.0`
+A complete changelog can be seen by running `git log v0.28.2..v0.29.0`
 in the git repository or by visiting either
-https://github.com/mpv-player/mpv/compare/v0.27.0...v0.28.0 or
-https://git.srsfckn.biz/mpv/log/?qt=range&q=v0.27.0..v0.28.0
-
+https://github.com/mpv-player/mpv/compare/v0.28.2...v0.29.0 or
+https://git.srsfckn.biz/mpv/log/?qt=range&q=v0.28.2..v0.29.0
@@ -24,8 +24,8 @@ def is_user_lib(objfile, libname):

 def otool(objfile):
    command = "otool -L %s | grep -e '\t' | awk '{ print $1 }'" % objfile
-    output  = subprocess.check_output(command, shell = True)
-    return filter(partial(is_user_lib, objfile), output.split())
+    output  = subprocess.check_output(command, shell = True, universal_newlines=True)
+    return set(filter(partial(is_user_lib, objfile), output.split()))

 def install_name_tool_change(old, new, objfile):
    subprocess.call(["install_name_tool", "-change", old, new, objfile])
@@ -35,7 +35,7 @@ def install_name_tool_id(name, objfile):

 def libraries(objfile, result = dict()):
    libs_list       = otool(objfile)
-    result[objfile] = set(libs_list)
+    result[objfile] = libs_list

    for lib in libs_list:
        if lib not in result:
@@ -80,15 +80,17 @@ def process_libraries(libs_dict, binary, processed = []):
            if p in libs_dict[src]:
                install_name_tool_change(p, lib_name(p), dst)

-    process_libraries(libs_dict, binary, ls)
+    return ls

 def main():
    binary = os.path.abspath(sys.argv[1])
    if not os.path.exists(lib_path(binary)):
        os.makedirs(lib_path(binary))
    libs = libraries(binary)
-    print(libs)
-    process_libraries(libs, binary)
+
+    libs_processed = process_libraries(libs, binary)
+    while libs_processed is not None:
+        libs_processed = process_libraries(libs, binary, libs_processed)

 if __name__ == "__main__":
    main()
@@ -7,7 +7,9 @@
 -- Add at most 5000 * 2 files when starting a file (before + after).
 MAXENTRIES = 5000

+local msg = require 'mp.msg'
 local options = require 'mp.options'
+local utils = require 'mp.utils'

 o = {
    disabled = false
@@ -25,8 +27,6 @@ EXTENSIONS = Set {
    'mp3', 'wav', 'ogm', 'flac', 'm4a', 'wma', 'ogg', 'opus',
 }

-mputils = require 'mp.utils'
-
 function add_files_at(index, files)
    index = index - 1
    local oldcount = mp.get_property_number("playlist-count", 1)
@@ -84,21 +84,35 @@ function alnumcomp(x, y)
    return #xt < #yt
 end

+local autoloaded = nil
+
 function find_and_add_entries()
    local path = mp.get_property("path", "")
-    local dir, filename = mputils.split_path(path)
-    if o.disabled or #dir == 0 then
+    local dir, filename = utils.split_path(path)
+    msg.trace(("dir: %s, filename: %s"):format(dir, filename))
+    if o.disabled then
+        msg.verbose("stopping: autoload disabled")
+        return
+    elseif #dir == 0 then
+        msg.verbose("stopping: not a local path")
        return
    end
+
    local pl_count = mp.get_property_number("playlist-count", 1)
-    if (pl_count > 1 and autoload == nil) or
+    -- check if this is a manually made playlist
+    if (pl_count > 1 and autoloaded == nil) or
       (pl_count == 1 and EXTENSIONS[string.lower(get_extension(filename))] == nil) then
        return
    else
-        autoload = true
+        autoloaded = true
    end

-    local files = mputils.readdir(dir, "files")
+    local pl = mp.get_property_native("playlist", {})
+    local pl_current = mp.get_property_number("playlist-pos-1", 1)
+    msg.trace(("playlist-pos-1: %s, playlist: %s"):format(pl_current,
+        utils.to_string(pl)))
+
+    local files = utils.readdir(dir, "files")
    if files == nil then
        return
    end
@@ -118,8 +132,6 @@ function find_and_add_entries()
        dir = ""
    end

-    local pl = mp.get_property_native("playlist", {})
-    local pl_current = mp.get_property_number("playlist-pos", 0) + 1
    -- Find the current pl entry (dir+"/"+filename) in the sorted dir list
    local current
    for i = 1, #files do
@@ -131,6 +143,7 @@ function find_and_add_entries()
    if current == nil then
        return
    end
+    msg.trace("current file position in files: "..current)

    local append = {[-1] = {}, [1] = {}}
    for direction = -1, 1, 2 do -- 2 iterations, with direction = -1 and +1
@@ -144,6 +157,7 @@ function find_and_add_entries()
            local filepath = dir .. file
            if pl_e then
                -- If there's a playlist entry, and it's the same file, stop.
+                msg.trace(pl_e.filename.." == "..filepath.." ?")
                if pl_e.filename == filepath then
                    break
                end
@@ -151,11 +165,11 @@ function find_and_add_entries()

            if direction == -1 then
                if pl_current == 1 then -- never add additional entries in the middle
-                    mp.msg.info("Prepending " .. file)
+                    msg.info("Prepending " .. file)
                    table.insert(append[-1], 1, filepath)
                end
            else
-                mp.msg.info("Adding " .. file)
+                msg.info("Adding " .. file)
                table.insert(append[1], filepath)
            end
        end
@@ -213,6 +213,7 @@ elements_matroska = (
            'Tag*, 7373, sub', (
                'Targets, 63c0, sub', (
                    'TargetTypeValue, 68ca, uint',
+                    'TargetType, 63ca, str',
                    'TargetTrackUID, 63c5, uint',
                    'TargetEditionUID, 63c9, uint',
                    'TargetChapterUID, 63c4, uint',
@@ -221,7 +222,8 @@ elements_matroska = (
                'SimpleTag*, 67c8, sub', (
                    'TagName, 45a3, str',
                    'TagLanguage, 447a, str',
-                    'TagString, 4487, str'
+                    'TagString, 4487, str',
+                    'TagDefault, 4484, uint',
                ),
            ),
        ),
@@ -29,11 +29,11 @@ my @opts = parse_main_opts('--list-options', '^ (\-\-[^\s\*]*)\*?\s*(.*)');

 die "Couldn't find any options" unless (@opts);

-my @ao = parse_opts('--ao=help', '^  ([^\s\:]*)\s*: (.*)');
-my @vo = parse_opts('--vo=help', '^  ([^\s\:]*)\s*: (.*)');
+my @ao = parse_opts('--ao=help', '^  ([^\s\:]*)\s*(.*)');
+my @vo = parse_opts('--vo=help', '^  ([^\s\:]*)\s*(.*)');

-my @af = parse_opts('--af=help', '^  ([^\s\:]*)\s*: (.*)');
-my @vf = parse_opts('--vf=help', '^  ([^\s\:]*)\s*: (.*)');
+my @af = parse_opts('--af=help', '^  ([^\s\:]*)\s*(.*)');
+my @vf = parse_opts('--vf=help', '^  ([^\s\:]*)\s*(.*)');

 my @protos = parse_opts('--list-protocols', '^ ([^\s]*)');

@@ -1 +1 @@
-0.28.0
+0.29.1
@@ -2,6 +2,7 @@ branches:
  only:
    - master
    - appveyor
+    - ci

 environment:
  global:
@@ -15,11 +16,11 @@ test: off

 install:
  # Update core packages
-  - C:\msys64\usr\bin\pacman -Syyuu --noconfirm
+  - C:\msys64\usr\bin\pacman -Syyuu --noconfirm --noprogressbar --ask=20
  # Update non-core packages
-  - C:\msys64\usr\bin\pacman -Suu --noconfirm
+  - C:\msys64\usr\bin\pacman -Suu --noconfirm --noprogressbar --ask=20
  # Install required MSYS2 packages
-  - C:\msys64\usr\bin\pacman -S --noconfirm --needed base-devel perl python pkg-config
+  - C:\msys64\usr\bin\pacman -S --noconfirm --noprogressbar --needed base-devel perl python pkg-config
  # Now MSYS2 is up to date, do the rest of the install from a bash script
  - C:\msys64\usr\bin\bash -lc "cd && exec \"$APPVEYOR_BUILD_FOLDER\"/TOOLS/appveyor-install.sh"
 build_script:
@@ -1,39 +0,0 @@
-#pragma once
-
-#include <stdbool.h>
-
-#include "chmap.h"
-
-struct mp_aconverter;
-struct mp_aframe;
-struct mpv_global;
-struct mp_log;
-
-struct mp_resample_opts {
-    int filter_size;
-    int phase_shift;
-    int linear;
-    double cutoff;
-    int normalize;
-    int allow_passthrough;
-    char **avopts;
-};
-
-#define MP_RESAMPLE_OPTS_DEF {  \
-    .filter_size = 16,          \
-    .cutoff      = 0.0,         \
-    .phase_shift = 10,          \
-    .normalize   = -1,          \
-    }
-
-struct mp_aconverter *mp_aconverter_create(struct mpv_global *global,
-                                           struct mp_log *log,
-                                           const struct mp_resample_opts *opts);
-bool mp_aconverter_reconfig(struct mp_aconverter *p,
-                    int in_rate, int in_format, struct mp_chmap in_channels,
-                    int out_rate, int out_format, struct mp_chmap out_channels);
-void mp_aconverter_flush(struct mp_aconverter *p);
-void mp_aconverter_set_speed(struct mp_aconverter *p, double speed);
-bool mp_aconverter_write_input(struct mp_aconverter *p, struct mp_aframe *in);
-struct mp_aframe *mp_aconverter_read_output(struct mp_aconverter *p, bool *eof);
-double mp_aconverter_get_latency(struct mp_aconverter *p);
@@ -32,6 +32,11 @@ struct mp_aframe {
    // We support spdif formats, which are allocated as AV_SAMPLE_FMT_S16.
    int format;
    double pts;
+    double speed;
+};
+
+struct avframe_opaque {
+    double speed;
 };

 static void free_frame(void *ptr)
@@ -43,11 +48,11 @@ static void free_frame(void *ptr)
 struct mp_aframe *mp_aframe_create(void)
 {
    struct mp_aframe *frame = talloc_zero(NULL, struct mp_aframe);
-    frame->pts = MP_NOPTS_VALUE;
    frame->av_frame = av_frame_alloc();
    if (!frame->av_frame)
        abort();
    talloc_set_destructor(frame, free_frame);
+    mp_aframe_reset(frame);
    return frame;
 }

@@ -61,6 +66,7 @@ struct mp_aframe *mp_aframe_new_ref(struct mp_aframe *frame)
    dst->chmap = frame->chmap;
    dst->format = frame->format;
    dst->pts = frame->pts;
+    dst->speed = frame->speed;

    if (mp_aframe_is_allocated(frame)) {
        if (av_frame_ref(dst->av_frame, frame->av_frame) < 0)
@@ -80,6 +86,7 @@ void mp_aframe_reset(struct mp_aframe *frame)
    frame->chmap.num = 0;
    frame->format = 0;
    frame->pts = MP_NOPTS_VALUE;
+    frame->speed = 1.0;
 }

 // Remove all actual audio data and leave only the metadata.
@@ -120,6 +127,11 @@ struct mp_aframe *mp_aframe_from_avframe(struct AVFrame *av_frame)
        mp_chmap_from_channels(&frame->chmap, av_frame->channels);
 #endif

+    if (av_frame->opaque_ref) {
+        struct avframe_opaque *op = (void *)av_frame->opaque_ref->data;
+        frame->speed = op->speed;
+    }
+
    return frame;
 }

@@ -137,6 +149,16 @@ struct AVFrame *mp_aframe_to_avframe(struct mp_aframe *frame)
    if (!mp_chmap_is_lavc(&frame->chmap))
        return NULL;

+    if (!frame->av_frame->opaque_ref && frame->speed != 1.0) {
+        frame->av_frame->opaque_ref =
+            av_buffer_alloc(sizeof(struct avframe_opaque));
+        if (!frame->av_frame->opaque_ref)
+            return NULL;
+
+        struct avframe_opaque *op = (void *)frame->av_frame->opaque_ref->data;
+        op->speed = frame->speed;
+    }
+
    return av_frame_clone(frame->av_frame);
 }

@@ -183,6 +205,7 @@ void mp_aframe_config_copy(struct mp_aframe *dst, struct mp_aframe *src)
 void mp_aframe_copy_attributes(struct mp_aframe *dst, struct mp_aframe *src)
 {
    dst->pts = src->pts;
+    dst->speed = src->speed;

    int rate = dst->av_frame->sample_rate;

@@ -316,6 +339,37 @@ void mp_aframe_set_pts(struct mp_aframe *frame, double pts)
    frame->pts = pts;
 }

+// Set a speed factor. This is multiplied with the sample rate to get the
+// "effective" samplerate (mp_aframe_get_effective_rate()), which will be used
+// to do PTS calculations. If speed!=1.0, the PTS values always refer to the
+// original PTS (before changing speed), and if you want reasonably continuous
+// PTS between frames, you need to use the effective samplerate.
+void mp_aframe_set_speed(struct mp_aframe *frame, double factor)
+{
+    frame->speed = factor;
+}
+
+// Adjust current speed factor.
+void mp_aframe_mul_speed(struct mp_aframe *frame, double factor)
+{
+    frame->speed *= factor;
+}
+
+double mp_aframe_get_speed(struct mp_aframe *frame)
+{
+    return frame->speed;
+}
+
+// Matters for speed changed frames (such as a frame which has been resampled
+// to play at a different speed).
+// Return the sample rate at which the frame would have to be played to result
+// in the same duration as the original frame before the speed change.
+// This is used for A/V sync.
+double mp_aframe_get_effective_rate(struct mp_aframe *frame)
+{
+    return mp_aframe_get_rate(frame) / frame->speed;
+}
+
 // Return number of data pointers.
 int mp_aframe_get_planes(struct mp_aframe *frame)
 {
@@ -339,6 +393,18 @@ int mp_aframe_get_total_plane_samples(struct mp_aframe *frame)
            ? 1 : mp_aframe_get_channels(frame));
 }

+char *mp_aframe_format_str_buf(char *buf, size_t buf_size, struct mp_aframe *fmt)
+{
+    char ch[128];
+    mp_chmap_to_str_buf(ch, sizeof(ch), &fmt->chmap);
+    char *hr_ch = mp_chmap_to_str_hr(&fmt->chmap);
+    if (strcmp(hr_ch, ch) != 0)
+        mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch);
+    snprintf(buf, buf_size, "%dHz %s %dch %s", fmt->av_frame->sample_rate,
+             ch, fmt->chmap.num, af_fmt_to_str(fmt->format));
+    return buf;
+}
+
 // Set data to the audio after the given number of samples (i.e. slice it).
 void mp_aframe_skip_samples(struct mp_aframe *f, int samples)
 {
@@ -352,35 +418,38 @@ void mp_aframe_skip_samples(struct mp_aframe *f, int samples)
    f->av_frame->nb_samples -= samples;

    if (f->pts != MP_NOPTS_VALUE)
-        f->pts += samples / (double)mp_aframe_get_rate(f);
+        f->pts += samples / mp_aframe_get_effective_rate(f);
 }

 // Return the timestamp of the sample just after the end of this frame.
 double mp_aframe_end_pts(struct mp_aframe *f)
 {
-    int rate = mp_aframe_get_rate(f);
-    if (f->pts == MP_NOPTS_VALUE || rate < 1)
+    double rate = mp_aframe_get_effective_rate(f);
+    if (f->pts == MP_NOPTS_VALUE || rate <= 0)
        return MP_NOPTS_VALUE;
-    return f->pts + f->av_frame->nb_samples / (double)rate;
+    return f->pts + f->av_frame->nb_samples / rate;
 }

 // Return the duration in seconds of the frame (0 if invalid).
 double mp_aframe_duration(struct mp_aframe *f)
 {
-    int rate = mp_aframe_get_rate(f);
-    if (rate < 1)
+    double rate = mp_aframe_get_effective_rate(f);
+    if (rate <= 0)
        return 0;
-    return f->av_frame->nb_samples / (double)rate;
+    return f->av_frame->nb_samples / rate;
 }

 // Clip the given frame to the given timestamp range. Adjusts the frame size
 // and timestamp.
+// Refuses to change spdif frames.
 void mp_aframe_clip_timestamps(struct mp_aframe *f, double start, double end)
 {
    double f_end = mp_aframe_end_pts(f);
-    int rate = mp_aframe_get_rate(f);
+    double rate = mp_aframe_get_effective_rate(f);
    if (f_end == MP_NOPTS_VALUE)
        return;
+    if (af_fmt_is_spdif(mp_aframe_get_format(f)))
+        return;
    if (end != MP_NOPTS_VALUE) {
        if (f_end >= end) {
            if (f->pts >= end) {
@@ -405,6 +474,52 @@ void mp_aframe_clip_timestamps(struct mp_aframe *f, double start, double end)
    }
 }

+bool mp_aframe_copy_samples(struct mp_aframe *dst, int dst_offset,
+                            struct mp_aframe *src, int src_offset,
+                            int samples)
+{
+    if (!mp_aframe_config_equals(dst, src))
+        return false;
+
+    if (mp_aframe_get_size(dst) < dst_offset + samples ||
+        mp_aframe_get_size(src) < src_offset + samples)
+        return false;
+
+    uint8_t **s = mp_aframe_get_data_ro(src);
+    uint8_t **d = mp_aframe_get_data_rw(dst);
+    if (!s || !d)
+        return false;
+
+    int planes = mp_aframe_get_planes(dst);
+    size_t sstride = mp_aframe_get_sstride(dst);
+
+    for (int n = 0; n < planes; n++) {
+        memcpy(d[n] + dst_offset * sstride, s[n] + src_offset * sstride,
+               samples * sstride);
+    }
+
+    return true;
+}
+
+bool mp_aframe_set_silence(struct mp_aframe *f, int offset, int samples)
+{
+    if (mp_aframe_get_size(f) < offset + samples)
+        return false;
+
+    int format = mp_aframe_get_format(f);
+    uint8_t **d = mp_aframe_get_data_rw(f);
+    if (!d)
+        return false;
+
+    int planes = mp_aframe_get_planes(f);
+    size_t sstride = mp_aframe_get_sstride(f);
+
+    for (int n = 0; n < planes; n++)
+        af_fill_silence(d[n] + offset * sstride, samples * sstride, format);
+
+    return true;
+}
+
 struct mp_aframe_pool {
    AVBufferPool *avpool;
    int element_size;
@@ -1,6 +1,8 @@
 #pragma once

 #include <stdint.h>
+#include <stddef.h>
+#include <stdbool.h>

 struct mp_aframe;
 struct AVFrame;
@@ -34,21 +36,32 @@ int mp_aframe_get_channels(struct mp_aframe *frame);
 int mp_aframe_get_rate(struct mp_aframe *frame);
 int mp_aframe_get_size(struct mp_aframe *frame);
 double mp_aframe_get_pts(struct mp_aframe *frame);
+double mp_aframe_get_speed(struct mp_aframe *frame);
+double mp_aframe_get_effective_rate(struct mp_aframe *frame);

 bool mp_aframe_set_format(struct mp_aframe *frame, int format);
 bool mp_aframe_set_chmap(struct mp_aframe *frame, struct mp_chmap *in);
 bool mp_aframe_set_rate(struct mp_aframe *frame, int rate);
 bool mp_aframe_set_size(struct mp_aframe *frame, int samples);
 void mp_aframe_set_pts(struct mp_aframe *frame, double pts);
+void mp_aframe_set_speed(struct mp_aframe *frame, double factor);
+void mp_aframe_mul_speed(struct mp_aframe *frame, double factor);

 int mp_aframe_get_planes(struct mp_aframe *frame);
 int mp_aframe_get_total_plane_samples(struct mp_aframe *frame);
 size_t mp_aframe_get_sstride(struct mp_aframe *frame);

+char *mp_aframe_format_str_buf(char *buf, size_t buf_size, struct mp_aframe *fmt);
+#define mp_aframe_format_str(fmt) mp_aframe_format_str_buf((char[32]){0}, 32, (fmt))
+
 void mp_aframe_skip_samples(struct mp_aframe *f, int samples);
 double mp_aframe_end_pts(struct mp_aframe *f);
 double mp_aframe_duration(struct mp_aframe *f);
 void mp_aframe_clip_timestamps(struct mp_aframe *f, double start, double end);
+bool mp_aframe_copy_samples(struct mp_aframe *dst, int dst_offset,
+                            struct mp_aframe *src, int src_offset,
+                            int samples);
+bool mp_aframe_set_silence(struct mp_aframe *f, int offset, int samples);

 struct mp_aframe_pool;
 struct mp_aframe_pool *mp_aframe_pool_create(void *ta_parent);
@@ -1,625 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdint.h>
-#include <limits.h>
-#include <stdlib.h>
-#include <assert.h>
-
-#include <libavutil/buffer.h>
-#include <libavutil/frame.h>
-#include <libavutil/mem.h>
-#include <libavutil/version.h>
-
-#include "mpv_talloc.h"
-#include "common/common.h"
-#include "fmt-conversion.h"
-#include "audio.h"
-#include "aframe.h"
-
-static void update_redundant_info(struct mp_audio *mpa)
-{
-    assert(mp_chmap_is_empty(&mpa->channels) ||
-           mp_chmap_is_valid(&mpa->channels));
-    mpa->nch = mpa->channels.num;
-    mpa->bps = af_fmt_to_bytes(mpa->format);
-    if (af_fmt_is_planar(mpa->format)) {
-        mpa->spf = 1;
-        mpa->num_planes = mpa->nch;
-        mpa->sstride = mpa->bps;
-    } else {
-        mpa->spf = mpa->nch;
-        mpa->num_planes = 1;
-        mpa->sstride = mpa->bps * mpa->nch;
-    }
-}
-
-void mp_audio_set_format(struct mp_audio *mpa, int format)
-{
-    mpa->format = format;
-    update_redundant_info(mpa);
-}
-
-void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels)
-{
-    mp_chmap_from_channels(&mpa->channels, num_channels);
-    update_redundant_info(mpa);
-}
-
-void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap)
-{
-    mpa->channels = *chmap;
-    update_redundant_info(mpa);
-}
-
-void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src)
-{
-    dst->format = src->format;
-    dst->channels = src->channels;
-    dst->rate = src->rate;
-    update_redundant_info(dst);
-}
-
-bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b)
-{
-    return a->format == b->format && a->rate == b->rate &&
-           mp_chmap_equals(&a->channels, &b->channels);
-}
-
-bool mp_audio_config_valid(const struct mp_audio *mpa)
-{
-    return mp_chmap_is_valid(&mpa->channels) && af_fmt_is_valid(mpa->format)
-        && mpa->rate >= 1 && mpa->rate < 10000000;
-}
-
-char *mp_audio_config_to_str_buf(char *buf, size_t buf_sz, struct mp_audio *mpa)
-{
-    char ch[128];
-    mp_chmap_to_str_buf(ch, sizeof(ch), &mpa->channels);
-    char *hr_ch = mp_chmap_to_str_hr(&mpa->channels);
-    if (strcmp(hr_ch, ch) != 0)
-        mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch);
-    snprintf(buf, buf_sz, "%dHz %s %dch %s", mpa->rate,
-             ch, mpa->channels.num, af_fmt_to_str(mpa->format));
-    return buf;
-}
-
-void mp_audio_force_interleaved_format(struct mp_audio *mpa)
-{
-    if (af_fmt_is_planar(mpa->format))
-        mp_audio_set_format(mpa, af_fmt_from_planar(mpa->format));
-}
-
-// Return used size of a plane. (The size is the same for all planes.)
-int mp_audio_psize(struct mp_audio *mpa)
-{
-    return mpa->samples * mpa->sstride;
-}
-
-void mp_audio_set_null_data(struct mp_audio *mpa)
-{
-    for (int n = 0; n < MP_NUM_CHANNELS; n++) {
-        mpa->planes[n] = NULL;
-        mpa->allocated[n] = NULL;
-    }
-    mpa->samples = 0;
-}
-
-static int get_plane_size(const struct mp_audio *mpa, int samples)
-{
-    if (samples < 0 || !mp_audio_config_valid(mpa))
-        return -1;
-    if (samples >= INT_MAX / mpa->sstride)
-        return -1;
-    return MPMAX(samples * mpa->sstride, 1);
-}
-
-static void mp_audio_destructor(void *ptr)
-{
-    struct mp_audio *mpa = ptr;
-    for (int n = 0; n < MP_NUM_CHANNELS; n++)
-        av_buffer_unref(&mpa->allocated[n]);
-}
-
-/* Reallocate the data stored in mpa->planes[n] so that enough samples are
- * available on every plane. The previous data is kept (for the smallest
- * common number of samples before/after resize).
- *
- * This also makes sure the resulting buffer is writable (even in the case
- * the buffer has the correct size).
- *
- * mpa->samples is not set or used.
- *
- * This function is flexible enough to handle format and channel layout
- * changes. In these cases, all planes are reallocated as needed. Unused
- * planes are freed.
- *
- * mp_audio_realloc(mpa, 0) will still yield non-NULL for mpa->data[n].
- *
- * Allocated data is implicitly freed on talloc_free(mpa).
- */
-void mp_audio_realloc(struct mp_audio *mpa, int samples)
-{
-    int size = get_plane_size(mpa, samples);
-    if (size < 0)
-        abort(); // oom or invalid parameters
-    if (!mp_audio_is_writeable(mpa)) {
-        for (int n = 0; n < MP_NUM_CHANNELS; n++) {
-            av_buffer_unref(&mpa->allocated[n]);
-            mpa->planes[n] = NULL;
-        }
-    }
-    for (int n = 0; n < mpa->num_planes; n++) {
-        if (!mpa->allocated[n] || size != mpa->allocated[n]->size) {
-            if (av_buffer_realloc(&mpa->allocated[n], size) < 0)
-                abort(); // OOM
-        }
-        mpa->planes[n] = mpa->allocated[n]->data;
-    }
-    for (int n = mpa->num_planes; n < MP_NUM_CHANNELS; n++) {
-        av_buffer_unref(&mpa->allocated[n]);
-        mpa->planes[n] = NULL;
-    }
-    talloc_set_destructor(mpa, mp_audio_destructor);
-}
-
-// Like mp_audio_realloc(), but only reallocate if the audio grows in size.
-// If the buffer is reallocated, also preallocate.
-void mp_audio_realloc_min(struct mp_audio *mpa, int samples)
-{
-    if (samples > mp_audio_get_allocated_size(mpa) || !mp_audio_is_writeable(mpa)) {
-        size_t alloc = ta_calc_prealloc_elems(samples);
-        if (alloc > INT_MAX)
-            abort(); // oom
-        mp_audio_realloc(mpa, alloc);
-    }
-}
-
-/* Get the size allocated for the data, in number of samples. If the allocated
- * size isn't on sample boundaries (e.g. after format changes), the returned
- * sample number is a rounded down value.
- *
- * Note that this only works in situations where mp_audio_realloc() also works!
- */
-int mp_audio_get_allocated_size(struct mp_audio *mpa)
-{
-    int size = 0;
-    for (int n = 0; n < mpa->num_planes; n++) {
-        for (int i = 0; i < MP_NUM_CHANNELS && mpa->allocated[i]; i++) {
-            uint8_t *start = mpa->allocated[i]->data;
-            uint8_t *end = start + mpa->allocated[i]->size;
-            uint8_t *plane = mpa->planes[n];
-            if (plane >= start && plane < end) {
-                int s = MPMIN((end - plane) / mpa->sstride, INT_MAX);
-                size = n == 0 ? s : MPMIN(size, s);
-                goto next;
-            }
-        }
-        return 0; // plane is not covered by any buffer
-    next: ;
-    }
-    return size;
-}
-
-// Clear the samples [start, start + length) with silence.
-void mp_audio_fill_silence(struct mp_audio *mpa, int start, int length)
-{
-    assert(start >= 0 && length >= 0 && start + length <= mpa->samples);
-    int offset = start * mpa->sstride;
-    int size = length * mpa->sstride;
-    for (int n = 0; n < mpa->num_planes; n++) {
-        if (n > 0 && mpa->planes[n] == mpa->planes[0])
-            continue; // silly optimization for special cases
-        af_fill_silence((char *)mpa->planes[n] + offset, size, mpa->format);
-    }
-}
-
-// All integer parameters are in samples.
-// dst and src can overlap.
-void mp_audio_copy(struct mp_audio *dst, int dst_offset,
-                   struct mp_audio *src, int src_offset, int length)
-{
-    assert(mp_audio_config_equals(dst, src));
-    assert(length >= 0);
-    assert(dst_offset >= 0 && dst_offset + length <= dst->samples);
-    assert(src_offset >= 0 && src_offset + length <= src->samples);
-
-    for (int n = 0; n < dst->num_planes; n++) {
-        memmove((char *)dst->planes[n] + dst_offset * dst->sstride,
-                (char *)src->planes[n] + src_offset * src->sstride,
-                length * dst->sstride);
-    }
-}
-
-// Copy fields that describe characteristics of the audio frame, but which are
-// not part of the core format (format/channels/rate), and not part of the
-// data (samples).
-void mp_audio_copy_attributes(struct mp_audio *dst, struct mp_audio *src)
-{
-    // nothing yet
-}
-
-// Set data to the audio after the given number of samples (i.e. slice it).
-void mp_audio_skip_samples(struct mp_audio *data, int samples)
-{
-    assert(samples >= 0 && samples <= data->samples);
-
-    for (int n = 0; n < data->num_planes; n++)
-        data->planes[n] = (uint8_t *)data->planes[n] + samples * data->sstride;
-
-    data->samples -= samples;
-
-    if (data->pts != MP_NOPTS_VALUE)
-        data->pts += samples / (double)data->rate;
-}
-
-// Return the timestamp of the sample just after the end of this frame.
-double mp_audio_end_pts(struct mp_audio *f)
-{
-    if (f->pts == MP_NOPTS_VALUE || f->rate < 1)
-        return MP_NOPTS_VALUE;
-    return f->pts + f->samples / (double)f->rate;
-}
-
-// Clip the given frame to the given timestamp range. Adjusts the frame size
-// and timestamp.
-void mp_audio_clip_timestamps(struct mp_audio *f, double start, double end)
-{
-    double f_end = mp_audio_end_pts(f);
-    if (f_end == MP_NOPTS_VALUE)
-        return;
-    if (end != MP_NOPTS_VALUE) {
-        if (f_end >= end) {
-            if (f->pts >= end) {
-                f->samples = 0;
-            } else {
-                int new = (end - f->pts) * f->rate;
-                f->samples = MPCLAMP(new, 0, f->samples);
-            }
-        }
-    }
-    if (start != MP_NOPTS_VALUE) {
-        if (f->pts < start) {
-            if (f_end <= start) {
-                f->samples = 0;
-                f->pts = f_end;
-            } else {
-                int skip = (start - f->pts) * f->rate;
-                skip = MPCLAMP(skip, 0, f->samples);
-                mp_audio_skip_samples(f, skip);
-            }
-        }
-    }
-}
-
-
-// Return false if the frame data is shared, true otherwise.
-// Will return true for non-refcounted frames.
-bool mp_audio_is_writeable(struct mp_audio *data)
-{
-    bool ok = true;
-    for (int n = 0; n < MP_NUM_CHANNELS; n++) {
-        if (data->allocated[n])
-            ok &= av_buffer_is_writable(data->allocated[n]);
-    }
-    return ok;
-}
-
-static void mp_audio_steal_data(struct mp_audio *dst, struct mp_audio *src)
-{
-    talloc_set_destructor(dst, mp_audio_destructor);
-    mp_audio_destructor(dst);
-    *dst = *src;
-    talloc_set_destructor(src, NULL);
-    talloc_free(src);
-}
-
-// Make sure the frame owns the audio data, and if not, copy the data.
-// Return negative value on failure (which means it can't be made writeable).
-// Non-refcounted frames are always considered writeable.
-int mp_audio_make_writeable(struct mp_audio *data)
-{
-    if (!mp_audio_is_writeable(data)) {
-        struct mp_audio *new = talloc(NULL, struct mp_audio);
-        *new = *data;
-        mp_audio_set_null_data(new); // use format only
-        mp_audio_realloc(new, data->samples);
-        new->samples = data->samples;
-        mp_audio_copy(new, 0, data, 0, data->samples);
-        mp_audio_steal_data(data, new);
-    }
-    return 0;
-}
-
-struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe)
-{
-    AVFrame *tmp = NULL;
-    struct mp_audio *new = talloc_zero(NULL, struct mp_audio);
-    talloc_set_destructor(new, mp_audio_destructor);
-
-    mp_audio_set_format(new, af_from_avformat(avframe->format));
-
-    struct mp_chmap lavc_chmap;
-    mp_chmap_from_lavc(&lavc_chmap, avframe->channel_layout);
-
-#if LIBAVUTIL_VERSION_MICRO >= 100
-    // FFmpeg being stupid POS again
-    if (lavc_chmap.num != avframe->channels)
-        mp_chmap_from_channels(&lavc_chmap, avframe->channels);
-#endif
-
-    new->rate = avframe->sample_rate;
-
-    mp_audio_set_channels(new, &lavc_chmap);
-
-    // Force refcounted frame.
-    if (!avframe->buf[0]) {
-        tmp = av_frame_alloc();
-        if (!tmp)
-            goto fail;
-        if (av_frame_ref(tmp, avframe) < 0)
-            goto fail;
-        avframe = tmp;
-    }
-
-    // If we can't handle the format (e.g. too many channels), bail out.
-    if (!mp_audio_config_valid(new))
-        goto fail;
-
-    for (int n = 0; n < AV_NUM_DATA_POINTERS + avframe->nb_extended_buf; n++) {
-        AVBufferRef *buf = n < AV_NUM_DATA_POINTERS ? avframe->buf[n]
-                            : avframe->extended_buf[n - AV_NUM_DATA_POINTERS];
-        if (!buf)
-            break;
-        if (n >= MP_NUM_CHANNELS)
-            goto fail;
-        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->extended_data[n];
-    new->samples = avframe->nb_samples;
-
-    return new;
-
-fail:
-    talloc_free(new);
-    av_frame_free(&tmp);
-    return NULL;
-}
-
-struct mp_audio *mp_audio_from_aframe(struct mp_aframe *aframe)
-{
-    if (!aframe)
-        return NULL;
-
-    struct AVFrame *av = mp_aframe_get_raw_avframe(aframe);
-    struct mp_audio *res = mp_audio_from_avframe(av);
-    if (!res)
-        return NULL;
-    struct mp_chmap chmap = {0};
-    mp_aframe_get_chmap(aframe, &chmap);
-    mp_audio_set_channels(res, &chmap);
-    mp_audio_set_format(res, mp_aframe_get_format(aframe));
-    res->pts = mp_aframe_get_pts(aframe);
-    return res;
-}
-
-void mp_audio_config_from_aframe(struct mp_audio *dst, struct mp_aframe *src)
-{
-    *dst = (struct mp_audio){0};
-    struct mp_chmap chmap = {0};
-    mp_aframe_get_chmap(src, &chmap);
-    mp_audio_set_channels(dst, &chmap);
-    mp_audio_set_format(dst, mp_aframe_get_format(src));
-    dst->rate = mp_aframe_get_rate(src);
-}
-
-struct mp_aframe *mp_audio_to_aframe(struct mp_audio *mpa)
-{
-    if (!mpa)
-        return NULL;
-
-    struct mp_aframe *aframe = mp_aframe_create();
-    struct AVFrame *av = mp_aframe_get_raw_avframe(aframe);
-    mp_aframe_set_format(aframe, mpa->format);
-    mp_aframe_set_chmap(aframe, &mpa->channels);
-    mp_aframe_set_rate(aframe, mpa->rate);
-
-    // bullshit it into ffmpeg-compatible parameters
-    struct mp_audio mpb = *mpa;
-    struct mp_chmap chmap;
-    mp_chmap_set_unknown(&chmap, mpb.channels.num);
-    mp_audio_set_channels(&mpb, &chmap);
-    if (af_fmt_is_spdif(mpb.format))
-        mp_audio_set_format(&mpb, AF_FORMAT_S16);
-
-    // put the reference into av, which magically puts it into aframe
-    // aframe keeps its parameters, so the bullshit doesn't matter
-    if (mp_audio_to_avframe(&mpb, av) < 0) {
-        talloc_free(aframe);
-        return NULL;
-    }
-    return aframe;
-}
-
-int mp_audio_to_avframe(struct mp_audio *frame, struct AVFrame *avframe)
-{
-    av_frame_unref(avframe);
-
-    avframe->nb_samples = frame->samples;
-    avframe->format = af_to_avformat(frame->format);
-    if (avframe->format == AV_SAMPLE_FMT_NONE)
-        goto fail;
-
-    avframe->channel_layout = mp_chmap_to_lavc(&frame->channels);
-    if (!avframe->channel_layout)
-        goto fail;
-#if LIBAVUTIL_VERSION_MICRO >= 100
-    // FFmpeg being a stupid POS again
-    avframe->channels = frame->channels.num;
-#endif
-    avframe->sample_rate = frame->rate;
-
-    if (frame->num_planes > AV_NUM_DATA_POINTERS) {
-        avframe->extended_data =
-            av_mallocz_array(frame->num_planes, sizeof(avframe->extended_data[0]));
-        int extbufs = frame->num_planes - AV_NUM_DATA_POINTERS;
-        avframe->extended_buf =
-            av_mallocz_array(extbufs, sizeof(avframe->extended_buf[0]));
-        if (!avframe->extended_data || !avframe->extended_buf)
-            goto fail;
-        avframe->nb_extended_buf = extbufs;
-    }
-
-    for (int p = 0; p < frame->num_planes; p++)
-        avframe->extended_data[p] = frame->planes[p];
-    avframe->linesize[0] = frame->samples * frame->sstride;
-
-    for (int p = 0; p < AV_NUM_DATA_POINTERS; p++)
-        avframe->data[p] = avframe->extended_data[p];
-
-    for (int p = 0; p < frame->num_planes; p++) {
-        if (!frame->allocated[p])
-            break;
-        AVBufferRef *nref = av_buffer_ref(frame->allocated[p]);
-        if (!nref)
-            goto fail;
-        if (p < AV_NUM_DATA_POINTERS) {
-            avframe->buf[p] = nref;
-        } else {
-            avframe->extended_buf[p - AV_NUM_DATA_POINTERS] = nref;
-        }
-    }
-
-    // Force refcounted frame.
-    if (!avframe->buf[0]) {
-        AVFrame *tmp = av_frame_alloc();
-        if (!tmp)
-            goto fail;
-        if (av_frame_ref(tmp, avframe) < 0)
-            goto fail;
-        av_frame_free(&avframe);
-        avframe = tmp;
-    }
-
-    return 0;
-
-fail:
-    av_frame_unref(avframe);
-    return -1;
-}
-
-// Returns NULL on failure. The input is always unreffed.
-struct AVFrame *mp_audio_to_avframe_and_unref(struct mp_audio *frame)
-{
-    struct AVFrame *avframe = av_frame_alloc();
-    if (!avframe)
-        goto fail;
-
-    if (mp_audio_to_avframe(frame, avframe) < 0)
-        goto fail;
-
-    talloc_free(frame);
-    return avframe;
-
-fail:
-    av_frame_free(&avframe);
-    talloc_free(frame);
-    return NULL;
-}
-
-struct mp_audio_pool {
-    AVBufferPool *avpool;
-    int element_size;
-};
-
-struct mp_audio_pool *mp_audio_pool_create(void *ta_parent)
-{
-    return talloc_zero(ta_parent, struct mp_audio_pool);
-}
-
-static void mp_audio_pool_destructor(void *p)
-{
-    struct mp_audio_pool *pool = p;
-    av_buffer_pool_uninit(&pool->avpool);
-}
-
-// Allocate data using the given format and number of samples.
-// Returns NULL on error.
-struct mp_audio *mp_audio_pool_get(struct mp_audio_pool *pool,
-                                   const struct mp_audio *fmt, int samples)
-{
-    int size = get_plane_size(fmt, samples);
-    if (size < 0)
-        return NULL;
-    if (!pool->avpool || size > pool->element_size) {
-        size_t alloc = ta_calc_prealloc_elems(size);
-        if (alloc >= INT_MAX)
-            return NULL;
-        av_buffer_pool_uninit(&pool->avpool);
-        pool->element_size = alloc;
-        pool->avpool = av_buffer_pool_init(pool->element_size, NULL);
-        if (!pool->avpool)
-            return NULL;
-        talloc_set_destructor(pool, mp_audio_pool_destructor);
-    }
-    struct mp_audio *new = talloc_ptrtype(NULL, new);
-    talloc_set_destructor(new, mp_audio_destructor);
-    *new = *fmt;
-    mp_audio_set_null_data(new);
-    new->samples = samples;
-    for (int n = 0; n < new->num_planes; n++) {
-        new->allocated[n] = av_buffer_pool_get(pool->avpool);
-        if (!new->allocated[n]) {
-            talloc_free(new);
-            return NULL;
-        }
-        new->planes[n] = new->allocated[n]->data;
-    }
-    return new;
-}
-
-// Return a copy of the given frame.
-// Returns NULL on error.
-struct mp_audio *mp_audio_pool_new_copy(struct mp_audio_pool *pool,
-                                        struct mp_audio *frame)
-{
-    struct mp_audio *new = mp_audio_pool_get(pool, frame, frame->samples);
-    if (new) {
-        mp_audio_copy(new, 0, frame, 0, new->samples);
-        mp_audio_copy_attributes(new, frame);
-    }
-    return new;
-}
-
-// Exactly like mp_audio_make_writeable(), but get the data from the pool.
-int mp_audio_pool_make_writeable(struct mp_audio_pool *pool,
-                                 struct mp_audio *data)
-{
-    if (mp_audio_is_writeable(data))
-        return 0;
-    struct mp_audio *new = mp_audio_pool_get(pool, data, data->samples);
-    if (!new)
-        return -1;
-    mp_audio_copy(new, 0, data, 0, data->samples);
-    mp_audio_copy_attributes(new, data);
-    mp_audio_steal_data(data, new);
-    return 0;
-}
@@ -1,102 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#ifndef MP_AUDIO_H
-#define MP_AUDIO_H
-
-#include "format.h"
-#include "chmap.h"
-
-// Audio data chunk
-struct mp_audio {
-    int samples;        // number of samples in data (per channel)
-    void *planes[MP_NUM_CHANNELS]; // data buffer (one per plane)
-    int rate;           // sample rate
-    struct mp_chmap channels; // channel layout, use mp_audio_set_*() to set
-    int format; // format (AF_FORMAT_...), use mp_audio_set_format() to set
-    // Redundant fields, for convenience
-    int sstride;        // distance between 2 samples in bytes on a plane
-                        //  interleaved: bps * nch
-                        //  planar:      bps
-    int nch;            // number of channels (redundant with chmap)
-    int spf;            // sub-samples per sample on each plane
-    int num_planes;     // number of planes
-    int bps;            // size of sub-samples (af_fmt_to_bytes(format))
-
-    double pts;         // currently invalid within the filter chain
-
-    // --- private
-    // These do not necessarily map directly to planes[]. They can have
-    // different order or count. There shouldn't be more buffers than planes.
-    // If allocated[n] is NULL, allocated[n+1] must also be NULL.
-    struct AVBufferRef *allocated[MP_NUM_CHANNELS];
-};
-
-void mp_audio_set_format(struct mp_audio *mpa, int format);
-void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels);
-void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap);
-void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src);
-bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b);
-bool mp_audio_config_valid(const struct mp_audio *mpa);
-
-char *mp_audio_config_to_str_buf(char *buf, size_t buf_sz, struct mp_audio *mpa);
-#define mp_audio_config_to_str(m) mp_audio_config_to_str_buf((char[64]){0}, 64, (m))
-
-void mp_audio_force_interleaved_format(struct mp_audio *mpa);
-
-int mp_audio_psize(struct mp_audio *mpa);
-
-void mp_audio_set_null_data(struct mp_audio *mpa);
-
-void mp_audio_realloc(struct mp_audio *mpa, int samples);
-void mp_audio_realloc_min(struct mp_audio *mpa, int samples);
-int mp_audio_get_allocated_size(struct mp_audio *mpa);
-
-void mp_audio_fill_silence(struct mp_audio *mpa, int start, int length);
-
-void mp_audio_copy(struct mp_audio *dst, int dst_offset,
-                   struct mp_audio *src, int src_offset, int length);
-void mp_audio_copy_attributes(struct mp_audio *dst, struct mp_audio *src);
-void mp_audio_skip_samples(struct mp_audio *data, int samples);
-void mp_audio_clip_timestamps(struct mp_audio *f, double start, double end);
-double mp_audio_end_pts(struct mp_audio *data);
-
-bool mp_audio_is_writeable(struct mp_audio *data);
-int mp_audio_make_writeable(struct mp_audio *data);
-
-struct AVFrame;
-struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe);
-struct AVFrame *mp_audio_to_avframe_and_unref(struct mp_audio *frame);
-int mp_audio_to_avframe(struct mp_audio *frame, struct AVFrame *avframe);
-
-struct mp_aframe;
-struct mp_audio *mp_audio_from_aframe(struct mp_aframe *aframe);
-void mp_audio_config_from_aframe(struct mp_audio *dst, struct mp_aframe *src);
-struct mp_aframe *mp_audio_to_aframe(struct mp_audio *mpa);
-
-struct mp_audio_pool;
-struct mp_audio_pool *mp_audio_pool_create(void *ta_parent);
-struct mp_audio *mp_audio_pool_get(struct mp_audio_pool *pool,
-                                   const struct mp_audio *fmt, int samples);
-struct mp_audio *mp_audio_pool_new_copy(struct mp_audio_pool *pool,
-                                        struct mp_audio *frame);
-int mp_audio_pool_make_writeable(struct mp_audio_pool *pool,
-                                 struct mp_audio *frame);
-
-#include "filter/af.h"
-
-#endif
@@ -1,49 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#ifndef MPLAYER_AD_H
-#define MPLAYER_AD_H
-
-#include "common/codecs.h"
-#include "demux/stheader.h"
-#include "demux/demux.h"
-
-#include "audio/format.h"
-#include "audio/aframe.h"
-#include "dec_audio.h"
-
-struct mp_decoder_list;
-
-/* interface of audio decoder drivers */
-struct ad_functions {
-    const char *name;
-    void (*add_decoders)(struct mp_decoder_list *list);
-    int (*init)(struct dec_audio *da, const char *decoder);
-    void (*uninit)(struct dec_audio *da);
-    int (*control)(struct dec_audio *da, int cmd, void *arg);
-    // Return whether or not the packet has been consumed.
-    bool (*send_packet)(struct dec_audio *da, struct demux_packet *pkt);
-    // Return whether decoding is still going on (false if EOF was reached).
-    // Never returns false & *out set, but can return true with !*out.
-    bool (*receive_frame)(struct dec_audio *da, struct mp_aframe **out);
-};
-
-enum ad_ctrl {
-    ADCTRL_RESET = 1,   // flush and reset state, e.g. after seeking
-};
-
-#endif /* MPLAYER_AD_H */
@@ -27,27 +27,30 @@
 #include <libavutil/intreadwrite.h>

 #include "mpv_talloc.h"
-
-#include "config.h"
+#include "audio/aframe.h"
+#include "audio/fmt-conversion.h"
 #include "common/av_common.h"
 #include "common/codecs.h"
+#include "common/global.h"
 #include "common/msg.h"
+#include "demux/packet.h"
+#include "demux/stheader.h"
+#include "filters/f_decoder_wrapper.h"
+#include "filters/filter_internal.h"
 #include "options/options.h"

-#include "ad.h"
-#include "audio/fmt-conversion.h"
-
 struct priv {
    AVCodecContext *avctx;
    AVFrame *avframe;
-    bool force_channel_map;
+    struct mp_chmap force_channel_map;
    uint32_t skip_samples, trim_samples;
    bool preroll_done;
    double next_pts;
    AVRational codec_timebase;
-};
+    bool eof_returned;

-static void uninit(struct dec_audio *da);
+    struct mp_decoder public;
+};

 #define OPT_BASE_STRUCT struct ad_lavc_params
 struct ad_lavc_params {
@@ -73,26 +76,24 @@ const struct m_sub_options ad_lavc_conf = {
    },
 };

-static int init(struct dec_audio *da, const char *decoder)
+static bool init(struct mp_filter *da, struct mp_codec_params *codec,
+                 const char *decoder)
 {
-    struct MPOpts *mpopts = da->opts;
+    struct priv *ctx = da->priv;
+    struct MPOpts *mpopts = da->global->opts;
    struct ad_lavc_params *opts = mpopts->ad_lavc_params;
    AVCodecContext *lavc_context;
    AVCodec *lavc_codec;
-    struct mp_codec_params *c = da->codec;

-    struct priv *ctx = talloc_zero(NULL, struct priv);
-    da->priv = ctx;
+    ctx->codec_timebase = mp_get_codec_timebase(codec);

-    ctx->codec_timebase = mp_get_codec_timebase(da->codec);
-
-    ctx->force_channel_map = c->force_channels;
+    if (codec->force_channels)
+        ctx->force_channel_map = codec->channels;

    lavc_codec = avcodec_find_decoder_by_name(decoder);
    if (!lavc_codec) {
        MP_ERR(da, "Cannot find codec '%s' in libavcodec...\n", decoder);
-        uninit(da);
-        return 0;
+        return false;
    }

    lavc_context = avcodec_alloc_context3(lavc_codec);
@@ -121,10 +122,9 @@ static int init(struct dec_audio *da, const char *decoder)

    mp_set_avopts(da->log, lavc_context, opts->avopts);

-    if (mp_set_avctx_codec_headers(lavc_context, c) < 0) {
+    if (mp_set_avctx_codec_headers(lavc_context, codec) < 0) {
        MP_ERR(da, "Could not set decoder parameters.\n");
-        uninit(da);
-        return 0;
+        return false;
    }

    mp_set_avcodec_threads(da->log, lavc_context, opts->threads);
@@ -132,41 +132,35 @@ static int init(struct dec_audio *da, const char *decoder)
    /* open it */
    if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
        MP_ERR(da, "Could not open codec.\n");
-        uninit(da);
-        return 0;
+        return false;
    }

    ctx->next_pts = MP_NOPTS_VALUE;

-    return 1;
+    return true;
 }

-static void uninit(struct dec_audio *da)
+static void destroy(struct mp_filter *da)
 {
    struct priv *ctx = da->priv;
-    if (!ctx)
-        return;

    avcodec_free_context(&ctx->avctx);
    av_frame_free(&ctx->avframe);
 }

-static int control(struct dec_audio *da, int cmd, void *arg)
+static void reset(struct mp_filter *da)
 {
    struct priv *ctx = da->priv;
-    switch (cmd) {
-    case ADCTRL_RESET:
-        avcodec_flush_buffers(ctx->avctx);
-        ctx->skip_samples = 0;
-        ctx->trim_samples = 0;
-        ctx->preroll_done = false;
-        ctx->next_pts = MP_NOPTS_VALUE;
-        return CONTROL_TRUE;
-    }
-    return CONTROL_UNKNOWN;
+
+    avcodec_flush_buffers(ctx->avctx);
+    ctx->skip_samples = 0;
+    ctx->trim_samples = 0;
+    ctx->preroll_done = false;
+    ctx->next_pts = MP_NOPTS_VALUE;
+    ctx->eof_returned = false;
 }

-static bool send_packet(struct dec_audio *da, struct demux_packet *mpkt)
+static bool send_packet(struct mp_filter *da, struct demux_packet *mpkt)
 {
    struct priv *priv = da->priv;
    AVCodecContext *avctx = priv->avctx;
@@ -190,7 +184,7 @@ static bool send_packet(struct dec_audio *da, struct demux_packet *mpkt)
    return true;
 }

-static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
+static bool receive_frame(struct mp_filter *da, struct mp_frame *out)
 {
    struct priv *priv = da->priv;
    AVCodecContext *avctx = priv->avctx;
@@ -200,7 +194,8 @@ static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
    if (ret == AVERROR_EOF) {
        // If flushing was initialized earlier and has ended now, make it start
        // over in case we get new packets at some point in the future.
-        control(da, ADCTRL_RESET, NULL);
+        // (Dont' reset the filter itself, we want to keep other state.)
+        avcodec_flush_buffers(priv->avctx);
        return false;
    } else if (ret < 0 && ret != AVERROR(EAGAIN)) {
        MP_ERR(da, "Error decoding audio.\n");
@@ -220,8 +215,8 @@ static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
    if (!mpframe)
        return true;

-    if (priv->force_channel_map)
-        mp_aframe_set_chmap(mpframe, &da->codec->channels);
+    if (priv->force_channel_map.num)
+        mp_aframe_set_chmap(mpframe, &priv->force_channel_map);

    if (out_pts == MP_NOPTS_VALUE)
        out_pts = priv->next_pts;
@@ -257,24 +252,57 @@ static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
        priv->trim_samples -= trim;
    }

-    *out = mpframe;
+    *out = MAKE_FRAME(MP_FRAME_AUDIO, mpframe);

    av_frame_unref(priv->avframe);

    return true;
 }

+static void process(struct mp_filter *ad)
+{
+    struct priv *priv = ad->priv;
+
+    lavc_process(ad, &priv->eof_returned, send_packet, receive_frame);
+}
+
+static const struct mp_filter_info ad_lavc_filter = {
+    .name = "ad_lavc",
+    .priv_size = sizeof(struct priv),
+    .process = process,
+    .reset = reset,
+    .destroy = destroy,
+};
+
+static struct mp_decoder *create(struct mp_filter *parent,
+                                 struct mp_codec_params *codec,
+                                 const char *decoder)
+{
+    struct mp_filter *da = mp_filter_create(parent, &ad_lavc_filter);
+    if (!da)
+        return NULL;
+
+    mp_filter_add_pin(da, MP_PIN_IN, "in");
+    mp_filter_add_pin(da, MP_PIN_OUT, "out");
+
+    da->log = mp_log_new(da, parent->log, NULL);
+
+    struct priv *priv = da->priv;
+    priv->public.f = da;
+
+    if (!init(da, codec, decoder)) {
+        talloc_free(da);
+        return NULL;
+    }
+    return &priv->public;
+}
+
 static void add_decoders(struct mp_decoder_list *list)
 {
    mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO);
 }

-const struct ad_functions ad_lavc = {
-    .name = "lavc",
+const struct mp_decoder_fns ad_lavc = {
+    .create = create,
    .add_decoders = add_decoders,
-    .init = init,
-    .uninit = uninit,
-    .control = control,
-    .send_packet = send_packet,
-    .receive_frame = receive_frame,
 };
@@ -24,11 +24,16 @@
 #include <libavcodec/avcodec.h>
 #include <libavutil/opt.h>

-#include "config.h"
-#include "common/msg.h"
+#include "audio/aframe.h"
+#include "audio/format.h"
 #include "common/av_common.h"
+#include "common/codecs.h"
+#include "common/msg.h"
+#include "demux/packet.h"
+#include "demux/stheader.h"
+#include "filters/f_decoder_wrapper.h"
+#include "filters/filter_internal.h"
 #include "options/options.h"
-#include "ad.h"

 #define OUTBUF_SIZE 65536

@@ -43,8 +48,8 @@ struct spdifContext {
    struct mp_aframe *fmt;
    int              sstride;
    struct mp_aframe_pool *pool;
-    bool             got_eof;
-    struct demux_packet *queued_packet;
+
+    struct mp_decoder public;
 };

 static int write_packet(void *p, uint8_t *buf, int buf_size)
@@ -62,7 +67,8 @@ static int write_packet(void *p, uint8_t *buf, int buf_size)
    return buf_size;
 }

-static void uninit(struct dec_audio *da)
+// (called on both filter destruction _and_ if lavf fails to init)
+static void destroy(struct mp_filter *da)
 {
    struct spdifContext *spdif_ctx = da->priv;
    AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;
@@ -74,27 +80,11 @@ static void uninit(struct dec_audio *da)
            av_freep(&lavf_ctx->pb->buffer);
        av_freep(&lavf_ctx->pb);
        avformat_free_context(lavf_ctx);
-        talloc_free(spdif_ctx->queued_packet);
        spdif_ctx->lavf_ctx = NULL;
    }
 }

-static int init(struct dec_audio *da, const char *decoder)
-{
-    struct spdifContext *spdif_ctx = talloc_zero(NULL, struct spdifContext);
-    da->priv = spdif_ctx;
-    spdif_ctx->log = da->log;
-    spdif_ctx->use_dts_hd = da->opts->dtshd;
-    spdif_ctx->pool = mp_aframe_pool_create(spdif_ctx);
-
-    if (strcmp(decoder, "spdif_dts_hd") == 0)
-        spdif_ctx->use_dts_hd = true;
-
-    spdif_ctx->codec_id = mp_codec_to_av_codec_id(da->codec->codec);
-    return spdif_ctx->codec_id != AV_CODEC_ID_NONE;
-}
-
-static void determine_codec_params(struct dec_audio *da, AVPacket *pkt,
+static void determine_codec_params(struct mp_filter *da, AVPacket *pkt,
                                   int *out_profile, int *out_rate)
 {
    struct spdifContext *spdif_ctx = da->priv;
@@ -157,7 +147,7 @@ done:
        MP_WARN(da, "Failed to parse codec profile.\n");
 }

-static int init_filter(struct dec_audio *da, AVPacket *pkt)
+static int init_filter(struct mp_filter *da, AVPacket *pkt)
 {
    struct spdifContext *spdif_ctx = da->priv;

@@ -271,39 +261,36 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
    return 0;

 fail:
-    uninit(da);
+    destroy(da);
+    mp_filter_internal_mark_failed(da);
    return -1;
 }

-
-static bool send_packet(struct dec_audio *da, struct demux_packet *mpkt)
+static void process(struct mp_filter *da)
 {
    struct spdifContext *spdif_ctx = da->priv;

-    if (spdif_ctx->queued_packet || spdif_ctx->got_eof)
-        return false;
+    if (!mp_pin_can_transfer_data(da->ppins[1], da->ppins[0]))
+        return;

-    spdif_ctx->queued_packet = mpkt ? demux_copy_packet(mpkt) : NULL;
-    spdif_ctx->got_eof = !mpkt;
-    return true;
-}
-
-static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
-{
-    struct spdifContext *spdif_ctx = da->priv;
-
-    if (spdif_ctx->got_eof) {
-        spdif_ctx->got_eof = false;
-        return false;
+    struct mp_frame inframe = mp_pin_out_read(da->ppins[0]);
+    if (inframe.type == MP_FRAME_EOF) {
+        mp_pin_in_write(da->ppins[1], inframe);
+        return;
+    } else if (inframe.type != MP_FRAME_PACKET) {
+        if (inframe.type) {
+            MP_ERR(da, "unknown frame type\n");
+            mp_filter_internal_mark_failed(da);
+        }
+        return;
    }

-    if (!spdif_ctx->queued_packet)
-        return true;
-
-    double pts = spdif_ctx->queued_packet->pts;
+    struct demux_packet *mpkt = inframe.data;
+    struct mp_aframe *out = NULL;
+    double pts = mpkt->pts;

    AVPacket pkt;
-    mp_set_av_packet(&pkt, spdif_ctx->queued_packet, NULL);
+    mp_set_av_packet(&pkt, mpkt, NULL);
    pkt.pts = pkt.dts = 0;
    if (!spdif_ctx->lavf_ctx) {
        if (init_filter(da, &pkt) < 0)
@@ -317,39 +304,29 @@ static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
        goto done;
    }

-    *out = mp_aframe_new_ref(spdif_ctx->fmt);
+    out = mp_aframe_new_ref(spdif_ctx->fmt);
    int samples = spdif_ctx->out_buffer_len / spdif_ctx->sstride;
-    if (mp_aframe_pool_allocate(spdif_ctx->pool, *out, samples) < 0) {
-        TA_FREEP(out);
+    if (mp_aframe_pool_allocate(spdif_ctx->pool, out, samples) < 0) {
+        TA_FREEP(&out);
        goto done;
    }

-    uint8_t **data = mp_aframe_get_data_rw(*out);
+    uint8_t **data = mp_aframe_get_data_rw(out);
    if (!data) {
-        TA_FREEP(out);
+        TA_FREEP(&out);
        goto done;
    }

    memcpy(data[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
-    mp_aframe_set_pts(*out, pts);
+    mp_aframe_set_pts(out, pts);

 done:
-    talloc_free(spdif_ctx->queued_packet);
-    spdif_ctx->queued_packet = NULL;
-    return true;
-}
-
-static int control(struct dec_audio *da, int cmd, void *arg)
-{
-    struct spdifContext *spdif_ctx = da->priv;
-    switch (cmd) {
-    case ADCTRL_RESET:
-        talloc_free(spdif_ctx->queued_packet);
-        spdif_ctx->queued_packet = NULL;
-        spdif_ctx->got_eof = false;
-        return CONTROL_TRUE;
+    talloc_free(mpkt);
+    if (out) {
+        mp_pin_in_write(da->ppins[1], MAKE_FRAME(MP_FRAME_AUDIO, out));
+    } else {
+        mp_filter_internal_mark_failed(da);
    }
-    return CONTROL_UNKNOWN;
 }

 static const int codecs[] = {
@@ -401,17 +378,49 @@ struct mp_decoder_list *select_spdif_codec(const char *codec, const char *pref)
    const char *suffix_name = dts_hd_allowed ? "dts_hd" : codec;
    char name[80];
    snprintf(name, sizeof(name), "spdif_%s", suffix_name);
-    mp_add_decoder(list, "spdif", codec, name,
+    mp_add_decoder(list, codec, name,
                   "libavformat/spdifenc audio pass-through decoder");
    return list;
 }

-const struct ad_functions ad_spdif = {
-    .name = "spdif",
-    .add_decoders = NULL,
-    .init = init,
-    .uninit = uninit,
-    .control = control,
-    .send_packet = send_packet,
-    .receive_frame = receive_frame,
+static const struct mp_filter_info ad_spdif_filter = {
+    .name = "ad_spdif",
+    .priv_size = sizeof(struct spdifContext),
+    .process = process,
+    .destroy = destroy,
+};
+
+static struct mp_decoder *create(struct mp_filter *parent,
+                                 struct mp_codec_params *codec,
+                                 const char *decoder)
+{
+    struct mp_filter *da = mp_filter_create(parent, &ad_spdif_filter);
+    if (!da)
+        return NULL;
+
+    mp_filter_add_pin(da, MP_PIN_IN, "in");
+    mp_filter_add_pin(da, MP_PIN_OUT, "out");
+
+    da->log = mp_log_new(da, parent->log, NULL);
+
+    struct spdifContext *spdif_ctx = da->priv;
+    spdif_ctx->log = da->log;
+    spdif_ctx->pool = mp_aframe_pool_create(spdif_ctx);
+    spdif_ctx->public.f = da;
+
+    if (strcmp(decoder, "spdif_dts_hd") == 0)
+        spdif_ctx->use_dts_hd = true;
+
+    spdif_ctx->codec_id = mp_codec_to_av_codec_id(codec->codec);
+
+
+    if (spdif_ctx->codec_id == AV_CODEC_ID_NONE) {
+        talloc_free(da);
+        return NULL;
+    }
+    return &spdif_ctx->public;
+}
+
+const struct mp_decoder_fns ad_spdif = {
+    .create = create,
 };
@@ -1,290 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <math.h>
-#include <assert.h>
-
-#include <libavutil/mem.h>
-
-#include "demux/codec_tags.h"
-
-#include "common/codecs.h"
-#include "common/msg.h"
-#include "common/recorder.h"
-#include "misc/bstr.h"
-#include "options/options.h"
-
-#include "stream/stream.h"
-#include "demux/demux.h"
-
-#include "demux/stheader.h"
-
-#include "dec_audio.h"
-#include "ad.h"
-#include "audio/format.h"
-
-extern const struct ad_functions ad_lavc;
-
-// Not a real codec - specially treated.
-extern const struct ad_functions ad_spdif;
-
-static const struct ad_functions * const ad_drivers[] = {
-    &ad_lavc,
-    NULL
-};
-
-static void uninit_decoder(struct dec_audio *d_audio)
-{
-    audio_reset_decoding(d_audio);
-    if (d_audio->ad_driver) {
-        MP_VERBOSE(d_audio, "Uninit audio decoder.\n");
-        d_audio->ad_driver->uninit(d_audio);
-    }
-    d_audio->ad_driver = NULL;
-    talloc_free(d_audio->priv);
-    d_audio->priv = NULL;
-}
-
-static int init_audio_codec(struct dec_audio *d_audio, const char *decoder)
-{
-    if (!d_audio->ad_driver->init(d_audio, decoder)) {
-        MP_VERBOSE(d_audio, "Audio decoder init failed.\n");
-        d_audio->ad_driver = NULL;
-        uninit_decoder(d_audio);
-        return 0;
-    }
-
-    return 1;
-}
-
-struct mp_decoder_list *audio_decoder_list(void)
-{
-    struct mp_decoder_list *list = talloc_zero(NULL, struct mp_decoder_list);
-    for (int i = 0; ad_drivers[i] != NULL; i++)
-        ad_drivers[i]->add_decoders(list);
-    return list;
-}
-
-static struct mp_decoder_list *audio_select_decoders(struct dec_audio *d_audio)
-{
-    struct MPOpts *opts = d_audio->opts;
-    const char *codec = d_audio->codec->codec;
-
-    struct mp_decoder_list *list = audio_decoder_list();
-    struct mp_decoder_list *new =
-        mp_select_decoders(d_audio->log, list, codec, opts->audio_decoders);
-    if (d_audio->try_spdif && codec) {
-        struct mp_decoder_list *spdif =
-            select_spdif_codec(codec, opts->audio_spdif);
-        mp_append_decoders(spdif, new);
-        talloc_free(new);
-        new = spdif;
-    }
-    talloc_free(list);
-    return new;
-}
-
-static const struct ad_functions *find_driver(const char *name)
-{
-    for (int i = 0; ad_drivers[i] != NULL; i++) {
-        if (strcmp(ad_drivers[i]->name, name) == 0)
-            return ad_drivers[i];
-    }
-    if (strcmp(name, "spdif") == 0)
-        return &ad_spdif;
-    return NULL;
-}
-
-int audio_init_best_codec(struct dec_audio *d_audio)
-{
-    uninit_decoder(d_audio);
-    assert(!d_audio->ad_driver);
-
-    struct mp_decoder_entry *decoder = NULL;
-    struct mp_decoder_list *list = audio_select_decoders(d_audio);
-
-    mp_print_decoders(d_audio->log, MSGL_V, "Codec list:", list);
-
-    for (int n = 0; n < list->num_entries; n++) {
-        struct mp_decoder_entry *sel = &list->entries[n];
-        const struct ad_functions *driver = find_driver(sel->family);
-        if (!driver)
-            continue;
-        MP_VERBOSE(d_audio, "Opening audio decoder %s\n", sel->decoder);
-        d_audio->ad_driver = driver;
-        if (init_audio_codec(d_audio, sel->decoder)) {
-            decoder = sel;
-            break;
-        }
-        MP_WARN(d_audio, "Audio decoder init failed for %s\n", sel->decoder);
-    }
-
-    if (d_audio->ad_driver) {
-        d_audio->decoder_desc =
-            talloc_asprintf(d_audio, "%s (%s)", decoder->decoder, decoder->desc);
-        MP_VERBOSE(d_audio, "Selected audio codec: %s\n", d_audio->decoder_desc);
-    } else {
-        MP_ERR(d_audio, "Failed to initialize an audio decoder for codec '%s'.\n",
-               d_audio->codec->codec);
-    }
-
-    talloc_free(list);
-    return !!d_audio->ad_driver;
-}
-
-void audio_uninit(struct dec_audio *d_audio)
-{
-    if (!d_audio)
-        return;
-    uninit_decoder(d_audio);
-    talloc_free(d_audio);
-}
-
-void audio_reset_decoding(struct dec_audio *d_audio)
-{
-    if (d_audio->ad_driver)
-        d_audio->ad_driver->control(d_audio, ADCTRL_RESET, NULL);
-    d_audio->pts = MP_NOPTS_VALUE;
-    talloc_free(d_audio->current_frame);
-    d_audio->current_frame = NULL;
-    talloc_free(d_audio->packet);
-    d_audio->packet = NULL;
-    talloc_free(d_audio->new_segment);
-    d_audio->new_segment = NULL;
-    d_audio->start = d_audio->end = MP_NOPTS_VALUE;
-}
-
-static void fix_audio_pts(struct dec_audio *da)
-{
-    if (!da->current_frame)
-        return;
-
-    double frame_pts = mp_aframe_get_pts(da->current_frame);
-    if (frame_pts != MP_NOPTS_VALUE) {
-        if (da->pts != MP_NOPTS_VALUE)
-            MP_STATS(da, "value %f audio-pts-err", da->pts - frame_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 || fabs(da->pts - frame_pts) > 0.001)
-            da->pts = frame_pts;
-    }
-
-    if (da->pts == MP_NOPTS_VALUE && da->header->missing_timestamps)
-        da->pts = 0;
-
-    mp_aframe_set_pts(da->current_frame, da->pts);
-
-    if (da->pts != MP_NOPTS_VALUE)
-        da->pts += mp_aframe_duration(da->current_frame);
-}
-
-static bool is_new_segment(struct dec_audio *da, struct demux_packet *p)
-{
-    return p->segmented &&
-        (p->start != da->start || p->end != da->end || p->codec != da->codec);
-}
-
-void audio_work(struct dec_audio *da)
-{
-    if (da->current_frame || !da->ad_driver)
-        return;
-
-    if (!da->packet && !da->new_segment &&
-        demux_read_packet_async(da->header, &da->packet) == 0)
-    {
-        da->current_state = DATA_WAIT;
-        return;
-    }
-
-    if (da->packet && is_new_segment(da, da->packet)) {
-        assert(!da->new_segment);
-        da->new_segment = da->packet;
-        da->packet = NULL;
-    }
-
-    if (da->ad_driver->send_packet(da, da->packet)) {
-        if (da->recorder_sink)
-            mp_recorder_feed_packet(da->recorder_sink, da->packet);
-
-        talloc_free(da->packet);
-        da->packet = NULL;
-    }
-
-    bool progress = da->ad_driver->receive_frame(da, &da->current_frame);
-
-    da->current_state = da->current_frame ? DATA_OK : DATA_AGAIN;
-    if (!progress)
-        da->current_state = DATA_EOF;
-
-    fix_audio_pts(da);
-
-    bool segment_end = da->current_state == DATA_EOF;
-
-    if (da->current_frame) {
-        mp_aframe_clip_timestamps(da->current_frame, da->start, da->end);
-        double frame_pts = mp_aframe_get_pts(da->current_frame);
-        if (frame_pts != MP_NOPTS_VALUE && da->start != MP_NOPTS_VALUE)
-            segment_end = frame_pts >= da->end;
-        if (mp_aframe_get_size(da->current_frame) == 0) {
-            talloc_free(da->current_frame);
-            da->current_frame = NULL;
-        }
-    }
-
-    // If there's a new segment, start it as soon as we're drained/finished.
-    if (segment_end && da->new_segment) {
-        struct demux_packet *new_segment = da->new_segment;
-        da->new_segment = NULL;
-
-        if (da->codec == new_segment->codec) {
-            audio_reset_decoding(da);
-        } else {
-            da->codec = new_segment->codec;
-            da->ad_driver->uninit(da);
-            da->ad_driver = NULL;
-            audio_init_best_codec(da);
-        }
-
-        da->start = new_segment->start;
-        da->end = new_segment->end;
-
-        da->packet = new_segment;
-        da->current_state = DATA_AGAIN;
-    }
-}
-
-// Fetch an audio frame decoded with audio_work(). Returns one of:
-//  DATA_OK:    *out_frame is set to a new image
-//  DATA_WAIT:  waiting for demuxer; will receive a wakeup signal
-//  DATA_EOF:   end of file, no more frames to be expected
-//  DATA_AGAIN: dropped frame or something similar
-int audio_get_frame(struct dec_audio *da, struct mp_aframe **out_frame)
-{
-    *out_frame = NULL;
-    if (da->current_frame) {
-        *out_frame = da->current_frame;
-        da->current_frame = NULL;
-        return DATA_OK;
-    }
-    if (da->current_state == DATA_OK)
-        return DATA_AGAIN;
-    return da->current_state;
-}
@@ -1,66 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#ifndef MPLAYER_DEC_AUDIO_H
-#define MPLAYER_DEC_AUDIO_H
-
-#include "audio/chmap.h"
-#include "audio/aframe.h"
-#include "demux/demux.h"
-#include "demux/stheader.h"
-
-struct mp_decoder_list;
-
-struct dec_audio {
-    struct mp_log *log;
-    struct MPOpts *opts;
-    struct mpv_global *global;
-    const struct ad_functions *ad_driver;
-    struct sh_stream *header;
-    struct mp_codec_params *codec;
-    char *decoder_desc;
-
-    bool try_spdif;
-
-    struct mp_recorder_sink *recorder_sink;
-
-    // For free use by the ad_driver
-    void *priv;
-
-    // Strictly internal (dec_audio.c).
-
-    double pts; // endpts of previous frame
-    double start, end;
-    struct demux_packet *packet;
-    struct demux_packet *new_segment;
-    struct mp_aframe *current_frame;
-    int current_state;
-};
-
-struct mp_decoder_list *audio_decoder_list(void);
-int audio_init_best_codec(struct dec_audio *d_audio);
-void audio_uninit(struct dec_audio *d_audio);
-
-void audio_work(struct dec_audio *d_audio);
-int audio_get_frame(struct dec_audio *d_audio, struct mp_aframe **out_frame);
-
-void audio_reset_decoding(struct dec_audio *d_audio);
-
-// ad_spdif.c
-struct mp_decoder_list *select_spdif_codec(const char *codec, const char *pref);
-
-#endif /* MPLAYER_DEC_AUDIO_H */
@@ -1,821 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include "config.h"
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <assert.h>
-
-#include "common/common.h"
-#include "common/global.h"
-
-#include "options/m_option.h"
-#include "options/m_config.h"
-
-#include "audio/audio_buffer.h"
-#include "af.h"
-
-// Static list of filters
-extern const struct af_info af_info_format;
-extern const struct af_info af_info_lavcac3enc;
-extern const struct af_info af_info_lavrresample;
-extern const struct af_info af_info_scaletempo;
-extern const struct af_info af_info_lavfi;
-extern const struct af_info af_info_lavfi_bridge;
-extern const struct af_info af_info_rubberband;
-
-static const struct af_info *const filter_list[] = {
-    &af_info_format,
-    &af_info_lavcac3enc,
-    &af_info_lavrresample,
-#if HAVE_RUBBERBAND
-    &af_info_rubberband,
-#endif
-    &af_info_scaletempo,
-    &af_info_lavfi,
-    &af_info_lavfi_bridge,
-    NULL
-};
-
-static bool get_desc(struct m_obj_desc *dst, int index)
-{
-    if (index >= MP_ARRAY_SIZE(filter_list) - 1)
-        return false;
-    const struct af_info *af = filter_list[index];
-    *dst = (struct m_obj_desc) {
-        .name = af->name,
-        .description = af->info,
-        .priv_size = af->priv_size,
-        .priv_defaults = af->priv_defaults,
-        .options = af->options,
-        .p = af,
-    };
-    return true;
-}
-
-const struct m_obj_list af_obj_list = {
-    .get_desc = get_desc,
-    .description = "audio filters",
-    .allow_disable_entries = true,
-    .allow_unknown_entries = true,
-    .aliases = {
-        {"force",     "format"},
-        {0}
-    },
-};
-
-static void af_forget_frames(struct af_instance *af)
-{
-    for (int n = 0; n < af->num_out_queued; n++)
-        talloc_free(af->out_queued[n]);
-    af->num_out_queued = 0;
-}
-
-static void af_chain_forget_frames(struct af_stream *s)
-{
-    for (struct af_instance *cur = s->first; cur; cur = cur->next)
-        af_forget_frames(cur);
-}
-
-static void af_copy_unset_fields(struct mp_audio *dst, struct mp_audio *src)
-{
-    if (dst->format == AF_FORMAT_UNKNOWN)
-        mp_audio_set_format(dst, src->format);
-    if (dst->nch == 0)
-        mp_audio_set_channels(dst, &src->channels);
-    if (dst->rate == 0)
-        dst->rate = src->rate;
-}
-
-static int input_control(struct af_instance* af, int cmd, void* arg)
-{
-    switch (cmd) {
-    case AF_CONTROL_REINIT:
-        assert(arg == &((struct af_stream *)af->priv)->input);
-        return AF_OK;
-    }
-    return AF_UNKNOWN;
-}
-
-static int output_control(struct af_instance* af, int cmd, void* arg)
-{
-    struct af_stream *s = af->priv;
-    struct mp_audio *output = &s->output;
-    struct mp_audio *filter_output = &s->filter_output;
-
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-
-        *filter_output = *output;
-        af_copy_unset_fields(filter_output, in);
-        *in = *filter_output;
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
-    }
-    }
-    return AF_UNKNOWN;
-}
-
-static int dummy_filter(struct af_instance *af, struct mp_audio *frame)
-{
-    af_add_output_frame(af, frame);
-    return 0;
-}
-
-/* Function for creating a new filter of type name.The name may
-contain the commandline parameters for the filter */
-static struct af_instance *af_create(struct af_stream *s, char *name,
-                                     char **args)
-{
-    const char *lavfi_name = NULL;
-    char **lavfi_args = NULL;
-    struct m_obj_desc desc;
-    if (!m_obj_list_find(&desc, &af_obj_list, bstr0(name))) {
-        if (!m_obj_list_find(&desc, &af_obj_list, bstr0("lavfi-bridge"))) {
-            MP_ERR(s, "Couldn't find audio filter '%s'.\n", name);
-            return NULL;
-        }
-        lavfi_name = name;
-        lavfi_args = args;
-        args = NULL;
-        if (strncmp(lavfi_name, "lavfi-", 6) == 0)
-            lavfi_name += 6;
-    }
-    MP_VERBOSE(s, "Adding filter %s \n", name);
-
-    struct af_instance *af = talloc_zero(NULL, struct af_instance);
-    *af = (struct af_instance) {
-        .full_name = talloc_strdup(af, name),
-        .info = desc.p,
-        .data = talloc_zero(af, struct mp_audio),
-        .log = mp_log_new(af, s->log, name),
-        .opts = s->opts,
-        .global = s->global,
-        .out_pool = mp_audio_pool_create(af),
-    };
-    struct m_config *config =
-        m_config_from_obj_desc_and_args(af, s->log, NULL, &desc,
-                                        name, s->opts->af_defs, args);
-    if (!config)
-        goto error;
-    if (lavfi_name) {
-        // Pass the filter arguments as proper sub-options to the bridge filter.
-        struct m_config_option *name_opt = m_config_get_co(config, bstr0("name"));
-        assert(name_opt);
-        assert(name_opt->opt->type == &m_option_type_string);
-        if (m_config_set_option_raw(config, name_opt, &lavfi_name, 0) < 0)
-            goto error;
-        struct m_config_option *opts = m_config_get_co(config, bstr0("opts"));
-        assert(opts);
-        assert(opts->opt->type == &m_option_type_keyvalue_list);
-        if (m_config_set_option_raw(config, opts, &lavfi_args, 0) < 0)
-            goto error;
-        af->full_name = talloc_asprintf(af, "%s (lavfi)", af->full_name);
-    }
-    af->priv = config->optstruct;
-
-    // Initialize the new filter
-    if (af->info->open(af) != AF_OK)
-        goto error;
-
-    return af;
-
-error:
-    MP_ERR(s, "Couldn't create or open audio filter '%s'\n", name);
-    talloc_free(af);
-    return NULL;
-}
-
-/* Create and insert a new filter of type name before the filter in the
-   argument. This function can be called during runtime, the return
-   value is the new filter */
-static struct af_instance *af_prepend(struct af_stream *s,
-                                      struct af_instance *af,
-                                      char *name, char **args)
-{
-    if (!af)
-        af = s->last;
-    if (af == s->first)
-        af = s->first->next;
-    // Create the new filter and make sure it is OK
-    struct af_instance *new = af_create(s, name, args);
-    if (!new)
-        return NULL;
-    // Update pointers
-    new->next = af;
-    new->prev = af->prev;
-    af->prev = new;
-    new->prev->next = new;
-    return new;
-}
-
-// Uninit and remove the filter "af"
-static void af_remove(struct af_stream *s, struct af_instance *af)
-{
-    if (!af)
-        return;
-
-    if (af == s->first || af == s->last)
-        return;
-
-    // Print friendly message
-    MP_VERBOSE(s, "Removing filter %s \n", af->info->name);
-
-    // Detach pointers
-    af->prev->next = af->next;
-    af->next->prev = af->prev;
-
-    if (af->uninit)
-        af->uninit(af);
-    af_forget_frames(af);
-    talloc_free(af);
-}
-
-static void remove_auto_inserted_filters(struct af_stream *s)
-{
-repeat:
-    for (struct af_instance *af = s->first; af; af = af->next) {
-        if (af->auto_inserted) {
-            af_remove(s, af);
-            goto repeat;
-        }
-    }
-}
-
-static void af_print_filter_chain(struct af_stream *s, struct af_instance *at,
-                                  int msg_level)
-{
-    MP_MSG(s, msg_level, "Audio filter chain:\n");
-
-    struct af_instance *af = s->first;
-    while (af) {
-        char b[128] = {0};
-        mp_snprintf_cat(b, sizeof(b), "  [%s] ", af->full_name);
-        if (af->label)
-            mp_snprintf_cat(b, sizeof(b), "\"%s\" ", af->label);
-        if (af->data)
-            mp_snprintf_cat(b, sizeof(b), "%s", mp_audio_config_to_str(af->data));
-        if (af->auto_inserted)
-            mp_snprintf_cat(b, sizeof(b), " [a]");
-        if (af == at)
-            mp_snprintf_cat(b, sizeof(b), " <-");
-        MP_MSG(s, msg_level, "%s\n", b);
-
-        af = af->next;
-    }
-
-    MP_MSG(s, msg_level, "  [ao] %s\n", mp_audio_config_to_str(&s->output));
-}
-
-static void reset_formats(struct af_stream *s)
-{
-    struct mp_audio none = {0};
-    for (struct af_instance *af = s->first; af; af = af->next) {
-        if (af != s->first && af != s->last)
-            mp_audio_copy_config(af->data, &none);
-    }
-}
-
-static int filter_reinit(struct af_instance *af)
-{
-    struct af_instance *prev = af->prev;
-    assert(prev);
-
-    // Check if this is the first filter
-    struct mp_audio in = *prev->data;
-    // Reset just in case...
-    mp_audio_set_null_data(&in);
-
-    if (!mp_audio_config_valid(&in))
-        return AF_ERROR;
-
-    af->fmt_in = in;
-    int rv = af->control(af, AF_CONTROL_REINIT, &in);
-    if (rv == AF_OK && !mp_audio_config_equals(&in, prev->data))
-        rv = AF_FALSE; // conversion filter needed
-    if (rv == AF_FALSE)
-        af->fmt_in = in;
-
-    if (rv == AF_OK) {
-        if (!mp_audio_config_valid(af->data))
-            return AF_ERROR;
-        af->fmt_out = *af->data;
-    }
-
-    return rv;
-}
-
-static int filter_reinit_with_conversion(struct af_stream *s, struct af_instance *af)
-{
-    int rv = filter_reinit(af);
-
-    // Conversion filter is needed
-    if (rv == AF_FALSE) {
-        // First try if we can change the output format of the previous
-        // filter to the input format the current filter is expecting.
-        struct mp_audio in = af->fmt_in;
-        if (af->prev != s->first && !mp_audio_config_equals(af->prev->data, &in)) {
-            // This should have been successful (because it succeeded
-            // before), even if just reverting to the old output format.
-            mp_audio_copy_config(af->prev->data, &in);
-            rv = filter_reinit(af->prev);
-            if (rv != AF_OK)
-                return rv;
-        }
-        if (!mp_audio_config_equals(af->prev->data, &in)) {
-            // Retry with conversion filter added.
-            struct af_instance *new =
-                af_prepend(s, af, "lavrresample", NULL);
-            if (!new)
-                return AF_ERROR;
-            new->auto_inserted = true;
-            mp_audio_copy_config(new->data, &in);
-            rv = filter_reinit(new);
-            if (rv != AF_OK)
-                af_remove(s, new);
-        }
-        if (rv == AF_OK)
-            rv = filter_reinit(af);
-    }
-
-    return rv;
-}
-
-static int af_find_output_conversion(struct af_stream *s, struct mp_audio *cfg)
-{
-    assert(mp_audio_config_valid(&s->output));
-    assert(s->initialized > 0);
-
-    if (mp_chmap_equals_reordered(&s->input.channels, &s->output.channels))
-        return AF_ERROR;
-
-    // Heuristic to detect point of conversion. If it looks like something
-    // more complicated is going on, better bail out.
-    // We expect that the last filter converts channels.
-    struct af_instance *conv = s->last->prev;
-    if (!conv->auto_inserted)
-        return AF_ERROR;
-    if (!(mp_chmap_equals_reordered(&conv->fmt_in.channels, &s->input.channels) &&
-          mp_chmap_equals_reordered(&conv->fmt_out.channels, &s->output.channels)))
-        return AF_ERROR;
-    // Also, should be the only one which does auto conversion.
-    for (struct af_instance *af = s->first->next; af != s->last; af = af->next)
-    {
-        if (af != conv && af->auto_inserted &&
-            !mp_chmap_equals_reordered(&af->fmt_in.channels, &af->fmt_out.channels))
-            return AF_ERROR;
-    }
-    // And not if it's the only filter.
-    if (conv->prev == s->first && conv->next == s->last)
-        return AF_ERROR;
-
-    *cfg = s->output;
-    return AF_OK;
-}
-
-// Return AF_OK on success or AF_ERROR on failure.
-static int af_do_reinit(struct af_stream *s, bool second_pass)
-{
-    struct mp_audio convert_early = {0};
-    if (second_pass) {
-        // If a channel conversion happens, and it is done by an auto-inserted
-        // filter, then insert a filter to convert it early. Otherwise, do
-        // nothing and return immediately.
-        if (af_find_output_conversion(s, &convert_early) != AF_OK)
-            return AF_OK;
-    }
-
-    remove_auto_inserted_filters(s);
-    af_chain_forget_frames(s);
-    reset_formats(s);
-    s->first->fmt_in = s->first->fmt_out = s->input;
-
-    if (mp_audio_config_valid(&convert_early)) {
-        struct af_instance *new = af_prepend(s, s->first, "lavrresample", NULL);
-        if (!new)
-            return AF_ERROR;
-        new->auto_inserted = true;
-        mp_audio_copy_config(new->data, &convert_early);
-        int rv = filter_reinit(new);
-        if (rv != AF_DETACH && rv != AF_OK)
-            return AF_ERROR;
-        MP_VERBOSE(s, "Moving up output conversion.\n");
-    }
-
-    // Start with the second filter, as the first filter is the special input
-    // filter which needs no initialization.
-    struct af_instance *af = s->first->next;
-    while (af) {
-        int rv = filter_reinit_with_conversion(s, af);
-
-        switch (rv) {
-        case AF_OK:
-            af = af->next;
-            break;
-        case AF_FALSE: {
-            // If the format conversion is (probably) caused by spdif, then
-            // (as a feature) drop the filter, instead of failing hard.
-            int fmt_in1 = af->prev->data->format;
-            int fmt_in2 = af->fmt_in.format;
-            if (af_fmt_is_valid(fmt_in1) && af_fmt_is_valid(fmt_in2)) {
-                bool spd1 = af_fmt_is_spdif(fmt_in1);
-                bool spd2 = af_fmt_is_spdif(fmt_in2);
-                if (spd1 != spd2 && af->next) {
-                    MP_WARN(af, "Filter %s apparently cannot be used due to "
-                                "spdif passthrough - removing it.\n",
-                                af->info->name);
-                    struct af_instance *aft = af->prev;
-                    af_remove(s, af);
-                    af = aft->next;
-                    continue;
-                }
-            }
-            goto negotiate_error;
-        }
-        case AF_DETACH: { // Filter is redundant and wants to be unloaded
-            struct af_instance *aft = af->prev; // never NULL
-            af_remove(s, af);
-            af = aft->next;
-            break;
-        }
-        default:
-            MP_ERR(s, "Reinitialization did not work, "
-                   "audio filter '%s' returned error code %i\n",
-                   af->info->name, rv);
-            goto error;
-        }
-    }
-
-    /* Set previously unset fields in s->output to those of the filter chain
-     * output. This is used to make the output format fixed, and even if you
-     * insert new filters or change the input format, the output format won't
-     * change. (Audio outputs generally can't change format at runtime.) */
-    af_copy_unset_fields(&s->output, &s->filter_output);
-    if (mp_audio_config_equals(&s->output, &s->filter_output)) {
-        s->initialized = 1;
-        af_print_filter_chain(s, NULL, MSGL_V);
-        return AF_OK;
-    }
-
-    goto error;
-
-negotiate_error:
-    MP_ERR(s, "Unable to convert audio input format to output format.\n");
-error:
-    s->initialized = -1;
-    af_print_filter_chain(s, af, MSGL_ERR);
-    return AF_ERROR;
-}
-
-static int af_reinit(struct af_stream *s)
-{
-    int r = af_do_reinit(s, false);
-    if (r == AF_OK && mp_audio_config_valid(&s->output)) {
-        r = af_do_reinit(s, true);
-        if (r != AF_OK) {
-            MP_ERR(s, "Failed second pass filter negotiation.\n");
-            r = af_do_reinit(s, false);
-        }
-    }
-    return r;
-}
-
-// Uninit and remove all filters
-void af_uninit(struct af_stream *s)
-{
-    while (s->first->next && s->first->next != s->last)
-        af_remove(s, s->first->next);
-    af_chain_forget_frames(s);
-    s->initialized = 0;
-}
-
-struct af_stream *af_new(struct mpv_global *global)
-{
-    struct af_stream *s = talloc_zero(NULL, struct af_stream);
-    s->log = mp_log_new(s, global->log, "!af");
-
-    static const struct af_info in = { .name = "in" };
-    s->first = talloc(s, struct af_instance);
-    *s->first = (struct af_instance) {
-        .full_name = "in",
-        .info = &in,
-        .log = s->log,
-        .control = input_control,
-        .filter_frame = dummy_filter,
-        .priv = s,
-        .data = &s->input,
-    };
-
-    static const struct af_info out = { .name = "out" };
-    s->last = talloc(s, struct af_instance);
-    *s->last = (struct af_instance) {
-        .full_name = "out",
-        .info = &out,
-        .log = s->log,
-        .control = output_control,
-        .filter_frame = dummy_filter,
-        .priv = s,
-        .data = &s->filter_output,
-    };
-
-    s->first->next = s->last;
-    s->last->prev = s->first;
-    s->opts = global->opts;
-    s->global = global;
-    return s;
-}
-
-void af_destroy(struct af_stream *s)
-{
-    af_uninit(s);
-    talloc_free(s);
-}
-
-/* Initialize the stream "s". This function creates a new filter list
-   if necessary according to the values set in input and output. Input
-   and output should contain the format of the current movie and the
-   format of the preferred output respectively. The function is
-   reentrant i.e. if called with an already initialized stream the
-   stream will be reinitialized.
-   If one of the preferred output parameters is 0 the one that needs
-   no conversion is used (i.e. the output format in the last filter).
-   The return value is 0 if success and -1 if failure */
-int af_init(struct af_stream *s)
-{
-    // Precaution in case caller is misbehaving
-    mp_audio_set_null_data(&s->input);
-    mp_audio_set_null_data(&s->output);
-
-    // Check if this is the first call
-    if (s->first->next == s->last) {
-        // Add all filters in the list (if there are any)
-        struct m_obj_settings *list = s->opts->af_settings;
-        for (int i = 0; list && list[i].name; i++) {
-            if (!list[i].enabled)
-                continue;
-            struct af_instance *af =
-                af_prepend(s, s->last, list[i].name, list[i].attribs);
-            if (!af) {
-                af_uninit(s);
-                s->initialized = -1;
-                return -1;
-            }
-            af->label = talloc_strdup(af, list[i].label);
-        }
-    }
-
-    if (af_reinit(s) != AF_OK) {
-        // Something is stuffed audio out will not work
-        MP_ERR(s, "Could not create audio filter chain.\n");
-        return -1;
-    }
-    return 0;
-}
-
-/* Add filter during execution. This function adds the filter "name"
-   to the stream s. The filter will be inserted somewhere nice in the
-   list of filters. The return value is a pointer to the new filter,
-   If the filter couldn't be added the return value is NULL. */
-struct af_instance *af_add(struct af_stream *s, char *name, char *label,
-                           char **args)
-{
-    assert(label);
-
-    if (af_find_by_label(s, label))
-        return NULL;
-
-    struct af_instance *new = af_prepend(s, s->last, name, args);
-    if (!new)
-        return NULL;
-    new->label = talloc_strdup(new, label);
-
-    // Reinitialize the filter list
-    if (af_reinit(s) != AF_OK) {
-        af_remove_by_label(s, label);
-        return NULL;
-    }
-    return af_find_by_label(s, label);
-}
-
-struct af_instance *af_find_by_label(struct af_stream *s, char *label)
-{
-    for (struct af_instance *af = s->first; af; af = af->next) {
-        if (af->label && strcmp(af->label, label) == 0)
-            return af;
-    }
-    return NULL;
-}
-
-/* Remove the first filter that matches this name. Return number of filters
- * removed (0, 1), or a negative error code if reinit after removing failed.
- */
-int af_remove_by_label(struct af_stream *s, char *label)
-{
-    struct af_instance *af = af_find_by_label(s, label);
-    if (!af)
-        return 0;
-    af_remove(s, af);
-    if (af_reinit(s) != AF_OK) {
-        af_uninit(s);
-        af_init(s);
-        return -1;
-    }
-    return 1;
-}
-
-/* Calculate the total delay [seconds of output] caused by the filters */
-double af_calc_delay(struct af_stream *s)
-{
-    struct af_instance *af = s->first;
-    double delay = 0.0;
-    while (af) {
-        delay += af->delay;
-        for (int n = 0; n < af->num_out_queued; n++)
-            delay += af->out_queued[n]->samples / (double)af->data->rate;
-        af = af->next;
-    }
-    return delay;
-}
-
-/* Send control to all filters, starting with the last until one accepts the
- * command with AF_OK. Return the accepting filter. */
-struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg)
-{
-    int res = AF_UNKNOWN;
-    struct af_instance *filt = s->last;
-    while (filt) {
-        res = filt->control(filt, cmd, arg);
-        if (res == AF_OK)
-            return filt;
-        filt = filt->prev;
-    }
-    return NULL;
-}
-
-/* Send control to all filters. Never stop, even if a filter returns AF_OK. */
-void af_control_all(struct af_stream *s, int cmd, void *arg)
-{
-    for (struct af_instance *af = s->first; af; af = af->next)
-        af->control(af, cmd, arg);
-}
-
-int af_control_by_label(struct af_stream *s, int cmd, void *arg, bstr label)
-{
-    char *label_str = bstrdup0(NULL, label);
-    struct af_instance *cur = af_find_by_label(s, label_str);
-    talloc_free(label_str);
-    if (cur) {
-        return cur->control ? cur->control(cur, cmd, arg) : CONTROL_NA;
-    } else {
-        return CONTROL_UNKNOWN;
-    }
-}
-
-int af_send_command(struct af_stream *s, char *label, char *cmd, char *arg)
-{
-    char *args[2] = {cmd, arg};
-    if (strcmp(label, "all") == 0) {
-        af_control_all(s, AF_CONTROL_COMMAND, args);
-        return 0;
-    } else {
-        return af_control_by_label(s, AF_CONTROL_COMMAND, args, bstr0(label));
-    }
-}
-
-// Used by filters to add a filtered frame to the output queue.
-// Ownership of frame is transferred from caller to the filter chain.
-void af_add_output_frame(struct af_instance *af, struct mp_audio *frame)
-{
-    if (frame) {
-        assert(mp_audio_config_equals(&af->fmt_out, frame));
-        MP_TARRAY_APPEND(af, af->out_queued, af->num_out_queued, frame);
-    }
-}
-
-static bool af_has_output_frame(struct af_instance *af)
-{
-    if (!af->num_out_queued && af->filter_out) {
-        if (af->filter_out(af) < 0)
-            MP_ERR(af, "Error filtering frame.\n");
-    }
-    return af->num_out_queued > 0;
-}
-
-static struct mp_audio *af_dequeue_output_frame(struct af_instance *af)
-{
-    struct mp_audio *res = NULL;
-    if (af_has_output_frame(af)) {
-        res = af->out_queued[0];
-        MP_TARRAY_REMOVE_AT(af->out_queued, af->num_out_queued, 0);
-    }
-    return res;
-}
-
-static void read_remaining(struct af_instance *af)
-{
-    int num_frames;
-    do {
-        num_frames = af->num_out_queued;
-        if (!af->filter_out || af->filter_out(af) < 0)
-            break;
-    } while (num_frames != af->num_out_queued);
-}
-
-static int af_do_filter(struct af_instance *af, struct mp_audio *frame)
-{
-    if (frame)
-        assert(mp_audio_config_equals(&af->fmt_in, frame));
-    int r = af->filter_frame(af, frame);
-    if (r < 0)
-        MP_ERR(af, "Error filtering frame.\n");
-    return r;
-}
-
-// Input a frame into the filter chain. Ownership of frame is transferred.
-// Return >= 0 on success, < 0 on failure (even if output frames were produced)
-int af_filter_frame(struct af_stream *s, struct mp_audio *frame)
-{
-    assert(frame);
-    if (s->initialized < 1) {
-        talloc_free(frame);
-        return -1;
-    }
-    return af_do_filter(s->first, frame);
-}
-
-// Output the next queued frame (if any) from the full filter chain.
-// The frame can be retrieved with af_read_output_frame().
-//  eof: if set, assume there's no more input i.e. af_filter_frame() will
-//       not be called (until reset) - flush all internally delayed frames
-//  returns: -1: error, 0: no output, 1: output available
-int af_output_frame(struct af_stream *s, bool eof)
-{
-    if (s->last->num_out_queued)
-        return 1;
-    if (s->initialized < 1)
-        return -1;
-    while (1) {
-        struct af_instance *last = NULL;
-        for (struct af_instance * cur = s->first; cur; cur = cur->next) {
-            // Flush remaining frames on EOF, but do that only if the previous
-            // filters have been flushed (i.e. they have no more output).
-            if (eof && !last) {
-                read_remaining(cur);
-                int r = af_do_filter(cur, NULL);
-                if (r < 0)
-                    return r;
-            }
-            if (af_has_output_frame(cur))
-                last = cur;
-        }
-        if (!last)
-            return 0;
-        if (!last->next)
-            return 1;
-        int r = af_do_filter(last->next, af_dequeue_output_frame(last));
-        if (r < 0)
-            return r;
-    }
-}
-
-struct mp_audio *af_read_output_frame(struct af_stream *s)
-{
-    if (!s->last->num_out_queued)
-        af_output_frame(s, false);
-    return af_dequeue_output_frame(s->last);
-}
-
-void af_unread_output_frame(struct af_stream *s, struct mp_audio *frame)
-{
-    struct af_instance *af = s->last;
-    MP_TARRAY_INSERT_AT(af, af->out_queued, af->num_out_queued, 0, frame);
-}
-
-// Make sure the caller can change data referenced by the frame.
-// Return negative error code on failure (i.e. you can't write).
-int af_make_writeable(struct af_instance *af, struct mp_audio *frame)
-{
-    return mp_audio_pool_make_writeable(af->out_pool, frame);
-}
-
-void af_seek_reset(struct af_stream *s)
-{
-    af_control_all(s, AF_CONTROL_RESET, NULL);
-    af_chain_forget_frames(s);
-}
@@ -1,161 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#ifndef MPLAYER_AF_H
-#define MPLAYER_AF_H
-
-#include <stdio.h>
-#include <stdbool.h>
-#include <sys/types.h>
-
-#include "config.h"
-#if !(HAVE_LIBAF && HAVE_GPL)
-#error "libaf/GPL disabled"
-#endif
-
-#include "options/options.h"
-#include "audio/format.h"
-#include "audio/chmap.h"
-#include "audio/audio.h"
-#include "common/msg.h"
-#include "common/common.h"
-
-struct af_instance;
-struct mpv_global;
-
-// Number of channels
-#define AF_NCH MP_NUM_CHANNELS
-
-// Flags for af->filter()
-#define AF_FILTER_FLAG_EOF 1
-
-/* Audio filter information not specific for current instance, but for
-   a specific filter */
-struct af_info {
-    const char *info;
-    const char *name;
-    int (*open)(struct af_instance *vf);
-    int priv_size;
-    const void *priv_defaults;
-    const struct m_option *options;
-};
-
-// Linked list of audio filters
-struct af_instance {
-    const struct af_info *info;
-    char *full_name;
-    struct mp_log *log;
-    struct MPOpts *opts;
-    struct mpv_global *global;
-    int (*control)(struct af_instance *af, int cmd, void *arg);
-    void (*uninit)(struct af_instance *af);
-    /* Feed a frame. The frame is NULL if EOF was reached, and the filter
-     * should drain all remaining buffered data.
-     * Use af_add_output_frame() to output data. The optional filter_out
-     * callback can be set to produce output frames gradually.
-     */
-    int (*filter_frame)(struct af_instance *af, struct mp_audio *frame);
-    int (*filter_out)(struct af_instance *af);
-    void *priv;
-    struct mp_audio *data; // configuration and buffer for outgoing data stream
-
-    struct af_instance *next;
-    struct af_instance *prev;
-    double delay; /* Delay caused by the filter, in seconds of audio consumed
-                   * without corresponding output */
-    bool auto_inserted; // inserted by af.c, such as conversion filters
-    char *label;
-
-    struct mp_audio fmt_in, fmt_out;
-
-    struct mp_audio **out_queued;
-    int num_out_queued;
-
-    struct mp_audio_pool *out_pool;
-};
-
-// Current audio stream
-struct af_stream {
-    int initialized; // 0: no, 1: yes, -1: attempted to, but failed
-
-    // The first and last filter in the list
-    struct af_instance *first;
-    struct af_instance *last;
-    // The user sets the input format (what the decoder outputs), and sets some
-    // or all fields in output to the output format the AO accepts.
-    struct mp_audio input;
-    struct mp_audio output;
-    struct mp_audio filter_output;
-
-    struct mp_log *log;
-    struct MPOpts *opts;
-    struct mpv_global *global;
-};
-
-// Return values
-#define AF_DETACH   (CONTROL_OK + 1)
-#define AF_OK       CONTROL_OK
-#define AF_TRUE     CONTROL_TRUE
-#define AF_FALSE    CONTROL_FALSE
-#define AF_UNKNOWN  CONTROL_UNKNOWN
-#define AF_ERROR    CONTROL_ERROR
-
-// Parameters for af_control_*
-enum af_control {
-    AF_CONTROL_REINIT = 1,
-    AF_CONTROL_RESET,
-    AF_CONTROL_SET_PLAYBACK_SPEED,
-    AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE,
-    AF_CONTROL_GET_METADATA,
-    AF_CONTROL_COMMAND,
-};
-
-// Argument for AF_CONTROL_SET_PAN_LEVEL
-typedef struct af_control_ext_s {
-    void* arg;  // Argument
-    int ch;     // Chanel number
-} af_control_ext_t;
-
-struct af_stream *af_new(struct mpv_global *global);
-void af_destroy(struct af_stream *s);
-int af_init(struct af_stream *s);
-void af_uninit(struct af_stream *s);
-struct af_instance *af_add(struct af_stream *s, char *name, char *label,
-                           char **args);
-int af_remove_by_label(struct af_stream *s, char *label);
-struct af_instance *af_find_by_label(struct af_stream *s, char *label);
-struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg);
-void af_control_all(struct af_stream *s, int cmd, void *arg);
-int af_control_by_label(struct af_stream *s, int cmd, void *arg, bstr label);
-void af_seek_reset(struct af_stream *s);
-int af_send_command(struct af_stream *s, char *label, char *cmd, char *arg);
-
-void af_add_output_frame(struct af_instance *af, struct mp_audio *frame);
-int af_filter_frame(struct af_stream *s, struct mp_audio *frame);
-int af_output_frame(struct af_stream *s, bool eof);
-struct mp_audio *af_read_output_frame(struct af_stream *s);
-void af_unread_output_frame(struct af_stream *s, struct mp_audio *frame);
-int af_make_writeable(struct af_instance *af, struct mp_audio *frame);
-
-double af_calc_delay(struct af_stream *s);
-
-int af_test_output(struct af_instance *af, struct mp_audio *out);
-
-int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma);
-float af_softclip(float a);
-
-#endif /* MPLAYER_AF_H */
@@ -15,18 +15,14 @@
 * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
 */

-#include <stdlib.h>
-
-#include <libavutil/common.h>
-
+#include "audio/aframe.h"
+#include "audio/format.h"
+#include "filters/f_autoconvert.h"
+#include "filters/filter_internal.h"
+#include "filters/user_filters.h"
 #include "options/m_option.h"

-#include "audio/format.h"
-#include "af.h"
-
-struct priv {
-    struct m_config *config;
-
+struct f_opts {
    int in_format;
    int in_srate;
    struct m_channels in_channels;
@@ -37,98 +33,109 @@ struct priv {
    int fail;
 };

-static void force_in_params(struct af_instance *af, struct mp_audio *in)
+struct priv {
+    struct f_opts *opts;
+    struct mp_pin *in_pin;
+};
+
+static void process(struct mp_filter *f)
 {
-    struct priv *priv = af->priv;
+    struct priv *p = f->priv;
+
+    if (!mp_pin_can_transfer_data(f->ppins[1], p->in_pin))
+        return;

-    if (priv->in_format != AF_FORMAT_UNKNOWN)
-        mp_audio_set_format(in, priv->in_format);
+    struct mp_frame frame = mp_pin_out_read(p->in_pin);

-    if (priv->in_channels.num_chmaps > 0)
-        mp_audio_set_channels(in, &priv->in_channels.chmaps[0]);
+    if (p->opts->fail) {
+        MP_ERR(f, "Failing on purpose.\n");
+        goto error;
+    }

-    if (priv->in_srate)
-        in->rate = priv->in_srate;
-}
+    if (frame.type == MP_FRAME_EOF) {
+        mp_pin_in_write(f->ppins[1], frame);
+        return;
+    }

-static void force_out_params(struct af_instance *af, struct mp_audio *out)
-{
-    struct priv *priv = af->priv;
+    if (frame.type != MP_FRAME_AUDIO) {
+        MP_ERR(f, "audio frame expected\n");
+        goto error;
+    }

-    if (priv->out_format != AF_FORMAT_UNKNOWN)
-        mp_audio_set_format(out, priv->out_format);
+    struct mp_aframe *in = frame.data;

-    if (priv->out_channels.num_chmaps > 0)
-        mp_audio_set_channels(out, &priv->out_channels.chmaps[0]);
+    if (p->opts->out_channels.num_chmaps > 0) {
+        if (!mp_aframe_set_chmap(in, &p->opts->out_channels.chmaps[0])) {
+            MP_ERR(f, "could not force output channels\n");
+            goto error;
+        }
+    }

-    if (priv->out_srate)
-        out->rate = priv->out_srate;
-}
+    if (p->opts->out_srate)
+        mp_aframe_set_rate(in, p->opts->out_srate);

-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    struct priv *priv = af->priv;
+    mp_pin_in_write(f->ppins[1], frame);
+    return;

-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-        struct mp_audio *out = af->data;
+error:
+    mp_frame_unref(&frame);
+    mp_filter_internal_mark_failed(f);
+}

-        force_in_params(af, in);
-        mp_audio_copy_config(out, in);
-        force_out_params(af, out);
+static const struct mp_filter_info af_format_filter = {
+    .name = "format",
+    .priv_size = sizeof(struct priv),
+    .process = process,
+};

-        if (in->nch != out->nch || in->bps != out->bps) {
-            MP_ERR(af, "Forced input/output formats are incompatible.\n");
-            return AF_ERROR;
-        }
+static struct mp_filter *af_format_create(struct mp_filter *parent,
+                                              void *options)
+{
+    struct mp_filter *f = mp_filter_create(parent, &af_format_filter);
+    if (!f) {
+        talloc_free(options);
+        return NULL;
+    }

-        if (priv->fail) {
-            MP_ERR(af, "Failing on purpose.\n");
-            return AF_ERROR;
-        }
+    struct priv *p = f->priv;
+    p->opts = talloc_steal(p, options);

-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
-    }
-    }
-    return AF_UNKNOWN;
-}
+    mp_filter_add_pin(f, MP_PIN_IN, "in");
+    mp_filter_add_pin(f, MP_PIN_OUT, "out");

-static int filter(struct af_instance *af, struct mp_audio *data)
-{
-    if (data)
-        mp_audio_copy_config(data, af->data);
-    af_add_output_frame(af, data);
-    return 0;
-}
+    struct mp_autoconvert *conv = mp_autoconvert_create(f);
+    if (!conv)
+        abort();

-static int af_open(struct af_instance *af)
-{
-    af->control = control;
-    af->filter_frame = filter;
+    if (p->opts->in_format)
+        mp_autoconvert_add_afmt(conv, p->opts->in_format);
+    if (p->opts->in_srate)
+        mp_autoconvert_add_srate(conv, p->opts->in_srate);
+    if (p->opts->in_channels.num_chmaps > 0)
+        mp_autoconvert_add_chmap(conv, &p->opts->in_channels.chmaps[0]);

-    force_in_params(af, af->data);
-    force_out_params(af, af->data);
+    mp_pin_connect(conv->f->pins[0], f->ppins[0]);
+    p->in_pin = conv->f->pins[1];

-    return AF_OK;
+    return f;
 }

-#define OPT_BASE_STRUCT struct priv
-
-const struct af_info af_info_format = {
-    .info = "Force audio format",
-    .name = "format",
-    .open = af_open,
-    .priv_size = sizeof(struct priv),
-    .options = (const struct m_option[]) {
-        OPT_AUDIOFORMAT("format", in_format, 0),
-        OPT_INTRANGE("srate", in_srate, 0, 1000, 8*48000),
-        OPT_CHANNELS("channels", in_channels, 0, .min = 1),
-        OPT_AUDIOFORMAT("out-format", out_format, 0),
-        OPT_INTRANGE("out-srate", out_srate, 0, 1000, 8*48000),
-        OPT_CHANNELS("out-channels", out_channels, 0, .min = 1),
-        OPT_FLAG("fail", fail, 0),
-        {0}
+#define OPT_BASE_STRUCT struct f_opts
+
+const struct mp_user_filter_entry af_format = {
+    .desc = {
+        .name = "format",
+        .description = "Force audio format",
+        .priv_size = sizeof(struct f_opts),
+        .options = (const struct m_option[]) {
+            OPT_AUDIOFORMAT("format", in_format, 0),
+            OPT_INTRANGE("srate", in_srate, 0, 1000, 8*48000),
+            OPT_CHANNELS("channels", in_channels, 0, .min = 1),
+            OPT_INTRANGE("out-srate", out_srate, 0, 1000, 8*48000),
+            OPT_CHANNELS("out-channels", out_channels, 0, .min = 1),
+            OPT_FLAG("fail", fail, 0),
+            {0}
+        },
    },
+    .create = af_format_create,
 };
@@ -31,14 +31,17 @@
 #include <libavutil/bswap.h>
 #include <libavutil/mem.h>

-#include "config.h"
-
-#include "common/av_common.h"
-#include "common/common.h"
-#include "af.h"
-#include "audio/audio_buffer.h"
+#include "audio/aframe.h"
 #include "audio/chmap_sel.h"
 #include "audio/fmt-conversion.h"
+#include "audio/format.h"
+#include "common/av_common.h"
+#include "common/common.h"
+#include "filters/f_autoconvert.h"
+#include "filters/f_utils.h"
+#include "filters/filter_internal.h"
+#include "filters/user_filters.h"
+#include "options/m_option.h"


 #define AC3_MAX_CHANNELS 6
@@ -49,173 +52,89 @@ const uint16_t ac3_bitrate_tab[19] = {
    160, 192, 224, 256, 320, 384, 448, 512, 576, 640
 };

-// Data for specific instances of this filter
-typedef struct af_ac3enc_s {
+struct f_opts {
+    int add_iec61937_header;
+    int bit_rate;
+    int min_channel_num;
+    char *encoder;
+    char **avopts;
+};
+
+struct priv {
+    struct f_opts *opts;
+
+    struct mp_pin *in_pin;
+    struct mp_aframe *cur_format;
+    struct mp_aframe *in_frame;
+    struct mp_aframe_pool *out_pool;
+
    struct AVCodec        *lavc_acodec;
    struct AVCodecContext *lavc_actx;
    int bit_rate;
-    struct mp_audio *input;     // frame passed to libavcodec
-    struct mp_audio *pending;   // unconsumed input data
-    int in_samples;     // samples of input per AC3 frame
    int out_samples;    // upper bound on encoded output per AC3 frame
-    int64_t encoder_buffered;
+};

-    int cfg_add_iec61937_header;
-    int cfg_bit_rate;
-    int cfg_min_channel_num;
-    char *cfg_encoder;
-    char **cfg_avopts;
-} af_ac3enc_t;
-
-// fmt carries the input format. Change it to the best next-possible format
-// the encoder likely accepts.
-static void select_encode_format(AVCodecContext *c, struct mp_audio *fmt)
+static bool reinit(struct mp_filter *f)
 {
-    int formats[AF_FORMAT_COUNT];
-    af_get_best_sample_formats(fmt->format, formats);
+    struct priv *s = f->priv;

-    for (int n = 0; formats[n]; n++) {
-        const enum AVSampleFormat *lf = c->codec->sample_fmts;
-        for (int i = 0; lf && lf[i] != AV_SAMPLE_FMT_NONE; i++) {
-            int mpfmt = af_from_avformat(lf[i]);
-            if (mpfmt && mpfmt == formats[n]) {
-                mp_audio_set_format(fmt, mpfmt);
-                goto done_fmt;
-            }
-        }
-    }
-done_fmt: ;
+    mp_aframe_reset(s->cur_format);

-    int rate =
-        af_select_best_samplerate(fmt->rate, c->codec->supported_samplerates);
-    if (rate > 0)
-        fmt->rate = rate;
-
-    struct mp_chmap_sel sel = {0};
-    const uint64_t *lch = c->codec->channel_layouts;
-    for (int n = 0; lch && lch[n]; n++) {
-        struct mp_chmap chmap = {0};
-        mp_chmap_from_lavc(&chmap, lch[n]);
-        mp_chmap_sel_add_map(&sel, &chmap);
-    }
-    struct mp_chmap res = fmt->channels;
-    mp_chmap_sel_adjust(&sel, &res);
-    if (!mp_chmap_is_empty(&res))
-        mp_audio_set_channels(fmt, &res);
-}
-
-// Initialization and runtime control
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    af_ac3enc_t *s  = af->priv;
    static const int default_bit_rate[AC3_MAX_CHANNELS+1] = \
        {0, 96000, 192000, 256000, 384000, 448000, 448000};

-    switch (cmd){
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-
-        if (!af_fmt_is_pcm(in->format) || in->nch < s->cfg_min_channel_num)
-            return AF_DETACH;
-
-        // At least currently, the AC3 encoder doesn't export sample rates.
-        in->rate = 48000;
-        select_encode_format(s->lavc_actx, in);
-
-        af->data->rate = in->rate;
-        mp_audio_set_format(af->data, AF_FORMAT_S_AC3);
-        mp_audio_set_num_channels(af->data, 2);
-
-        if (!mp_audio_config_equals(in, &orig_in))
-            return AF_FALSE;
-
-        if (s->cfg_add_iec61937_header) {
-            s->out_samples = AC3_FRAME_SIZE;
-        } else {
-            s->out_samples = AC3_MAX_CODED_FRAME_SIZE / af->data->sstride;
-        }
-
-        mp_audio_copy_config(s->input, in);
-
-        talloc_free(s->pending);
-        s->pending = NULL;
-
-        MP_DBG(af, "reinit: %d, %d, %d.\n", in->nch, in->rate, s->in_samples);
-
-        int bit_rate = s->bit_rate ? s->bit_rate : default_bit_rate[in->nch];
-
-        if (s->lavc_actx->channels != in->nch ||
-            s->lavc_actx->sample_rate != in->rate ||
-            s->lavc_actx->bit_rate != bit_rate)
-        {
-            avcodec_close(s->lavc_actx);
-
-            // Put sample parameters
-            s->lavc_actx->sample_fmt = af_to_avformat(in->format);
-            s->lavc_actx->channels = in->nch;
-            s->lavc_actx->channel_layout = mp_chmap_to_lavc(&in->channels);
-            s->lavc_actx->sample_rate = in->rate;
-            s->lavc_actx->bit_rate = bit_rate;
-
-            if (avcodec_open2(s->lavc_actx, s->lavc_acodec, NULL) < 0) {
-                MP_ERR(af, "Couldn't open codec %s, br=%d.\n", "ac3", bit_rate);
-                return AF_ERROR;
-            }
-
-            if (s->lavc_actx->frame_size < 1) {
-                MP_ERR(af, "encoder didn't specify input frame size\n");
-                return AF_ERROR;
-            }
-        }
-        s->in_samples = s->lavc_actx->frame_size;
-        mp_audio_realloc(s->input, s->in_samples);
-        s->input->samples = 0;
-        s->encoder_buffered = 0;
-        return AF_OK;
+    if (s->opts->add_iec61937_header) {
+        s->out_samples = AC3_FRAME_SIZE;
+    } else {
+        s->out_samples = AC3_MAX_CODED_FRAME_SIZE /
+                         mp_aframe_get_sstride(s->in_frame);
    }
-    case AF_CONTROL_RESET:
-        if (avcodec_is_open(s->lavc_actx))
-            avcodec_flush_buffers(s->lavc_actx);
-        talloc_free(s->pending);
-        s->pending = NULL;
-        s->input->samples = 0;
-        s->encoder_buffered = 0;
-        return AF_OK;
+
+    int format = mp_aframe_get_format(s->in_frame);
+    int rate = mp_aframe_get_rate(s->in_frame);
+    struct mp_chmap chmap = {0};
+    mp_aframe_get_chmap(s->in_frame, &chmap);
+
+    int bit_rate = s->bit_rate;
+    if (!bit_rate && chmap.num < AC3_MAX_CHANNELS + 1)
+        bit_rate = default_bit_rate[chmap.num];
+
+    avcodec_close(s->lavc_actx);
+
+    // Put sample parameters
+    s->lavc_actx->sample_fmt = af_to_avformat(format);
+    s->lavc_actx->channels = chmap.num;
+    s->lavc_actx->channel_layout = mp_chmap_to_lavc(&chmap);
+    s->lavc_actx->sample_rate = rate;
+    s->lavc_actx->bit_rate = bit_rate;
+
+    if (avcodec_open2(s->lavc_actx, s->lavc_acodec, NULL) < 0) {
+        MP_ERR(f, "Couldn't open codec %s, br=%d.\n", "ac3", bit_rate);
+        return false;
    }
-    return AF_UNKNOWN;
+
+    if (s->lavc_actx->frame_size < 1) {
+        MP_ERR(f, "encoder didn't specify input frame size\n");
+        return false;
+    }
+
+    mp_aframe_config_copy(s->cur_format, s->in_frame);
+    return true;
 }

-// Deallocate memory
-static void uninit(struct af_instance* af)
+static void reset(struct mp_filter *f)
 {
-    af_ac3enc_t *s = af->priv;
+    struct priv *s = f->priv;

-    if (s) {
-        avcodec_free_context(&s->lavc_actx);
-        talloc_free(s->pending);
-    }
+    TA_FREEP(&s->in_frame);
 }

-static void update_delay(struct af_instance *af)
+static void destroy(struct mp_filter *f)
 {
-    af_ac3enc_t *s = af->priv;
-    af->delay = ((s->pending ? s->pending->samples : 0) + s->input->samples +
-                  s->encoder_buffered) / (double)s->input->rate;
-}
+    struct priv *s = f->priv;

-static int filter_frame(struct af_instance *af, struct mp_audio *audio)
-{
-    af_ac3enc_t *s = af->priv;
-
-    // filter_output must have been called until no output was produced.
-    if (s->pending && s->pending->samples)
-        MP_ERR(af, "broken data flow\n");
-
-    talloc_free(s->pending);
-    s->pending = audio;
-    update_delay(af);
-    return 0;
+    reset(f);
+    avcodec_free_context(&s->lavc_actx);
 }

 static void swap_16(uint16_t *ptr, size_t size)
@@ -224,105 +143,84 @@ static void swap_16(uint16_t *ptr, size_t size)
        ptr[n] = av_bswap16(ptr[n]);
 }

-// Copy data from input frame to encode frame (because libavcodec wants a full
-// AC3 frame for encoding, while filter input frames can be smaller or larger).
-// Return true if the frame is complete.
-static bool fill_buffer(struct af_instance *af)
+static void process(struct mp_filter *f)
 {
-    af_ac3enc_t *s = af->priv;
+    struct priv *s = f->priv;

-    af->delay = 0;
-
-    if (s->pending) {
-        if (!mp_audio_is_writeable(s->input))
-            assert(s->input->samples == 0); // we can't have sent a partial frame
-        mp_audio_realloc_min(s->input, s->in_samples);
-        int copy = MPMIN(s->in_samples - s->input->samples, s->pending->samples);
-        s->input->samples += copy;
-        mp_audio_copy(s->input, s->input->samples - copy, s->pending, 0, copy);
-        mp_audio_skip_samples(s->pending, copy);
-    }
-    update_delay(af);
-    return s->input->samples >= s->in_samples;
-}
-
-// Return <0 on error, 0 on need more input, 1 on success (and *frame set).
-// To actually advance the read pointer, set s->input->samples=0 afterwards.
-static int read_input_frame(struct af_instance *af, AVFrame *frame)
-{
-    af_ac3enc_t *s = af->priv;
-    if (!fill_buffer(af))
-        return 0; // need more input
-
-    if (mp_audio_to_avframe(s->input, frame) < 0)
-        return -1;
-
-    return 1;
-}
-
-static int filter_out(struct af_instance *af)
-{
-    af_ac3enc_t *s = af->priv;
-
-    if (!s->pending)
-        return 0;
-
-    AVFrame *frame = av_frame_alloc();
-    if (!frame) {
-        MP_FATAL(af, "Could not allocate memory \n");
-        return -1;
-    }
-    int err = -1;
+    if (!mp_pin_in_needs_data(f->ppins[1]))
+        return;

+    bool err = true;
+    struct mp_aframe *out = NULL;
    AVPacket pkt = {0};
    av_init_packet(&pkt);

    // Send input as long as it wants.
    while (1) {
-        err = read_input_frame(af, frame);
-        if (err < 0)
+        if (avcodec_is_open(s->lavc_actx)) {
+            int lavc_ret = avcodec_receive_packet(s->lavc_actx, &pkt);
+            if (lavc_ret >= 0)
+                break;
+            if (lavc_ret < 0 && lavc_ret != AVERROR(EAGAIN)) {
+                MP_FATAL(f, "Encode failed (receive).\n");
+                goto done;
+            }
+        }
+        AVFrame *frame = NULL;
+        struct mp_frame input = mp_pin_out_read(s->in_pin);
+        // The following code assumes no sample data buffering in the encoder.
+        if (input.type == MP_FRAME_EOF) {
+            mp_pin_in_write(f->ppins[1], input);
+            return;
+        } else if (input.type == MP_FRAME_AUDIO) {
+            TA_FREEP(&s->in_frame);
+            s->in_frame = input.data;
+            frame = mp_frame_to_av(input, NULL);
+            if (!frame)
+                goto done;
+            if (mp_aframe_get_channels(s->in_frame) < s->opts->min_channel_num) {
+                // Just pass it through.
+                s->in_frame = NULL;
+                mp_pin_in_write(f->ppins[1], input);
+                return;
+            }
+            if (!mp_aframe_config_equals(s->in_frame, s->cur_format)) {
+                if (!reinit(f))
+                    goto done;
+            }
+        } else if (input.type) {
            goto done;
-        if (err == 0)
-            break;
-        err = -1;
+        } else {
+            return; // no data yet
+        }
        int lavc_ret = avcodec_send_frame(s->lavc_actx, frame);
-        // On EAGAIN, we're supposed to read remaining output.
-        if (lavc_ret == AVERROR(EAGAIN))
-            break;
-        if (lavc_ret < 0) {
-            MP_FATAL(af, "Encode failed.\n");
+        av_frame_free(&frame);
+        if (lavc_ret < 0 && lavc_ret != AVERROR(EAGAIN)) {
+            MP_FATAL(f, "Encode failed (send).\n");
            goto done;
        }
-        s->encoder_buffered += s->input->samples;
-        s->input->samples = 0;
-    }
-    int lavc_ret = avcodec_receive_packet(s->lavc_actx, &pkt);
-    if (lavc_ret == AVERROR(EAGAIN)) {
-        // Need to buffer more input.
-        err = 0;
-        goto done;
-    }
-    if (lavc_ret < 0) {
-        MP_FATAL(af, "Encode failed.\n");
-        goto done;
    }

-    MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
-           pkt.size, s->pending->samples + s->input->samples);
-
-    s->encoder_buffered -= AC3_FRAME_SIZE;
-
-    struct mp_audio *out =
-        mp_audio_pool_get(af->out_pool, af->data, s->out_samples);
-    if (!out)
+    if (!s->in_frame)
        goto done;
-    mp_audio_copy_attributes(out, s->pending);
+
+    out = mp_aframe_create();
+    mp_aframe_set_format(out, AF_FORMAT_S_AC3);
+    mp_aframe_set_chmap(out, &(struct mp_chmap)MP_CHMAP_INIT_STEREO);
+    mp_aframe_set_rate(out, 48000);
+
+    if (mp_aframe_pool_allocate(s->out_pool, out, s->out_samples) < 0)
+        goto done;
+
+    int sstride = mp_aframe_get_sstride(out);
+
+    mp_aframe_copy_attributes(out, s->in_frame);

    int frame_size = pkt.size;
    int header_len = 0;
    char hdr[8];

-    if (s->cfg_add_iec61937_header && pkt.size > 5) {
+    if (s->opts->add_iec61937_header && pkt.size > 5) {
        int bsmod = pkt.data[5] & 0x7;
        int len = frame_size;

@@ -336,48 +234,69 @@ static int filter_out(struct af_instance *af)
        AV_WL16(hdr + 6, len << 3); // number of bits in payload
    }

-    if (frame_size > out->samples * out->sstride)
+    if (frame_size > s->out_samples * sstride)
        abort();

-    char *buf = (char *)out->planes[0];
+    uint8_t **planes = mp_aframe_get_data_rw(out);
+    if (!planes)
+        goto done;
+    char *buf = planes[0];
    memcpy(buf, hdr, header_len);
    memcpy(buf + header_len, pkt.data, pkt.size);
    memset(buf + header_len + pkt.size, 0,
           frame_size - (header_len + pkt.size));
    swap_16((uint16_t *)(buf + header_len), pkt.size / 2);
-    out->samples = frame_size / out->sstride;
-    af_add_output_frame(af, out);
+    mp_aframe_set_size(out, frame_size / sstride);
+    mp_pin_in_write(f->ppins[1], MAKE_FRAME(MP_FRAME_AUDIO, out));
+    out = NULL;

    err = 0;
 done:
    av_packet_unref(&pkt);
-    av_frame_free(&frame);
-    update_delay(af);
-    return err;
+    talloc_free(out);
+    if (err)
+        mp_filter_internal_mark_failed(f);
 }

-static int af_open(struct af_instance* af){
+static const struct mp_filter_info af_lavcac3enc_filter = {
+    .name = "lavcac3enc",
+    .priv_size = sizeof(struct priv),
+    .process = process,
+    .reset = reset,
+    .destroy = destroy,
+};

-    af_ac3enc_t *s = af->priv;
-    af->control=control;
-    af->uninit=uninit;
-    af->filter_frame = filter_frame;
-    af->filter_out = filter_out;
+static struct mp_filter *af_lavcac3enc_create(struct mp_filter *parent,
+                                              void *options)
+{
+    struct mp_filter *f = mp_filter_create(parent, &af_lavcac3enc_filter);
+    if (!f) {
+        talloc_free(options);
+        return NULL;
+    }

-    s->lavc_acodec = avcodec_find_encoder_by_name(s->cfg_encoder);
+    mp_filter_add_pin(f, MP_PIN_IN, "in");
+    mp_filter_add_pin(f, MP_PIN_OUT, "out");
+
+    struct priv *s = f->priv;
+    s->opts = talloc_steal(s, options);
+    s->cur_format = talloc_steal(s, mp_aframe_create());
+    s->out_pool = mp_aframe_pool_create(s);
+
+    s->lavc_acodec = avcodec_find_encoder_by_name(s->opts->encoder);
    if (!s->lavc_acodec) {
-        MP_ERR(af, "Couldn't find encoder %s.\n", s->cfg_encoder);
-        return AF_ERROR;
+        MP_ERR(f, "Couldn't find encoder %s.\n", s->opts->encoder);
+        goto error;
    }

    s->lavc_actx = avcodec_alloc_context3(s->lavc_acodec);
    if (!s->lavc_actx) {
-        MP_ERR(af, "Audio LAVC, couldn't allocate context!\n");
-        return AF_ERROR;
+        MP_ERR(f, "Audio LAVC, couldn't allocate context!\n");
+        goto error;
    }

-    if (mp_set_avopts(af->log, s->lavc_actx, s->cfg_avopts) < 0)
-        return AF_ERROR;
+    if (mp_set_avopts(f->log, s->lavc_actx, s->opts->avopts) < 0)
+        goto error;

    // For this one, we require the decoder to expert lists of all supported
    // parameters. (Not all decoders do that, but the ones we're interested
@@ -385,50 +304,85 @@ static int af_open(struct af_instance* af){
    if (!s->lavc_acodec->sample_fmts ||
        !s->lavc_acodec->channel_layouts)
    {
-        MP_ERR(af, "Audio encoder doesn't list supported parameters.\n");
-        return AF_ERROR;
+        MP_ERR(f, "Audio encoder doesn't list supported parameters.\n");
+        goto error;
    }

-    s->input = talloc_zero(s, struct mp_audio);
-
-    if (s->cfg_bit_rate) {
+    if (s->opts->bit_rate) {
        int i;
        for (i = 0; i < 19; i++) {
-            if (ac3_bitrate_tab[i] == s->cfg_bit_rate) {
+            if (ac3_bitrate_tab[i] == s->opts->bit_rate) {
                s->bit_rate = ac3_bitrate_tab[i] * 1000;
                break;
            }
        }
        if (i >= 19) {
-            MP_WARN(af, "unable set unsupported bitrate %d, using default "
+            MP_WARN(f, "unable set unsupported bitrate %d, using default "
                    "bitrate (check manpage to see supported bitrates).\n",
-                    s->cfg_bit_rate);
+                    s->opts->bit_rate);
        }
    }

-    return AF_OK;
+    struct mp_autoconvert *conv = mp_autoconvert_create(f);
+    if (!conv)
+        abort();
+
+    const enum AVSampleFormat *lf = s->lavc_acodec->sample_fmts;
+    for (int i = 0; lf && lf[i] != AV_SAMPLE_FMT_NONE; i++) {
+        int mpfmt = af_from_avformat(lf[i]);
+        if (mpfmt)
+            mp_autoconvert_add_afmt(conv, mpfmt);
+    }
+
+    const uint64_t *lch = s->lavc_acodec->channel_layouts;
+    for (int n = 0; lch && lch[n]; n++) {
+        struct mp_chmap chmap = {0};
+        mp_chmap_from_lavc(&chmap, lch[n]);
+        if (mp_chmap_is_valid(&chmap))
+            mp_autoconvert_add_chmap(conv, &chmap);
+    }
+
+    // At least currently, the AC3 encoder doesn't export sample rates.
+    mp_autoconvert_add_srate(conv, 48000);
+
+    mp_pin_connect(conv->f->pins[0], f->ppins[0]);
+
+    struct mp_filter *fs = mp_fixed_aframe_size_create(f, AC3_FRAME_SIZE, true);
+    if (!fs)
+        abort();
+
+    mp_pin_connect(fs->pins[0], conv->f->pins[1]);
+    s->in_pin = fs->pins[1];
+
+    return f;
+
+error:
+    talloc_free(f);
+    return NULL;
 }

-#define OPT_BASE_STRUCT struct af_ac3enc_s
+#define OPT_BASE_STRUCT struct f_opts

-const struct af_info af_info_lavcac3enc = {
-    .info = "runtime encode to ac3 using libavcodec",
-    .name = "lavcac3enc",
-    .open = af_open,
-    .priv_size = sizeof(struct af_ac3enc_s),
-    .priv_defaults = &(const struct af_ac3enc_s){
-        .cfg_add_iec61937_header = 1,
-        .cfg_bit_rate = 640,
-        .cfg_min_channel_num = 3,
-        .cfg_encoder = "ac3",
-    },
-    .options = (const struct m_option[]) {
-        OPT_FLAG("tospdif", cfg_add_iec61937_header, 0),
-        OPT_CHOICE_OR_INT("bitrate", cfg_bit_rate, 0, 32, 640,
-                          ({"auto", 0}, {"default", 0})),
-        OPT_INTRANGE("minch", cfg_min_channel_num, 0, 2, 6),
-        OPT_STRING("encoder", cfg_encoder, 0),
-        OPT_KEYVALUELIST("o", cfg_avopts, 0),
-        {0}
+const struct mp_user_filter_entry af_lavcac3enc = {
+    .desc = {
+        .description = "runtime encode to ac3 using libavcodec",
+        .name = "lavcac3enc",
+        .priv_size = sizeof(OPT_BASE_STRUCT),
+        .priv_defaults = &(const OPT_BASE_STRUCT) {
+            .add_iec61937_header = 1,
+            .bit_rate = 640,
+            .min_channel_num = 3,
+            .encoder = "ac3",
+        },
+        .options = (const struct m_option[]) {
+            OPT_FLAG("tospdif", add_iec61937_header, 0),
+            OPT_CHOICE_OR_INT("bitrate", bit_rate, 0, 32, 640,
+                            ({"auto", 0}, {"default", 0})),
+            OPT_INTRANGE("minch", min_channel_num, 0, 2, 6),
+            OPT_STRING("encoder", encoder, 0),
+            OPT_KEYVALUELIST("o", avopts, 0),
+            {0}
+        },
    },
+    .create = af_lavcac3enc_create,
 };
@@ -1,413 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * Filter graph creation code taken from FFmpeg ffplay.c (LGPL 2.1 or later)
- *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdlib.h>
-#include <stdint.h>
-#include <stdio.h>
-#include <assert.h>
-
-#include <libavutil/avstring.h>
-#include <libavutil/mem.h>
-#include <libavutil/mathematics.h>
-#include <libavutil/rational.h>
-#include <libavutil/samplefmt.h>
-#include <libavutil/time.h>
-#include <libavutil/opt.h>
-#include <libavfilter/avfilter.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-
-#include "config.h"
-
-#include "audio/format.h"
-#include "audio/fmt-conversion.h"
-#include "af.h"
-
-#include "common/av_common.h"
-#include "common/tags.h"
-
-#include "options/m_option.h"
-
-// FFmpeg and Libav have slightly different APIs, just enough to cause us
-// unnecessary pain. <Expletive deleted.>
-#if LIBAVFILTER_VERSION_MICRO < 100
-#define graph_parse(graph, filters, inputs, outputs, log_ctx) \
-    avfilter_graph_parse(graph, filters, inputs, outputs, log_ctx)
-#define avfilter_graph_send_command(a, b, c, d, e, f, g) -1
-#else
-#define graph_parse(graph, filters, inputs, outputs, log_ctx) \
-    avfilter_graph_parse_ptr(graph, filters, &(inputs), &(outputs), log_ctx)
-#endif
-
-struct priv {
-    // Single filter bridge, instead of a graph.
-    bool is_bridge;
-
-    AVFilterGraph *graph;
-    AVFilterContext *in;
-    AVFilterContext *out;
-
-    int64_t samples_in;
-
-    AVRational timebase_out;
-
-    bool eof;
-
-    struct mp_tags *metadata;
-
-    // options
-    char *cfg_graph;
-    char **cfg_avopts;
-    char *cfg_filter_name;
-    char **cfg_filter_opts;
-};
-
-static void destroy_graph(struct af_instance *af)
-{
-    struct priv *p = af->priv;
-    avfilter_graph_free(&p->graph);
-    p->in = p->out = NULL;
-    p->samples_in = 0;
-    p->eof = false;
-}
-
-static bool recreate_graph(struct af_instance *af, struct mp_audio *config)
-{
-    void *tmp = talloc_new(NULL);
-    struct priv *p = af->priv;
-    AVFilterContext *in = NULL, *out = NULL;
-    bool ok = false;
-
-    if (!p->is_bridge && bstr0(p->cfg_graph).len == 0) {
-        MP_FATAL(af, "lavfi: no filter graph set\n");
-        return false;
-    }
-
-    destroy_graph(af);
-
-    AVFilterGraph *graph = avfilter_graph_alloc();
-    if (!graph)
-        goto error;
-
-    if (mp_set_avopts(af->log, graph, p->cfg_avopts) < 0)
-        goto error;
-
-    AVFilterInOut *outputs = avfilter_inout_alloc();
-    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    if (!outputs || !inputs)
-        goto error;
-
-    char *src_args = talloc_asprintf(tmp,
-        "sample_rate=%d:sample_fmt=%s:time_base=%d/%d:"
-        "channel_layout=0x%"PRIx64,  config->rate,
-        av_get_sample_fmt_name(af_to_avformat(config->format)),
-        1, config->rate, mp_chmap_to_lavc(&config->channels));
-
-    if (avfilter_graph_create_filter(&in, avfilter_get_by_name("abuffer"),
-                                     "src", src_args, NULL, graph) < 0)
-        goto error;
-
-    if (avfilter_graph_create_filter(&out, avfilter_get_by_name("abuffersink"),
-                                     "out", NULL, NULL, graph) < 0)
-        goto error;
-
-    if (p->is_bridge) {
-        AVFilterContext *filter = avfilter_graph_alloc_filter(graph,
-                        avfilter_get_by_name(p->cfg_filter_name), "filter");
-        if (!filter)
-            goto error;
-
-        if (mp_set_avopts(af->log, filter->priv, p->cfg_filter_opts) < 0)
-            goto error;
-
-        if (avfilter_init_str(filter, NULL) < 0)
-            goto error;
-
-        // Yep, we have to manually link those filters.
-        if (filter->nb_inputs != 1 ||
-            avfilter_pad_get_type(filter->input_pads, 0) != AVMEDIA_TYPE_AUDIO ||
-            filter->nb_outputs != 1 ||
-            avfilter_pad_get_type(filter->output_pads, 0) != AVMEDIA_TYPE_AUDIO)
-        {
-            MP_ERR(af, "The filter is required to have 1 audio input pad and "
-                       "1 audio output pad.\n");
-            goto error;
-        }
-        if (avfilter_link(in, 0, filter, 0) < 0 ||
-            avfilter_link(filter, 0, out, 0) < 0)
-        {
-            MP_ERR(af, "Failed to link filter.\n");
-            goto error;
-        }
-    } else {
-        MP_VERBOSE(af, "lavfi: create graph: '%s'\n", p->cfg_graph);
-
-        outputs->name = av_strdup("in");
-        outputs->filter_ctx = in;
-
-        inputs->name = av_strdup("out");
-        inputs->filter_ctx = out;
-
-        if (graph_parse(graph, p->cfg_graph, inputs, outputs, NULL) < 0)
-            goto error;
-    }
-
-    if (avfilter_graph_config(graph, NULL) < 0)
-        goto error;
-
-    p->in = in;
-    p->out = out;
-    p->graph = graph;
-
-    assert(out->nb_inputs == 1);
-    assert(in->nb_outputs == 1);
-
-    ok = true;
-error:
-
-    if (!ok) {
-        MP_FATAL(af, "Can't configure libavfilter graph.\n");
-        avfilter_graph_free(&graph);
-    }
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-    talloc_free(tmp);
-    return ok;
-}
-
-static void reset(struct af_instance *af)
-{
-    if (!recreate_graph(af, &af->fmt_in))
-        MP_FATAL(af, "Can't recreate libavfilter filter after a seek reset.\n");
-}
-
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    struct priv *p = af->priv;
-
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-        struct mp_audio *out = af->data;
-
-        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
-
-        if (!recreate_graph(af, in))
-            return AF_ERROR;
-
-        AVFilterLink *l_out = p->out->inputs[0];
-
-        out->rate = l_out->sample_rate;
-
-        mp_audio_set_format(out, af_from_avformat(l_out->format));
-
-        struct mp_chmap out_cm;
-        mp_chmap_from_lavc(&out_cm, l_out->channel_layout);
-        mp_audio_set_channels(out, &out_cm);
-
-        if (!mp_audio_config_valid(out) || out->channels.num > AV_NUM_DATA_POINTERS)
-            return AF_ERROR;
-
-        p->timebase_out = l_out->time_base;
-
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
-    }
-    case AF_CONTROL_COMMAND: {
-        if (!p->graph)
-            break;
-        char **args = arg;
-        return avfilter_graph_send_command(p->graph, "all",
-                                           args[0], args[1], &(char){0}, 0, 0)
-                >= 0 ? CONTROL_OK : CONTROL_ERROR;
-    }
-    case AF_CONTROL_GET_METADATA:
-        if (p->metadata) {
-            *(struct mp_tags *)arg = *p->metadata;
-            return CONTROL_OK;
-        }
-        return CONTROL_NA;
-    case AF_CONTROL_RESET:
-        reset(af);
-        return AF_OK;
-    }
-    return AF_UNKNOWN;
-}
-
-static void get_metadata_from_av_frame(struct af_instance *af, AVFrame *frame)
-{
-#if LIBAVUTIL_VERSION_MICRO >= 100
-    struct priv *p = af->priv;
-    if (!p->metadata)
-        p->metadata = talloc_zero(p, struct mp_tags);
-
-    mp_tags_copy_from_av_dictionary(p->metadata, frame->metadata);
-#endif
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *data)
-{
-    struct priv *p = af->priv;
-    AVFrame *frame = NULL;
-
-    if (p->eof && data)
-        reset(af);
-
-    if (!p->graph)
-        goto error;
-
-    if (!data) {
-        if (p->eof)
-            return 0;
-        p->eof = true;
-    }
-
-    if (data) {
-        frame = mp_audio_to_avframe_and_unref(data);
-        data = NULL;
-        if (!frame)
-            goto error;
-
-        // Timebase is 1/sample_rate
-        frame->pts = p->samples_in;
-        p->samples_in += frame->nb_samples;
-    }
-
-    if (av_buffersrc_add_frame(p->in, frame) < 0)
-        goto error;
-
-    av_frame_free(&frame);
-    talloc_free(data);
-    return 0;
-error:
-    av_frame_free(&frame);
-    talloc_free(data);
-    return -1;
-}
-
-static int filter_out(struct af_instance *af)
-{
-    struct priv *p = af->priv;
-
-    if (!p->graph)
-        goto error;
-
-    AVFrame *frame = av_frame_alloc();
-    if (!frame)
-        goto error;
-
-    int err = av_buffersink_get_frame(p->out, frame);
-    if (err == AVERROR(EAGAIN) || err == AVERROR_EOF) {
-        // Not an error situation - no more output buffers in queue.
-        // AVERROR_EOF means we shouldn't even give the filter more
-        // input, but we don't handle that completely correctly.
-        av_frame_free(&frame);
-        p->eof |= err == AVERROR_EOF;
-        return 0;
-    }
-
-    struct mp_audio *out = mp_audio_from_avframe(frame);
-    if (!out)
-        goto error;
-
-    mp_audio_copy_config(out, af->data);
-
-    if (frame->pts != AV_NOPTS_VALUE) {
-        double in_time = p->samples_in / (double)af->fmt_in.rate;
-        double out_time = frame->pts * av_q2d(p->timebase_out);
-        // Need pts past the last output sample.
-        out_time += out->samples / (double)out->rate;
-
-        af->delay = in_time - out_time;
-    }
-
-    get_metadata_from_av_frame(af, frame);
-    af_add_output_frame(af, out);
-    av_frame_free(&frame);
-    return 0;
-error:
-    av_frame_free(&frame);
-    return -1;
-}
-
-static void uninit(struct af_instance *af)
-{
-    destroy_graph(af);
-}
-
-static int af_open(struct af_instance *af)
-{
-    struct priv *p = af->priv;
-
-    af->control = control;
-    af->uninit = uninit;
-    af->filter_frame = filter_frame;
-    af->filter_out = filter_out;
-
-    if (p->is_bridge) {
-        if (!p->cfg_filter_name) {
-            MP_ERR(af, "Filter name not set!\n");
-            return 0;
-        }
-        if (!avfilter_get_by_name(p->cfg_filter_name)) {
-            MP_ERR(af, "libavfilter filter '%s' not found!\n", p->cfg_filter_name);
-            return 0;
-        }
-    }
-    return AF_OK;
-}
-
-#define OPT_BASE_STRUCT struct priv
-
-const struct af_info af_info_lavfi = {
-    .info = "libavfilter bridge",
-    .name = "lavfi",
-    .open = af_open,
-    .priv_size = sizeof(struct priv),
-    .options = (const struct m_option[]) {
-        OPT_STRING("graph", cfg_graph, 0),
-        OPT_KEYVALUELIST("o", cfg_avopts, 0),
-        {0}
-    },
-};
-
-const struct af_info af_info_lavfi_bridge = {
-    .info = "libavfilter bridge (explicit options)",
-    .name = "lavfi-bridge",
-    .open = af_open,
-    .priv_size = sizeof(struct priv),
-    .priv_defaults = &(const struct priv){
-        .is_bridge = true,
-    },
-    .options = (const struct m_option[]) {
-        OPT_STRING("name", cfg_filter_name, M_OPT_MIN, .min = 1),
-        OPT_KEYVALUELIST("opts", cfg_filter_opts, 0),
-        OPT_STRING("graph", cfg_graph, 0),
-        OPT_KEYVALUELIST("o", cfg_avopts, 0),
-        {0}
-    },
-};
@@ -32,142 +32,81 @@

 #include "common/av_common.h"
 #include "common/msg.h"
+#include "filters/f_swresample.h"
+#include "filters/filter_internal.h"
+#include "filters/user_filters.h"
+#include "options/m_config.h"
 #include "options/m_option.h"
-#include "audio/filter/af.h"
-#include "audio/fmt-conversion.h"
-#include "osdep/endian.h"
-#include "audio/aconverter.h"
+#include "options/options.h"

 struct af_resample {
    int allow_detach;
-    double playback_speed;
    struct mp_resample_opts opts;
-    struct mp_aconverter *converter;
+    int global_normalize;
 };

-static int control(struct af_instance *af, int cmd, void *arg)
+static void set_defaults(struct mpv_global *global, void *p)
 {
-    struct af_resample *s = af->priv;
+    struct af_resample *s = p;

-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio *out = af->data;
-        struct mp_audio orig_in = *in;
+    struct mp_resample_opts *opts = &s->opts;

-        if (((out->rate    == in->rate) || (out->rate == 0)) &&
-            (out->format   == in->format) &&
-            (mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
-            s->allow_detach && s->playback_speed == 1.0)
-            return AF_DETACH;
+    struct mp_resample_opts *src_opts =
+        mp_get_config_group(s, global, &resample_conf);

-        if (out->rate == 0)
-            out->rate = in->rate;
+    s->global_normalize = src_opts->normalize;

-        if (mp_chmap_is_empty(&out->channels))
-            mp_audio_set_channels(out, &in->channels);
+    assert(!opts->avopts); // we don't set a default value, so it must be NULL

-        if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
-            mp_audio_set_format(in, AF_FORMAT_FLOAT);
-        if (af_to_avformat(out->format) == AV_SAMPLE_FMT_NONE)
-            mp_audio_set_format(out, in->format);
+    *opts = *src_opts;

-        int r = ((in->format == orig_in.format) &&
-                mp_chmap_equals(&in->channels, &orig_in.channels))
-                ? AF_OK : AF_FALSE;
-
-        if (r == AF_OK) {
-            if (!mp_aconverter_reconfig(s->converter,
-                    in->rate, in->format, in->channels,
-                    out->rate, out->format, out->channels))
-                r = AF_ERROR;
-        }
-        return r;
-    }
-    case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
-        s->playback_speed = *(double *)arg;
-        return AF_OK;
-    }
-    case AF_CONTROL_RESET:
-        mp_aconverter_flush(s->converter);
-        return AF_OK;
-    }
-    return AF_UNKNOWN;
-}
-
-static void uninit(struct af_instance *af)
-{
-    struct af_resample *s = af->priv;
-
-    talloc_free(s->converter);
-}
-
-static int filter(struct af_instance *af, struct mp_audio *in)
-{
-    struct af_resample *s = af->priv;
-
-    mp_aconverter_set_speed(s->converter, s->playback_speed);
-
-    af->filter_out(af);
-
-    struct mp_aframe *aframe = mp_audio_to_aframe(in);
-    if (!aframe && in)
-        return -1;
-    talloc_free(in);
-    bool ok = mp_aconverter_write_input(s->converter, aframe);
-    if (!ok)
-        talloc_free(aframe);
-
-    return ok ? 0 : -1;
-}
-
-static int filter_out(struct af_instance *af)
-{
-    struct af_resample *s = af->priv;
-    bool eof;
-    struct mp_aframe *out = mp_aconverter_read_output(s->converter, &eof);
-    if (out)
-        af_add_output_frame(af, mp_audio_from_aframe(out));
-    talloc_free(out);
-    af->delay = mp_aconverter_get_latency(s->converter);
-    return 0;
-}
-
-static int af_open(struct af_instance *af)
-{
-    struct af_resample *s = af->priv;
-
-    af->control = control;
-    af->uninit  = uninit;
-    af->filter_frame = filter;
-    af->filter_out = filter_out;
-
-    s->converter = mp_aconverter_create(af->global, af->log, &s->opts);
-
-    return AF_OK;
+    opts->avopts = NULL;
+    struct m_option dummy = {.type = &m_option_type_keyvalue_list};
+    m_option_copy(&dummy, &opts->avopts, &src_opts->avopts);
 }

 #define OPT_BASE_STRUCT struct af_resample

-const struct af_info af_info_lavrresample = {
-    .info = "Sample frequency conversion using libavresample",
-    .name = "lavrresample",
-    .open = af_open,
-    .priv_size = sizeof(struct af_resample),
-    .priv_defaults = &(const struct af_resample) {
-        .opts = MP_RESAMPLE_OPTS_DEF,
-        .playback_speed = 1.0,
-        .allow_detach = 1,
-    },
-    .options = (const struct m_option[]) {
-        OPT_INTRANGE("filter-size", opts.filter_size, 0, 0, 32),
-        OPT_INTRANGE("phase-shift", opts.phase_shift, 0, 0, 30),
-        OPT_FLAG("linear", opts.linear, 0),
-        OPT_DOUBLE("cutoff", opts.cutoff, M_OPT_RANGE, .min = 0, .max = 1),
-        OPT_FLAG("detach", allow_detach, 0),
-        OPT_CHOICE("normalize", opts.normalize, 0,
-                   ({"no", 0}, {"yes", 1}, {"auto", -1})),
-        OPT_KEYVALUELIST("o", opts.avopts, 0),
-        {0}
+static struct mp_filter *af_lavrresample_create(struct mp_filter *parent,
+                                                void *options)
+{
+    struct af_resample *s = options;
+
+    if (s->opts.normalize < 0)
+        s->opts.normalize = s->global_normalize;
+
+    struct mp_swresample *swr = mp_swresample_create(parent, &s->opts);
+    if (!swr)
+        abort();
+
+    MP_WARN(swr->f, "This filter is deprecated! Use the --audio-resample- options"
+            " to customize resampling, or the --af=aresample filter.\n");
+
+    talloc_free(s);
+    return swr->f;
+}
+
+const struct mp_user_filter_entry af_lavrresample = {
+    .desc = {
+        .description = "Sample frequency conversion using libavresample",
+        .name = "lavrresample",
+        .priv_size = sizeof(struct af_resample),
+        .priv_defaults = &(const struct af_resample) {
+            .opts = MP_RESAMPLE_OPTS_DEF,
+            .allow_detach = 1,
+        },
+        .options = (const struct m_option[]) {
+            OPT_INTRANGE("filter-size", opts.filter_size, 0, 0, 32),
+            OPT_INTRANGE("phase-shift", opts.phase_shift, 0, 0, 30),
+            OPT_FLAG("linear", opts.linear, 0),
+            OPT_DOUBLE("cutoff", opts.cutoff, M_OPT_RANGE, .min = 0, .max = 1),
+            OPT_FLAG("detach", allow_detach, 0), // does nothing
+            OPT_CHOICE("normalize", opts.normalize, 0,
+                    ({"no", 0}, {"yes", 1}, {"auto", -1})),
+            OPT_KEYVALUELIST("o", opts.avopts, 0),
+            {0}
+        },
+        .set_defaults = set_defaults,
    },
+    .create = af_lavrresample_create,
 };
@@ -20,232 +20,350 @@

 #include <rubberband/rubberband-c.h>

+#include "audio/aframe.h"
+#include "audio/format.h"
 #include "common/common.h"
-#include "af.h"
+#include "filters/f_autoconvert.h"
+#include "filters/filter_internal.h"
+#include "filters/user_filters.h"
+#include "options/m_option.h"
+
+// command line options
+struct f_opts {
+    int transients, detector, phase, window,
+        smoothing, formant, pitch, channels;
+    double scale;
+};

 struct priv {
+    struct f_opts *opts;
+
+    struct mp_pin *in_pin;
+    struct mp_aframe *cur_format;
+    struct mp_aframe_pool *out_pool;
+    bool sent_final;
    RubberBandState rubber;
    double speed;
    double pitch;
-    struct mp_audio *pending;
-    bool needs_reset;
+    struct mp_aframe *pending;
    // Estimate how much librubberband has buffered internally.
    // I could not find a way to do this with the librubberband API.
    double rubber_delay;
-    // command line options
-    int opt_transients, opt_detector, opt_phase, opt_window,
-        opt_smoothing, opt_formant, opt_pitch, opt_channels;
 };

-static void update_speed(struct af_instance *af, double new_speed)
+static void update_speed(struct priv *p, double new_speed)
 {
-    struct priv *p = af->priv;
-
    p->speed = new_speed;
-    rubberband_set_time_ratio(p->rubber, 1.0 / p->speed);
+    if (p->rubber)
+        rubberband_set_time_ratio(p->rubber, 1.0 / p->speed);
 }

-static void update_pitch(struct af_instance *af, double new_pitch)
+static bool update_pitch(struct priv *p, double new_pitch)
 {
-    struct priv *p = af->priv;
+    if (new_pitch < 0.01 || new_pitch > 100.0)
+        return false;

    p->pitch = new_pitch;
-    rubberband_set_pitch_scale(p->rubber, p->pitch);
+    if (p->rubber)
+        rubberband_set_pitch_scale(p->rubber, p->pitch);
+    return true;
 }

-static int control(struct af_instance *af, int cmd, void *arg)
+static bool init_rubberband(struct mp_filter *f)
 {
-    struct priv *p = af->priv;
+    struct priv *p = f->priv;

-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-        struct mp_audio *out = af->data;
+    assert(!p->rubber);
+    assert(p->pending);

-        in->format = AF_FORMAT_FLOATP;
-        mp_audio_copy_config(out, in);
+    int opts = p->opts->transients | p->opts->detector | p->opts->phase |
+               p->opts->window | p->opts->smoothing | p->opts->formant |
+               p->opts->pitch | p-> opts->channels |
+               RubberBandOptionProcessRealTime;

-        if (p->rubber)
-            rubberband_delete(p->rubber);
+    int rate = mp_aframe_get_rate(p->pending);
+    int channels = mp_aframe_get_channels(p->pending);
+    if (mp_aframe_get_format(p->pending) != AF_FORMAT_FLOATP)
+        return false;

-        int opts = p->opt_transients | p->opt_detector | p->opt_phase |
-                   p->opt_window | p->opt_smoothing | p->opt_formant |
-                   p->opt_pitch | p-> opt_channels |
-                   RubberBandOptionProcessRealTime;
-
-        p->rubber = rubberband_new(in->rate, in->channels.num, opts, 1.0, 1.0);
-        if (!p->rubber) {
-            MP_FATAL(af, "librubberband initialization failed.\n");
-            return AF_ERROR;
-        }
-
-        update_speed(af, p->speed);
-        update_pitch(af, p->pitch);
-        control(af, AF_CONTROL_RESET, NULL);
-
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
+    p->rubber = rubberband_new(rate, channels, opts, 1.0, 1.0);
+    if (!p->rubber) {
+        MP_FATAL(f, "librubberband initialization failed.\n");
+        return false;
    }
-    case AF_CONTROL_SET_PLAYBACK_SPEED: {
-        update_speed(af, *(double *)arg);
-        return AF_OK;
-    }
-    case AF_CONTROL_RESET:
-        if (p->rubber)
-            rubberband_reset(p->rubber);
-        talloc_free(p->pending);
-        p->pending = NULL;
-        p->rubber_delay = 0;
-        return AF_OK;
-    case AF_CONTROL_COMMAND: {
-        char **args = arg;
-        if (!strcmp(args[0], "set-pitch")) {
-            char *endptr;
-            double pitch = strtod(args[1], &endptr);
-            if (*endptr || pitch < 0.01 || pitch > 100.0)
-                return CONTROL_ERROR;
-            update_pitch(af, pitch);
-            return CONTROL_OK;
-        } else {
-            return CONTROL_ERROR;
-        }
-    }
-    }
-    return AF_UNKNOWN;
+
+    mp_aframe_config_copy(p->cur_format, p->pending);
+
+    update_speed(p, p->speed);
+    update_pitch(p, p->pitch);
+
+    return true;
 }

-static int filter_frame(struct af_instance *af, struct mp_audio *data)
+static void process(struct mp_filter *f)
 {
-    struct priv *p = af->priv;
+    struct priv *p = f->priv;

-    talloc_free(p->pending);
-    p->pending = data;
+    if (!mp_pin_in_needs_data(f->ppins[1]))
+        return;

-    return 0;
-}
-
-static int filter_out(struct af_instance *af)
-{
-    struct priv *p = af->priv;
-
-    while (rubberband_available(p->rubber) <= 0) {
+    while (!p->rubber || !p->pending || rubberband_available(p->rubber) <= 0) {
        const float *dummy[MP_NUM_CHANNELS] = {0};
        const float **in_data = dummy;
        size_t in_samples = 0;
-        if (p->pending) {
-            if (!p->pending->samples)
-                break;

-            // recover from previous EOF
-            if (p->needs_reset) {
-                rubberband_reset(p->rubber);
-                p->rubber_delay = 0;
+        bool eof = false;
+        if (!p->pending || !mp_aframe_get_size(p->pending)) {
+            struct mp_frame frame = mp_pin_out_read(p->in_pin);
+            if (frame.type == MP_FRAME_AUDIO) {
+                TA_FREEP(&p->pending);
+                p->pending = frame.data;
+            } else if (frame.type == MP_FRAME_EOF) {
+                eof = true;
+            } else if (frame.type) {
+                MP_ERR(f, "unexpected frame type\n");
+                goto error;
+            } else {
+                return; // no new data yet
            }
-            p->needs_reset = false;
+        }
+        assert(p->pending || eof);

-            size_t needs = rubberband_get_samples_required(p->rubber);
-            in_data = (void *)&p->pending->planes;
-            in_samples = MPMIN(p->pending->samples, needs);
+        if (!p->rubber) {
+            if (!p->pending) {
+                mp_pin_in_write(f->ppins[1], MP_EOF_FRAME);
+                return;
+            }
+            if (!init_rubberband(f))
+                goto error;
        }

-        if (p->needs_reset)
-            break; // previous EOF
-        p->needs_reset = !p->pending; // EOF
+        bool format_change =
+            p->pending && !mp_aframe_config_equals(p->pending, p->cur_format);
+
+        if (p->pending && !format_change) {
+            size_t needs = rubberband_get_samples_required(p->rubber);
+            uint8_t **planes = mp_aframe_get_data_ro(p->pending);
+            int num_planes = mp_aframe_get_planes(p->pending);
+            for (int n = 0; n < num_planes; n++)
+                in_data[n] = (void *)planes[n];
+            in_samples = MPMIN(mp_aframe_get_size(p->pending), needs);
+        }
+
+        bool final = format_change || eof;
+        if (!p->sent_final)
+            rubberband_process(p->rubber, in_data, in_samples, final);
+        p->sent_final |= final;

-        rubberband_process(p->rubber, in_data, in_samples, p->needs_reset);
        p->rubber_delay += in_samples;

-        if (!p->pending)
-            break;
-        mp_audio_skip_samples(p->pending, in_samples);
+        if (p->pending && !format_change)
+            mp_aframe_skip_samples(p->pending, in_samples);
+
+        if (rubberband_available(p->rubber) > 0) {
+            if (eof)
+                mp_pin_out_repeat_eof(p->in_pin); // drain more next time
+        } else {
+            if (eof) {
+                mp_pin_in_write(f->ppins[1], MP_EOF_FRAME);
+                rubberband_reset(p->rubber);
+                p->rubber_delay = 0;
+                TA_FREEP(&p->pending);
+                p->sent_final = false;
+                return;
+            } else if (format_change) {
+                // go on with proper reinit on the next iteration
+                rubberband_delete(p->rubber);
+                p->sent_final = false;
+                p->rubber = NULL;
+            }
+        }
    }

+    assert(p->pending);
+
    int out_samples = rubberband_available(p->rubber);
    if (out_samples > 0) {
-        struct mp_audio *out =
-            mp_audio_pool_get(af->out_pool, af->data, out_samples);
-        if (!out)
-            return -1;
-        if (p->pending)
-            mp_audio_copy_config(out, p->pending);
+        struct mp_aframe *out = mp_aframe_new_ref(p->cur_format);
+        if (mp_aframe_pool_allocate(p->out_pool, out, out_samples) < 0) {
+            talloc_free(out);
+            goto error;
+        }

-        float **out_data = (void *)&out->planes;
-        out->samples = rubberband_retrieve(p->rubber, out_data, out->samples);
-        p->rubber_delay -= out->samples * p->speed;
+        mp_aframe_copy_attributes(out, p->pending);

-        af_add_output_frame(af, out);
+        float *out_data[MP_NUM_CHANNELS] = {0};
+        uint8_t **planes = mp_aframe_get_data_rw(out);
+        assert(planes);
+        int num_planes = mp_aframe_get_planes(out);
+        for (int n = 0; n < num_planes; n++)
+            out_data[n] = (void *)planes[n];
+
+        out_samples = rubberband_retrieve(p->rubber, out_data, out_samples);
+
+        if (!out_samples) {
+            mp_filter_internal_mark_progress(f); // unexpected, just try again
+            talloc_free(out);
+            return;
+        }
+
+        mp_aframe_set_size(out, out_samples);
+
+        p->rubber_delay -= out_samples * p->speed;
+
+        double pts = mp_aframe_get_pts(p->pending);
+        if (pts != MP_NOPTS_VALUE) {
+            // Note: rubberband_get_latency() does not do what you'd expect.
+            double delay = p->rubber_delay / mp_aframe_get_effective_rate(out);
+            mp_aframe_set_pts(out, pts - delay);
+        }
+
+        mp_aframe_mul_speed(out, p->speed);
+
+        mp_pin_in_write(f->ppins[1], MAKE_FRAME(MP_FRAME_AUDIO, out));
    }

-    int delay_samples = p->rubber_delay;
-    if (p->pending)
-        delay_samples += p->pending->samples;
-    af->delay = delay_samples / (af->data->rate * p->speed);
-
-    return 0;
+    return;
+error:
+    mp_filter_internal_mark_failed(f);
 }

-static void uninit(struct af_instance *af)
+static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
 {
-    struct priv *p = af->priv;
+    struct priv *p = f->priv;
+
+    switch (cmd->type) {
+    case MP_FILTER_COMMAND_TEXT: {
+        char *endptr = NULL;
+        double pitch = p->pitch;
+        if (!strcmp(cmd->cmd, "set-pitch")) {
+            pitch = strtod(cmd->arg, &endptr);
+            if (*endptr)
+                return false;
+            return update_pitch(p, pitch);
+        } else if (!strcmp(cmd->cmd, "multiply-pitch")) {
+            double mult = strtod(cmd->arg, &endptr);
+            if (*endptr || mult <= 0)
+                return false;
+            pitch *= mult;
+            return update_pitch(p, pitch);
+        }
+        return false;
+    }
+    case MP_FILTER_COMMAND_SET_SPEED:
+        update_speed(p, cmd->speed);
+        return true;
+    }
+
+    return false;
+}
+
+static void reset(struct mp_filter *f)
+{
+    struct priv *p = f->priv;
+
+    if (p->rubber)
+        rubberband_reset(p->rubber);
+    p->rubber_delay = 0;
+    p->sent_final = false;
+    TA_FREEP(&p->pending);
+}
+
+static void destroy(struct mp_filter *f)
+{
+    struct priv *p = f->priv;

    if (p->rubber)
        rubberband_delete(p->rubber);
    talloc_free(p->pending);
 }

-static int af_open(struct af_instance *af)
+static const struct mp_filter_info af_rubberband_filter = {
+    .name = "rubberband",
+    .priv_size = sizeof(struct priv),
+    .process = process,
+    .command = command,
+    .reset = reset,
+    .destroy = destroy,
+};
+
+static struct mp_filter *af_rubberband_create(struct mp_filter *parent,
+                                              void *options)
 {
-    af->control = control;
-    af->filter_frame = filter_frame;
-    af->filter_out = filter_out;
-    af->uninit = uninit;
-    return AF_OK;
+    struct mp_filter *f = mp_filter_create(parent, &af_rubberband_filter);
+    if (!f) {
+        talloc_free(options);
+        return NULL;
+    }
+
+    mp_filter_add_pin(f, MP_PIN_IN, "in");
+    mp_filter_add_pin(f, MP_PIN_OUT, "out");
+
+    struct priv *p = f->priv;
+    p->opts = talloc_steal(p, options);
+    p->speed = 1.0;
+    p->pitch = p->opts->scale;
+    p->cur_format = talloc_steal(p, mp_aframe_create());
+    p->out_pool = mp_aframe_pool_create(p);
+
+    struct mp_autoconvert *conv = mp_autoconvert_create(f);
+    if (!conv)
+        abort();
+
+    mp_autoconvert_add_afmt(conv, AF_FORMAT_FLOATP);
+
+    mp_pin_connect(conv->f->pins[0], f->ppins[0]);
+    p->in_pin = conv->f->pins[1];
+
+    return f;
 }

-#define OPT_BASE_STRUCT struct priv
-const struct af_info af_info_rubberband = {
-    .info = "Pitch conversion with librubberband",
-    .name = "rubberband",
-    .open = af_open,
-    .priv_size = sizeof(struct priv),
-    .priv_defaults = &(const struct priv) {
-        .speed = 1.0,
-        .pitch = 1.0,
-        .opt_pitch = RubberBandOptionPitchHighConsistency,
-        .opt_transients = RubberBandOptionTransientsMixed,
-        .opt_formant = RubberBandOptionFormantPreserved,
-        .opt_channels = RubberBandOptionChannelsTogether,
-    },
-    .options = (const struct m_option[]) {
-        OPT_CHOICE("transients", opt_transients, 0,
-                   ({"crisp", RubberBandOptionTransientsCrisp},
-                    {"mixed", RubberBandOptionTransientsMixed},
-                    {"smooth", RubberBandOptionTransientsSmooth})),
-        OPT_CHOICE("detector", opt_detector, 0,
-                   ({"compound", RubberBandOptionDetectorCompound},
-                    {"percussive", RubberBandOptionDetectorPercussive},
-                    {"soft", RubberBandOptionDetectorSoft})),
-        OPT_CHOICE("phase", opt_phase, 0,
-                   ({"laminar", RubberBandOptionPhaseLaminar},
-                    {"independent", RubberBandOptionPhaseIndependent})),
-        OPT_CHOICE("window", opt_window, 0,
-                   ({"standard", RubberBandOptionWindowStandard},
-                    {"short", RubberBandOptionWindowShort},
-                    {"long", RubberBandOptionWindowLong})),
-        OPT_CHOICE("smoothing", opt_smoothing, 0,
-                   ({"off", RubberBandOptionSmoothingOff},
-                    {"on", RubberBandOptionSmoothingOn})),
-        OPT_CHOICE("formant", opt_formant, 0,
-                   ({"shifted", RubberBandOptionFormantShifted},
-                    {"preserved", RubberBandOptionFormantPreserved})),
-        OPT_CHOICE("pitch", opt_pitch, 0,
-                   ({"quality", RubberBandOptionPitchHighQuality},
-                    {"speed", RubberBandOptionPitchHighSpeed},
-                    {"consistency", RubberBandOptionPitchHighConsistency})),
-        OPT_CHOICE("channels", opt_channels, 0,
-                   ({"apart", RubberBandOptionChannelsApart},
-                    {"together", RubberBandOptionChannelsTogether})),
-        OPT_DOUBLE("pitch-scale", pitch, M_OPT_RANGE, .min = 0.01, .max = 100),
-        {0}
+#define OPT_BASE_STRUCT struct f_opts
+
+const struct mp_user_filter_entry af_rubberband = {
+    .desc = {
+        .description = "Pitch conversion with librubberband",
+        .name = "rubberband",
+        .priv_size = sizeof(OPT_BASE_STRUCT),
+        .priv_defaults = &(const OPT_BASE_STRUCT) {
+            .scale = 1.0,
+            .pitch = RubberBandOptionPitchHighConsistency,
+            .transients = RubberBandOptionTransientsMixed,
+            .formant = RubberBandOptionFormantPreserved,
+            .channels = RubberBandOptionChannelsTogether,
+        },
+        .options = (const struct m_option[]) {
+            OPT_CHOICE("transients", transients, 0,
+                    ({"crisp", RubberBandOptionTransientsCrisp},
+                     {"mixed", RubberBandOptionTransientsMixed},
+                     {"smooth", RubberBandOptionTransientsSmooth})),
+            OPT_CHOICE("detector", detector, 0,
+                    ({"compound", RubberBandOptionDetectorCompound},
+                     {"percussive", RubberBandOptionDetectorPercussive},
+                     {"soft", RubberBandOptionDetectorSoft})),
+            OPT_CHOICE("phase", phase, 0,
+                    ({"laminar", RubberBandOptionPhaseLaminar},
+                     {"independent", RubberBandOptionPhaseIndependent})),
+            OPT_CHOICE("window", window, 0,
+                    ({"standard", RubberBandOptionWindowStandard},
+                     {"short", RubberBandOptionWindowShort},
+                     {"long", RubberBandOptionWindowLong})),
+            OPT_CHOICE("smoothing", smoothing, 0,
+                    ({"off", RubberBandOptionSmoothingOff},
+                     {"on", RubberBandOptionSmoothingOn})),
+            OPT_CHOICE("formant", formant, 0,
+                    ({"shifted", RubberBandOptionFormantShifted},
+                     {"preserved", RubberBandOptionFormantPreserved})),
+            OPT_CHOICE("pitch", pitch, 0,
+                    ({"quality", RubberBandOptionPitchHighQuality},
+                     {"speed", RubberBandOptionPitchHighSpeed},
+                     {"consistency", RubberBandOptionPitchHighConsistency})),
+            OPT_CHOICE("channels", channels, 0,
+                    ({"apart", RubberBandOptionChannelsApart},
+                     {"together", RubberBandOptionChannelsTogether})),
+            OPT_DOUBLE("pitch-scale", scale, M_OPT_RANGE, .min = 0.01, .max = 100),
+            {0}
+        },
    },
+    .create = af_rubberband_create,
 };
@@ -35,14 +35,33 @@
 #include <limits.h>
 #include <assert.h>

+#include "audio/aframe.h"
+#include "audio/format.h"
 #include "common/common.h"
-
-#include "af.h"
+#include "filters/f_autoconvert.h"
+#include "filters/filter_internal.h"
+#include "filters/user_filters.h"
 #include "options/m_option.h"

-// Data for specific instances of this filter
-typedef struct af_scaletempo_s
-{
+struct f_opts {
+    float scale_nominal;
+    float ms_stride;
+    float ms_search;
+    float percent_overlap;
+#define SCALE_TEMPO 1
+#define SCALE_PITCH 2
+    int speed_opt;
+};
+
+struct priv {
+    struct f_opts *opts;
+
+    struct mp_pin *in_pin;
+    struct mp_aframe *cur_format;
+    struct mp_aframe_pool *out_pool;
+    double current_pts;
+    struct mp_aframe *in;
+
    // stride
    float scale;
    float speed;
@@ -62,29 +81,23 @@ typedef struct af_scaletempo_s
    int bytes_standing;
    void *buf_overlap;
    void *table_blend;
-    void (*output_overlap)(struct af_scaletempo_s *s, void *out_buf,
+    void (*output_overlap)(struct priv *s, void *out_buf,
                           int bytes_off);
    // best overlap
    int frames_search;
    int num_channels;
    void *buf_pre_corr;
    void *table_window;
-    int (*best_overlap_offset)(struct af_scaletempo_s *s);
-    // command line
-    float scale_nominal;
-    float ms_stride;
-    float percent_overlap;
-    float ms_search;
-#define SCALE_TEMPO 1
-#define SCALE_PITCH 2
-    int speed_opt;
-} af_scaletempo_t;
+    int (*best_overlap_offset)(struct priv *s);
+};

-static int fill_queue(struct af_instance *af, struct mp_audio *data, int offset)
+static bool reinit(struct mp_filter *f);
+
+// Return whether it got enough data for filtering.
+static bool fill_queue(struct priv *s)
 {
-    af_scaletempo_t *s = af->priv;
-    int bytes_in = (data ? mp_audio_psize(data) : 0) - offset;
-    int offset_unchanged = offset;
+    int bytes_in = s->in ? mp_aframe_get_size(s->in) * s->bytes_per_frame : 0;
+    int offset = 0;

    if (s->bytes_to_slide > 0) {
        if (s->bytes_to_slide < s->bytes_queued) {
@@ -103,21 +116,27 @@ static int fill_queue(struct af_instance *af, struct mp_audio *data, int offset)
        }
    }

-    if (bytes_in > 0) {
-        int bytes_copy = MPMIN(s->bytes_queue - s->bytes_queued, bytes_in);
-        assert(bytes_copy >= 0);
-        memcpy(s->buf_queue + s->bytes_queued,
-               (int8_t *)data->planes[0] + offset, bytes_copy);
+    int bytes_needed = s->bytes_queue - s->bytes_queued;
+    assert(bytes_needed >= 0);
+
+    int bytes_copy = MPMIN(bytes_needed, bytes_in);
+    if (bytes_copy > 0) {
+        uint8_t **planes = mp_aframe_get_data_ro(s->in);
+        memcpy(s->buf_queue + s->bytes_queued, planes[0] + offset, bytes_copy);
        s->bytes_queued += bytes_copy;
        offset += bytes_copy;
+        bytes_needed -= bytes_copy;
    }

-    return offset - offset_unchanged;
+    if (s->in)
+        mp_aframe_skip_samples(s->in, offset / s->bytes_per_frame);
+
+    return bytes_needed == 0;
 }

 #define UNROLL_PADDING (4 * 4)

-static int best_overlap_offset_float(af_scaletempo_t *s)
+static int best_overlap_offset_float(struct priv *s)
 {
    float best_corr = INT_MIN;
    int best_off = 0;
@@ -146,7 +165,7 @@ static int best_overlap_offset_float(af_scaletempo_t *s)
    return best_off * 4 * s->num_channels;
 }

-static int best_overlap_offset_s16(af_scaletempo_t *s)
+static int best_overlap_offset_s16(struct priv *s)
 {
    int64_t best_corr = INT64_MIN;
    int best_off = 0;
@@ -183,7 +202,7 @@ static int best_overlap_offset_s16(af_scaletempo_t *s)
    return best_off * 2 * s->num_channels;
 }

-static void output_overlap_float(af_scaletempo_t *s, void *buf_out,
+static void output_overlap_float(struct priv *s, void *buf_out,
                                 int bytes_off)
 {
    float *pout = buf_out;
@@ -196,7 +215,7 @@ static void output_overlap_float(af_scaletempo_t *s, void *buf_out,
    }
 }

-static void output_overlap_s16(af_scaletempo_t *s, void *buf_out,
+static void output_overlap_s16(struct priv *s, void *buf_out,
                               int bytes_off)
 {
    int16_t *pout = buf_out;
@@ -209,29 +228,77 @@ static void output_overlap_s16(af_scaletempo_t *s, void *buf_out,
    }
 }

-static int filter(struct af_instance *af, struct mp_audio *data)
+static void process(struct mp_filter *f)
 {
-    af_scaletempo_t *s = af->priv;
+    struct priv *s = f->priv;

-    if (s->scale == 1.0) {
-        af->delay = 0;
-        af_add_output_frame(af, data);
-        return 0;
+    if (!mp_pin_in_needs_data(f->ppins[1]))
+        return;
+
+    struct mp_aframe *out = NULL;
+
+    bool drain = false;
+    bool is_eof = false;
+    if (!s->in) {
+        struct mp_frame frame = mp_pin_out_read(s->in_pin);
+        if (!frame.type)
+            return; // no input yet
+        if (frame.type != MP_FRAME_AUDIO && frame.type != MP_FRAME_EOF) {
+            MP_ERR(f, "unexpected frame type\n");
+            goto error;
+        }
+
+        s->in = frame.type == MP_FRAME_AUDIO ? frame.data : NULL;
+        is_eof = drain = !s->in;
+
+        // EOF before it was even initialized once.
+        if (is_eof && !mp_aframe_config_is_valid(s->cur_format)) {
+            mp_pin_in_write(f->ppins[1], MP_EOF_FRAME);
+            return;
+        }
+
+        if (s->in && !mp_aframe_config_equals(s->in, s->cur_format)) {
+            if (s->bytes_queued) {
+                // Drain remaining data before executing the format change.
+                MP_VERBOSE(f, "draining\n");
+                mp_pin_out_unread(s->in_pin, frame);
+                s->in = NULL;
+                drain = true;
+            } else {
+                if (!reinit(f)) {
+                    MP_ERR(f, "initialization failed\n");
+                    goto error;
+                }
+            }
+        }
+
+        if (s->in)
+            s->current_pts = mp_aframe_end_pts(s->in);
    }

-    int in_samples = data ? data->samples : 0;
-    struct mp_audio *out = mp_audio_pool_get(af->out_pool, af->data,
-        ((int)(in_samples / s->frames_stride_scaled) + 1) * s->frames_stride);
-    if (!out) {
-        talloc_free(data);
-        return -1;
+    if (!fill_queue(s) && !drain) {
+        TA_FREEP(&s->in);
+        mp_pin_out_request_data_next(s->in_pin);
+        return;
    }
-    if (data)
-        mp_audio_copy_attributes(out, data);

-    int offset_in = fill_queue(af, data, 0);
-    int8_t *pout = out->planes[0];
-    while (s->bytes_queued >= s->bytes_queue) {
+    int max_out_samples = s->bytes_stride / s->bytes_per_frame;
+    if (drain)
+        max_out_samples += s->bytes_queued;
+
+    out = mp_aframe_new_ref(s->cur_format);
+    if (mp_aframe_pool_allocate(s->out_pool, out, max_out_samples) < 0)
+        goto error;
+
+    if (s->in)
+        mp_aframe_copy_attributes(out, s->in);
+
+    uint8_t **out_planes = mp_aframe_get_data_rw(out);
+    if (!out_planes)
+        goto error;
+    int8_t *pout = out_planes[0];
+    int out_offset = 0;
+    if (s->bytes_queued >= s->bytes_queue) {
        int ti;
        float tf;
        int bytes_off = 0;
@@ -240,12 +307,12 @@ static int filter(struct af_instance *af, struct mp_audio *data)
        if (s->output_overlap) {
            if (s->best_overlap_offset)
                bytes_off = s->best_overlap_offset(s);
-            s->output_overlap(s, pout, bytes_off);
+            s->output_overlap(s, pout + out_offset, bytes_off);
        }
-        memcpy(pout + s->bytes_overlap,
+        memcpy(pout + out_offset + s->bytes_overlap,
               s->buf_queue + bytes_off + s->bytes_overlap,
               s->bytes_standing);
-        pout += s->bytes_stride;
+        out_offset += s->bytes_stride;

        // input stride
        memcpy(s->buf_overlap,
@@ -255,240 +322,303 @@ static int filter(struct af_instance *af, struct mp_audio *data)
        ti = (int)tf;
        s->frames_stride_error = tf - ti;
        s->bytes_to_slide = ti * s->bytes_per_frame;
-
-        offset_in += fill_queue(af, data, offset_in);
    }
+    // Drain remaining buffered data.
+    if (drain && s->bytes_queued) {
+        memcpy(pout + out_offset, s->buf_queue, s->bytes_queued);
+        out_offset += s->bytes_queued;
+        s->bytes_queued = 0;
+    }
+    mp_aframe_set_size(out, out_offset / s->bytes_per_frame);

    // This filter can have a negative delay when scale > 1:
    // output corresponding to some length of input can be decided and written
    // after receiving only a part of that input.
-    af->delay = (s->bytes_queued - s->bytes_to_slide) / s->scale
-                / out->sstride / out->rate;
+    float delay = (out_offset * s->speed + s->bytes_queued - s->bytes_to_slide) /
+                    s->bytes_per_frame / mp_aframe_get_effective_rate(out)
+                  + (s->in ? mp_aframe_duration(s->in) : 0);

-    out->samples = (pout - (int8_t *)out->planes[0]) / out->sstride;
-    talloc_free(data);
-    if (out->samples) {
-        af_add_output_frame(af, out);
-    } else {
-        talloc_free(out);
+    if (s->current_pts != MP_NOPTS_VALUE)
+        mp_aframe_set_pts(out, s->current_pts - delay);
+
+    mp_aframe_mul_speed(out, s->speed);
+
+    if (!mp_aframe_get_size(out))
+        TA_FREEP(&out);
+
+    if (is_eof && out) {
+        mp_pin_out_repeat_eof(s->in_pin);
+    } else if (is_eof && !out) {
+        mp_pin_in_write(f->ppins[1], MP_EOF_FRAME);
+    } else if (!is_eof && !out) {
+        mp_pin_out_request_data_next(s->in_pin);
    }
-    return 0;
+
+    if (out)
+        mp_pin_in_write(f->ppins[1], MAKE_FRAME(MP_FRAME_AUDIO, out));
+
+    return;
+
+error:
+    TA_FREEP(&s->in);
+    talloc_free(out);
+    mp_filter_internal_mark_failed(f);
 }

-static void update_speed(struct af_instance *af, float speed)
+static void update_speed(struct priv *s, float speed)
 {
-    af_scaletempo_t *s = af->priv;
-
    s->speed = speed;

-    double factor = (s->speed_opt & SCALE_PITCH) ? 1.0 / s->speed : s->speed;
-    s->scale = factor * s->scale_nominal;
+    double factor = (s->opts->speed_opt & SCALE_PITCH) ? 1.0 / s->speed : s->speed;
+    s->scale = factor * s->opts->scale_nominal;

    s->frames_stride_scaled = s->scale * s->frames_stride;
    s->frames_stride_error = MPMIN(s->frames_stride_error, s->frames_stride_scaled);
 }

-// Initialization and runtime control
-static int control(struct af_instance *af, int cmd, void *arg)
+static bool reinit(struct mp_filter *f)
 {
-    af_scaletempo_t *s = af->priv;
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *data = (struct mp_audio *)arg;
-        float srate  = data->rate / 1000.0;
-        int nch = data->nch;
-        int use_int = 0;
+    struct priv *s = f->priv;

-        mp_audio_force_interleaved_format(data);
-        mp_audio_copy_config(af->data, data);
+    mp_aframe_reset(s->cur_format);

-        if (data->format == AF_FORMAT_S16) {
-            use_int = 1;
-        } else {
-            mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-        }
-        int bps = af->data->bps;
+    float srate  = mp_aframe_get_rate(s->in) / 1000.0;
+    int nch = mp_aframe_get_channels(s->in);
+    int format = mp_aframe_get_format(s->in);

-        s->frames_stride        = srate * s->ms_stride;
-        s->bytes_stride         = s->frames_stride * bps * nch;
-        af->delay = 0;
-
-        update_speed(af, s->speed);
-
-        int frames_overlap = s->frames_stride * s->percent_overlap;
-        if (frames_overlap <= 0) {
-            s->bytes_standing   = s->bytes_stride;
-            s->samples_standing = s->bytes_standing / bps;
-            s->output_overlap   = NULL;
-            s->bytes_overlap    = 0;
-        } else {
-            s->samples_overlap  = frames_overlap * nch;
-            s->bytes_overlap    = frames_overlap * nch * bps;
-            s->bytes_standing   = s->bytes_stride - s->bytes_overlap;
-            s->samples_standing = s->bytes_standing / bps;
-            s->buf_overlap      = realloc(s->buf_overlap, s->bytes_overlap);
-            s->table_blend      = realloc(s->table_blend, s->bytes_overlap * 4);
-            if (!s->buf_overlap || !s->table_blend) {
-                MP_FATAL(af, "Out of memory\n");
-                return AF_ERROR;
-            }
-            memset(s->buf_overlap, 0, s->bytes_overlap);
-            if (use_int) {
-                int32_t *pb = s->table_blend;
-                int64_t blend = 0;
-                for (int i = 0; i < frames_overlap; i++) {
-                    int32_t v = blend / frames_overlap;
-                    for (int j = 0; j < nch; j++)
-                        *pb++ = v;
-                    blend += 65536; // 2^16
-                }
-                s->output_overlap = output_overlap_s16;
-            } else {
-                float *pb = s->table_blend;
-                for (int i = 0; i < frames_overlap; i++) {
-                    float v = i / (float)frames_overlap;
-                    for (int j = 0; j < nch; j++)
-                        *pb++ = v;
-                }
-                s->output_overlap = output_overlap_float;
-            }
-        }
-
-        s->frames_search = (frames_overlap > 1) ? srate * s->ms_search : 0;
-        if (s->frames_search <= 0)
-            s->best_overlap_offset = NULL;
-        else {
-            if (use_int) {
-                int64_t t = frames_overlap;
-                int32_t n = 8589934588LL / (t * t); // 4 * (2^31 - 1) / t^2
-                s->buf_pre_corr = realloc(s->buf_pre_corr,
-                                          s->bytes_overlap * 2 + UNROLL_PADDING);
-                s->table_window = realloc(s->table_window,
-                                          s->bytes_overlap * 2 - nch * bps * 2);
-                if (!s->buf_pre_corr || !s->table_window) {
-                    MP_FATAL(af, "Out of memory\n");
-                    return AF_ERROR;
-                }
-                memset((char *)s->buf_pre_corr + s->bytes_overlap * 2, 0,
-                       UNROLL_PADDING);
-                int32_t *pw = s->table_window;
-                for (int i = 1; i < frames_overlap; i++) {
-                    int32_t v = (i * (t - i) * n) >> 15;
-                    for (int j = 0; j < nch; j++)
-                        *pw++ = v;
-                }
-                s->best_overlap_offset = best_overlap_offset_s16;
-            } else {
-                s->buf_pre_corr = realloc(s->buf_pre_corr, s->bytes_overlap);
-                s->table_window = realloc(s->table_window,
-                                          s->bytes_overlap - nch * bps);
-                if (!s->buf_pre_corr || !s->table_window) {
-                    MP_FATAL(af, "Out of memory\n");
-                    return AF_ERROR;
-                }
-                float *pw = s->table_window;
-                for (int i = 1; i < frames_overlap; i++) {
-                    float v = i * (frames_overlap - i);
-                    for (int j = 0; j < nch; j++)
-                        *pw++ = v;
-                }
-                s->best_overlap_offset = best_overlap_offset_float;
-            }
-        }
-
-        s->bytes_per_frame = bps * nch;
-        s->num_channels    = nch;
-
-        s->bytes_queue = (s->frames_search + s->frames_stride + frames_overlap)
-                         * bps * nch;
-        s->buf_queue = realloc(s->buf_queue, s->bytes_queue + UNROLL_PADDING);
-        if (!s->buf_queue) {
-            MP_FATAL(af, "Out of memory\n");
-            return AF_ERROR;
-        }
-
-        s->bytes_queued = 0;
-        s->bytes_to_slide = 0;
-
-        MP_DBG(af, ""
-               "%.2f stride_in, %i stride_out, %i standing, "
-               "%i overlap, %i search, %i queue, %s mode\n",
-               s->frames_stride_scaled,
-               (int)(s->bytes_stride / nch / bps),
-               (int)(s->bytes_standing / nch / bps),
-               (int)(s->bytes_overlap / nch / bps),
-               s->frames_search,
-               (int)(s->bytes_queue / nch / bps),
-               (use_int ? "s16" : "float"));
-
-        return af_test_output(af, (struct mp_audio *)arg);
+    int use_int = 0;
+    if (format == AF_FORMAT_S16) {
+        use_int = 1;
+    } else if (format != AF_FORMAT_FLOAT) {
+        return false;
    }
-    case AF_CONTROL_SET_PLAYBACK_SPEED: {
-        double speed = *(double *)arg;
-        if (s->speed_opt & SCALE_TEMPO) {
-            if (s->speed_opt & SCALE_PITCH)
-                break;
-            update_speed(af, speed);
-        } else if (s->speed_opt & SCALE_PITCH) {
-            update_speed(af, speed);
-            break; // do not signal OK
+    int bps = use_int ? 2 : 4;
+
+    s->frames_stride        = srate * s->opts->ms_stride;
+    s->bytes_stride         = s->frames_stride * bps * nch;
+
+    update_speed(s, s->speed);
+
+    int frames_overlap = s->frames_stride * s->opts->percent_overlap;
+    if (frames_overlap <= 0) {
+        s->bytes_standing   = s->bytes_stride;
+        s->samples_standing = s->bytes_standing / bps;
+        s->output_overlap   = NULL;
+        s->bytes_overlap    = 0;
+    } else {
+        s->samples_overlap  = frames_overlap * nch;
+        s->bytes_overlap    = frames_overlap * nch * bps;
+        s->bytes_standing   = s->bytes_stride - s->bytes_overlap;
+        s->samples_standing = s->bytes_standing / bps;
+        s->buf_overlap      = realloc(s->buf_overlap, s->bytes_overlap);
+        s->table_blend      = realloc(s->table_blend, s->bytes_overlap * 4);
+        if (!s->buf_overlap || !s->table_blend) {
+            MP_FATAL(f, "Out of memory\n");
+            return false;
        }
-        return AF_OK;
-    }
-    case AF_CONTROL_RESET:
-        s->bytes_queued = 0;
-        s->bytes_to_slide = 0;
-        s->frames_stride_error = 0;
        memset(s->buf_overlap, 0, s->bytes_overlap);
+        if (use_int) {
+            int32_t *pb = s->table_blend;
+            int64_t blend = 0;
+            for (int i = 0; i < frames_overlap; i++) {
+                int32_t v = blend / frames_overlap;
+                for (int j = 0; j < nch; j++)
+                    *pb++ = v;
+                blend += 65536; // 2^16
+            }
+            s->output_overlap = output_overlap_s16;
+        } else {
+            float *pb = s->table_blend;
+            for (int i = 0; i < frames_overlap; i++) {
+                float v = i / (float)frames_overlap;
+                for (int j = 0; j < nch; j++)
+                    *pb++ = v;
+            }
+            s->output_overlap = output_overlap_float;
+        }
    }
-    return AF_UNKNOWN;
+
+    s->frames_search = (frames_overlap > 1) ? srate * s->opts->ms_search : 0;
+    if (s->frames_search <= 0)
+        s->best_overlap_offset = NULL;
+    else {
+        if (use_int) {
+            int64_t t = frames_overlap;
+            int32_t n = 8589934588LL / (t * t); // 4 * (2^31 - 1) / t^2
+            s->buf_pre_corr = realloc(s->buf_pre_corr,
+                                        s->bytes_overlap * 2 + UNROLL_PADDING);
+            s->table_window = realloc(s->table_window,
+                                        s->bytes_overlap * 2 - nch * bps * 2);
+            if (!s->buf_pre_corr || !s->table_window) {
+                MP_FATAL(f, "Out of memory\n");
+                return false;
+            }
+            memset((char *)s->buf_pre_corr + s->bytes_overlap * 2, 0,
+                    UNROLL_PADDING);
+            int32_t *pw = s->table_window;
+            for (int i = 1; i < frames_overlap; i++) {
+                int32_t v = (i * (t - i) * n) >> 15;
+                for (int j = 0; j < nch; j++)
+                    *pw++ = v;
+            }
+            s->best_overlap_offset = best_overlap_offset_s16;
+        } else {
+            s->buf_pre_corr = realloc(s->buf_pre_corr, s->bytes_overlap);
+            s->table_window = realloc(s->table_window,
+                                        s->bytes_overlap - nch * bps);
+            if (!s->buf_pre_corr || !s->table_window) {
+                MP_FATAL(f, "Out of memory\n");
+                return false;
+            }
+            float *pw = s->table_window;
+            for (int i = 1; i < frames_overlap; i++) {
+                float v = i * (frames_overlap - i);
+                for (int j = 0; j < nch; j++)
+                    *pw++ = v;
+            }
+            s->best_overlap_offset = best_overlap_offset_float;
+        }
+    }
+
+    s->bytes_per_frame = bps * nch;
+    s->num_channels    = nch;
+
+    s->bytes_queue = (s->frames_search + s->frames_stride + frames_overlap)
+                        * bps * nch;
+    s->buf_queue = realloc(s->buf_queue, s->bytes_queue + UNROLL_PADDING);
+    if (!s->buf_queue) {
+        MP_FATAL(f, "Out of memory\n");
+        return false;
+    }
+
+    s->bytes_queued = 0;
+    s->bytes_to_slide = 0;
+
+    MP_DBG(f, ""
+           "%.2f stride_in, %i stride_out, %i standing, "
+           "%i overlap, %i search, %i queue, %s mode\n",
+           s->frames_stride_scaled,
+           (int)(s->bytes_stride / nch / bps),
+           (int)(s->bytes_standing / nch / bps),
+           (int)(s->bytes_overlap / nch / bps),
+           s->frames_search,
+           (int)(s->bytes_queue / nch / bps),
+           (use_int ? "s16" : "float"));
+
+    mp_aframe_config_copy(s->cur_format, s->in);
+
+    return true;
 }

-// Deallocate memory
-static void uninit(struct af_instance *af)
+static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
 {
-    af_scaletempo_t *s = af->priv;
+    struct priv *s = f->priv;
+
+    if (cmd->type == MP_FILTER_COMMAND_SET_SPEED) {
+        if (s->opts->speed_opt & SCALE_TEMPO) {
+            if (s->opts->speed_opt & SCALE_PITCH)
+                return false;
+            update_speed(s, cmd->speed);
+            return true;
+        } else if (s->opts->speed_opt & SCALE_PITCH) {
+            update_speed(s, cmd->speed);
+            return false; // do not signal OK
+        }
+    }
+
+    return false;
+}
+
+static void reset(struct mp_filter *f)
+{
+    struct priv *s = f->priv;
+
+    s->current_pts = MP_NOPTS_VALUE;
+    s->bytes_queued = 0;
+    s->bytes_to_slide = 0;
+    s->frames_stride_error = 0;
+    memset(s->buf_overlap, 0, s->bytes_overlap);
+    TA_FREEP(&s->in);
+}
+
+static void destroy(struct mp_filter *f)
+{
+    struct priv *s = f->priv;
    free(s->buf_queue);
    free(s->buf_overlap);
    free(s->buf_pre_corr);
    free(s->table_blend);
    free(s->table_window);
+    TA_FREEP(&s->in);
+    mp_filter_free_children(f);
 }

-// Allocate memory and set function pointers
-static int af_open(struct af_instance *af)
-{
-    af->control   = control;
-    af->uninit    = uninit;
-    af->filter_frame = filter;
-    return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_scaletempo_t
-
-const struct af_info af_info_scaletempo = {
-    .info = "Scale audio tempo while maintaining pitch",
+static const struct mp_filter_info af_scaletempo_filter = {
    .name = "scaletempo",
-    .open = af_open,
-    .priv_size = sizeof(af_scaletempo_t),
-    .priv_defaults = &(const af_scaletempo_t) {
-        .ms_stride = 60,
-        .percent_overlap = .20,
-        .ms_search = 14,
-        .speed_opt = SCALE_TEMPO,
-        .speed = 1.0,
-        .scale_nominal = 1.0,
+    .priv_size = sizeof(struct priv),
+    .process = process,
+    .command = command,
+    .reset = reset,
+    .destroy = destroy,
+};
+
+static struct mp_filter *af_scaletempo_create(struct mp_filter *parent,
+                                              void *options)
+{
+    struct mp_filter *f = mp_filter_create(parent, &af_scaletempo_filter);
+    if (!f) {
+        talloc_free(options);
+        return NULL;
+    }
+
+    mp_filter_add_pin(f, MP_PIN_IN, "in");
+    mp_filter_add_pin(f, MP_PIN_OUT, "out");
+
+    struct priv *s = f->priv;
+    s->opts = talloc_steal(s, options);
+    s->speed = 1.0;
+    s->cur_format = talloc_steal(s, mp_aframe_create());
+    s->out_pool = mp_aframe_pool_create(s);
+
+    struct mp_autoconvert *conv = mp_autoconvert_create(f);
+    if (!conv)
+        abort();
+
+    mp_autoconvert_add_afmt(conv, AF_FORMAT_S16);
+    mp_autoconvert_add_afmt(conv, AF_FORMAT_FLOAT);
+
+    mp_pin_connect(conv->f->pins[0], f->ppins[0]);
+    s->in_pin = conv->f->pins[1];
+
+    return f;
+}
+
+#define OPT_BASE_STRUCT struct f_opts
+
+const struct mp_user_filter_entry af_scaletempo = {
+    .desc = {
+        .description = "Scale audio tempo while maintaining pitch",
+        .name = "scaletempo",
+        .priv_size = sizeof(OPT_BASE_STRUCT),
+        .priv_defaults = &(const OPT_BASE_STRUCT) {
+            .ms_stride = 60,
+            .percent_overlap = .20,
+            .ms_search = 14,
+            .speed_opt = SCALE_TEMPO,
+            .scale_nominal = 1.0,
+        },
+        .options = (const struct m_option[]) {
+            OPT_FLOAT("scale", scale_nominal, M_OPT_MIN, .min = 0.01),
+            OPT_FLOAT("stride", ms_stride, M_OPT_MIN, .min = 0.01),
+            OPT_FLOAT("overlap", percent_overlap, M_OPT_RANGE, .min = 0, .max = 1),
+            OPT_FLOAT("search", ms_search, M_OPT_MIN, .min = 0),
+            OPT_CHOICE("speed", speed_opt, 0,
+                       ({"pitch", SCALE_PITCH},
+                        {"tempo", SCALE_TEMPO},
+                        {"none", 0},
+                        {"both", SCALE_TEMPO | SCALE_PITCH})),
+            {0}
+        },
    },
-    .options = (const struct m_option[]) {
-        OPT_FLOAT("scale", scale_nominal, M_OPT_MIN, .min = 0.01),
-        OPT_FLOAT("stride", ms_stride, M_OPT_MIN, .min = 0.01),
-        OPT_FLOAT("overlap", percent_overlap, M_OPT_RANGE, .min = 0, .max = 1),
-        OPT_FLOAT("search", ms_search, M_OPT_MIN, .min = 0),
-        OPT_CHOICE("speed", speed_opt, 0,
-                   ({"pitch", SCALE_PITCH},
-                    {"tempo", SCALE_TEMPO},
-                    {"none", 0},
-                    {"both", SCALE_TEMPO | SCALE_PITCH})),
-        {0}
-    },
+    .create = af_scaletempo_create,
 };
@@ -1,72 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <math.h>
-#include <string.h>
-
-#include "common/common.h"
-#include "af.h"
-
-/* Convert from ms to sample time */
-int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
-{
-  int i = 0;
-  // Sanity check
-  if(!in || !out)
-    return AF_ERROR;
-
-  for(i=0;i<n;i++)
-    out[i]=(int)((float)rate * MPCLAMP(in[i],mi,ma)/1000.0);
-
-  return AF_OK;
-}
-
-/*
- * test if output format matches
- *  af: audio filter
- *  out: needed format, will be overwritten by available
- *       format if they do not match
- *  returns: AF_FALSE if formats do not match, AF_OK if they match
- *
- * compares the format, rate and nch values of af->data with out
- * Note: logically, *out=*af->data always happens, because out contains the
- *       format only, no actual audio data or memory allocations. *out always
- *       contains the parameters from af->data after the function returns.
- */
-int af_test_output(struct af_instance* af, struct mp_audio* out)
-{
-  if((af->data->format != out->format) ||
-     (af->data->bps    != out->bps)    ||
-     (af->data->rate   != out->rate)   ||
-     !mp_chmap_equals(&af->data->channels, &out->channels)){
-    *out = *af->data;
-    return AF_FALSE;
-  }
-  return AF_OK;
-}
-
-/* Soft clipping, the sound of a dream, thanks to Jon Wattes
-   post to Musicdsp.org */
-float af_softclip(float a)
-{
-    if (a >= M_PI/2)
-        return 1.0;
-    else if (a <= -M_PI/2)
-        return -1.0;
-    else
-        return sin(a);
-}
@@ -55,12 +55,6 @@ bool af_fmt_is_int(int format)
    return format && !af_fmt_is_spdif(format) && !af_fmt_is_float(format);
 }

-// false for interleaved and AF_FORMAT_UNKNOWN
-bool af_fmt_is_planar(int format)
-{
-    return format && af_fmt_to_planar(format) == format;
-}
-
 bool af_fmt_is_spdif(int format)
 {
    return af_format_sample_alignment(format) > 1;
@@ -79,23 +73,30 @@ static const int planar_formats[][2] = {
    {AF_FORMAT_DOUBLEP, AF_FORMAT_DOUBLE},
 };

+bool af_fmt_is_planar(int format)
+{
+    for (int n = 0; n < MP_ARRAY_SIZE(planar_formats); n++) {
+        if (planar_formats[n][0] == format)
+            return true;
+    }
+    return false;
+}
+
 // Return the planar format corresponding to the given format.
-// If the format is already planar, return it.
-// Return 0 if there's no equivalent.
+// If the format is already planar or if there's no equivalent,
+// return it.
 int af_fmt_to_planar(int format)
 {
    for (int n = 0; n < MP_ARRAY_SIZE(planar_formats); n++) {
        if (planar_formats[n][1] == format)
            return planar_formats[n][0];
-        if (planar_formats[n][0] == format)
-            return format;
    }
-    return 0;
+    return format;
 }

 // Return the interleaved format corresponding to the given format.
-// If the format is already interleaved, return it.
-// Always succeeds if format is actually planar; otherwise return 0.
+// If the format is already interleaved or if there's no equivalent,
+// return it.
 int af_fmt_from_planar(int format)
 {
    for (int n = 0; n < MP_ARRAY_SIZE(planar_formats); n++) {
@@ -134,17 +135,6 @@ const char *af_fmt_to_str(int format)
    return "??";
 }

-int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate)
-{
-    assert(!af_fmt_is_planar(format));
-    int bps      = af_fmt_to_bytes(format);
-    int framelen = channels * bps;
-    int bytes    = seconds  * bps * samplerate;
-    if (bytes % framelen)
-        bytes += framelen - (bytes % framelen);
-    return bytes;
-}
-
 void af_fill_silence(void *dst, size_t bytes, int format)
 {
    memset(dst, af_fmt_is_unsigned(format) ? 0x80 : 0, bytes);
@@ -154,7 +144,7 @@ void af_fill_silence(void *dst, size_t bytes, int format)
 // If the formats are equal, 1024 is returned. If they are gravely incompatible
 // (like s16<->ac3), INT_MIN is returned. If there is implied loss of precision
 // (like s16->s8), a value <0 is returned.
-static int af_format_conversion_score(int dst_format, int src_format)
+int af_format_conversion_score(int dst_format, int src_format)
 {
    if (dst_format == AF_FORMAT_UNKNOWN || src_format == AF_FORMAT_UNKNOWN)
        return INT_MIN;
@@ -209,10 +199,11 @@ static int cmp_entry(const void *a, const void *b)
 // and the list is terminated with 0 (AF_FORMAT_UNKNOWN).
 // Keep in mind that this also returns formats with flipped interleaving
 // (e.g. for s16, it returns [s16, s16p, ...]).
-void af_get_best_sample_formats(int src_format, int out_formats[AF_FORMAT_COUNT])
+// out_formats must be an int[AF_FORMAT_COUNT + 1] array.
+void af_get_best_sample_formats(int src_format, int *out_formats)
 {
    int num = 0;
-    struct entry e[AF_FORMAT_COUNT];
+    struct entry e[AF_FORMAT_COUNT + 1];
    for (int fmt = 1; fmt < AF_FORMAT_COUNT; fmt++) {
        int score = af_format_conversion_score(fmt, src_format);
        if (score > INT_MIN)
@@ -64,12 +64,10 @@ bool af_fmt_is_pcm(int format);
 int af_fmt_to_planar(int format);
 int af_fmt_from_planar(int format);

-// Amount of bytes that contain audio of the given duration, aligned to frames.
-int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate);
-
 void af_fill_silence(void *dst, size_t bytes, int format);

-void af_get_best_sample_formats(int src_format, int out_formats[AF_FORMAT_COUNT]);
+void af_get_best_sample_formats(int src_format, int *out_formats);
+int af_format_conversion_score(int dst_format, int src_format);
 int af_select_best_samplerate(int src_sampelrate, const int *available);

 int af_format_sample_alignment(int format);
@@ -82,7 +82,7 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #if HAVE_OPENSLES
    &audio_out_opensles,
 #endif
-#if HAVE_SDL1 || HAVE_SDL2
+#if HAVE_SDL2
    &audio_out_sdl,
 #endif
 #if HAVE_SNDIO
@@ -93,9 +93,7 @@ static const struct ao_driver * const audio_out_drivers[] = {
    &audio_out_coreaudio_exclusive,
 #endif
    &audio_out_pcm,
-#if HAVE_ENCODING
    &audio_out_lavc,
-#endif
 #if HAVE_RSOUND
    &audio_out_rsound,
 #endif
@@ -232,6 +230,7 @@ static struct ao *ao_init(bool probing, struct mpv_global *global,
    if (ao->device_buffer)
        MP_VERBOSE(ao, "device buffer: %d samples.\n", ao->device_buffer);
    ao->buffer = MPMAX(ao->device_buffer, ao->def_buffer * ao->samplerate);
+    ao->buffer = MPMAX(ao->buffer, 1);

    int align = af_format_sample_alignment(ao->format);
    ao->buffer = (ao->buffer + align - 1) / align * align;
@@ -421,7 +420,7 @@ int ao_query_and_reset_events(struct ao *ao, int events)
    return atomic_fetch_and(&ao->events_, ~(unsigned)events) & events;
 }

-static void ao_add_events(struct ao *ao, int events)
+void ao_add_events(struct ao *ao, int events)
 {
    atomic_fetch_or(&ao->events_, events);
    ao->wakeup_cb(ao->wakeup_ctx);
@@ -48,6 +48,7 @@ enum aocontrol {
 enum {
    AO_EVENT_RELOAD = 1,
    AO_EVENT_HOTPLUG = 2,
+    AO_EVENT_INITIAL_UNBLOCK = 4,
 };

 enum {
@@ -104,6 +105,8 @@ void ao_resume(struct ao *ao);
 void ao_drain(struct ao *ao);
 bool ao_eof_reached(struct ao *ao);
 int ao_query_and_reset_events(struct ao *ao, int events);
+void ao_add_events(struct ao *ao, int events);
+void ao_unblock(struct ao *ao);
 void ao_request_reload(struct ao *ao);
 void ao_hotplug_event(struct ao *ao);

@@ -30,6 +30,7 @@
 #include <sys/time.h>
 #include <stdlib.h>
 #include <stdarg.h>
+#include <limits.h>
 #include <math.h>
 #include <string.h>

@@ -42,8 +43,11 @@

 #include <alsa/asoundlib.h>

-#define HAVE_CHMAP_API \
-    (defined(SND_CHMAP_API_VERSION) && SND_CHMAP_API_VERSION >= (1 << 16))
+#if defined(SND_CHMAP_API_VERSION) && SND_CHMAP_API_VERSION >= (1 << 16)
+#define HAVE_CHMAP_API 1
+#else
+#define HAVE_CHMAP_API 0
+#endif

 #include "ao.h"
 #include "internal.h"
@@ -56,6 +60,8 @@ struct ao_alsa_opts {
    int resample;
    int ni;
    int ignore_chmap;
+    int buffer_time;
+    int frags;
 };

 #define OPT_BASE_STRUCT struct ao_alsa_opts
@@ -67,6 +73,8 @@ static const struct m_sub_options ao_alsa_conf = {
        OPT_INTRANGE("alsa-mixer-index", mixer_index, 0, 0, 99),
        OPT_FLAG("alsa-non-interleaved", ni, 0),
        OPT_FLAG("alsa-ignore-chmap", ignore_chmap, 0),
+        OPT_INTRANGE("alsa-buffer-time", buffer_time, 0, 0, INT_MAX),
+        OPT_INTRANGE("alsa-periods", frags, 0, 0, INT_MAX),
        {0}
    },
    .defaults = &(const struct ao_alsa_opts) {
@@ -74,6 +82,8 @@ static const struct m_sub_options ao_alsa_conf = {
        .mixer_name = "Master",
        .mixer_index = 0,
        .ni = 0,
+        .buffer_time = 100000,
+        .frags = 4,
    },
    .size = sizeof(struct ao_alsa_opts),
 };
@@ -96,9 +106,6 @@ struct priv {
    struct ao_alsa_opts *opts;
 };

-#define BUFFER_TIME 250000  // 250ms
-#define FRAGCOUNT 16
-
 #define CHECK_ALSA_ERROR(message) \
    do { \
        if (err < 0) { \
@@ -518,7 +525,7 @@ static int set_chmap(struct ao *ao, struct mp_chmap *dev_chmap, int num_channels

 #endif /* else HAVE_CHMAP_API */

-static void dump_hw_params(struct ao *ao, int msglevel, const char *msg,
+static void dump_hw_params(struct ao *ao, const char *msg,
                           snd_pcm_hw_params_t *hw_params)
 {
    struct priv *p = ao->priv;
@@ -530,7 +537,7 @@ static void dump_hw_params(struct ao *ao, int msglevel, const char *msg,
    char *tmp = NULL;
    size_t tmp_s = snd_output_buffer_string(p->output, &tmp);
    if (tmp)
-        mp_msg(ao->log, msglevel, "%s---\n%.*s---\n", msg, (int)tmp_s, tmp);
+        mp_msg(ao->log, MSGL_DEBUG, "%s---\n%.*s---\n", msg, (int)tmp_s, tmp);
    snd_output_flush(p->output);
 }

@@ -645,6 +652,7 @@ alsa_error: ;
 static int init_device(struct ao *ao, int mode)
 {
    struct priv *p = ao->priv;
+    struct ao_alsa_opts *opts = p->opts;
    int ret = INIT_DEVICE_ERR_GENERIC;
    char *tmp;
    size_t tmp_s;
@@ -678,7 +686,7 @@ static int init_device(struct ao *ao, int mode)
    err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
    CHECK_ALSA_ERROR("Unable to get initial parameters");

-    dump_hw_params(ao, MSGL_DEBUG, "Start HW params:\n", alsa_hwparams);
+    dump_hw_params(ao, "Start HW params:\n", alsa_hwparams);

    // Some ALSA drivers have broken delay reporting, so disable the ALSA
    // resampling plugin by default.
@@ -686,7 +694,7 @@ static int init_device(struct ao *ao, int mode)
        err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
        CHECK_ALSA_ERROR("Unable to disable resampling");
    }
-    dump_hw_params(ao, MSGL_DEBUG, "HW params after rate:\n", alsa_hwparams);
+    dump_hw_params(ao, "HW params after rate:\n", alsa_hwparams);

    snd_pcm_access_t access = af_fmt_is_planar(ao->format)
                                    ? SND_PCM_ACCESS_RW_NONINTERLEAVED
@@ -698,10 +706,10 @@ static int init_device(struct ao *ao, int mode)
        err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access);
    }
    CHECK_ALSA_ERROR("Unable to set access type");
-    dump_hw_params(ao, MSGL_DEBUG, "HW params after access:\n", alsa_hwparams);
+    dump_hw_params(ao, "HW params after access:\n", alsa_hwparams);

    bool found_format = false;
-    int try_formats[AF_FORMAT_COUNT];
+    int try_formats[AF_FORMAT_COUNT + 1];
    af_get_best_sample_formats(ao->format, try_formats);
    for (int n = 0; try_formats[n] && !found_format; n++) {
        int mp_format = try_formats[n];
@@ -741,7 +749,7 @@ static int init_device(struct ao *ao, int mode)

    err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
    CHECK_ALSA_ERROR("Unable to set format");
-    dump_hw_params(ao, MSGL_DEBUG, "HW params after format:\n", alsa_hwparams);
+    dump_hw_params(ao, "HW params after format:\n", alsa_hwparams);

    // Stereo, or mono if input is 1 channel.
    struct mp_chmap reduced;
@@ -770,7 +778,7 @@ static int init_device(struct ao *ao, int mode)
    err = snd_pcm_hw_params_set_channels_near
            (p->alsa, alsa_hwparams, &num_channels);
    CHECK_ALSA_ERROR("Unable to set channels");
-    dump_hw_params(ao, MSGL_DEBUG, "HW params after channels:\n", alsa_hwparams);
+    dump_hw_params(ao, "HW params after channels:\n", alsa_hwparams);

    if (num_channels > MP_NUM_CHANNELS) {
        MP_FATAL(ao, "Too many audio channels (%d).\n", num_channels);
@@ -780,32 +788,35 @@ static int init_device(struct ao *ao, int mode)
    err = snd_pcm_hw_params_set_rate_near
            (p->alsa, alsa_hwparams, &ao->samplerate, NULL);
    CHECK_ALSA_ERROR("Unable to set samplerate-2");
-    dump_hw_params(ao, MSGL_DEBUG, "HW params after rate-2:\n", alsa_hwparams);
+    dump_hw_params(ao, "HW params after rate-2:\n", alsa_hwparams);

    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");
-    if (err >= 0) {
+    err = 0;
+    if (opts->buffer_time) {
+        err = snd_pcm_hw_params_set_buffer_time_near
+                (p->alsa, alsa_hwparams, &(unsigned int){opts->buffer_time}, NULL);
+        CHECK_ALSA_WARN("Unable to set buffer time near");
+    }
+    if (err >= 0 && opts->frags) {
        err = snd_pcm_hw_params_set_periods_near
-                    (p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
+                    (p->alsa, alsa_hwparams, &(unsigned int){opts->frags}, NULL);
        CHECK_ALSA_WARN("Unable to set periods");
    }
    if (err < 0)
        snd_pcm_hw_params_copy(alsa_hwparams, hwparams_backup);

-    dump_hw_params(ao, MSGL_V, "Going to set final HW params:\n", alsa_hwparams);
+    dump_hw_params(ao, "Going to set final HW params:\n", alsa_hwparams);

    /* finally install hardware parameters */
    err = snd_pcm_hw_params(p->alsa, alsa_hwparams);
    ret = INIT_DEVICE_ERR_HWPARAMS;
    CHECK_ALSA_ERROR("Unable to set hw-parameters");
    ret = INIT_DEVICE_ERR_GENERIC;
-    dump_hw_params(ao, MSGL_DEBUG, "Final HW params:\n", alsa_hwparams);
+    dump_hw_params(ao, "Final HW params:\n", alsa_hwparams);

    if (set_chmap(ao, &dev_chmap, num_channels) < 0)
        goto alsa_error;
@@ -845,11 +856,6 @@ static int init_device(struct ao *ao, int mode)
            (p->alsa, alsa_swparams, p->outburst);
    CHECK_ALSA_ERROR("Unable to set start threshold");

-    /* disable underrun reporting */
-    err = snd_pcm_sw_params_set_stop_threshold
-            (p->alsa, alsa_swparams, boundary);
-    CHECK_ALSA_ERROR("Unable to set stop threshold");
-
    /* play silence when there is an underrun */
    err = snd_pcm_sw_params_set_silence_size
            (p->alsa, alsa_swparams, boundary);
@@ -933,17 +939,32 @@ static void drain(struct ao *ao)
 static int get_space(struct ao *ao)
 {
    struct priv *p = ao->priv;
-    snd_pcm_status_t *status;
-    int err;

-    snd_pcm_status_alloca(&status);
+    // in case of pausing or the device still being configured,
+    // just return our buffer size.
+    if (p->paused || snd_pcm_state(p->alsa) == SND_PCM_STATE_SETUP)
+        return p->buffersize;

-    err = snd_pcm_status(p->alsa, status);
-    if (!check_device_present(ao, err))
+    snd_pcm_sframes_t space = snd_pcm_avail(p->alsa);
+    if (space < 0) {
+        if (space == -EPIPE) {
+            MP_WARN(ao, "ALSA XRUN hit, attempting to recover...\n");
+            int err = snd_pcm_prepare(p->alsa);
+            CHECK_ALSA_ERROR("Unable to recover from under/overrun!");
+            return p->buffersize;
+        }
+
+        MP_ERR(ao, "Error received from snd_pcm_avail "
+                   "(%ld, %s with ALSA state %s)!\n",
+               space, snd_strerror(space),
+               snd_pcm_state_name(snd_pcm_state(p->alsa)));
+
+        // request a reload of the AO if device is not present,
+        // then error out.
+        check_device_present(ao, space);
        goto alsa_error;
-    CHECK_ALSA_ERROR("cannot get pcm status");
+    }

-    unsigned space = snd_pcm_status_get_avail(status);
    if (space > p->buffersize) // Buffer underrun?
        space = p->buffersize;
    return space / p->outburst * p->outburst;
@@ -1106,6 +1127,11 @@ static int play(struct ao *ao, void **data, int samples, int flags)
        } else if (res < 0) {
            if (res == -ESTRPIPE) {  /* suspend */
                resume_device(ao);
+            } else if (res == -EPIPE) {
+                // For some reason, writing a smaller fragment at the end
+                // immediately underruns.
+                if (!(flags & AOPLAY_FINAL_CHUNK))
+                    MP_WARN(ao, "Device underrun detected.\n");
            } else {
                MP_ERR(ao, "Write error: %s\n", snd_strerror(res));
            }
@@ -143,7 +143,8 @@ connect_to_outports(struct ao *ao)
    if (!port_name)
        port_flags |= JackPortIsPhysical;

-    matching_ports = jack_get_ports(p->client, port_name, NULL, port_flags);
+    const char *port_type = JACK_DEFAULT_AUDIO_TYPE; // exclude MIDI ports
+    matching_ports = jack_get_ports(p->client, port_name, port_type, port_flags);

    if (!matching_ports || !matching_ports[0]) {
        MP_FATAL(ao, "no ports to connect to\n");
@@ -40,8 +40,8 @@
 #include "common/encode_lavc.h"

 struct priv {
-    AVStream *stream;
-    AVCodecContext *codec;
+    struct encoder_context *enc;
+
    int pcmhack;
    int aframesize;
    int aframecount;
@@ -53,16 +53,17 @@ struct priv {
    double expected_next_pts;

    AVRational worst_time_base;
-    int worst_time_base_is_stream;

    bool shutdown;
 };

-static bool supports_format(AVCodec *codec, int format)
+static void encode(struct ao *ao, double apts, void **data);
+
+static bool supports_format(const AVCodec *codec, int format)
 {
    for (const enum AVSampleFormat *sampleformat = codec->sample_fmts;
         sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
-         ++sampleformat)
+         sampleformat++)
    {
        if (af_from_avformat(*sampleformat) == format)
            return true;
@@ -70,9 +71,9 @@ static bool supports_format(AVCodec *codec, int format)
    return false;
 }

-static void select_format(struct ao *ao, AVCodec *codec)
+static void select_format(struct ao *ao, const AVCodec *codec)
 {
-    int formats[AF_FORMAT_COUNT];
+    int formats[AF_FORMAT_COUNT + 1];
    af_get_best_sample_formats(ao->format, formats);

    for (int n = 0; formats[n]; n++) {
@@ -83,72 +84,64 @@ static void select_format(struct ao *ao, AVCodec *codec)
    }
 }

+static void on_ready(void *ptr)
+{
+    struct ao *ao = ptr;
+
+    ao_add_events(ao, AO_EVENT_INITIAL_UNBLOCK);
+}
+
 // open & setup audio device
 static int init(struct ao *ao)
 {
-    struct priv *ac = talloc_zero(ao, struct priv);
-    AVCodec *codec;
+    struct priv *ac = ao->priv;

-    ao->priv = ac;
-
-    if (!encode_lavc_available(ao->encode_lavc_ctx)) {
-        MP_ERR(ao, "the option --o (output file) must be specified\n");
+    ac->enc = encoder_context_alloc(ao->encode_lavc_ctx, STREAM_AUDIO, ao->log);
+    if (!ac->enc)
        return -1;
-    }
+    talloc_steal(ac, ac->enc);

-    pthread_mutex_lock(&ao->encode_lavc_ctx->lock);
-
-    if (encode_lavc_alloc_stream(ao->encode_lavc_ctx,
-                                 AVMEDIA_TYPE_AUDIO,
-                                 &ac->stream, &ac->codec) < 0) {
-      MP_ERR(ao, "could not get a new audio stream\n");
-      goto fail;
-    }
-
-    codec = ao->encode_lavc_ctx->ac;
+    AVCodecContext *encoder = ac->enc->encoder;
+    const AVCodec *codec = encoder->codec;

    int samplerate = af_select_best_samplerate(ao->samplerate,
                                               codec->supported_samplerates);
    if (samplerate > 0)
        ao->samplerate = samplerate;

-    // TODO: Remove this redundancy with encode_lavc_alloc_stream also
-    // setting the time base.
-    // Using codec->time_bvase is deprecated, but needed for older lavf.
-    ac->stream->time_base.num = 1;
-    ac->stream->time_base.den = ao->samplerate;
-    ac->codec->time_base.num = 1;
-    ac->codec->time_base.den = ao->samplerate;
+    encoder->time_base.num = 1;
+    encoder->time_base.den = ao->samplerate;

-    ac->codec->sample_rate = ao->samplerate;
+    encoder->sample_rate = ao->samplerate;

    struct mp_chmap_sel sel = {0};
    mp_chmap_sel_add_any(&sel);
    if (!ao_chmap_sel_adjust2(ao, &sel, &ao->channels, false))
        goto fail;
    mp_chmap_reorder_to_lavc(&ao->channels);
-    ac->codec->channels = ao->channels.num;
-    ac->codec->channel_layout = mp_chmap_to_lavc(&ao->channels);
+    encoder->channels = ao->channels.num;
+    encoder->channel_layout = mp_chmap_to_lavc(&ao->channels);

-    ac->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
+    encoder->sample_fmt = AV_SAMPLE_FMT_NONE;

    select_format(ao, codec);

    ac->sample_size = af_fmt_to_bytes(ao->format);
-    ac->codec->sample_fmt = af_to_avformat(ao->format);
-    ac->codec->bits_per_raw_sample = ac->sample_size * 8;
+    encoder->sample_fmt = af_to_avformat(ao->format);
+    encoder->bits_per_raw_sample = ac->sample_size * 8;

-    if (encode_lavc_open_codec(ao->encode_lavc_ctx, ac->codec) < 0)
+    if (!encoder_init_codec_and_muxer(ac->enc, on_ready, ao))
        goto fail;

    ac->pcmhack = 0;
-    if (ac->codec->frame_size <= 1)
-        ac->pcmhack = av_get_bits_per_sample(ac->codec->codec_id) / 8;
+    if (encoder->frame_size <= 1)
+        ac->pcmhack = av_get_bits_per_sample(encoder->codec_id) / 8;

-    if (ac->pcmhack)
+    if (ac->pcmhack) {
        ac->aframesize = 16384; // "enough"
-    else
-        ac->aframesize = ac->codec->frame_size;
+    } else {
+        ac->aframesize = encoder->frame_size;
+    }

    // enough frames for at least 0.25 seconds
    ac->framecount = ceil(ao->samplerate * 0.25 / ac->aframesize);
@@ -165,7 +158,6 @@ static int init(struct ao *ao)
    if (ao->channels.num > AV_NUM_DATA_POINTERS)
        goto fail;

-    pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
    return 0;

 fail:
@@ -175,34 +167,22 @@ fail:
 }

 // close audio device
-static void encode(struct ao *ao, double apts, void **data);
 static void uninit(struct ao *ao)
 {
    struct priv *ac = ao->priv;
    struct encode_lavc_context *ectx = ao->encode_lavc_ctx;

-    if (!ac || ac->shutdown)
-        return;
-
-    pthread_mutex_lock(&ectx->lock);
-
-    if (!encode_lavc_start(ectx)) {
-        MP_WARN(ao, "not even ready to encode audio at end -> dropped\n");
-        pthread_mutex_unlock(&ectx->lock);
-        return;
-    }
-
-    if (ac->stream) {
+    if (!ac->shutdown) {
        double outpts = ac->expected_next_pts;
-        if (!ectx->options->rawts && ectx->options->copyts)
+
+        pthread_mutex_lock(&ectx->lock);
+        if (!ac->enc->options->rawts)
            outpts += ectx->discontinuity_pts_offset;
-        outpts += encode_lavc_getoffset(ectx, ac->codec);
+        pthread_mutex_unlock(&ectx->lock);
+
+        outpts += encoder_get_offset(ac->enc);
        encode(ao, outpts, NULL);
    }
-
-    pthread_mutex_unlock(&ectx->lock);
-
-    ac->shutdown = true;
 }

 // return: how many samples can be played without blocking
@@ -213,106 +193,21 @@ static int get_space(struct ao *ao)
    return ac->aframesize * ac->framecount;
 }

-static void write_packet(struct ao *ao, AVPacket *packet)
-{
-    // TODO: Can we unify this with the equivalent video code path?
-    struct priv *ac = ao->priv;
-
-    packet->stream_index = ac->stream->index;
-    if (packet->pts != AV_NOPTS_VALUE) {
-        packet->pts = av_rescale_q(packet->pts,
-                                   ac->codec->time_base,
-                                   ac->stream->time_base);
-    } else {
-        // Do we need this at all? Better be safe than sorry...
-        MP_WARN(ao, "encoder lost pts, why?\n");
-        if (ac->savepts != MP_NOPTS_VALUE) {
-            packet->pts = av_rescale_q(ac->savepts,
-                                       ac->codec->time_base,
-                                       ac->stream->time_base);
-        }
-    }
-    if (packet->dts != AV_NOPTS_VALUE) {
-        packet->dts = av_rescale_q(packet->dts,
-                                   ac->codec->time_base,
-                                   ac->stream->time_base);
-    }
-    if (packet->duration > 0) {
-        packet->duration = av_rescale_q(packet->duration,
-                                        ac->codec->time_base,
-                                        ac->stream->time_base);
-    }
-
-    ac->savepts = AV_NOPTS_VALUE;
-
-    if (encode_lavc_write_frame(ao->encode_lavc_ctx,
-                                ac->stream, packet) < 0) {
-        MP_ERR(ao, "error writing at %d %d/%d\n",
-               (int) packet->pts,
-               ac->stream->time_base.num,
-               ac->stream->time_base.den);
-        return;
-    }
-}
-
-static void encode_audio_and_write(struct ao *ao, AVFrame *frame)
-{
-    // TODO: Can we unify this with the equivalent video code path?
-    struct priv *ac = ao->priv;
-    AVPacket packet = {0};
-
-    int status = avcodec_send_frame(ac->codec, frame);
-    if (status < 0) {
-        MP_ERR(ao, "error encoding at %d %d/%d\n",
-               frame ? (int) frame->pts : -1,
-               ac->codec->time_base.num,
-               ac->codec->time_base.den);
-        return;
-    }
-    for (;;) {
-        av_init_packet(&packet);
-        status = avcodec_receive_packet(ac->codec, &packet);
-        if (status == AVERROR(EAGAIN)) { // No more packets for now.
-            if (frame == NULL) {
-                MP_ERR(ao, "sent flush frame, got EAGAIN");
-            }
-            break;
-        }
-        if (status == AVERROR_EOF) { // No more packets, ever.
-            if (frame != NULL) {
-                MP_ERR(ao, "sent audio frame, got EOF");
-            }
-            break;
-        }
-        if (status < 0) {
-            MP_ERR(ao, "error encoding at %d %d/%d\n",
-                   frame ? (int) frame->pts : -1,
-                   ac->codec->time_base.num,
-                   ac->codec->time_base.den);
-            break;
-        }
-        if (frame) {
-            if (ac->savepts == AV_NOPTS_VALUE)
-                ac->savepts = frame->pts;
-        }
-        encode_lavc_write_stats(ao->encode_lavc_ctx, ac->codec);
-        write_packet(ao, &packet);
-        av_packet_unref(&packet);
-    }
-}
-
 // must get exactly ac->aframesize amount of data
 static void encode(struct ao *ao, double apts, void **data)
 {
    struct priv *ac = ao->priv;
    struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
+    AVCodecContext *encoder = ac->enc->encoder;
    double realapts = ac->aframecount * (double) ac->aframesize /
                      ao->samplerate;

    ac->aframecount++;

+    pthread_mutex_lock(&ectx->lock);
    if (data)
        ectx->audio_pts_offset = realapts - apts;
+    pthread_mutex_unlock(&ectx->lock);

    if(data) {
        AVFrame *frame = av_frame_alloc();
@@ -326,31 +221,27 @@ static void encode(struct ao *ao, double apts, void **data)

        frame->linesize[0] = frame->nb_samples * ao->sstride;

-        if (ectx->options->rawts || ectx->options->copyts) {
-            // real audio pts
-            frame->pts = floor(apts * ac->codec->time_base.den / ac->codec->time_base.num + 0.5);
-        } else {
-            // audio playback time
-            frame->pts = floor(realapts * ac->codec->time_base.den / ac->codec->time_base.num + 0.5);
-        }
+        frame->pts = rint(apts * av_q2d(av_inv_q(encoder->time_base)));

-        int64_t frame_pts = av_rescale_q(frame->pts, ac->codec->time_base, ac->worst_time_base);
+        int64_t frame_pts = av_rescale_q(frame->pts, encoder->time_base,
+                                         ac->worst_time_base);
        if (ac->lastpts != AV_NOPTS_VALUE && frame_pts <= ac->lastpts) {
            // this indicates broken video
            // (video pts failing to increase fast enough to match audio)
            MP_WARN(ao, "audio frame pts went backwards (%d <- %d), autofixed\n",
                    (int)frame->pts, (int)ac->lastpts);
            frame_pts = ac->lastpts + 1;
-            frame->pts = av_rescale_q(frame_pts, ac->worst_time_base, ac->codec->time_base);
+            frame->pts = av_rescale_q(frame_pts, ac->worst_time_base,
+                                      encoder->time_base);
        }
        ac->lastpts = frame_pts;

-        frame->quality = ac->codec->global_quality;
-        encode_audio_and_write(ao, frame);
+        frame->quality = encoder->global_quality;
+        encoder_encode(ac->enc, frame);
        av_frame_free(&frame);
+    } else {
+        encoder_encode(ac->enc, NULL);
    }
-    else
-        encode_audio_and_write(ao, NULL);
 }

 // this should round samples down to frame sizes
@@ -358,20 +249,15 @@ static void encode(struct ao *ao, double apts, void **data)
 static int play(struct ao *ao, void **data, int samples, int flags)
 {
    struct priv *ac = ao->priv;
+    struct encoder_context *enc = ac->enc;
    struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
    int bufpos = 0;
    double nextpts;
-    double outpts;
    int orig_samples = samples;

+    // for ectx PTS fields
    pthread_mutex_lock(&ectx->lock);

-    if (!encode_lavc_start(ectx)) {
-        MP_WARN(ao, "not ready yet for encoding audio\n");
-        pthread_mutex_unlock(&ectx->lock);
-        return 0;
-    }
-
    double pts = ectx->last_audio_in_pts;
    pts += ectx->samples_since_last_pts / (double)ao->samplerate;

@@ -393,60 +279,15 @@ static int play(struct ao *ao, void **data, int samples, int flags)
       samples = (bytelen + extralen) / ao->sstride;
    }

-    if (pts == MP_NOPTS_VALUE) {
-        MP_WARN(ao, "frame without pts, please report; synthesizing pts instead\n");
-        // synthesize pts from previous expected next pts
-        pts = ac->expected_next_pts;
-    }
-
-    if (ac->worst_time_base.den == 0) {
-        //if (ac->codec->time_base.num / ac->codec->time_base.den >= ac->stream->time_base.num / ac->stream->time_base.den)
-        if (ac->codec->time_base.num * (double) ac->stream->time_base.den >=
-                ac->stream->time_base.num * (double) ac->codec->time_base.den) {
-            MP_VERBOSE(ao, "NOTE: using codec time base (%d/%d) for pts "
-                       "adjustment; the stream base (%d/%d) is not worse.\n",
-                       (int)ac->codec->time_base.num,
-                       (int)ac->codec->time_base.den,
-                       (int)ac->stream->time_base.num,
-                       (int)ac->stream->time_base.den);
-            ac->worst_time_base = ac->codec->time_base;
-            ac->worst_time_base_is_stream = 0;
-        } else {
-            MP_WARN(ao, "NOTE: not using codec time base (%d/%d) for pts "
-                    "adjustment; the stream base (%d/%d) is worse.\n",
-                    (int)ac->codec->time_base.num,
-                    (int)ac->codec->time_base.den,
-                    (int)ac->stream->time_base.num,
-                    (int)ac->stream->time_base.den);
-            ac->worst_time_base = ac->stream->time_base;
-            ac->worst_time_base_is_stream = 1;
-        }
-
-        // NOTE: we use the following "axiom" of av_rescale_q:
-        // if time base A is worse than time base B, then
-        //   av_rescale_q(av_rescale_q(x, A, B), B, A) == x
-        // this can be proven as long as av_rescale_q rounds to nearest, which
-        // it currently does
-
-        // av_rescale_q(x, A, B) * B = "round x*A to nearest multiple of B"
-        // and:
-        //    av_rescale_q(av_rescale_q(x, A, B), B, A) * A
-        // == "round av_rescale_q(x, A, B)*B to nearest multiple of A"
-        // == "round 'round x*A to nearest multiple of B' to nearest multiple of A"
-        //
-        // assume this fails. Then there is a value of x*A, for which the
-        // nearest multiple of B is outside the range [(x-0.5)*A, (x+0.5)*A[.
-        // Absurd, as this range MUST contain at least one multiple of B.
-    }
-
-    // Fix and apply the discontinuity pts offset.
-    if (!ectx->options->rawts && ectx->options->copyts) {
-        // fix the discontinuity pts offset
+    double outpts = pts;
+    if (!enc->options->rawts) {
+        // Fix and apply the discontinuity pts offset.
        nextpts = pts;
        if (ectx->discontinuity_pts_offset == MP_NOPTS_VALUE) {
            ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
-        }
-        else if (fabs(nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts) > 30) {
+        } else if (fabs(nextpts + ectx->discontinuity_pts_offset -
+                        ectx->next_in_pts) > 30)
+        {
            MP_WARN(ao, "detected an unexpected discontinuity (pts jumped by "
                    "%f seconds)\n",
                    nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts);
@@ -455,12 +296,11 @@ static int play(struct ao *ao, void **data, int samples, int flags)

        outpts = pts + ectx->discontinuity_pts_offset;
    }
-    else {
-        outpts = pts;
-    }
+
+    pthread_mutex_unlock(&ectx->lock);

    // Shift pts by the pts offset first.
-    outpts += encode_lavc_getoffset(ectx, ac->codec);
+    outpts += encoder_get_offset(enc);

    while (samples - bufpos >= ac->aframesize) {
        void *start[MP_NUM_CHANNELS] = {0};
@@ -473,8 +313,10 @@ static int play(struct ao *ao, void **data, int samples, int flags)
    // Calculate expected pts of next audio frame (input side).
    ac->expected_next_pts = pts + bufpos / (double) ao->samplerate;

+    pthread_mutex_lock(&ectx->lock);
+
    // Set next allowed input pts value (input side).
-    if (!ectx->options->rawts && ectx->options->copyts) {
+    if (!enc->options->rawts) {
        nextpts = ac->expected_next_pts + ectx->discontinuity_pts_offset;
        if (nextpts > ectx->next_in_pts)
            ectx->next_in_pts = nextpts;
@@ -488,13 +330,11 @@ static int play(struct ao *ao, void **data, int samples, int flags)
    pthread_mutex_unlock(&ectx->lock);

    if (flags & AOPLAY_FINAL_CHUNK) {
-        if (bufpos < orig_samples) {
+        if (bufpos < orig_samples)
            MP_ERR(ao, "did not write enough data at the end\n");
-        }
    } else {
-        if (bufpos > orig_samples) {
+        if (bufpos > orig_samples)
            MP_ERR(ao, "audio buffer overflow (should never happen)\n");
-        }
    }

    return taken;
@@ -509,6 +349,8 @@ const struct ao_driver audio_out_lavc = {
    .encode = true,
    .description = "audio encoding using libavcodec",
    .name      = "lavc",
+    .initially_blocked = true,
+    .priv_size = sizeof(struct priv),
    .init      = init,
    .uninit    = uninit,
    .get_space = get_space,
@@ -58,6 +58,7 @@ struct priv {
    int outburst;       // samples

    struct m_channels channel_layouts;
+    int format;
 };

 static void drain(struct ao *ao)
@@ -88,6 +89,9 @@ static int init(struct ao *ao)
 {
    struct priv *priv = ao->priv;

+    if (priv->format)
+        ao->format = priv->format;
+
    ao->untimed = priv->untimed;

    struct mp_chmap_sel sel = {.tmp = ao};
@@ -241,6 +245,7 @@ const struct ao_driver audio_out_null = {
        OPT_FLAG("broken-eof", broken_eof, 0),
        OPT_FLAG("broken-delay", broken_delay, 0),
        OPT_CHANNELS("channel-layouts", channel_layouts, 0),
+        OPT_AUDIOFORMAT("format", format, 0),
        {0}
    },
    .options_prefix = "ao-null",
@@ -24,19 +24,6 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include <inttypes.h>
-
-#ifdef __APPLE__
-#ifndef AL_FORMAT_MONO_FLOAT32
-#define AL_FORMAT_MONO_FLOAT32 0x10010
-#endif
-#ifndef AL_FORMAT_STEREO_FLOAT32
-#define AL_FORMAT_STEREO_FLOAT32 0x10011
-#endif
-#ifndef AL_FORMAT_MONO_DOUBLE_EXT
-#define AL_FORMAT_MONO_DOUBLE_EXT 0x10012
-#endif
-#include <OpenAL/MacOSX_OALExtensions.h>
-#else
 #ifdef OPENAL_AL_H
 #include <OpenAL/alc.h>
 #include <OpenAL/al.h>
@@ -46,7 +33,6 @@
 #include <AL/al.h>
 #include <AL/alext.h>
 #endif
-#endif // __APPLE__

 #include "common/msg.h"

@@ -57,19 +43,22 @@
 #include "options/m_option.h"

 #define MAX_CHANS MP_NUM_CHANNELS
-#define NUM_BUF 128
-#define CHUNK_SAMPLES 256
-static ALuint buffers[MAX_CHANS][NUM_BUF];
-static ALuint sources[MAX_CHANS];
+#define MAX_BUF 128
+#define MAX_SAMPLES 32768
+static ALuint buffers[MAX_BUF];
+static ALuint buffer_size[MAX_BUF];
+static ALuint source;

-static int cur_buf[MAX_CHANS];
-static int unqueue_buf[MAX_CHANS];
+static int cur_buf;
+static int unqueue_buf;

 static struct ao *ao_data;

 struct priv {
    ALenum al_format;
-    int chunk_size;
+    int num_buffers;
+    int num_samples;
+    int direct_channels;
 };

 static void reset(struct ao *ao);
@@ -89,53 +78,99 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
        vol->left = vol->right = volume * 100;
        return CONTROL_TRUE;
    }
+    case AOCONTROL_GET_MUTE:
+    case AOCONTROL_SET_MUTE: {
+        bool mute = *(bool *)arg;
+
+        // openal has no mute control, only gain.
+        // Thus reverse the muted state to get required gain
+        ALfloat al_mute = (ALfloat)(!mute);
+        if (cmd == AOCONTROL_SET_MUTE) {
+            alSourcef(source, AL_GAIN, al_mute);
+        }
+        alGetSourcef(source, AL_GAIN, &al_mute);
+        *(bool *)arg = !((bool)al_mute);
+        return CONTROL_TRUE;
+    }
+
    case AOCONTROL_HAS_SOFT_VOLUME:
        return CONTROL_TRUE;
    }
    return CONTROL_UNKNOWN;
 }

-struct speaker {
-    int id;
-    float pos[3];
-};
-
-static const struct speaker speaker_pos[] = {
-    {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)
+static enum af_format get_supported_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:
+    case AF_FORMAT_U8:
+        if (alGetEnumValue((ALchar*)"AL_FORMAT_MONO8"))
+            return AF_FORMAT_U8;
+        break;
+
+    case AF_FORMAT_S16:
+        if (alGetEnumValue((ALchar*)"AL_FORMAT_MONO16"))
+            return AF_FORMAT_S16;
+        break;
+
+    case AF_FORMAT_S32:
+        if (strstr(alGetString(AL_RENDERER), "X-Fi") != NULL)
+            return AF_FORMAT_S32;
+        break;
+
+    case AF_FORMAT_FLOAT:
        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;
+            return AF_FORMAT_FLOAT;
        break;
    }
    return AL_FALSE;
 }

+static ALenum get_supported_layout(int format, int channels)
+{
+    const char *channel_str[] = {
+        [1] = "MONO",
+        [2] = "STEREO",
+        [4] = "QUAD",
+        [6] = "51CHN",
+        [7] = "61CHN",
+        [8] = "71CHN",
+    };
+    const char *format_str[] = {
+        [AF_FORMAT_U8] = "8",
+        [AF_FORMAT_S16] = "16",
+        [AF_FORMAT_S32] = "32",
+        [AF_FORMAT_FLOAT] = "_FLOAT32",
+    };
+    if (channel_str[channels] == NULL || format_str[format] == NULL)
+        return AL_FALSE;
+
+    char enum_name[32];
+    // AF_FORMAT_FLOAT uses same enum name as AF_FORMAT_S32 for multichannel
+    // playback, while it is different for mono and stereo.
+    // OpenAL Soft does not support AF_FORMAT_S32 and seems to reuse the names.
+    if (channels > 2 && format == AF_FORMAT_FLOAT)
+        format = AF_FORMAT_S32;
+    snprintf(enum_name, sizeof(enum_name), "AL_FORMAT_%s%s", channel_str[channels],
+             format_str[format]);
+
+    if (alGetEnumValue((ALchar*)enum_name)) {
+        return alGetEnumValue((ALchar*)enum_name);
+    }
+    return AL_FALSE;
+}
+
 // close audio device
 static void uninit(struct ao *ao)
 {
+    struct priv *p = ao->priv;
+    alSourceStop(source);
+    alSourcei(source, AL_BUFFER, 0);
+
+    alDeleteBuffers(p->num_buffers, buffers);
+    alDeleteSources(1, &source);
+
    ALCcontext *ctx = alcGetCurrentContext();
    ALCdevice *dev = alcGetContextsDevice(ctx);
-    reset(ao);
    alcMakeContextCurrent(NULL);
    alcDestroyContext(ctx);
    alcCloseDevice(dev);
@@ -150,30 +185,12 @@ static int init(struct ao *ao)
    ALCcontext *ctx = NULL;
    ALCint freq = 0;
    ALCint attribs[] = {ALC_FREQUENCY, ao->samplerate, 0, 0};
-    int i;
    struct priv *p = ao->priv;
    if (ao_data) {
        MP_FATAL(ao, "Not reentrant!\n");
        return -1;
    }
    ao_data = ao;
-    struct mp_chmap_sel sel = {0};
-    for (i = 0; speaker_pos[i].id != -1; i++)
-        mp_chmap_sel_add_speaker(&sel, speaker_pos[i].id);
-    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
-        goto err_out;
-    struct speaker speakers[MAX_CHANS];
-    for (i = 0; i < ao->channels.num; i++) {
-        speakers[i].id = -1;
-        for (int n = 0; speaker_pos[n].id >= 0; n++) {
-            if (speaker_pos[n].id == ao->channels.speaker[i])
-                speakers[i] = speaker_pos[n];
-        }
-        if (speakers[i].id < 0) {
-            MP_FATAL(ao, "Unknown channel layout\n");
-            goto err_out;
-        }
-    }
    char *dev_name = ao->device;
    dev = alcOpenDevice(dev_name && dev_name[0] ? dev_name : NULL);
    if (!dev) {
@@ -184,37 +201,74 @@ static int init(struct ao *ao)
    alcMakeContextCurrent(ctx);
    alListenerfv(AL_POSITION, position);
    alListenerfv(AL_ORIENTATION, direction);
-    alGenSources(ao->channels.num, sources);
-    for (i = 0; i < ao->channels.num; i++) {
-        cur_buf[i] = 0;
-        unqueue_buf[i] = 0;
-        alGenBuffers(NUM_BUF, buffers[i]);
-        alSourcefv(sources[i], AL_POSITION, speakers[i].pos);
-        alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
+
+    alGenSources(1, &source);
+    if (p->direct_channels && alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT")) {
+        alSourcei(source, alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT"), AL_TRUE);
    }
+
+    cur_buf = 0;
+    unqueue_buf = 0;
+    for (int i = 0; i < p->num_buffers; ++i) {
+        buffer_size[i] = 0;
+    }
+
+    alGenBuffers(p->num_buffers, buffers);
+
    alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
    if (alcGetError(dev) == ALC_NO_ERROR && freq)
        ao->samplerate = freq;

-    p->al_format = AL_FALSE;
-    int try_formats[AF_FORMAT_COUNT];
+    // Check sample format
+    int try_formats[AF_FORMAT_COUNT + 1];
+    enum af_format sample_format = 0;
    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) {
+        sample_format = get_supported_format(try_formats[n]);
+        if (sample_format != AF_FORMAT_UNKNOWN) {
            ao->format = try_formats[n];
            break;
        }
    }

-    if (p->al_format == AL_FALSE) {
+    if (sample_format == AF_FORMAT_UNKNOWN) {
        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);
-    ao->period_size = CHUNK_SAMPLES;
+    // Check if OpenAL driver supports the desired number of channels.
+    int num_channels = ao->channels.num;
+    do {
+        p->al_format = get_supported_layout(sample_format, num_channels);
+        if (p->al_format == AL_FALSE) {
+            num_channels = num_channels - 1;
+        }
+    } while (p->al_format == AL_FALSE && num_channels > 1);
+
+    // Request number of speakers for output from ao.
+    const struct mp_chmap possible_layouts[] = {
+        {0},                                        // empty
+        MP_CHMAP_INIT_MONO,                         // mono
+        MP_CHMAP_INIT_STEREO,                       // stereo
+        {0},                                        // 2.1
+        MP_CHMAP4(FL, FR, BL, BR),                  // 4.0
+        {0},                                        // 5.0
+        MP_CHMAP6(FL, FR, FC, LFE, BL, BR),         // 5.1
+        MP_CHMAP7(FL, FR, FC, LFE, SL, SR, BC),     // 6.1
+        MP_CHMAP8(FL, FR, FC, LFE, BL, BR, SL, SR), // 7.1
+    };
+    ao->channels = possible_layouts[num_channels];
+    if (!ao->channels.num)
+        mp_chmap_set_unknown(&ao->channels, num_channels);
+
+    if (p->al_format == AL_FALSE || !mp_chmap_is_valid(&ao->channels)) {
+        MP_FATAL(ao, "Can't find appropriate channel layout.\n");
+        uninit(ao);
+        goto err_out;
+    }
+
+    ao->period_size = p->num_samples;
    return 0;

 err_out:
@@ -225,30 +279,27 @@ err_out:
 static void drain(struct ao *ao)
 {
    ALint state;
-    alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
+    alGetSourcei(source, AL_SOURCE_STATE, &state);
    while (state == AL_PLAYING) {
        mp_sleep_us(10000);
-        alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
+        alGetSourcei(source, AL_SOURCE_STATE, &state);
    }
 }

-static void unqueue_buffers(void)
+static void unqueue_buffers(struct ao *ao)
 {
+    struct priv *q = ao->priv;
    ALint p;
-    int s;
-    for (s = 0; s < ao_data->channels.num; s++) {
-        int till_wrap = NUM_BUF - unqueue_buf[s];
-        alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
-        if (p >= till_wrap) {
-            alSourceUnqueueBuffers(sources[s], till_wrap,
-                                   &buffers[s][unqueue_buf[s]]);
-            unqueue_buf[s] = 0;
-            p -= till_wrap;
-        }
-        if (p) {
-            alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]);
-            unqueue_buf[s] += p;
-        }
+    int till_wrap = q->num_buffers - unqueue_buf;
+    alGetSourcei(source, AL_BUFFERS_PROCESSED, &p);
+    if (p >= till_wrap) {
+        alSourceUnqueueBuffers(source, till_wrap, &buffers[unqueue_buf]);
+        unqueue_buf = 0;
+        p -= till_wrap;
+    }
+    if (p) {
+        alSourceUnqueueBuffers(source, p, &buffers[unqueue_buf]);
+        unqueue_buf += p;
    }
 }

@@ -257,8 +308,8 @@ static void unqueue_buffers(void)
 */
 static void reset(struct ao *ao)
 {
-    alSourceStopv(ao->channels.num, sources);
-    unqueue_buffers();
+    alSourceStop(source);
+    unqueue_buffers(ao);
 }

 /**
@@ -266,7 +317,7 @@ static void reset(struct ao *ao)
 */
 static void audio_pause(struct ao *ao)
 {
-    alSourcePausev(ao->channels.num, sources);
+    alSourcePause(source);
 }

 /**
@@ -274,18 +325,19 @@ static void audio_pause(struct ao *ao)
 */
 static void audio_resume(struct ao *ao)
 {
-    alSourcePlayv(ao->channels.num, sources);
+    alSourcePlay(source);
 }

 static int get_space(struct ao *ao)
 {
+    struct priv *p = ao->priv;
    ALint queued;
-    unqueue_buffers();
-    alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
-    queued = NUM_BUF - queued - 3;
+    unqueue_buffers(ao);
+    alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
+    queued = p->num_buffers - queued;
    if (queued < 0)
        return 0;
-    return queued * CHUNK_SAMPLES;
+    return p->num_samples * queued;
 }

 /**
@@ -294,30 +346,66 @@ 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++) {
-            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;
-        }
+
+    int buffered_samples = 0;
+    int num = 0;
+    if (flags & AOPLAY_FINAL_CHUNK) {
+        num = 1;
+        buffered_samples = samples;
+    } else {
+        num = samples / p->num_samples;
+        buffered_samples = num * p->num_samples;
    }
-    alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
+
+    for (int i = 0; i < num; i++) {
+        char *d = *data;
+        if (flags & AOPLAY_FINAL_CHUNK) {
+            buffer_size[cur_buf] = samples;
+        } else {
+            buffer_size[cur_buf] = p->num_samples;
+        }
+        d += i * buffer_size[cur_buf] * ao->sstride;
+        alBufferData(buffers[cur_buf], p->al_format, d,
+            buffer_size[cur_buf] * ao->sstride, ao->samplerate);
+        alSourceQueueBuffers(source, 1, &buffers[cur_buf]);
+        cur_buf = (cur_buf + 1) % p->num_buffers;
+    }
+
+    ALint state;
+    alGetSourcei(source, AL_SOURCE_STATE, &state);
    if (state != AL_PLAYING) // checked here in case of an underrun
-        alSourcePlayv(ao->channels.num, sources);
-    return num * CHUNK_SAMPLES;
+        alSourcePlay(source);
+
+    return buffered_samples;
 }

 static double get_delay(struct ao *ao)
 {
+    struct priv *p = ao->priv;
    ALint queued;
-    unqueue_buffers();
-    alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
-    return queued * CHUNK_SAMPLES / (double)ao->samplerate;
+    unqueue_buffers(ao);
+    alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
+
+    double soft_source_latency = 0;
+    if(alIsExtensionPresent("AL_SOFT_source_latency")) {
+        ALdouble offsets[2];
+        LPALGETSOURCEDVSOFT alGetSourcedvSOFT = alGetProcAddress("alGetSourcedvSOFT");
+        alGetSourcedvSOFT(source, AL_SEC_OFFSET_LATENCY_SOFT, offsets);
+        // Additional latency to the play buffer, the remaining seconds to be
+        // played minus the offset (seconds already played)
+        soft_source_latency = offsets[1] - offsets[0];
+    } else {
+        float offset = 0;
+        alGetSourcef(source, AL_SEC_OFFSET, &offset);
+        soft_source_latency = -offset;
+    }
+
+    int queued_samples = 0;
+    for (int i = 0, index = cur_buf; i < queued; ++i) {
+        queued_samples += buffer_size[index];
+        index = (index + 1) % p->num_buffers;
+    }
+    return (queued_samples / (double)ao->samplerate) + soft_source_latency;
 }

 #define OPT_BASE_STRUCT struct priv
@@ -336,4 +424,16 @@ const struct ao_driver audio_out_openal = {
    .reset     = reset,
    .drain     = drain,
    .priv_size = sizeof(struct priv),
+    .priv_defaults = &(const struct priv) {
+        .num_buffers = 4,
+        .num_samples = 8192,
+        .direct_channels = 0,
+    },
+    .options = (const struct m_option[]) {
+        OPT_INTRANGE("num-buffers", num_buffers, 0, 2, MAX_BUF),
+        OPT_INTRANGE("num-samples", num_samples, 0, 256, MAX_SAMPLES),
+        OPT_FLAG("direct-channels", direct_channels, 0),
+        {0}
+    },
+    .options_prefix = "openal",
 };
@@ -35,12 +35,12 @@ struct priv {
    SLBufferQueueItf buffer_queue;
    SLEngineItf engine;
    SLPlayItf play;
-    char *curbuf, *buf1, *buf2;
+    char *buf;
    size_t buffer_size;
    pthread_mutex_t buffer_lock;
+    double audio_latency;

    int cfg_frames_per_buffer;
-    int cfg_sample_rate;
 };

 static const int fmtmap[][2] = {
@@ -69,9 +69,8 @@ static void uninit(struct ao *ao)

    pthread_mutex_destroy(&p->buffer_lock);

-    free(p->buf1);
-    free(p->buf2);
-    p->curbuf = p->buf1 = p->buf2 = NULL;
+    free(p->buf);
+    p->buf = NULL;
    p->buffer_size = 0;
 }

@@ -87,21 +86,20 @@ static void buffer_callback(SLBufferQueueItf buffer_queue, void *context)

    pthread_mutex_lock(&p->buffer_lock);

-    data[0] = p->curbuf;
+    data[0] = p->buf;
    delay = 2 * p->buffer_size / (double)ao->bps;
+    delay += p->audio_latency;
    ao_read_data(ao, data, p->buffer_size / ao->sstride,
        mp_time_us() + 1000000LL * delay);

-    res = (*buffer_queue)->Enqueue(buffer_queue, p->curbuf, p->buffer_size);
+    res = (*buffer_queue)->Enqueue(buffer_queue, p->buf, p->buffer_size);
    if (res != SL_RESULT_SUCCESS)
        MP_ERR(ao, "Failed to Enqueue: %d\n", res);
-    else
-        p->curbuf = (p->curbuf == p->buf1) ? p->buf2 : p->buf1;

    pthread_mutex_unlock(&p->buffer_lock);
 }

-#define DEFAULT_BUFFER_SIZE_MS 50
+#define DEFAULT_BUFFER_SIZE_MS 250

 #define CHK(stmt) \
    { \
@@ -131,12 +129,12 @@ static int init(struct ao *ao)
    CHK((*p->output_mix)->Realize(p->output_mix, SL_BOOLEAN_FALSE));

    locator_buffer_queue.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
-    locator_buffer_queue.numBuffers = 2;
+    locator_buffer_queue.numBuffers = 1;

    pcm.formatType = SL_DATAFORMAT_PCM;
    pcm.numChannels = 2;

-    int compatible_formats[AF_FORMAT_COUNT];
+    int compatible_formats[AF_FORMAT_COUNT + 1];
    af_get_best_sample_formats(ao->format, compatible_formats);
    pcm.bitsPerSample = 0;
    for (int i = 0; compatible_formats[i] && !pcm.bitsPerSample; ++i)
@@ -154,9 +152,6 @@ static int init(struct ao *ao)
    pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
    pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;

-    if (p->cfg_sample_rate)
-        ao->samplerate = p->cfg_sample_rate;
-
    // samplesPerSec is misnamed, actually it's samples per ms
    pcm.samplesPerSec = ao->samplerate * 1000;

@@ -166,10 +161,8 @@ static int init(struct ao *ao)
        ao->device_buffer = ao->samplerate * DEFAULT_BUFFER_SIZE_MS / 1000;
    p->buffer_size = ao->device_buffer * ao->channels.num *
        af_fmt_to_bytes(ao->format);
-    p->buf1 = calloc(1, p->buffer_size);
-    p->buf2 = calloc(1, p->buffer_size);
-    p->curbuf = p->buf1;
-    if (!p->buf1 || !p->buf2) {
+    p->buf = calloc(1, p->buffer_size);
+    if (!p->buf) {
        MP_ERR(ao, "Failed to allocate device buffer\n");
        goto error;
    }
@@ -188,16 +181,41 @@ static int init(struct ao *ao)
    audio_sink.pLocator = (void*)&locator_output_mix;
    audio_sink.pFormat = NULL;

-    SLboolean required[] = { SL_BOOLEAN_TRUE };
-    SLInterfaceID iid_array[] = { SL_IID_BUFFERQUEUE };
+    SLInterfaceID iid_array[] = { SL_IID_BUFFERQUEUE, SL_IID_ANDROIDCONFIGURATION };
+    SLboolean required[] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE };
    CHK((*p->engine)->CreateAudioPlayer(p->engine, &p->player, &audio_source,
-        &audio_sink, 1, iid_array, required));
+        &audio_sink, 2, iid_array, required));
+
    CHK((*p->player)->Realize(p->player, SL_BOOLEAN_FALSE));
    CHK((*p->player)->GetInterface(p->player, SL_IID_PLAY, (void*)&p->play));
    CHK((*p->player)->GetInterface(p->player, SL_IID_BUFFERQUEUE,
        (void*)&p->buffer_queue));
    CHK((*p->buffer_queue)->RegisterCallback(p->buffer_queue,
        buffer_callback, ao));
+    CHK((*p->play)->SetPlayState(p->play, SL_PLAYSTATE_PLAYING));
+
+    SLAndroidConfigurationItf android_config;
+    SLuint32 audio_latency = 0, value_size = sizeof(SLuint32);
+
+    SLint32 get_interface_result = (*p->player)->GetInterface(
+        p->player,
+        SL_IID_ANDROIDCONFIGURATION,
+        &android_config
+    );
+
+    if (get_interface_result == SL_RESULT_SUCCESS) {
+        SLint32 get_configuration_result = (*android_config)->GetConfiguration(
+            android_config,
+            (const SLchar *)"androidGetAudioLatency",
+            &value_size,
+            &audio_latency
+        );
+
+        if (get_configuration_result == SL_RESULT_SUCCESS) {
+            p->audio_latency = (double)audio_latency / 1000.0;
+            MP_INFO(ao, "Device latency is %f\n", p->audio_latency);
+        }
+    }

    return 1;
 error:
@@ -207,26 +225,15 @@ error:

 #undef CHK

-static void set_play_state(struct ao *ao, SLuint32 state)
-{
-    struct priv *p = ao->priv;
-    SLresult res = (*p->play)->SetPlayState(p->play, state);
-    if (res != SL_RESULT_SUCCESS)
-        MP_ERR(ao, "Failed to SetPlayState(%d): %d\n", state, res);
-}
-
 static void reset(struct ao *ao)
 {
-    set_play_state(ao, SL_PLAYSTATE_STOPPED);
+    struct priv *p = ao->priv;
+    (*p->buffer_queue)->Clear(p->buffer_queue);
 }

 static void resume(struct ao *ao)
 {
    struct priv *p = ao->priv;
-    set_play_state(ao, SL_PLAYSTATE_PLAYING);
-
-    // enqueue two buffers
-    buffer_callback(p->buffer_queue, ao);
    buffer_callback(p->buffer_queue, ao);
 }

@@ -242,8 +249,7 @@ const struct ao_driver audio_out_opensles = {

    .priv_size = sizeof(struct priv),
    .options = (const struct m_option[]) {
-        OPT_INTRANGE("frames-per-buffer", cfg_frames_per_buffer, 0, 1, 10000),
-        OPT_INTRANGE("sample-rate", cfg_sample_rate, 0, 1000, 100000),
+        OPT_INTRANGE("frames-per-buffer", cfg_frames_per_buffer, 0, 1, 96000),
        {0}
    },
    .options_prefix = "opensles",
@@ -56,7 +56,11 @@
 // Define to 0 if the device must be reopened to reset it (stop all playback,
 // clear the buffer), and the device should be closed when unused.
 // Define to 1 if SNDCTL_DSP_RESET should be used to reset without close.
-#define KEEP_DEVICE (defined(SNDCTL_DSP_RESET) && !defined(__NetBSD__))
+#if defined(SNDCTL_DSP_RESET) && !defined(__NetBSD__)
+#define KEEP_DEVICE 1
+#else
+#define KEEP_DEVICE 0
+#endif

 #define PATH_DEV_DSP "/dev/dsp"
 #define PATH_DEV_MIXER "/dev/mixer"
@@ -311,7 +315,7 @@ static int reopen_device(struct ao *ao, bool allow_format_changes)
        }
    }

-    int try_formats[AF_FORMAT_COUNT];
+    int try_formats[AF_FORMAT_COUNT + 1];
    af_get_best_sample_formats(format, try_formats);
    for (int n = 0; try_formats[n]; n++) {
        format = try_formats[n];
@@ -454,11 +454,11 @@ static int init(struct ao *ao)
    pa_stream_set_write_callback(priv->stream, stream_request_cb, ao);
    pa_stream_set_latency_update_callback(priv->stream,
                                          stream_latency_update_cb, ao);
-    int buf_size = af_fmt_seconds_to_bytes(ao->format, priv->cfg_buffer / 1000.0,
-                                           ao->channels.num, ao->samplerate);
+    uint32_t buf_size = ao->samplerate * (priv->cfg_buffer / 1000.0) *
+        af_fmt_to_bytes(ao->format) * ao->channels.num;
    pa_buffer_attr bufattr = {
        .maxlength = -1,
-        .tlength = buf_size > 0 ? buf_size : (uint32_t)-1,
+        .tlength = buf_size > 0 ? buf_size : -1,
        .prebuf = -1,
        .minreq = -1,
        .fragsize = -1,
@@ -831,7 +831,7 @@ const struct ao_driver audio_out_pulse = {
    .list_devs = list_devs,
    .priv_size = sizeof(struct priv),
    .priv_defaults = &(const struct priv) {
-        .cfg_buffer = 250,
+        .cfg_buffer = 100,
    },
    .options = (const struct m_option[]) {
        OPT_STRING("host", cfg_host, 0),
@@ -113,8 +113,7 @@ static int init(struct ao *ao)

    ao->format = af_fmt_from_planar(ao->format);

-    SDL_AudioSpec desired, obtained;
-
+    SDL_AudioSpec desired = {0};
    desired.format = AUDIO_S16SYS;
    for (int n = 0; fmtmap[n][0]; n++) {
        if (ao->format == fmtmap[n][0]) {
@@ -124,8 +123,10 @@ static int init(struct ao *ao)
    }
    desired.freq = ao->samplerate;
    desired.channels = ao->channels.num;
-    desired.samples = MPMIN(32768, ceil_power_of_two(ao->samplerate *
-                                                     priv->buflen));
+    if (priv->buflen) {
+        desired.samples = MPMIN(32768, ceil_power_of_two(ao->samplerate *
+                                                         priv->buflen));
+    }
    desired.callback = audio_callback;
    desired.userdata = ao;

@@ -134,7 +135,7 @@ static int init(struct ao *ao)
               (int) desired.freq, (int) desired.channels,
               (int) desired.format, (int) desired.samples);

-    obtained = desired;
+    SDL_AudioSpec obtained = desired;
    if (SDL_OpenAudio(&desired, &obtained)) {
        if (!ao->probing)
            MP_ERR(ao, "could not open audio: %s\n", SDL_GetError());
@@ -97,7 +97,7 @@ static const struct wasapi_sample_fmt wasapi_formats[] = {
 static void wasapi_get_best_sample_formats(
    int src_format, struct wasapi_sample_fmt *out_formats)
 {
-    int mp_formats[AF_FORMAT_COUNT];
+    int mp_formats[AF_FORMAT_COUNT + 1];
    af_get_best_sample_formats(src_format, mp_formats);
    for (int n = 0; mp_formats[n]; n++) {
        for (int i = 0; wasapi_formats[i].mp_format; i++) {
@@ -130,6 +130,10 @@ struct ao_driver {
    const char *name;
    // Description shown with --ao=help.
    const char *description;
+    // This requires waiting for a AO_EVENT_INITIAL_UNBLOCK event before the
+    // first play() call is done. Encode mode uses this, and push mode
+    // respects it automatically (don't use with pull mode).
+    bool initially_blocked;
    // Init the device using ao->format/ao->channels/ao->samplerate. If the
    // device doesn't accept these parameters, you can attempt to negotiate
    // fallback parameters, and set the ao format fields accordingly.
@@ -56,6 +56,7 @@ struct ao_push_state {
    bool still_playing;
    bool need_wakeup;
    bool paused;
+    bool initial_unblocked;

    // Whether the current buffer contains the complete audio.
    bool final_chunk;
@@ -357,7 +358,8 @@ static void *playthread(void *arg)
    mpthread_set_name("ao");
    pthread_mutex_lock(&p->lock);
    while (!p->terminate) {
-        bool playing = !p->paused || ao->stream_silence;
+        bool blocked = ao->driver->initially_blocked && !p->initial_unblocked;
+        bool playing = (!p->paused || ao->stream_silence) && !blocked;
        if (playing)
            ao_play_data(ao);

@@ -502,6 +504,19 @@ int ao_play_silence(struct ao *ao, int samples)
    return ao->driver->play(ao, (void **)p->silence, samples, 0);
 }

+void ao_unblock(struct ao *ao)
+{
+    if (ao->api == &ao_api_push) {
+        struct ao_push_state *p = ao->api_priv;
+        pthread_mutex_lock(&p->lock);
+        p->need_wakeup = true;
+        p->initial_unblocked = true;
+        wakeup_playthread(ao);
+        pthread_cond_signal(&p->wakeup);
+        pthread_mutex_unlock(&p->lock);
+    }
+}
+
 #ifndef __MINGW32__

 #include <poll.h>
@@ -5,10 +5,10 @@
 from __future__ import print_function
 import os, sys, stat, hashlib, subprocess

-WAFRELEASE = "waf-1.9.8"
+WAFRELEASE = "waf-2.0.9"
 WAFURLS    = ["https://waf.io/" + WAFRELEASE,
              "http://www.freehackers.org/~tnagy/release/" + WAFRELEASE]
-SHA256HASH = "167dc42bab6d5bd823b798af195420319cb5c9b571e00db7d83df2a0fe1f4dbf"
+SHA256HASH = "2a8e0816f023995e557f79ea8940d322bec18f286917c8f9a6fa2dc3875dfa48"

 if os.path.exists("waf"):
    wafver = subprocess.check_output([sys.executable, './waf', '--version']).decode()
@@ -0,0 +1,34 @@
+#!/bin/sh
+set -e
+
+_mingw_sysroot=/usr/$TARGET/sysroot
+_mingw_prefix=$_mingw_sysroot/mingw
+_mingw_exec_prefix=$_mingw_prefix
+_mingw_libdir=$_mingw_exec_prefix/lib
+_mingw_datadir=$_mingw_prefix/share
+
+export PKG_CONFIG_PATH="$_mingw_libdir/pkgconfig:$_mingw_datadir/pkgconfig";
+export CC=$TARGET-gcc
+export CXX=$TARGET-g++
+export AR=$TARGET-ar
+export NM=$TARGET-nm
+export RANLIB=$TARGET-ranlib
+export CFLAGS="-O2 -mtune=intel -g -ggdb -pipe -Wall --param=ssp-buffer-size=4 -mms-bitfields -fmessage-length=0 -D_FORTIFY_SOURCE=2 -fexceptions -fasynchronous-unwind-tables -fstack-protector-strong -fno-ident"
+export LDFLAGS="-Wl,--no-keep-memory -fstack-protector-strong"
+
+python3 ./waf configure \
+    --enable-static-build \
+    --enable-libmpv-shared \
+    --enable-lua \
+    --enable-javascript \
+    --enable-libarchive \
+    --enable-libass \
+    --enable-libbluray \
+    --enable-dvdread \
+    --enable-dvdnav \
+    --enable-uchardet \
+    --enable-vulkan \
+    --enable-shaderc \
+    --enable-rubberband \
+    --enable-lcms2
+python3 ./waf build --verbose
@@ -0,0 +1,16 @@
+#!/bin/sh
+set -e
+
+python3 ./waf configure \
+  --enable-cdda          \
+  --enable-dvbin         \
+  --enable-dvdnav        \
+  --enable-dvdread       \
+  --enable-libarchive    \
+  --enable-libmpv-shared \
+  --enable-libsmbclient  \
+  --enable-manpage-build \
+  --enable-shaderc       \
+  --enable-vulkan        \
+  --enable-zsh-comp
+python3 ./waf build --verbose
@@ -26,6 +26,7 @@
 #include <libavutil/error.h>
 #include <libavutil/cpu.h>
 #include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>

 #include "config.h"

@@ -33,6 +34,7 @@
 #include "common/msg.h"
 #include "demux/packet.h"
 #include "demux/stheader.h"
+#include "misc/bstr.h"
 #include "video/fmt-conversion.h"
 #include "av_common.h"
 #include "codecs.h"
@@ -217,33 +219,47 @@ void mp_set_avcodec_threads(struct mp_log *l, AVCodecContext *avctx, int threads
    avctx->thread_count = threads;
 }

-void mp_add_lavc_decoders(struct mp_decoder_list *list, enum AVMediaType type)
+static void add_codecs(struct mp_decoder_list *list, enum AVMediaType type,
+                       bool decoders)
 {
-    AVCodec *cur = NULL;
+    void *iter = NULL;
    for (;;) {
-        cur = av_codec_next(cur);
+        const AVCodec *cur = av_codec_iterate(&iter);
        if (!cur)
            break;
-        if (av_codec_is_decoder(cur) && cur->type == type) {
-            mp_add_decoder(list, "lavc", mp_codec_from_av_codec_id(cur->id),
+        if (av_codec_is_decoder(cur) == decoders &&
+            (type == AVMEDIA_TYPE_UNKNOWN || cur->type == type))
+        {
+            mp_add_decoder(list, mp_codec_from_av_codec_id(cur->id),
                           cur->name, cur->long_name);
        }
    }
 }

+void mp_add_lavc_decoders(struct mp_decoder_list *list, enum AVMediaType type)
+{
+    add_codecs(list, type, true);
+}
+
 // (Abuses the decoder list data structures.)
 void mp_add_lavc_encoders(struct mp_decoder_list *list)
 {
-    AVCodec *cur = NULL;
+    add_codecs(list, AVMEDIA_TYPE_UNKNOWN, false);
+}
+
+char **mp_get_lavf_demuxers(void)
+{
+    char **list = NULL;
+    void *iter = NULL;
+    int num = 0;
    for (;;) {
-        cur = av_codec_next(cur);
+        const AVInputFormat *cur = av_demuxer_iterate(&iter);
        if (!cur)
            break;
-        if (av_codec_is_encoder(cur)) {
-            mp_add_decoder(list, "lavc", mp_codec_from_av_codec_id(cur->id),
-                           cur->name, cur->long_name);
-        }
+        MP_TARRAY_APPEND(NULL, list, num, talloc_strdup(NULL, cur->name));
    }
+    MP_TARRAY_APPEND(NULL, list, num, NULL);
+    return list;
 }

 int mp_codec_to_av_codec_id(const char *codec)
@@ -297,7 +313,7 @@ void mp_avdict_print_unset(struct mp_log *log, int msgl, AVDictionary *dict)
 // to the name of the n-th parameter.
 static void resolve_positional_arg(void *avobj, char **name)
 {
-    if (!*name || (*name)[0] != '@')
+    if (!*name || (*name)[0] != '@' || !avobj)
        return;

    char *end = NULL;
@@ -328,12 +344,19 @@ static void resolve_positional_arg(void *avobj, char **name)
 // Options which fail to set (error or not found) are printed to log.
 // Returns: >=0 success, <0 failed to set an option
 int mp_set_avopts(struct mp_log *log, void *avobj, char **kv)
+{
+    return mp_set_avopts_pos(log, avobj, avobj, kv);
+}
+
+// Like mp_set_avopts(), but the posargs argument is used to resolve positional
+// arguments. If posargs==NULL, positional args are disabled.
+int mp_set_avopts_pos(struct mp_log *log, void *avobj, void *posargs, char **kv)
 {
    int success = 0;
    for (int n = 0; kv && kv[n * 2]; n++) {
        char *k = kv[n * 2 + 0];
        char *v = kv[n * 2 + 1];
-        resolve_positional_arg(avobj, &k);
+        resolve_positional_arg(posargs, &k);
        int r = av_opt_set(avobj, k, v, AV_OPT_SEARCH_CHILDREN);
        if (r == AVERROR_OPTION_NOT_FOUND) {
            mp_err(log, "AVOption '%s' not found.\n", k);
@@ -347,51 +370,3 @@ int mp_set_avopts(struct mp_log *log, void *avobj, char **kv)
    }
    return success;
 }
-
-#if LIBAVUTIL_VERSION_MICRO >= 100
-AVFrameSideData *ffmpeg_garbage(AVFrame *frame,
-                                enum AVFrameSideDataType type,
-                                AVBufferRef *buf)
-{
-    AVFrameSideData *ret, **tmp;
-
-    if (!buf)
-        return NULL;
-
-    if (frame->nb_side_data > INT_MAX / sizeof(*frame->side_data) - 1)
-        goto fail;
-
-    tmp = av_realloc(frame->side_data,
-                     (frame->nb_side_data + 1) * sizeof(*frame->side_data));
-    if (!tmp)
-        goto fail;
-    frame->side_data = tmp;
-
-    ret = av_mallocz(sizeof(*ret));
-    if (!ret)
-        goto fail;
-
-    ret->buf = buf;
-    ret->data = ret->buf->data;
-    ret->size = buf->size;
-    ret->type = type;
-
-    frame->side_data[frame->nb_side_data++] = ret;
-
-    return ret;
-fail:
-    av_buffer_unref(&buf);
-    return NULL;
-}
-#else
-AVFrameSideData *ffmpeg_garbage(AVFrame *frame,
-                                enum AVFrameSideDataType type,
-                                AVBufferRef *buf)
-{
-    AVFrameSideData *sd = av_frame_new_side_data(frame, type, buf->size);
-    if (sd)
-        memcpy(sd->data, buf->data, buf->size);
-    av_buffer_unref(&buf);
-    return sd;
-}
-#endif
@@ -41,13 +41,12 @@ double mp_pts_from_av(int64_t av_pts, AVRational *tb);
 void mp_set_avcodec_threads(struct mp_log *l, AVCodecContext *avctx, int threads);
 void mp_add_lavc_decoders(struct mp_decoder_list *list, enum AVMediaType type);
 void mp_add_lavc_encoders(struct mp_decoder_list *list);
+char **mp_get_lavf_demuxers(void);
 int mp_codec_to_av_codec_id(const char *codec);
 const char *mp_codec_from_av_codec_id(int codec_id);
 void mp_set_avdict(struct AVDictionary **dict, char **kv);
 void mp_avdict_print_unset(struct mp_log *log, int msgl, struct AVDictionary *d);
 int mp_set_avopts(struct mp_log *log, void *avobj, char **kv);
-AVFrameSideData *ffmpeg_garbage(AVFrame *frame,
-                                enum AVFrameSideDataType type,
-                                AVBufferRef *buf);
+int mp_set_avopts_pos(struct mp_log *log, void *avobj, void *posargs, char **kv);

 #endif
@@ -93,10 +93,10 @@ static struct mp_log *get_av_log(void *ptr)
        AVCodecContext *s = ptr;
        if (s->codec) {
            if (s->codec->type == AVMEDIA_TYPE_AUDIO) {
-                if (s->codec->decode)
+                if (av_codec_is_decoder(s->codec))
                    return log_decaudio;
            } else if (s->codec->type == AVMEDIA_TYPE_VIDEO) {
-                if (s->codec->decode)
+                if (av_codec_is_decoder(s->codec))
                    return log_decvideo;
            }
        }
@@ -159,10 +159,7 @@ void init_libav(struct mpv_global *global)
    }
    pthread_mutex_unlock(&log_lock);

-    avcodec_register_all();
-    av_register_all();
    avformat_network_init();
-    avfilter_register_all();

 #if HAVE_LIBAVDEVICE
    avdevice_register_all();
@@ -21,11 +21,10 @@
 #include "common/msg.h"
 #include "codecs.h"

-void mp_add_decoder(struct mp_decoder_list *list, const char *family,
-                    const char *codec, const char *decoder, const char *desc)
+void mp_add_decoder(struct mp_decoder_list *list, const char *codec,
+                    const char *decoder, const char *desc)
 {
    struct mp_decoder_entry entry = {
-        .family = talloc_strdup(list, family),
        .codec = talloc_strdup(list, codec),
        .decoder = talloc_strdup(list, decoder),
        .desc = talloc_strdup(list, desc),
@@ -40,7 +39,7 @@ static void add_new(struct mp_decoder_list *to, struct mp_decoder_entry *entry,
 {
    if (!entry || (codec && strcmp(entry->codec, codec) != 0))
        return;
-    mp_add_decoder(to, entry->family, entry->codec, entry->decoder, entry->desc);
+    mp_add_decoder(to, entry->codec, entry->decoder, entry->desc);
 }

 // Select a decoder from the given list for the given codec. The selection
@@ -72,11 +71,6 @@ struct mp_decoder_list *mp_select_decoders(struct mp_log *log,
            stop = true;
            break;
        }
-        if (bstr_find0(entry, ":") >= 0) {
-            mp_err(log, "Codec family selection was removed. "
-                        "Pass the codec name directly.\n");
-            break;
-        }
        for (int n = 0; n < all->num_entries; n++) {
            struct mp_decoder_entry *cur = &all->entries[n];
            if (bstr_equals0(entry, cur->decoder))
@@ -21,7 +21,6 @@
 struct mp_log;

 struct mp_decoder_entry {
-    const char *family;         // decoder module (e.g. ad_lavc => "lavc")
    const char *codec;          // name of the codec (e.g. "mp3")
    const char *decoder;        // decoder name (e.g. "mp3float")
    const char *desc;           // human readable description
@@ -32,8 +31,8 @@ struct mp_decoder_list {
    int num_entries;
 };

-void mp_add_decoder(struct mp_decoder_list *list, const char *family,
-                    const char *codec, const char *decoder, const char *desc);
+void mp_add_decoder(struct mp_decoder_list *list, const char *codec,
+                    const char *decoder, const char *desc);

 struct mp_decoder_list *mp_select_decoders(struct mp_log *log,
                                           struct mp_decoder_list *all,
@@ -16,6 +16,7 @@
 */

 #include <stdarg.h>
+#include <math.h>
 #include <assert.h>

 #include <libavutil/common.h>
@@ -48,14 +49,17 @@ char *mp_format_time_fmt(const char *fmt, double time)
    time = time < 0 ? -time : time;
    long long int itime = time;
    long long int h, m, tm, s;
-    int ms;
+    int ms = lrint((time - itime) * 1000);
+    if (ms >= 1000) {
+        ms -= 1000;
+        itime += 1;
+    }
    s = itime;
    tm = s / 60;
    h = s / 3600;
    s -= h * 3600;
    m = s / 60;
    s -= m * 60;
-    ms = (time - itime) * 1000;
    char *res = talloc_strdup(NULL, "");
    while (*fmt) {
        if (fmt[0] == '%') {
@@ -68,6 +72,7 @@ char *mp_format_time_fmt(const char *fmt, double time)
            case 's': appendf(&res, "%s%lld", sign, itime); break;
            case 'S': appendf(&res, "%02lld", s); break;
            case 'T': appendf(&res, "%03d", ms); break;
+            case 'f': appendf(&res, "%f", time); break;
            case '%': appendf(&res, "%s", "%"); break;
            default: goto error;
            }
@@ -302,3 +307,14 @@ char *mp_tprintf_buf(char *buf, size_t buf_size, const char *format, ...)
    va_end(ap);
    return buf;
 }
+
+char **mp_dup_str_array(void *tctx, char **s)
+{
+    char **r = NULL;
+    int num_r = 0;
+    for (int n = 0; s && s[n]; n++)
+        MP_TARRAY_APPEND(tctx, r, num_r, talloc_strdup(tctx, s[n]));
+    if (r)
+        MP_TARRAY_APPEND(tctx, r, num_r, NULL);
+    return r;
+}
@@ -58,16 +58,9 @@ enum stream_type {
    STREAM_TYPE_COUNT,
 };

-enum {
-    DATA_OK     = 1,        // data is actually being returned
-    DATA_WAIT   = 0,        // async wait: check state again after next wakeup
-    DATA_AGAIN  = -2,       // repeat request (internal progress was made)
-    DATA_STARVE = -1,       // need input (might require to drain other outputs)
-    DATA_EOF    = -3,       // no more data available
-};
-
 extern const char mpv_version[];
 extern const char mpv_builddate[];
+extern const char mpv_copyright[];

 char *mp_format_time(double time, bool fractions);
 char *mp_format_time_fmt(const char *fmt, double time);
@@ -111,4 +104,17 @@ char *mp_tag_str_buf(char *buf, size_t buf_size, uint32_t tag);
 char *mp_tprintf_buf(char *buf, size_t buf_size, const char *format, ...)
    PRINTF_ATTRIBUTE(3, 4);

+char **mp_dup_str_array(void *tctx, char **s);
+
+// We generally do not handle allocation failure of small malloc()s. This would
+// create a large number of rarely tested code paths, which would probably
+// regress and cause security issues. We prefer to fail fast.
+// This macro generally behaves like an assert(), except it will make sure to
+// kill the process even with NDEBUG.
+#define MP_HANDLE_OOM(x) do {   \
+        assert(x);              \
+        if (!(x))               \
+            abort();            \
+    } while (0)
+
 #endif /* MPLAYER_MPCOMMON_H */
@@ -34,36 +34,32 @@ struct encode_opts {
    char *file;
    char *format;
    char **fopts;
-    float fps;
-    float maxfps;
    char *vcodec;
    char **vopts;
    char *acodec;
    char **aopts;
-    int harddup;
    float voffset;
    float aoffset;
-    int copyts;
    int rawts;
-    int autofps;
-    int neverdrop;
    int video_first;
    int audio_first;
-    int metadata;
+    int copy_metadata;
+    char **set_metadata;
+    char **remove_metadata;
 };

-// interface for mplayer.c
-struct encode_lavc_context *encode_lavc_init(struct encode_opts *options,
-                                             struct mpv_global *global);
-void encode_lavc_finish(struct encode_lavc_context *ctx);
-void encode_lavc_free(struct encode_lavc_context *ctx);
+// interface for player core
+struct encode_lavc_context *encode_lavc_init(struct mpv_global *global);
+bool encode_lavc_free(struct encode_lavc_context *ctx);
 void encode_lavc_discontinuity(struct encode_lavc_context *ctx);
 bool encode_lavc_showhelp(struct mp_log *log, struct encode_opts *options);
 int encode_lavc_getstatus(struct encode_lavc_context *ctx, char *buf, int bufsize, float relative_position);
-void encode_lavc_expect_stream(struct encode_lavc_context *ctx, int mt);
+void encode_lavc_expect_stream(struct encode_lavc_context *ctx,
+                               enum stream_type type);
+void encode_lavc_stream_eof(struct encode_lavc_context *ctx,
+                            enum stream_type type);
 void encode_lavc_set_metadata(struct encode_lavc_context *ctx,
                              struct mp_tags *metadata);
-void encode_lavc_set_video_fps(struct encode_lavc_context *ctx, float fps);
 void encode_lavc_set_audio_pts(struct encode_lavc_context *ctx, double pts);
 bool encode_lavc_didfail(struct encode_lavc_context *ctx); // check if encoding failed

@@ -31,43 +31,26 @@
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>

+#include "common/common.h"
 #include "encode.h"
 #include "video/csputils.h"

 struct encode_lavc_context {
+    // --- Immutable after init
    struct mpv_global *global;
    struct encode_opts *options;
    struct mp_log *log;
-    struct mp_tags *metadata;
+    struct encode_priv *priv;
+    AVOutputFormat *oformat;
+    const char *filename;

    // All entry points must be guarded with the lock. Functions called by
    // the playback core lock this automatically, but ao_lavc.c and vo_lavc.c
    // must lock manually before accessing state.
    pthread_mutex_t lock;

-    float vo_fps;
-
-    // FFmpeg contexts.
-    AVFormatContext *avc;
-    AVStream *vst;
-    AVStream *ast;
-    AVCodecContext *vcc;
-    AVCodecContext *acc;
-
-    // these are processed from the options
-    AVRational timebase;
-    AVCodec *vc;
-    AVCodec *ac;
-    AVDictionary *foptions;
-    AVDictionary *aoptions;
-    AVDictionary *voptions;
-
-    // values created during encoding
-    int header_written; // -1 means currently writing
-
    // sync to audio mode
    double audio_pts_offset;
-    double last_video_in_pts;

    double last_audio_in_pts;
    int64_t samples_since_last_pts;
@@ -75,51 +58,59 @@ struct encode_lavc_context {
    // anti discontinuity mode
    double next_in_pts;
    double discontinuity_pts_offset;
-
-    long long abytes;
-    long long vbytes;
-    struct stream *twopass_bytebuffer_a;
-    struct stream *twopass_bytebuffer_v;
-    double t0;
-    unsigned int frames;
-    double audioseconds;
-
-    bool expect_video;
-    bool expect_audio;
-    bool video_first;
-    bool audio_first;
-
-    // has encoding failed?
-    bool failed;
-    bool finished;
 };

-// interface for vo/ao drivers
-int encode_lavc_alloc_stream(struct encode_lavc_context *ctx,
-                             enum AVMediaType mt, AVStream **stream_out,
-                             AVCodecContext **codec_out);
-void encode_lavc_write_stats(struct encode_lavc_context *ctx,
-                             AVCodecContext *stream);
-int encode_lavc_write_frame(struct encode_lavc_context *ctx, AVStream *stream,
-                            AVPacket *packet);
-int encode_lavc_supports_pixfmt(struct encode_lavc_context *ctx, enum AVPixelFormat format);
-int encode_lavc_open_codec(struct encode_lavc_context *ctx,
-                           AVCodecContext *codec);
-int encode_lavc_available(struct encode_lavc_context *ctx);
-int encode_lavc_timesyncfailed(struct encode_lavc_context *ctx);
-int encode_lavc_start(struct encode_lavc_context *ctx); // returns 1 on success
-int encode_lavc_oformat_flags(struct encode_lavc_context *ctx);
-double encode_lavc_getoffset(struct encode_lavc_context *ctx,
-                             AVCodecContext *codec);
-void encode_lavc_fail(struct encode_lavc_context *ctx, const char *format, ...); // report failure of encoding
+// --- interface for vo/ao drivers

-bool encode_lavc_set_csp(struct encode_lavc_context *ctx,
-                         AVCodecContext *codec, enum mp_csp csp);
-bool encode_lavc_set_csp_levels(struct encode_lavc_context *ctx,
-                                AVCodecContext *codec, enum mp_csp_levels lev);
-enum mp_csp encode_lavc_get_csp(struct encode_lavc_context *ctx,
-                                AVCodecContext *codec);
-enum mp_csp_levels encode_lavc_get_csp_levels(struct encode_lavc_context *ctx,
-                                              AVCodecContext *codec);
+// Static information after encoder init. This never changes (even if there are
+// dynamic runtime changes, they have to work over AVPacket side data).
+// For use in encoder_context, most fields are copied from encoder_context.encoder
+// by encoder_init_codec_and_muxer().
+struct encoder_stream_info {
+    AVRational timebase; // timebase used by the encoder (in frames/out packets)
+    AVCodecParameters *codecpar;
+};
+
+// The encoder parts for each stream (no muxing parts included).
+// This is private to each stream.
+struct encoder_context {
+    struct mpv_global *global;
+    struct encode_opts *options;
+    struct mp_log *log;
+    AVOutputFormat *oformat;
+
+    // (avoid using this)
+    struct encode_lavc_context *encode_lavc_ctx;
+
+    enum stream_type type;
+
+    // (different access restrictions before/after encoder init)
+    struct encoder_stream_info info;
+    AVCodecContext *encoder;
+    struct mux_stream *mux_stream;
+
+    struct stream *twopass_bytebuffer;
+};
+
+// Free with talloc_free(). (Keep in mind actual deinitialization requires
+// sending a flush packet.)
+// This can fail and return NULL.
+struct encoder_context *encoder_context_alloc(struct encode_lavc_context *ctx,
+                                              enum stream_type type,
+                                              struct mp_log *log);
+
+// After setting your codec parameters on p->encoder, you call this to "open"
+// the encoder. This also initializes p->mux_stream. Returns false on failure.
+// on_ready is called as soon as the muxer has been initialized. Then you are
+// allowed to write packets with encoder_encode().
+// Warning: the on_ready callback is called asynchronously, so you need to
+// make sure to properly synchronize everything.
+bool encoder_init_codec_and_muxer(struct encoder_context *p,
+                                  void (*on_ready)(void *ctx), void *ctx);
+
+// Encode the frame and write the packet. frame is ref'ed as need.
+bool encoder_encode(struct encoder_context *p, AVFrame *frame);
+
+double encoder_get_offset(struct encoder_context *p);

 #endif
@@ -18,6 +18,7 @@
 #include <stddef.h>
 #include <limits.h>
 #include <strings.h>
+#include <assert.h>
 #include <libavutil/dict.h>
 #include "tags.h"
 #include "misc/bstr.h"
@@ -44,6 +45,24 @@ void mp_tags_set_bstr(struct mp_tags *tags, bstr key, bstr value)
    tags->num_keys++;
 }

+void mp_tags_remove_str(struct mp_tags *tags, const char *key)
+{
+    mp_tags_remove_bstr(tags, bstr0(key));
+}
+
+void mp_tags_remove_bstr(struct mp_tags *tags, bstr key)
+{
+    for (int n = 0; n < tags->num_keys; n++) {
+        if (bstrcasecmp0(key, tags->keys[n]) == 0) {
+            talloc_free(tags->keys[n]);
+            talloc_free(tags->values[n]);
+            int num_keys = tags->num_keys; // copy so it's only decremented once
+            MP_TARRAY_REMOVE_AT(tags->keys, num_keys, n);
+            MP_TARRAY_REMOVE_AT(tags->values, tags->num_keys, n);
+        }
+    }
+}
+
 char *mp_tags_get_str(struct mp_tags *tags, const char *key)
 {
    return mp_tags_get_bstr(tags, bstr0(key));
@@ -64,19 +83,27 @@ void mp_tags_clear(struct mp_tags *tags)
    talloc_free_children(tags);
 }

+
+
 struct mp_tags *mp_tags_dup(void *tparent, struct mp_tags *tags)
 {
    struct mp_tags *new = talloc_zero(tparent, struct mp_tags);
-    MP_RESIZE_ARRAY(new, new->keys,   tags->num_keys);
-    MP_RESIZE_ARRAY(new, new->values, tags->num_keys);
-    new->num_keys = tags->num_keys;
-    for (int n = 0; n < tags->num_keys; n++) {
-        new->keys[n] = talloc_strdup(new, tags->keys[n]);
-        new->values[n] = talloc_strdup(new, tags->values[n]);
-    }
+    mp_tags_replace(new, tags);
    return new;
 }

+void mp_tags_replace(struct mp_tags *dst, struct mp_tags *src)
+{
+    mp_tags_clear(dst);
+    MP_RESIZE_ARRAY(dst, dst->keys,   src->num_keys);
+    MP_RESIZE_ARRAY(dst, dst->values, src->num_keys);
+    dst->num_keys = src->num_keys;
+    for (int n = 0; n < src->num_keys; n++) {
+        dst->keys[n] = talloc_strdup(dst, src->keys[n]);
+        dst->values[n] = talloc_strdup(dst, src->values[n]);
+    }
+}
+
 // Return a copy of the tags, but containing only keys in list. Also forces
 // the order and casing of the keys (for cosmetic reasons).
 // A trailing '*' matches the rest.
@@ -1,6 +1,8 @@
 #ifndef MP_TAGS_H
 #define MP_TAGS_H

+#include <stdint.h>
+
 #include "misc/bstr.h"

 struct mp_tags {
@@ -11,10 +13,13 @@ struct mp_tags {

 void mp_tags_set_str(struct mp_tags *tags, const char *key, const char *value);
 void mp_tags_set_bstr(struct mp_tags *tags, bstr key, bstr value);
+void mp_tags_remove_str(struct mp_tags *tags, const char *key);
+void mp_tags_remove_bstr(struct mp_tags *tags, bstr key);
 char *mp_tags_get_str(struct mp_tags *tags, const char *key);
 char *mp_tags_get_bstr(struct mp_tags *tags, bstr key);
 void mp_tags_clear(struct mp_tags *tags);
 struct mp_tags *mp_tags_dup(void *tparent, struct mp_tags *tags);
+void mp_tags_replace(struct mp_tags *dst, struct mp_tags *src);
 struct mp_tags *mp_tags_filtered(void *tparent, struct mp_tags *tags, char **list);
 void mp_tags_merge(struct mp_tags *tags, struct mp_tags *src);
 struct AVDictionary;
@@ -24,3 +24,4 @@

 const char mpv_version[]  = "mpv " VERSION;
 const char mpv_builddate[] = BUILDDATE;
+const char mpv_copyright[] = MPVCOPYRIGHT;
@@ -75,7 +75,7 @@ static enum cue_command read_cmd(struct bstr *data, struct bstr *out_params)
            if (rest.len && !strchr(WHITESPACE, rest.start[0]))
                continue;
            if (out_params)
-                *out_params = rest;
+                *out_params = bstr_lstrip(rest);
            return cue_command_strings[n].command;
        }
    }
@@ -105,6 +105,14 @@ static char *read_quoted(void *talloc_ctx, struct bstr *data)
    return bstrto0(talloc_ctx, res);
 }

+static struct bstr strip_quotes(struct bstr data)
+{
+    bstr s = data;
+    if (bstr_eatstart0(&s, "\"") && bstr_eatend0(&s, "\""))
+        return s;
+    return data;
+}
+
 // Read a 2 digit unsigned decimal integer.
 // Return -1 on failure.
 static int read_int_2(struct bstr *data)
@@ -193,7 +201,7 @@ struct cue_file *mp_parse_cue(struct bstr data)
                [CUE_PERFORMER] = "performer",
            };
            struct mp_tags *tags = cur_track ? cur_track->tags : f->tags;
-            mp_tags_set_bstr(tags, bstr0(metanames[cmd]), param);
+            mp_tags_set_bstr(tags, bstr0(metanames[cmd]), strip_quotes(param));
            break;
        }
        case CUE_INDEX: {
@@ -53,6 +53,9 @@ struct demux_ctrl_reader_state {
    double ts_end; // approx. timestamp of end of buffered range
    int64_t total_bytes;
    int64_t fw_bytes;
+    double seeking; // current low level seek target, or NOPTS
+    int low_level_seeks; // number of started low level seeks
+    double ts_last; // approx. timestamp of demuxer position
    // Positions that can be seeked to without incurring the latency of a low
    // level seek.
    int num_seek_ranges;
@@ -232,8 +235,7 @@ typedef struct demuxer {

    // Since the demuxer can run in its own thread, and the stream is not
    // thread-safe, only the demuxer is allowed to access the stream directly.
-    // You can freely use demux_stream_control() to send STREAM_CTRLs, or use
-    // demux_pause() to get exclusive access to the stream.
+    // You can freely use demux_stream_control() to send STREAM_CTRLs.
    // Also note that the stream can get replaced if fully_read is set.
    struct stream *stream;
 } demuxer_t;
@@ -253,6 +255,8 @@ 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);
 bool demux_has_packet(struct sh_stream *sh);
+void demux_set_stream_wakeup_cb(struct sh_stream *sh,
+                                void (*cb)(void *ctx), void *ctx);
 struct demux_packet *demux_read_any_packet(struct demuxer *demuxer);

 struct sh_stream *demux_get_stream(struct demuxer *demuxer, int index);
@@ -282,6 +286,8 @@ void demux_set_ts_offset(struct demuxer *demuxer, double offset);

 int demux_control(struct demuxer *demuxer, int cmd, void *arg);

+void demux_block_reading(struct demuxer *demuxer, bool block);
+
 void demuxer_select_track(struct demuxer *demuxer, struct sh_stream *stream,
                          double ref_pts, bool selected);
 void demux_set_stream_autoselect(struct demuxer *demuxer, bool autoselect);
@@ -297,7 +303,7 @@ void demux_set_stream_tags(struct demuxer *demuxer, struct sh_stream *sh,

 int demux_stream_control(demuxer_t *demuxer, int ctrl, void *arg);

-void demux_changed(demuxer_t *demuxer, int events);
+void demux_metadata_changed(demuxer_t *demuxer);
 void demux_update(demuxer_t *demuxer);

 void demux_disable_cache(demuxer_t *demuxer);
@@ -312,4 +318,7 @@ bool demux_matroska_uid_cmp(struct matroska_segment_uid *a,

 const char *stream_type_name(enum stream_type type);

+void mp_packet_tags_unref(struct mp_packet_tags *tags);
+void mp_packet_tags_setref(struct mp_packet_tags **dst, struct mp_packet_tags *src);
+
 #endif /* MPLAYER_DEMUXER_H */
@@ -138,7 +138,7 @@ static struct tl_parts *parse_edl(bstr str)
            bstr type = param_vals[0]; // value, because no "="
            if (bstr_equals0(type, "mp4_dash")) {
                tl->dash = true;
-                if (bstr_equals0(param_names[1], "init"))
+                if (nparam > 1 && bstr_equals0(param_names[1], "init"))
                    tl->init_fragment_url = bstrto0(tl, param_vals[1]);
            }
            continue;
@@ -50,6 +50,12 @@
 #include "options/m_option.h"
 #include "options/path.h"

+#ifndef AV_DISPOSITION_TIMED_THUMBNAILS
+#define AV_DISPOSITION_TIMED_THUMBNAILS 0
+#endif
+#ifndef AV_DISPOSITION_STILL_IMAGE
+#define AV_DISPOSITION_STILL_IMAGE 0
+#endif

 #define INITIAL_PROBE_SIZE STREAM_BUFFER_SIZE
 #define PROBE_BUF_SIZE FFMIN(STREAM_MAX_BUFFER_SIZE, 2 * 1024 * 1024)
@@ -70,7 +76,6 @@ struct demux_lavf_opts {
    char *format;
    char **avopts;
    int hacks;
-    int genptsmode;
    char *sub_cp;
    int rtsp_transport;
 };
@@ -79,7 +84,7 @@ const struct m_sub_options demux_lavf_conf = {
    .opts = (const m_option_t[]) {
        OPT_INTRANGE("demuxer-lavf-probesize", probesize, 0, 32, INT_MAX),
        OPT_CHOICE("demuxer-lavf-probe-info", probeinfo, 0,
-                   ({"no", 0}, {"yes", 1}, {"auto", -1})),
+                   ({"no", 0}, {"yes", 1}, {"auto", -1}, {"nostreams", -2})),
        OPT_STRING("demuxer-lavf-format", format, 0),
        OPT_FLOATRANGE("demuxer-lavf-analyzeduration", analyzeduration, 0,
                       0, 3600),
@@ -89,8 +94,6 @@ const struct m_sub_options demux_lavf_conf = {
        OPT_INTRANGE("demuxer-lavf-probescore", probescore, 0,
                     1, AVPROBE_SCORE_MAX),
        OPT_FLAG("demuxer-lavf-hacks", hacks, 0),
-        OPT_CHOICE("demuxer-lavf-genpts-mode", genptsmode, 0,
-                   ({"lavf", 1}, {"no", 0})),
        OPT_KEYVALUELIST("demuxer-lavf-o", avopts, 0),
        OPT_STRING("sub-codepage", sub_cp, 0),
        OPT_CHOICE("rtsp-transport", rtsp_transport, 0,
@@ -133,6 +136,8 @@ struct format_hack {
    bool clear_filepos : 1;
    bool ignore_start : 1;
    bool fix_editlists : 1;
+    bool is_network : 1;
+    bool no_seek : 1;
 };

 #define BLACKLIST(fmt) {fmt, .ignore = true}
@@ -149,12 +154,16 @@ static const struct format_hack format_hacks[] = {
    {"mp3", NULL,         24, .max_probe = true},

    {"hls", .no_stream = true, .clear_filepos = true},
+    {"dash", .no_stream = true, .clear_filepos = true},
+    {"sdp", .clear_filepos = true, .is_network = true, .no_seek = true},
    {"mpeg", .use_stream_ids = true},
    {"mpegts", .use_stream_ids = true},

    {"mp4", .skipinfo = true, .fix_editlists = true},
    {"matroska", .skipinfo = true},

+    {"v4l2", .no_seek = true},
+
    // In theory, such streams might contain timestamps, but virtually none do.
    {"h264", .if_flags = AVFMT_NOTIMESTAMPS },
    {"hevc", .if_flags = AVFMT_NOTIMESTAMPS },
@@ -176,8 +185,6 @@ static const struct format_hack format_hacks[] = {
    BLACKLIST("bin"),
    // Useless, does not work with custom streams.
    BLACKLIST("image2"),
-    // Probably a security risk.
-    BLACKLIST("ffm"),
    // Image demuxers ("<name>_pipe" is detected explicitly)
    {"image2pipe", .image_format = true},
    {0}
@@ -253,9 +260,10 @@ static int64_t mp_seek(void *opaque, int64_t pos, int whence)
             whence == SEEK_CUR ? "cur" :
             whence == SEEK_SET ? "set" : "size");
    if (whence == SEEK_END || whence == AVSEEK_SIZE) {
-        int64_t end = stream_get_size(stream) + priv->init_fragment.len;
+        int64_t end = stream_get_size(stream);
        if (end < 0)
            return -1;
+        end += priv->init_fragment.len;
        if (whence == AVSEEK_SIZE)
            return end;
        pos += end;
@@ -310,8 +318,9 @@ static int64_t mp_read_seek(void *opaque, int stream_idx, int64_t ts, int flags)
 static void list_formats(struct demuxer *demuxer)
 {
    MP_INFO(demuxer, "Available lavf input formats:\n");
-    AVInputFormat *fmt = NULL;
-    while ((fmt = av_iformat_next(fmt)))
+    const AVInputFormat *fmt;
+    void *iter = NULL;
+    while ((fmt = av_demuxer_iterate(&iter)))
        MP_INFO(demuxer, "%15s : %s\n", fmt->name, fmt->long_name);
 }

@@ -421,6 +430,10 @@ static int lavf_check_file(demuxer_t *demuxer, enum demux_check check)
            int nsize = av_clip(avpd.buf_size * 2, INITIAL_PROBE_SIZE,
                                PROBE_BUF_SIZE);
            bstr buf = stream_peek(s, nsize);
+            if (demuxer->params && demuxer->params->init_fragment.len) {
+                buf = demuxer->params->init_fragment;
+                buf.len = MPMIN(buf.len, nsize);
+            }
            if (buf.len <= avpd.buf_size)
                final_probe = true;
            memcpy(avpd.buf, buf.start, buf.len);
@@ -434,10 +447,8 @@ static int lavf_check_file(demuxer_t *demuxer, enum demux_check check)
                       priv->avif->name, score, avpd.buf_size,
                       forced_format ? " (forced)" : "");

-            for (int n = 0; format_hacks[n].ff_name; n++) {
+            for (int n = 0; lavfdopts->hacks && format_hacks[n].ff_name; n++) {
                const struct format_hack *entry = &format_hacks[n];
-                if (!lavfdopts->hacks)
-                    continue;
                if (!matches_avinputformat_name(priv, entry->ff_name))
                    continue;
                if (entry->mime_type && strcasecmp(entry->mime_type, mime_type) != 0)
@@ -614,7 +625,9 @@ static void handle_new_stream(demuxer_t *demuxer, int i)
    case AVMEDIA_TYPE_VIDEO: {
        sh = demux_alloc_sh_stream(STREAM_VIDEO);

-        if (st->disposition & AV_DISPOSITION_ATTACHED_PIC) {
+        if ((st->disposition & AV_DISPOSITION_ATTACHED_PIC) &&
+            !(st->disposition & AV_DISPOSITION_TIMED_THUMBNAILS))
+        {
            sh->attached_picture =
                new_demux_packet_from_avpacket(&st->attached_pic);
            if (sh->attached_picture) {
@@ -706,6 +719,10 @@ static void handle_new_stream(demuxer_t *demuxer, int i)
            sh->default_track = true;
        if (st->disposition & AV_DISPOSITION_FORCED)
            sh->forced_track = true;
+        if (st->disposition & AV_DISPOSITION_DEPENDENT)
+            sh->dependent_track = true;
+        if (st->disposition & AV_DISPOSITION_STILL_IMAGE)
+            sh->still_image = true;
        if (priv->format_hack.use_stream_ids)
            sh->demuxer_id = st->id;
        AVDictionaryEntry *title = av_dict_get(st->metadata, "title", NULL, 0);
@@ -716,7 +733,7 @@ static void handle_new_stream(demuxer_t *demuxer, int i)
        if (!sh->title && st->disposition & AV_DISPOSITION_HEARING_IMPAIRED)
            sh->title = talloc_asprintf(sh, "hearing impaired");
        AVDictionaryEntry *lang = av_dict_get(st->metadata, "language", NULL, 0);
-        if (lang && lang->value)
+        if (lang && lang->value && strcmp(lang->value, "und") != 0)
            sh->lang = talloc_strdup(sh, lang->value);
        sh->hls_bitrate = dict_get_decimal(st->metadata, "variant_bitrate", 0);
        if (!sh->title && sh->hls_bitrate > 0)
@@ -737,13 +754,13 @@ static void add_new_streams(demuxer_t *demuxer)
        handle_new_stream(demuxer, priv->num_streams);
 }

-static void update_metadata(demuxer_t *demuxer, AVPacket *pkt)
+static void update_metadata(demuxer_t *demuxer)
 {
    lavf_priv_t *priv = demuxer->priv;
    if (priv->avfc->event_flags & AVFMT_EVENT_FLAG_METADATA_UPDATED) {
        mp_tags_copy_from_av_dictionary(demuxer->metadata, priv->avfc->metadata);
        priv->avfc->event_flags = 0;
-        demux_changed(demuxer, DEMUX_EVENT_METADATA);
+        demux_metadata_changed(demuxer);
    }

    for (int n = 0; n < priv->num_streams; n++) {
@@ -800,8 +817,6 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
    if (!avfc)
        return -1;

-    if (lavfdopts->genptsmode)
-        avfc->flags |= AVFMT_FLAG_GENPTS;
    if (index_mode != 1)
        avfc->flags |= AVFMT_FLAG_IGNIDX;

@@ -884,8 +899,11 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)

    priv->avfc = avfc;

-    bool probeinfo = lavfdopts->probeinfo < 0 ?
-                            !priv->format_hack.skipinfo : lavfdopts->probeinfo;
+    bool probeinfo = lavfdopts->probeinfo != 0;
+    switch (lavfdopts->probeinfo) {
+    case -2: probeinfo = priv->avfc->nb_streams == 0; break;
+    case -1: probeinfo = !priv->format_hack.skipinfo; break;
+    }
    if (demuxer->params && demuxer->params->skip_lavf_probing)
        probeinfo = false;
    if (probeinfo) {
@@ -909,7 +927,6 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
    add_new_streams(demuxer);

    mp_tags_copy_from_av_dictionary(demuxer->metadata, avfc->metadata);
-    update_metadata(demuxer, NULL);

    demuxer->ts_resets_possible =
        priv->avif_flags & (AVFMT_TS_DISCONT | AVFMT_NOTIMESTAMPS);
@@ -919,6 +936,14 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)

    demuxer->fully_read = priv->format_hack.fully_read;

+#ifdef AVFMTCTX_UNSEEKABLE
+    if (avfc->ctx_flags & AVFMTCTX_UNSEEKABLE)
+        demuxer->seekable = false;
+#endif
+
+    demuxer->is_network |= priv->format_hack.is_network;
+    demuxer->seekable &= !priv->format_hack.no_seek;
+
    if (priv->avfc->duration > 0) {
        demuxer->duration = (double)priv->avfc->duration / AV_TIME_BASE;
    } else {
@@ -939,6 +964,19 @@ static int demux_open_lavf(demuxer_t *demuxer, enum demux_check check)
            demuxer->duration = duration;
    }

+    // In some cases, libavformat will export bogus bullshit timestamps anyway,
+    // such as with mjpeg.
+    if (priv->avif_flags & AVFMT_NOTIMESTAMPS) {
+        MP_WARN(demuxer,
+                "This format is marked by FFmpeg as having no timestamps!\n"
+                "FFmpeg will likely make up its own broken timestamps. For\n"
+                "video streams you can correct this with:\n"
+                "    --no-correct-pts --fps=VALUE\n"
+                "with VALUE being the real framerate of the stream. You can\n"
+                "expect seeking and buffering estimation to be generally\n"
+                "broken as well.\n");
+    }
+
    return 0;
 }

@@ -959,7 +997,7 @@ static int demux_lavf_fill_buffer(demuxer_t *demux)
    }

    add_new_streams(demux);
-    update_metadata(demux, pkt);
+    update_metadata(demux);

    assert(pkt->stream_index >= 0 && pkt->stream_index < priv->num_streams);
    struct sh_stream *stream = priv->streams[pkt->stream_index];
@@ -1108,31 +1146,16 @@ 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);
+        update_metadata(demuxer);
        // Enforce metadata update even if no explicit METADATA_UPDATED since we switched program.
-        demux_changed(demuxer, DEMUX_EVENT_METADATA);
+        demux_metadata_changed(demuxer);

        return CONTROL_OK;
    }
    case DEMUXER_CTRL_RESYNC:
-        /* NOTE:
-         *
-         * We actually want to call ff_read_frame_flush() here, but it is
-         * internal.
-         *
-         * This function call seems to do the same for now.
-         *
-         * Once ff_read_frame_flush() is exported in some way, change this to
-         * call the new API instead of relying on av_seek_frame() to do this
-         * for us.
-         */
-        // avio_flush() is designed for write-only streams, and does the wrong
-        // thing when reading. Flush it manually instead.
        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(priv->stream);
-        av_seek_frame(priv->avfc, 0, stream_tell(priv->stream),
-                      AVSEEK_FLAG_BYTE);
+        avio_flush(priv->avfc->pb);
+        avformat_flush(priv->avfc);
        return CONTROL_OK;
    case DEMUXER_CTRL_REPLACE_STREAM:
        if (priv->own_stream)
@@ -193,7 +193,6 @@ typedef struct mkv_demuxer {
    struct header_elem {
        int32_t id;
        int64_t pos;
-        bool needed;
        bool parsed;
    } *headers;
    int num_headers;
@@ -207,7 +206,10 @@ typedef struct mkv_demuxer {

    bool eof_warning, keyframe_warning;

-    struct block_info tmp_block;
+    // Small queue of read but not yet returned packets. This is mostly
+    // temporary data, and not normally larger than 0 or 1 elements.
+    struct block_info *blocks;
+    int num_blocks;
 } mkv_demuxer_t;

 #define OPT_BASE_STRUCT struct demux_mkv_opts
@@ -251,6 +253,7 @@ const struct m_sub_options demux_mkv_conf = {

 static void probe_last_timestamp(struct demuxer *demuxer, int64_t start_pos);
 static void probe_first_timestamp(struct demuxer *demuxer);
+static int read_next_block_into_queue(demuxer_t *demuxer);
 static void free_block(struct block_info *block);

 #define AAC_SYNC_EXTENSION_TYPE 0x02b7
@@ -353,7 +356,7 @@ static bstr demux_mkv_decode(struct mp_log *log, mkv_track_t *track,
                    dest = NULL;
                    goto error;
                }
-                mp_dbg(log, "lzo decompression buffer too small.\n");
+                mp_trace(log, "lzo decompression buffer too small.\n");
                if (dstlen >= maxlen / 2) {
                    talloc_free(dest);
                    dest = NULL;
@@ -700,7 +703,10 @@ static void parse_trackentry(struct demuxer *demuxer,
    }
    track->uid = entry->track_uid;

-    track->name = talloc_strdup(track, entry->name);
+    if (entry->name) {
+        track->name = talloc_strdup(track, entry->name);
+        MP_VERBOSE(demuxer, "|  + Name: %s\n", track->name);
+    }

    track->type = entry->track_type;
    MP_VERBOSE(demuxer, "|  + Track type: ");
@@ -888,11 +894,11 @@ static int demux_mkv_read_cues(demuxer_t *demuxer)
            cue_index_add(demuxer, trackpos->cue_track, pos,
                          time, trackpos->cue_duration);
            mkv_d->index_has_durations |= trackpos->n_cue_duration > 0;
-            MP_DBG(demuxer, "|+ found cue point for track %"PRIu64": "
-                   "timecode %"PRIu64", filepos: %"PRIu64""
-                   "offset %"PRIu64", duration %"PRIu64"\n",
-                   trackpos->cue_track, time, pos,
-                   trackpos->cue_relative_position, trackpos->cue_duration);
+            MP_TRACE(demuxer, "|+ found cue point for track %"PRIu64": "
+                     "timecode %"PRIu64", filepos: %"PRIu64""
+                     "offset %"PRIu64", duration %"PRIu64"\n",
+                     trackpos->cue_track, time, pos,
+                     trackpos->cue_relative_position, trackpos->cue_duration);
        }
    }

@@ -1072,7 +1078,7 @@ static int demux_mkv_read_tags(demuxer_t *demuxer)
    if (ebml_read_element(s, &parse_ctx, &tags, &ebml_tags_desc) < 0)
        return -1;

-    mkv_d->tags = talloc_memdup(mkv_d, &tags, sizeof(tags));
+    mkv_d->tags = talloc_dup(mkv_d, &tags);
    talloc_steal(mkv_d->tags, parse_ctx.talloc_ctx);
    return 0;
 }
@@ -1226,8 +1232,8 @@ static int demux_mkv_read_seekhead(demuxer_t *demuxer)
            continue;
        }
        uint64_t pos = seek->seek_position + mkv_d->segment_start;
-        MP_DBG(demuxer, "Element 0x%"PRIx32" at %"PRIu64".\n",
-               seek->seek_id, pos);
+        MP_TRACE(demuxer, "Element 0x%"PRIx32" at %"PRIu64".\n",
+                 seek->seek_id, pos);
        get_header_element(demuxer, seek->seek_id, pos);
    }
 out:
@@ -1377,6 +1383,7 @@ static const char *const mkv_video_tags[][2] = {
    {"V_PRORES",            "prores"},
    {"V_MPEGH/ISO/HEVC",    "hevc"},
    {"V_SNOW",              "snow"},
+    {"V_AV1",               "av1"},
    {0}
 };

@@ -1581,6 +1588,7 @@ static const char *const mkv_audio_tags[][2] = {
    { "A_FLAC",                 "flac" },
    { "A_ALAC",                 "alac" },
    { "A_TTA1",                 "tta" },
+    { "A_MLP",                  "mlp" },
    { NULL },
 };

@@ -1753,7 +1761,7 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)

    const char *codec = sh_a->codec;
    if (!strcmp(codec, "mp2") || !strcmp(codec, "mp3") ||
-        !strcmp(codec, "truehd"))
+        !strcmp(codec, "truehd") || !strcmp(codec, "eac3"))
    {
        track->parse = true;
    } else if (!strcmp(codec, "flac")) {
@@ -1875,6 +1883,47 @@ static int demux_mkv_open_sub(demuxer_t *demuxer, mkv_track_t *track)
    return 0;
 }

+static void probe_x264_garbage(demuxer_t *demuxer)
+{
+    mkv_demuxer_t *mkv_d = demuxer->priv;
+
+    for (int n = 0; n < mkv_d->num_tracks; n++) {
+        mkv_track_t *track = mkv_d->tracks[n];
+        struct sh_stream *sh = track->stream;
+
+        if (!sh || sh->type != STREAM_VIDEO)
+            continue;
+
+        if (sh->codec->codec && strcmp(sh->codec->codec, "h264") != 0)
+            continue;
+
+        struct block_info *block = NULL;
+
+        // Find first block for this track.
+        // Restrict reading number of total packets. (Arbitrary to avoid bloat.)
+        for (int i = 0; i < 100; i++) {
+            if (i >= mkv_d->num_blocks && read_next_block_into_queue(demuxer) < 1)
+                break;
+            if (mkv_d->blocks[i].track == track) {
+                block = &mkv_d->blocks[i];
+                break;
+            }
+        }
+
+        if (!block || block->num_laces < 1)
+            continue;
+
+        bstr sblock = {block->laces[0]->data, block->laces[0]->size};
+        bstr nblock = demux_mkv_decode(demuxer->log, track, sblock, 1);
+
+        sh->codec->first_packet = new_demux_packet_from(nblock.start, nblock.len);
+        talloc_steal(mkv_d, sh->codec->first_packet);
+
+        if (nblock.start != sblock.start)
+            talloc_free(nblock.start);
+    }
+}
+
 static int read_ebml_header(demuxer_t *demuxer)
 {
    stream_t *s = demuxer->stream;
@@ -1890,7 +1939,7 @@ static int read_ebml_header(demuxer_t *demuxer)
                   "Assuming \"matroska\".\n");
    } else if (strcmp(ebml_master.doc_type, "matroska") != 0
        && strcmp(ebml_master.doc_type, "webm") != 0) {
-        MP_DBG(demuxer, "no head found\n");
+        MP_TRACE(demuxer, "no head found\n");
        talloc_free(parse_ctx.talloc_ctx);
        return 0;
    }
@@ -2030,7 +2079,6 @@ static int demux_mkv_open(demuxer_t *demuxer, enum demux_check check)
            }
            continue;
        }
-        elem->needed = true;
        only_cue = only_cue < 0 && elem->id == MATROSKA_ID_CUES;
    }

@@ -2045,7 +2093,7 @@ static int demux_mkv_open(demuxer_t *demuxer, enum demux_check check)
            struct header_elem *lowest = NULL;
            for (int n = 0; n < mkv_d->num_headers; n++) {
                struct header_elem *elem = &mkv_d->headers[n];
-                if (!elem->needed)
+                if (elem->parsed)
                    continue;
                if (!lowest || elem->pos < lowest->pos)
                    lowest = elem;
@@ -2054,7 +2102,6 @@ static int demux_mkv_open(demuxer_t *demuxer, enum demux_check check)
            if (!lowest)
                break;

-            lowest->needed = false;
            if (read_deferred_element(demuxer, lowest) < 0)
                return -1;
        }
@@ -2074,6 +2121,7 @@ static int demux_mkv_open(demuxer_t *demuxer, enum demux_check check)
    probe_first_timestamp(demuxer);
    if (mkv_d->opts->probe_duration)
        probe_last_timestamp(demuxer, start_pos);
+    probe_x264_garbage(demuxer);

    return 0;
 }
@@ -2285,7 +2333,9 @@ static void mkv_seek_reset(demuxer_t *demuxer)
        avcodec_free_context(&track->av_parser_codec);
    }

-    free_block(&mkv_d->tmp_block);
+    for (int n = 0; n < mkv_d->num_blocks; n++)
+        free_block(&mkv_d->blocks[n]);
+    mkv_d->num_blocks = 0;

    mkv_d->skip_to_timecode = INT64_MIN;
 }
@@ -2710,8 +2760,7 @@ static int read_block_group(demuxer_t *demuxer, int64_t end,
                                  &ebml_block_additions_desc) < 0)
                return -1;
            if (additions.n_block_more > 0) {
-                block->additions =
-                    talloc_memdup(NULL, &additions, sizeof(additions));
+                block->additions = talloc_dup(NULL, &additions);
                talloc_steal(block->additions, parse_ctx.talloc_ctx);
                parse_ctx.talloc_ctx = NULL;
            }
@@ -2736,16 +2785,11 @@ error:
    return -1;
 }

-static int read_next_block(demuxer_t *demuxer, struct block_info *block)
+static int read_next_block_into_queue(demuxer_t *demuxer)
 {
    mkv_demuxer_t *mkv_d = (mkv_demuxer_t *) demuxer->priv;
    stream_t *s = demuxer->stream;
-
-    if (mkv_d->tmp_block.num_laces) {
-        *block = mkv_d->tmp_block;
-        mkv_d->tmp_block = (struct block_info){0};
-        return 1;
-    }
+    struct block_info block = {0};

    while (1) {
        while (stream_tell(s) < mkv_d->cluster_end) {
@@ -2764,21 +2808,21 @@ static int read_next_block(demuxer_t *demuxer, struct block_info *block)
                end += stream_tell(s);
                if (end > mkv_d->cluster_end)
                    goto find_next_cluster;
-                int res = read_block_group(demuxer, end, block);
+                int res = read_block_group(demuxer, end, &block);
                if (res < 0)
                    goto find_next_cluster;
                if (res > 0)
-                    return 1;
+                    goto add_block;
                break;
            }

            case MATROSKA_ID_SIMPLEBLOCK: {
-                *block = (struct block_info){ .simple = true };
-                int res = read_block(demuxer, mkv_d->cluster_end, block);
+                block = (struct block_info){ .simple = true };
+                int res = read_block(demuxer, mkv_d->cluster_end, &block);
                if (res < 0)
                    goto find_next_cluster;
                if (res > 0)
-                    return 1;
+                    goto add_block;
                break;
            }

@@ -2828,6 +2872,29 @@ static int read_next_block(demuxer_t *demuxer, struct block_info *block)
        if (mkv_d->cluster_end != EBML_UINT_INVALID)
            mkv_d->cluster_end += stream_tell(s);
    }
+    assert(0); // unreachable
+
+add_block:
+    index_block(demuxer, &block);
+    MP_TARRAY_APPEND(mkv_d, mkv_d->blocks, mkv_d->num_blocks, block);
+    return 1;
+}
+
+static int read_next_block(demuxer_t *demuxer, struct block_info *block)
+{
+    mkv_demuxer_t *mkv_d = demuxer->priv;
+
+    if (!mkv_d->num_blocks) {
+        int res = read_next_block_into_queue(demuxer);
+        if (res < 1)
+            return res;
+
+        assert(mkv_d->num_blocks);
+    }
+
+    *block = mkv_d->blocks[0];
+    MP_TARRAY_REMOVE_AT(mkv_d->blocks, mkv_d->num_blocks, 0);
+    return 1;
 }

 static int demux_mkv_fill_buffer(demuxer_t *demuxer)
@@ -2839,7 +2906,6 @@ static int demux_mkv_fill_buffer(demuxer_t *demuxer)
        if (res < 0)
            return 0;
        if (res > 0) {
-            index_block(demuxer, &block);
            res = handle_block(demuxer, &block);
            free_block(&block);
            if (res > 0)
@@ -2887,7 +2953,6 @@ static int create_index_until(struct demuxer *demuxer, int64_t timecode)
            if (res < 0)
                break;
            if (res > 0) {
-                index_block(demuxer, &block);
                free_block(&block);
            }
            index = get_highest_index_entry(demuxer);
@@ -3113,7 +3178,7 @@ static void probe_last_timestamp(struct demuxer *demuxer, int64_t start_pos)
        }
    }

-    free_block(&mkv_d->tmp_block);
+    mkv_seek_reset(demuxer);

    int64_t last_ts[STREAM_TYPE_COUNT] = {0};
    while (1) {
@@ -3151,11 +3216,7 @@ static void probe_first_timestamp(struct demuxer *demuxer)
    if (!mkv_d->opts->probe_start_time)
        return;

-    struct block_info block;
-    if (read_next_block(demuxer, &block) > 0) {
-        index_block(demuxer, &block);
-        mkv_d->tmp_block = block;
-    }
+    read_next_block_into_queue(demuxer);

    demuxer->start_time = mkv_d->cluster_tc / 1e9;

@@ -24,6 +24,7 @@ static int try_open_file(struct demuxer *demux, enum demux_check check)
    if (!bstr_startswith0(bstr0(demux->filename), "null://") &&
        check != DEMUX_CHECK_REQUEST)
        return -1;
+    demux->seekable = true;
    return 0;
 }

@@ -85,7 +85,7 @@ static int demux_open_tv(demuxer_t *demuxer, enum demux_check check)
    /* get IMAGE FORMAT */
    int fourcc;
    funcs->control(tvh->priv, TVI_CONTROL_VID_GET_FORMAT, &fourcc);
-    if (fourcc == MP_FOURCC_MJPEG) {
+    if (fourcc == MP_FOURCC_MJPEG || fourcc == MP_FOURCC_JPEG) {
        sh_v->codec->codec = "mjpeg";
    } else {
        sh_v->codec->codec = "rawvideo";
@@ -327,7 +327,7 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
 {
    assert(type->type == EBML_TYPE_SUBELEMENTS);
    assert(level < 8);
-    MP_DBG(ctx, "%.*sParsing element %s\n", level, "       ", type->name);
+    MP_TRACE(ctx, "%.*sParsing element %s\n", level, "       ", type->name);

    char *s = target;
    uint8_t *end = data + size;
@@ -340,7 +340,7 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
        if (len > end - p)
            goto past_end_error;
        if (len < 0) {
-            MP_DBG(ctx, "Error parsing subelement id\n");
+            MP_ERR(ctx, "Error parsing subelement id\n");
            goto other_error;
        }
        p += len;
@@ -348,7 +348,7 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
        if (len > end - p)
            goto past_end_error;
        if (len < 0) {
-            MP_DBG(ctx, "Error parsing subelement length\n");
+            MP_ERR(ctx, "Error parsing subelement length\n");
            goto other_error;
        }
        p += len;
@@ -368,7 +368,7 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
        if (length > end - p) {
            if (field_idx >= 0 && type->fields[field_idx].desc->type
                != EBML_TYPE_SUBELEMENTS) {
-                MP_DBG(ctx, "Subelement content goes "
+                MP_ERR(ctx, "Subelement content goes "
                       "past end of containing element\n");
                goto other_error;
            }
@@ -381,7 +381,7 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
        continue;

    past_end_error:
-        MP_DBG(ctx, "Subelement headers go past end of containing element\n");
+        MP_ERR(ctx, "Subelement headers go past end of containing element\n");
    other_error:
        ctx->has_errors = true;
        end = startp;
@@ -437,20 +437,20 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
        int len;
        uint32_t id = ebml_parse_id(data, end - data, &len);
        if (len < 0 || len > end - data) {
-            MP_DBG(ctx, "Error parsing subelement\n");
+            MP_ERR(ctx, "Error parsing subelement\n");
            break;
        }
        data += len;
        uint64_t length = ebml_parse_length(data, end - data, &len);
        if (len < 0 || len > end - data) {
-            MP_DBG(ctx, "Error parsing subelement length\n");
+            MP_ERR(ctx, "Error parsing subelement length\n");
            break;
        }
        data += len;
        if (length > end - data) {
            // Try to parse what is possible from inside this partial element
            length = end - data;
-            MP_DBG(ctx, "Next subelement content goes "
+            MP_ERR(ctx, "Next subelement content goes "
                   "past end of containing element, will be truncated\n");
        }
        int field_idx = -1;
@@ -460,16 +460,17 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
                break;
            }
        if (field_idx < 0) {
-            if (id == 0xec)
-                MP_DBG(ctx, "%.*sIgnoring Void element "
-                       "size: %"PRIu64"\n", level+1, "        ", length);
-            else if (id == 0xbf)
-                MP_DBG(ctx, "%.*sIgnoring CRC-32 "
-                       "element size: %"PRIu64"\n", level+1, "        ",
-                       length);
-            else
+            if (id == 0xec) {
+                MP_TRACE(ctx, "%.*sIgnoring Void element "
+                         "size: %"PRIu64"\n", level+1, "        ", length);
+            } else if (id == 0xbf) {
+                MP_TRACE(ctx, "%.*sIgnoring CRC-32 "
+                         "element size: %"PRIu64"\n", level+1, "        ",
+                         length);
+            } else {
                MP_DBG(ctx, "Ignoring unrecognized "
                       "subelement. ID: %x size: %"PRIu64"\n", id, length);
+            }
            data += length;
            continue;
        }
@@ -485,20 +486,20 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
            continue;
        }
        if (*countptr > 0 && !multiple) {
-            MP_DBG(ctx, "Another subelement of type "
-                   "%x %s (size: %"PRIu64"). Only one allowed. Ignoring.\n",
-                   id, ed->name, length);
+            MP_WARN(ctx, "Another subelement of type "
+                    "%x %s (size: %"PRIu64"). Only one allowed. Ignoring.\n",
+                    id, ed->name, length);
            ctx->has_errors = true;
            data += length;
            continue;
        }
-        MP_DBG(ctx, "%.*sParsing %x %s size: %"PRIu64
-               " value: ", level+1, "        ", id, ed->name, length);
+        MP_TRACE(ctx, "%.*sParsing %x %s size: %"PRIu64
+                 " value: ", level+1, "        ", id, ed->name, length);

        char *fieldptr = s + fd->offset;
        switch (ed->type) {
        case EBML_TYPE_SUBELEMENTS:
-            MP_DBG(ctx, "subelements\n");
+            MP_TRACE(ctx, "subelements\n");
            char *subelptr;
            if (multiple) {
                char *array_start = (char *) *(generic_struct **) fieldptr;
@@ -517,29 +518,29 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
                subelptr = (fieldtype *) fieldptr
            GETPTR(uintptr, uint64_t);
            if (length < 1 || length > 8) {
-                MP_DBG(ctx, "uint invalid length %"PRIu64"\n", length);
+                MP_ERR(ctx, "uint invalid length %"PRIu64"\n", length);
                goto error;
            }
            *uintptr = ebml_parse_uint(data, length);
-            MP_DBG(ctx, "uint %"PRIu64"\n", *uintptr);
+            MP_TRACE(ctx, "uint %"PRIu64"\n", *uintptr);
            break;

        case EBML_TYPE_SINT:;
            int64_t *sintptr;
            GETPTR(sintptr, int64_t);
            if (length > 8) {
-                MP_DBG(ctx, "sint invalid length %"PRIu64"\n", length);
+                MP_ERR(ctx, "sint invalid length %"PRIu64"\n", length);
                goto error;
            }
            *sintptr = ebml_parse_sint(data, length);
-            MP_DBG(ctx, "sint %"PRId64"\n", *sintptr);
+            MP_TRACE(ctx, "sint %"PRId64"\n", *sintptr);
            break;

        case EBML_TYPE_FLOAT:;
            double *floatptr;
            GETPTR(floatptr, double);
            if (length != 0 && length != 4 && length != 8) {
-                MP_DBG(ctx, "float invalid length %"PRIu64"\n", length);
+                MP_ERR(ctx, "float invalid length %"PRIu64"\n", length);
                goto error;
            }
            *floatptr = ebml_parse_float(data, length);
@@ -554,7 +555,7 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
            char **strptr;
            GETPTR(strptr, char *);
            *strptr = talloc_strndup(ctx->talloc_ctx, data, length);
-            MP_DBG(ctx, "string \"%s\"\n", *strptr);
+            MP_TRACE(ctx, "string \"%s\"\n", *strptr);
            break;

        case EBML_TYPE_BINARY:;
@@ -566,7 +567,7 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
            GETPTR(binptr, struct bstr);
            binptr->start = data;
            binptr->len = length;
-            MP_DBG(ctx, "binary %zd bytes\n", binptr->len);
+            MP_TRACE(ctx, "binary %zd bytes\n", binptr->len);
            break;

        case EBML_TYPE_EBML_ID:;
@@ -574,10 +575,10 @@ static void ebml_parse_element(struct ebml_parse_ctx *ctx, void *target,
            GETPTR(idptr, uint32_t);
            *idptr = ebml_parse_id(data, end - data, &len);
            if (len != length) {
-                MP_DBG(ctx, "ebml_id broken value\n");
+                MP_ERR(ctx, "ebml_id broken value\n");
                goto error;
            }
-            MP_DBG(ctx, "ebml_id %x\n", (unsigned)*idptr);
+            MP_TRACE(ctx, "ebml_id %x\n", (unsigned)*idptr);
            break;
        default:
            abort();
@@ -27,6 +27,7 @@

 #include "common/av_common.h"
 #include "common/common.h"
+#include "demux.h"

 #include "packet.h"

@@ -34,6 +35,7 @@ static void packet_destroy(void *ptr)
 {
    struct demux_packet *dp = ptr;
    av_packet_unref(dp->avpacket);
+    mp_packet_tags_unref(dp->metadata);
 }

 // This actually preserves only data and side data, not PTS/DTS/pos/etc.
@@ -78,6 +80,8 @@ struct demux_packet *new_demux_packet_from_avpacket(struct AVPacket *avpkt)
 // (buf must include proper padding)
 struct demux_packet *new_demux_packet_from_buf(struct AVBufferRef *buf)
 {
+    if (!buf)
+        return NULL;
    AVPacket pkt = {
        .size = buf->size,
        .data = buf->data,
@@ -106,8 +110,8 @@ struct demux_packet *new_demux_packet(size_t len)
 void demux_packet_shorten(struct demux_packet *dp, size_t len)
 {
    assert(len <= dp->len);
-    dp->len = len;
-    memset(dp->buffer + dp->len, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+    av_shrink_packet(dp->avpacket, len);
+    dp->len = dp->avpacket->size;
 }

 void free_demux_packet(struct demux_packet *dp)
@@ -127,6 +131,7 @@ void demux_packet_copy_attribs(struct demux_packet *dst, struct demux_packet *sr
    dst->codec = src->codec;
    dst->keyframe = src->keyframe;
    dst->stream = src->stream;
+    mp_packet_tags_setref(&dst->metadata, src->metadata);
 }

 struct demux_packet *demux_copy_packet(struct demux_packet *dp)
@@ -44,6 +44,7 @@ typedef struct demux_packet {
    struct demux_packet *next;
    struct AVPacket *avpacket;   // keep the buffer allocation and sidedata
    double kf_seek_pts; // demux.c internal: seek pts for keyframe range
+    struct mp_packet_tags *metadata; // timed metadata (demux.c internal)
 } demux_packet_t;

 struct AVBufferRef;
@@ -45,6 +45,8 @@ struct sh_stream {
    char *lang;                 // language code
    bool default_track;         // container default track flag
    bool forced_track;          // container forced track flag
+    bool dependent_track;       // container dependent track flag
+    bool still_image;           // video stream contains still images
    int hls_bitrate;

    struct mp_tags *tags;
@@ -76,6 +78,11 @@ struct mp_codec_params {
    // Timestamp granularity for converting double<->rational timestamps.
    int native_tb_num, native_tb_den;

+    // Used by an obscure bug workaround mechanism. As an exception to the usual
+    // rules, demuxers are allowed to set this after adding the sh_stream, but
+    // only before the demuxer open call returns.
+    struct demux_packet *first_packet;
+
    // STREAM_AUDIO
    int samplerate;
    struct mp_chmap channels;
@@ -17,7 +17,6 @@ idle=yes
 terminal=no
 input-terminal=no
 osc=no
-ytdl=no
 input-default-bindings=no
 input-vo-keyboard=no
 # OSX/Cocoa global input hooks
@@ -44,6 +43,17 @@ correct-downscaling=yes
 sigmoid-upscaling=yes
 deband=yes

+[low-latency]
+audio-buffer=0          # minimize extra audio buffer (can lead to dropouts)
+vd-lavc-threads=1       # multithreaded decoding buffers extra frames
+cache-pause=no          # do not pause on underruns
+demuxer-lavf-o-add=fflags=+nobuffer # can help for weird reasons
+demuxer-lavf-probe-info=nostreams   # avoid probing unless absolutely needed
+demuxer-lavf-analyzeduration=0.1    # if it probes, reduce it
+video-sync=audio        # DS currently requires reading ahead a frame
+interpolation=no        # requires reference frames (more buffering)
+video-latency-hacks=yes # typically 1 or 2 video frame less latency
+
 # Compatibility alias (deprecated)
 [opengl-hq]
 profile=gpu-hq
@@ -31,7 +31,7 @@
 #
 # Examples:
 #   mpv --profile=enc-to-dvdpal --o=outfile.mpg infile.mkv
-#   mpv --profile=enc-f-avi --ofps=30 --o=outfile.avi infile.mkv
+#   mpv --profile=enc-f-avi --vf=fps=30 --o=outfile.avi infile.mkv
 #   mpv --profile=enc-v-mpeg4 --ovcopts-add=qscale=7 --profile=enc-a-mp3 --oacopts-add=b=320k --o=outfile.avi infile.mkv

 ################
@@ -105,24 +105,20 @@ ovcopts = speed=6,lag-in-frames=8,slices=2,threads=0,crf=18,qmin=0,qmax=36
 [enc-f-3gp]
 profile-desc = "H.263 + AAC (for 3GP)"
 of = 3gp
-ocopyts = yes
 profile = enc-v-h263
 profile = enc-a-aac
-ofopts-clr = yes
+ofopts = ""

 [enc-f-avi]
 profile-desc = "MPEG-4 + MP3 (for AVI)"
 of = avi
-ocopyts = no
-oautofps = yes
 profile = enc-v-mpeg4
 profile = enc-a-mp3
-ofopts-clr = yes
+ofopts = ""

 [enc-f-mp4]
 profile-desc = "H.264 + AAC (for MP4)"
 of = mp4
-ocopyts = yes
 profile = enc-v-h264
 profile = enc-a-aac
 ## equivalent to using qt-faststart tool
@@ -132,10 +128,9 @@ profile = enc-a-aac
 [enc-f-webm]
 profile-desc = "VP9 + Opus (for WebM)"
 of = webm
-ocopyts = yes
 profile = enc-v-vp9
 profile = enc-a-opus
-ofopts-clr = yes
+ofopts = ""

 ##################
 # target devices #
@@ -147,9 +142,7 @@ profile = enc-a-ac3
 of = dvd
 ofopts-append = packetsize=2048
 ofopts-append = muxrate=10080000
-ofps = 25
-oharddup = yes
-ovfirst = yes # dvdauthor needs this
+vf-add = fps=25
 audio-samplerate = 48000
 ovcopts = g=15,b=6000000,maxrate=9000000,minrate=0,bufsize=1835008

@@ -160,9 +153,7 @@ profile = enc-a-ac3
 of = dvd
 ofopts-append = packetsize=2048
 ofopts-append = muxrate=10080000
-ofps = 24000/1001
-oharddup = yes
-ovfirst = yes # dvdauthor needs this
+vf-add = fps="24000/1001"
 audio-samplerate = 48000
 ovcopts = g=18,b=6000000,maxrate=9000000,minrate=0,bufsize=1835008

@@ -177,12 +168,11 @@ ovcopts-append = maxrate=10000k
 ovcopts-append = bufsize=10000k
 ovcopts-append = rc_init_occupancy=9000k
 ovcopts-append = refs=5
-omaxfps = 30

 [enc-to-nok-6300]
 profile-desc = "3GP for Nokia 6300"
 profile = enc-f-3gp
-ofps = 25
+vf-add = fps=25
 vf-add = lavfi=graph="scale=176:144"
 audio-samplerate = 16000
 audio-channels = 1
@@ -191,7 +181,6 @@ oacopts-add = b=32k
 [enc-to-hp-slate-7]
 profile-desc = "MP4 for HP Slate 7 (1024x600, crazy aspect)"
 profile = enc-f-mp4
-omaxfps = 30
 ovcopts-add = profile=high
 # DW = 1024, DH = 600, DAR = 97:54 (=> SAR = 2425:2304)
 vf-add = lavfi=graph="scale=floor(min(1024*min(1\,dar/(97/54))\,in_w)/2+0.5)*2:floor(min(600*min((97/54)/dar\,1)\,in_h)/2+0.5)*2,setsar=sar=sar/(2425/2304)"
@@ -54,15 +54,30 @@
 # Skip to previous/next subtitle (subject to some restrictions; see manpage)
 #Ctrl+LEFT   no-osd sub-seek -1
 #Ctrl+RIGHT  no-osd sub-seek  1
+# Adjust timing to previous/next subtitle
+#Ctrl+Shift+LEFT sub-step -1
+#Ctrl+Shift+RIGHT sub-step 1
+# Move video rectangle
+#Alt+left  add video-pan-x  0.1
+#Alt+right add video-pan-x -0.1
+#Alt+up    add video-pan-y  0.1
+#Alt+down  add video-pan-y -0.1
+# Zoom/unzoom video
+#Alt++     add video-zoom   0.1
+#Alt+-     add video-zoom  -0.1
+# Reset video zoom/pan settings
+#Alt+BS set video-zoom 0 ; set video-pan-x 0 ; set video-pan-y 0
 #PGUP add chapter 1                     # skip to next chapter
 #PGDWN add chapter -1                   # skip to previous chapter
 #Shift+PGUP seek 600
 #Shift+PGDWN seek -600
-#[ multiply speed 0.9091                # scale playback speed
+#[ multiply speed 1/1.1                 # scale playback speed
 #] multiply speed 1.1
 #{ multiply speed 0.5
 #} multiply speed 2.0
 #BS set speed 1.0                       # reset speed to normal
+#Shift+BS revert-seek                   # undo previous (or marked) seek
+#Shift+Ctrl+BS revert-seek mark         # mark position for revert-seek
 #q quit
 #Q quit-watch-later
 #q {encode} quit 4
@@ -81,7 +96,8 @@
 #i script-binding stats/display-stats
 #I script-binding stats/display-stats-toggle
 #z add sub-delay -0.1                   # subtract 100 ms delay from subs
-#x add sub-delay +0.1                   # add
+#Z add sub-delay +0.1                   # add
+#x add sub-delay +0.1                   # same as previous binding (discouraged)
 #ctrl++ add audio-delay 0.100           # this changes audio/video sync
 #ctrl+- add audio-delay -0.100
 #9 add volume -2
@@ -103,7 +119,8 @@
 # toggle deinterlacer (automatically inserts or removes required filter)
 #d cycle deinterlace
 #r add sub-pos -1                       # move subtitles up
-#t add sub-pos +1                       #                down
+#R add sub-pos +1                       #                down
+#t add sub-pos +1                       # same as previous binding (discouraged)
 #v cycle sub-visibility
 # stretch SSA/ASS subtitles with anamorphic videos to match historical
 #V cycle sub-ass-vsfilter-aspect-compat
@@ -121,7 +138,8 @@
 #Ctrl+s async screenshot window         # ...with subtitles and OSD, and scaled
 #Alt+s screenshot each-frame            # automatically screenshot every frame
 #w add panscan -0.1                     # zoom out with -panscan 0 -fs
-#e add panscan +0.1                     #      in
+#W add panscan +0.1                     #      in
+#e add panscan +0.1                     # same as previous binding (discouraged)
 # cycle video aspect ratios; "-1" is the container aspect
 #A cycle-values video-aspect "16:9" "4:3" "2.35:1" "-1"
 #POWER quit
@@ -143,6 +161,9 @@
 #L cycle-values loop-file "inf" "no"    # toggle infinite looping
 #ctrl+c quit 4
 #DEL script-binding osc/visibility      # cycle OSC display
+#ctrl+h cycle-values hwdec "auto" "no"  # cycle hardware decoding
+#F8 show_text ${playlist}               # show playlist
+#F9 show_text ${track-list}             # show list of audio/sub streams

 # Apple Remote section
 #AR_PLAY cycle pause
@@ -173,11 +194,7 @@

 # ? add sub-scale +0.1                  # increase subtitle font size
 # ? add sub-scale -0.1                  # decrease subtitle font size
-# ? sub-step -1                         # immediately display next subtitle
-# ? sub-step +1                         #                     previous
 # ? cycle angle                         # switch DVD/Bluray angle
-# ? add balance -0.1                    # adjust audio balance in favor of left
-# ? add balance 0.1                     #                                  right
 # ? cycle sub-forced-only               # toggle DVD forced subs
 # ? cycle program                       # cycle transport stream programs
 # ? stop                                # stop playback (quit or enter idle mode)
@@ -4,6 +4,7 @@ Name=mpv Media Player
 Name[ca]=Reproductor multimèdia mpv
 Name[cs]=mpv přehrávač
 Name[da]=mpv-medieafspiller
+Name[fr]=Lecteur multimédia mpv
 Name[pl]=Odtwarzacz mpv
 Name[ru]=Проигрыватель mpv
 Name[zh_CN]=mpv 媒体播放器
@@ -11,6 +12,7 @@ Name[zh_TW]=mpv 媒體播放器
 GenericName=Multimedia player
 GenericName[cs]=Multimediální přehrávač
 GenericName[da]=Multimedieafspiller
+GenericName[fr]=Lecteur multimédia
 GenericName[zh_CN]=多媒体播放器
 GenericName[zh_TW]=多媒體播放器
 Comment=Play movies and songs
@@ -19,7 +21,7 @@ Comment[cs]=Přehrává filmy a hudbu
 Comment[da]=Afspil film og sange
 Comment[de]=Filme und Musik abspielen
 Comment[es]=Reproduzca vídeos y canciones
-Comment[fr]=Lecteur multimédia
+Comment[fr]=Lire des vidéos et des musiques
 Comment[it]=Lettore multimediale
 Comment[pl]=Odtwarzaj filmy i muzykę
 Comment[ru]=Воспроизвести фильмы и музыку
@@ -30,5 +32,5 @@ TryExec=mpv
 Exec=mpv --player-operation-mode=pseudo-gui -- %U
 Terminal=false
 Categories=AudioVideo;Audio;Video;Player;TV;
-MimeType=application/ogg;application/x-ogg;application/sdp;application/smil;application/x-smil;application/streamingmedia;application/x-streamingmedia;application/vnd.rn-realmedia;application/vnd.rn-realmedia-vbr;audio/aac;audio/x-aac;audio/m4a;audio/x-m4a;audio/mp1;audio/x-mp1;audio/mp2;audio/x-mp2;audio/mp3;audio/x-mp3;audio/mpeg;audio/x-mpeg;audio/mpegurl;audio/x-mpegurl;audio/mpg;audio/x-mpg;audio/rn-mpeg;audio/ogg;audio/scpls;audio/x-scpls;audio/vnd.rn-realaudio;audio/wav;audio/x-pn-windows-pcm;audio/x-realaudio;audio/x-pn-realaudio;audio/x-ms-wma;audio/x-pls;audio/x-wav;video/mpeg;video/x-mpeg;video/x-mpeg2;video/mp4;video/msvideo;video/x-msvideo;video/ogg;video/quicktime;video/vnd.rn-realvideo;video/x-ms-afs;video/x-ms-asf;video/x-ms-wmv;video/x-ms-wmx;video/x-ms-wvxvideo;video/x-avi;video/x-fli;video/x-flv;video/x-theora;video/x-matroska;video/webm;audio/x-flac;audio/x-vorbis+ogg;video/x-ogm+ogg;audio/x-shorten;audio/x-ape;audio/x-wavpack;audio/x-tta;audio/AMR;audio/ac3;video/mp2t;audio/flac;audio/mp4;
+MimeType=application/ogg;application/x-ogg;application/mxf;application/sdp;application/smil;application/x-smil;application/streamingmedia;application/x-streamingmedia;application/vnd.rn-realmedia;application/vnd.rn-realmedia-vbr;audio/aac;audio/x-aac;audio/vnd.dolby.heaac.1;audio/vnd.dolby.heaac.2;audio/aiff;audio/x-aiff;audio/m4a;audio/x-m4a;application/x-extension-m4a;audio/mp1;audio/x-mp1;audio/mp2;audio/x-mp2;audio/mp3;audio/x-mp3;audio/mpeg;audio/mpeg2;audio/mpeg3;audio/mpegurl;audio/x-mpegurl;audio/mpg;audio/x-mpg;audio/rn-mpeg;audio/musepack;audio/x-musepack;audio/ogg;audio/scpls;audio/x-scpls;audio/vnd.rn-realaudio;audio/wav;audio/x-pn-wav;audio/x-pn-windows-pcm;audio/x-realaudio;audio/x-pn-realaudio;audio/x-ms-wma;audio/x-pls;audio/x-wav;video/mpeg;video/x-mpeg2;video/x-mpeg3;video/mp4v-es;video/x-m4v;video/mp4;application/x-extension-mp4;video/divx;video/vnd.divx;video/msvideo;video/x-msvideo;video/ogg;video/quicktime;video/vnd.rn-realvideo;video/x-ms-afs;video/x-ms-asf;audio/x-ms-asf;application/vnd.ms-asf;video/x-ms-wmv;video/x-ms-wmx;video/x-ms-wvxvideo;video/x-avi;video/avi;video/x-flic;video/fli;video/x-flc;video/flv;video/x-flv;video/x-theora;video/x-theora+ogg;video/x-matroska;video/mkv;audio/x-matroska;application/x-matroska;video/webm;audio/webm;audio/vorbis;audio/x-vorbis;audio/x-vorbis+ogg;video/x-ogm;video/x-ogm+ogg;application/x-ogm;application/x-ogm-audio;application/x-ogm-video;application/x-shorten;audio/x-shorten;audio/x-ape;audio/x-wavpack;audio/x-tta;audio/AMR;audio/ac3;audio/eac3;audio/amr-wb;video/mp2t;audio/flac;audio/mp4;application/x-mpegurl;video/vnd.mpegurl;application/vnd.apple.mpegurl;audio/x-pn-au;video/3gp;video/3gpp;video/3gpp2;audio/3gpp;audio/3gpp2;video/dv;audio/dv;audio/opus;audio/vnd.dts;audio/vnd.dts.hd;audio/x-adpcm;application/x-cue;audio/m3u;
 X-KDE-Protocols=ftp,http,https,mms,rtmp,rtsp,sftp,smb
@@ -0,0 +1,363 @@
+#include <math.h>
+
+#include "common/common.h"
+#include "common/msg.h"
+#include "options/m_config.h"
+#include "options/options.h"
+#include "video/mp_image.h"
+
+#include "f_auto_filters.h"
+#include "f_swscale.h"
+#include "f_utils.h"
+#include "filter.h"
+#include "filter_internal.h"
+#include "user_filters.h"
+
+struct deint_priv {
+    struct mp_subfilter sub;
+    int prev_imgfmt;
+    int prev_setting;
+    struct m_config_cache *opts;
+};
+
+static void deint_process(struct mp_filter *f)
+{
+    struct deint_priv *p = f->priv;
+
+    if (!mp_subfilter_read(&p->sub))
+        return;
+
+    struct mp_frame frame = p->sub.frame;
+
+    if (mp_frame_is_signaling(frame)) {
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    if (frame.type != MP_FRAME_VIDEO) {
+        MP_ERR(f, "video input required!\n");
+        mp_filter_internal_mark_failed(f);
+        return;
+    }
+
+    m_config_cache_update(p->opts);
+    struct filter_opts *opts = p->opts->opts;
+
+    if (!opts->deinterlace)
+        mp_subfilter_destroy(&p->sub);
+
+    struct mp_image *img = frame.data;
+
+    if (img->imgfmt == p->prev_imgfmt && p->prev_setting == opts->deinterlace) {
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    if (!mp_subfilter_drain_destroy(&p->sub))
+        return;
+
+    assert(!p->sub.filter);
+
+    p->prev_imgfmt = img->imgfmt;
+    p->prev_setting = opts->deinterlace;
+    if (!p->prev_setting) {
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    if (img->imgfmt == IMGFMT_VDPAU) {
+        char *args[] = {"deint", "yes", NULL};
+        p->sub.filter =
+            mp_create_user_filter(f, MP_OUTPUT_CHAIN_VIDEO, "vdpaupp", args);
+    } else if (img->imgfmt == IMGFMT_VAAPI) {
+        p->sub.filter =
+            mp_create_user_filter(f, MP_OUTPUT_CHAIN_VIDEO, "vavpp", NULL);
+    } else if (img->imgfmt == IMGFMT_D3D11) {
+        p->sub.filter =
+            mp_create_user_filter(f, MP_OUTPUT_CHAIN_VIDEO, "d3d11vpp", NULL);
+    } else if (mp_sws_supports_input(img->imgfmt)) {
+        char *args[] = {"mode", "send_field", NULL};
+        p->sub.filter =
+            mp_create_user_filter(f, MP_OUTPUT_CHAIN_VIDEO, "yadif", args);
+    } else {
+        MP_ERR(f, "no deinterlace filter available for this format\n");
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    if (!p->sub.filter)
+        MP_ERR(f, "creating deinterlacer failed\n");
+
+    mp_subfilter_continue(&p->sub);
+}
+
+static void deint_reset(struct mp_filter *f)
+{
+    struct deint_priv *p = f->priv;
+
+    mp_subfilter_reset(&p->sub);
+}
+
+static void deint_destroy(struct mp_filter *f)
+{
+    struct deint_priv *p = f->priv;
+
+    mp_subfilter_reset(&p->sub);
+    TA_FREEP(&p->sub.filter);
+}
+
+static bool deint_command(struct mp_filter *f, struct mp_filter_command *cmd)
+{
+    struct deint_priv *p = f->priv;
+
+    if (cmd->type == MP_FILTER_COMMAND_IS_ACTIVE) {
+        cmd->is_active = !!p->sub.filter;
+        return true;
+    }
+    return false;
+}
+
+static const struct mp_filter_info deint_filter = {
+    .name = "deint",
+    .priv_size = sizeof(struct deint_priv),
+    .command = deint_command,
+    .process = deint_process,
+    .reset = deint_reset,
+    .destroy = deint_destroy,
+};
+
+struct mp_filter *mp_deint_create(struct mp_filter *parent)
+{
+    struct mp_filter *f = mp_filter_create(parent, &deint_filter);
+    if (!f)
+        return NULL;
+
+    struct deint_priv *p = f->priv;
+
+    p->sub.in = mp_filter_add_pin(f, MP_PIN_IN, "in");
+    p->sub.out = mp_filter_add_pin(f, MP_PIN_OUT, "out");
+
+    p->opts = m_config_cache_alloc(f, f->global, &filter_conf);
+
+    return f;
+}
+
+struct rotate_priv {
+    struct mp_subfilter sub;
+    int prev_rotate;
+    int prev_imgfmt;
+    int target_rotate;
+};
+
+static void rotate_process(struct mp_filter *f)
+{
+    struct rotate_priv *p = f->priv;
+
+    if (!mp_subfilter_read(&p->sub))
+        return;
+
+    struct mp_frame frame = p->sub.frame;
+
+    if (mp_frame_is_signaling(frame)) {
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    if (frame.type != MP_FRAME_VIDEO) {
+        MP_ERR(f, "video input required!\n");
+        return;
+    }
+
+    struct mp_image *img = frame.data;
+
+    if (img->params.rotate == p->prev_rotate &&
+        img->imgfmt == p->prev_imgfmt)
+    {
+        img->params.rotate = p->target_rotate;
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    if (!mp_subfilter_drain_destroy(&p->sub))
+        return;
+
+    assert(!p->sub.filter);
+
+    int rotate = p->prev_rotate = img->params.rotate;
+    p->target_rotate = rotate;
+    p->prev_imgfmt = img->imgfmt;
+
+    struct mp_stream_info *info = mp_filter_find_stream_info(f);
+    if (rotate == 0 || (info && info->rotate90 && !(rotate % 90))) {
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    if (!mp_sws_supports_input(img->imgfmt)) {
+        MP_ERR(f, "Video rotation with this format not supported\n");
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    double angle = rotate / 360.0 * M_PI * 2;
+    char *args[] = {"angle", mp_tprintf(30, "%f", angle),
+                    "ow", mp_tprintf(30, "rotw(%f)", angle),
+                    "oh", mp_tprintf(30, "roth(%f)", angle),
+                    NULL};
+    p->sub.filter =
+        mp_create_user_filter(f, MP_OUTPUT_CHAIN_VIDEO, "rotate", args);
+
+    if (p->sub.filter) {
+        MP_INFO(f, "Inserting rotation filter.\n");
+        p->target_rotate = 0;
+    } else {
+        MP_ERR(f, "could not create rotation filter\n");
+    }
+
+    mp_subfilter_continue(&p->sub);
+}
+
+static void rotate_reset(struct mp_filter *f)
+{
+    struct rotate_priv *p = f->priv;
+
+    mp_subfilter_reset(&p->sub);
+}
+
+static void rotate_destroy(struct mp_filter *f)
+{
+    struct rotate_priv *p = f->priv;
+
+    mp_subfilter_reset(&p->sub);
+    TA_FREEP(&p->sub.filter);
+}
+
+static bool rotate_command(struct mp_filter *f, struct mp_filter_command *cmd)
+{
+    struct rotate_priv *p = f->priv;
+
+    if (cmd->type == MP_FILTER_COMMAND_IS_ACTIVE) {
+        cmd->is_active = !!p->sub.filter;
+        return true;
+    }
+    return false;
+}
+
+static const struct mp_filter_info rotate_filter = {
+    .name = "autorotate",
+    .priv_size = sizeof(struct rotate_priv),
+    .command = rotate_command,
+    .process = rotate_process,
+    .reset = rotate_reset,
+    .destroy = rotate_destroy,
+};
+
+struct mp_filter *mp_autorotate_create(struct mp_filter *parent)
+{
+    struct mp_filter *f = mp_filter_create(parent, &rotate_filter);
+    if (!f)
+        return NULL;
+
+    struct rotate_priv *p = f->priv;
+    p->prev_rotate = -1;
+
+    p->sub.in = mp_filter_add_pin(f, MP_PIN_IN, "in");
+    p->sub.out = mp_filter_add_pin(f, MP_PIN_OUT, "out");
+
+    return f;
+}
+
+struct aspeed_priv {
+    struct mp_subfilter sub;
+    double cur_speed;
+};
+
+static void aspeed_process(struct mp_filter *f)
+{
+    struct aspeed_priv *p = f->priv;
+
+    if (!mp_subfilter_read(&p->sub))
+        return;
+
+    if (fabs(p->cur_speed - 1.0) < 1e-8) {
+        if (p->sub.filter)
+            MP_VERBOSE(f, "removing scaletempo\n");
+        if (!mp_subfilter_drain_destroy(&p->sub))
+            return;
+    } else if (!p->sub.filter) {
+        MP_VERBOSE(f, "adding scaletempo\n");
+        p->sub.filter =
+            mp_create_user_filter(f, MP_OUTPUT_CHAIN_AUDIO, "scaletempo", NULL);
+        if (!p->sub.filter) {
+            MP_ERR(f, "could not create scaletempo filter\n");
+            mp_subfilter_continue(&p->sub);
+            return;
+        }
+    }
+
+    if (p->sub.filter) {
+        struct mp_filter_command cmd = {
+            .type = MP_FILTER_COMMAND_SET_SPEED,
+            .speed = p->cur_speed,
+        };
+        mp_filter_command(p->sub.filter, &cmd);
+    }
+
+    mp_subfilter_continue(&p->sub);
+}
+
+static bool aspeed_command(struct mp_filter *f, struct mp_filter_command *cmd)
+{
+    struct aspeed_priv *p = f->priv;
+
+    if (cmd->type == MP_FILTER_COMMAND_SET_SPEED) {
+        p->cur_speed = cmd->speed;
+        return true;
+    }
+
+    if (cmd->type == MP_FILTER_COMMAND_IS_ACTIVE) {
+        cmd->is_active = !!p->sub.filter;
+        return true;
+    }
+
+    return false;
+}
+
+static void aspeed_reset(struct mp_filter *f)
+{
+    struct aspeed_priv *p = f->priv;
+
+    mp_subfilter_reset(&p->sub);
+}
+
+static void aspeed_destroy(struct mp_filter *f)
+{
+    struct aspeed_priv *p = f->priv;
+
+    mp_subfilter_reset(&p->sub);
+    TA_FREEP(&p->sub.filter);
+}
+
+static const struct mp_filter_info aspeed_filter = {
+    .name = "autoaspeed",
+    .priv_size = sizeof(struct aspeed_priv),
+    .command = aspeed_command,
+    .process = aspeed_process,
+    .reset = aspeed_reset,
+    .destroy = aspeed_destroy,
+};
+
+struct mp_filter *mp_autoaspeed_create(struct mp_filter *parent)
+{
+    struct mp_filter *f = mp_filter_create(parent, &aspeed_filter);
+    if (!f)
+        return NULL;
+
+    struct aspeed_priv *p = f->priv;
+    p->cur_speed = 1.0;
+
+    p->sub.in = mp_filter_add_pin(f, MP_PIN_IN, "in");
+    p->sub.out = mp_filter_add_pin(f, MP_PIN_OUT, "out");
+
+    return f;
+}
@@ -0,0 +1,13 @@
+#pragma once
+
+#include "filter.h"
+
+// A filter which inserts the required deinterlacing filter based on the
+// hardware decode mode and the deinterlace user option.
+struct mp_filter *mp_deint_create(struct mp_filter *parent);
+
+// Rotate according to mp_image.rotate and VO capabilities.
+struct mp_filter *mp_autorotate_create(struct mp_filter *parent);
+
+// Insert a filter that inserts scaletempo depending on speed settings.
+struct mp_filter *mp_autoaspeed_create(struct mp_filter *parent);
@@ -0,0 +1,473 @@
+#include "config.h"
+
+#include "audio/aframe.h"
+#include "audio/chmap_sel.h"
+#include "audio/format.h"
+#include "common/common.h"
+#include "common/msg.h"
+#include "video/hwdec.h"
+#include "video/mp_image.h"
+
+#include "f_autoconvert.h"
+#include "f_hwtransfer.h"
+#include "f_swresample.h"
+#include "f_swscale.h"
+#include "f_utils.h"
+#include "filter.h"
+#include "filter_internal.h"
+
+struct priv {
+    struct mp_log *log;
+
+    struct mp_subfilter sub;
+
+    bool force_update;
+
+    int *imgfmts;
+    int *subfmts;
+    int num_imgfmts;
+
+    // Enable special conversion for the final stage before the VO.
+    bool vo_convert;
+
+    // sws state
+    int in_imgfmt, in_subfmt;
+
+    int *afmts;
+    int num_afmts;
+    int *srates;
+    int num_srates;
+    struct mp_chmap_sel chmaps;
+
+    int in_afmt, in_srate;
+    struct mp_chmap in_chmap;
+
+    double audio_speed;
+    bool resampling_forced;
+
+    bool format_change_blocked;
+    bool format_change_cont;
+
+    struct mp_autoconvert public;
+};
+
+// Dummy filter for bundling sub-conversion filters.
+static const struct mp_filter_info convert_filter = {
+    .name = "convert",
+};
+
+// For hw decoding: thing which can convert between underlying surface formats.
+// The filter detects the needed target format from struct mp_hwdec_ctx.
+struct subfmt_conv {
+    int hw_imgfmt;
+    struct mp_filter *(*create)(struct mp_filter *parent);
+};
+
+static const struct subfmt_conv subfmt_converters[] = {
+#if HAVE_D3D_HWACCEL
+    {IMGFMT_D3D11, vf_d3d11_create_outconv},
+#endif
+    {0}
+};
+
+void mp_autoconvert_clear(struct mp_autoconvert *c)
+{
+    struct priv *p = c->f->priv;
+
+    p->num_imgfmts = 0;
+    p->num_afmts = 0;
+    p->num_srates = 0;
+    p->chmaps = (struct mp_chmap_sel){0};
+    p->force_update = true;
+}
+
+void mp_autoconvert_add_imgfmt(struct mp_autoconvert *c, int imgfmt, int subfmt)
+{
+    struct priv *p = c->f->priv;
+
+    MP_TARRAY_GROW(p, p->imgfmts, p->num_imgfmts);
+    MP_TARRAY_GROW(p, p->subfmts, p->num_imgfmts);
+
+    p->imgfmts[p->num_imgfmts] = imgfmt;
+    p->subfmts[p->num_imgfmts] = subfmt;
+
+    p->num_imgfmts += 1;
+    p->force_update = true;
+}
+
+void mp_autoconvert_add_vo_hwdec_subfmts(struct mp_autoconvert *c,
+                                         struct mp_hwdec_devices *devs)
+{
+    struct priv *p = c->f->priv;
+    assert(devs);
+
+    int prev_format = 0;
+
+    for (int n = 0; ; n++) {
+        struct mp_hwdec_ctx *ctx = hwdec_devices_get_n(devs, n);
+        if (!ctx)
+            break;
+        if (!ctx->hw_imgfmt || !ctx->supported_formats)
+            continue;
+        // Very hacky: don't let d3d11-egl-rgb overwrite d3d11-egl
+        if (ctx->hw_imgfmt == prev_format)
+            continue;
+        prev_format = ctx->hw_imgfmt;
+        // Stupidity: VOs export imgfmt only, so subfmt is always 0. Remove it
+        // to fix it up.
+        for (int i = 0; i < p->num_imgfmts; i++) {
+            if (p->imgfmts[i] != ctx->hw_imgfmt)
+                continue;
+
+            int count = p->num_imgfmts;
+            MP_TARRAY_REMOVE_AT(p->imgfmts, count, i);
+            count = p->num_imgfmts;
+            MP_TARRAY_REMOVE_AT(p->subfmts, count, i);
+            p->num_imgfmts -= 1;
+            break;
+        }
+        for (int i = 0; ctx->supported_formats[i]; i++)
+            mp_autoconvert_add_imgfmt(c, ctx->hw_imgfmt, ctx->supported_formats[i]);
+    }
+
+    p->vo_convert = true;
+}
+
+void mp_autoconvert_add_afmt(struct mp_autoconvert *c, int afmt)
+{
+    struct priv *p = c->f->priv;
+
+    MP_TARRAY_APPEND(p, p->afmts, p->num_afmts, afmt);
+    p->force_update = true;
+}
+
+void mp_autoconvert_add_chmap(struct mp_autoconvert *c, struct mp_chmap *chmap)
+{
+    struct priv *p = c->f->priv;
+
+    mp_chmap_sel_add_map(&p->chmaps, chmap);
+    p->force_update = true;
+}
+
+void mp_autoconvert_add_srate(struct mp_autoconvert *c, int rate)
+{
+    struct priv *p = c->f->priv;
+
+    MP_TARRAY_APPEND(p, p->srates, p->num_srates, rate);
+    // Some other API we call expects a 0-terminated sample rates array.
+    MP_TARRAY_GROW(p, p->srates, p->num_srates);
+    p->srates[p->num_srates] = 0;
+    p->force_update = true;
+}
+
+static void handle_video_frame(struct mp_filter *f)
+{
+    struct priv *p = f->priv;
+
+    struct mp_image *img = p->sub.frame.data;
+
+    if (p->force_update)
+        p->in_imgfmt = p->in_subfmt = 0;
+
+    if (img->imgfmt == p->in_imgfmt && img->params.hw_subfmt == p->in_subfmt) {
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    if (!mp_subfilter_drain_destroy(&p->sub)) {
+        p->in_imgfmt = p->in_subfmt = 0;
+        return;
+    }
+
+    p->in_imgfmt = img->params.imgfmt;
+    p->in_subfmt = img->params.hw_subfmt;
+    p->force_update = false;
+
+    if (!p->num_imgfmts) {
+        mp_subfilter_continue(&p->sub);
+        return;
+    }
+
+    bool different_subfmt = false;
+
+    for (int n = 0; n < p->num_imgfmts; n++) {
+        bool samefmt = img->params.imgfmt == p->imgfmts[n];
+        bool samesubffmt = img->params.hw_subfmt == p->subfmts[n];
+        if (samefmt && !samesubffmt)
+            different_subfmt = true;
+        if (samefmt && (samesubffmt || !p->subfmts[n])) {
+            mp_subfilter_continue(&p->sub);
+            return;
+        }
+    }
+
+    struct mp_stream_info *info = mp_filter_find_stream_info(f);
+
+    struct mp_filter *conv = mp_filter_create(f, &convert_filter);
+    mp_filter_add_pin(conv, MP_PIN_IN, "in");
+    mp_filter_add_pin(conv, MP_PIN_OUT, "out");
+
+    struct mp_filter *filters[2] = {0};
+    bool need_sws = true;
+
+    int *fmts = p->imgfmts;
+    int num_fmts = p->num_imgfmts;
+
+    // Source is sw, all targets are hw -> try to upload.
+    bool sw_to_hw = !IMGFMT_IS_HWACCEL(img->imgfmt);
+    for (int n = 0; n < num_fmts; n++)
+        sw_to_hw &= IMGFMT_IS_HWACCEL(fmts[n]);
+
+    if (sw_to_hw && num_fmts > 0) {
+        // We can probably use this! Very lazy and very approximate.
+        struct mp_hwupload *upload = mp_hwupload_create(conv, fmts[0]);
+        if (upload) {
+            MP_INFO(p, "HW-uploading to %s\n", mp_imgfmt_to_name(fmts[0]));
+            filters[1] = upload->f;
+            fmts = upload->upload_fmts;
+            num_fmts = upload->num_upload_fmts;
+        }
+    } else if (p->vo_convert && different_subfmt && info && info->hwdec_devs) {
+        for (int n = 0; subfmt_converters[n].hw_imgfmt; n++) {
+            if (subfmt_converters[n].hw_imgfmt == img->imgfmt) {
+                MP_INFO(p, "Using HW sub-conversion.\n");
+                filters[1] = subfmt_converters[n].create(conv);
+                if (filters[1]) {
+                    need_sws = false;
+                    break;
+                }
+            }
+        }
+    }
+
+    if (need_sws) {
+        // Create a new conversion filter.
+        struct mp_sws_filter *sws = mp_sws_filter_create(conv);
+        if (!sws) {
+            MP_ERR(p, "error creating conversion filter\n");
+            return;
+        }
+
+        int out = mp_sws_find_best_out_format(img->imgfmt,  fmts, num_fmts);
+        if (!out) {
+            MP_ERR(p, "can't find video conversion for %s/%s\n",
+                   mp_imgfmt_to_name(img->imgfmt),
+                   mp_imgfmt_to_name(img->params.hw_subfmt));
+            talloc_free(conv);
+            mp_filter_internal_mark_failed(f);
+            return;
+        }
+
+        if (out == img->imgfmt) {
+            // Can happen if hwupload goes to same format.
+            talloc_free(sws->f);
+        } else {
+            sws->out_format = out;
+            MP_INFO(p, "Converting %s -> %s\n", mp_imgfmt_to_name(img->imgfmt),
+                    mp_imgfmt_to_name(sws->out_format));
+            filters[0] = sws->f;
+        }
+    }
+
+    mp_chain_filters(conv->ppins[0], conv->ppins[1], filters, 2);
+
+    p->sub.filter = conv;
+    mp_subfilter_continue(&p->sub);
+}
+
+static void handle_audio_frame(struct mp_filter *f)
+{
+    struct priv *p = f->priv;
+
+    struct mp_aframe *aframe = p->sub.frame.data;
+
+    int afmt = mp_aframe_get_format(aframe);
+    int srate = mp_aframe_get_rate(aframe);
+    struct mp_chmap chmap = {0};
+    mp_aframe_get_chmap(aframe, &chmap);
+
+    if (p->resampling_forced && !af_fmt_is_pcm(afmt)) {
+        MP_WARN(p, "ignoring request to resample non-PCM audio for speed change\n");
+        p->resampling_forced = false;
+    }
+
+    bool format_change = afmt != p->in_afmt ||
+                         srate != p->in_srate ||
+                         !mp_chmap_equals(&chmap, &p->in_chmap) ||
+                         p->force_update;
+
+    if (!format_change && (!p->resampling_forced || p->sub.filter))
+        goto cont;
+
+    if (!mp_subfilter_drain_destroy(&p->sub))
+        return;
+
+    if (format_change && p->public.on_audio_format_change) {
+        if (p->format_change_blocked)
+            return;
+
+        if (!p->format_change_cont) {
+            p->format_change_blocked = true;
+            p->public.
+                on_audio_format_change(p->public.on_audio_format_change_opaque);
+            return;
+        }
+        p->format_change_cont = false;
+    }
+
+    p->in_afmt = afmt;
+    p->in_srate = srate;
+    p->in_chmap = chmap;
+    p->force_update = false;
+
+    int out_afmt = 0;
+    int best_score = 0;
+    for (int n = 0; n < p->num_afmts; n++) {
+        int score = af_format_conversion_score(p->afmts[n], afmt);
+        if (!out_afmt || score > best_score) {
+            best_score = score;
+            out_afmt = p->afmts[n];
+        }
+    }
+    if (!out_afmt)
+        out_afmt = afmt;
+
+    // (The p->srates array is 0-terminated already.)
+    int out_srate = af_select_best_samplerate(srate, p->srates);
+    if (out_srate <= 0)
+        out_srate = p->num_srates ? p->srates[0] : srate;
+
+    struct mp_chmap out_chmap = chmap;
+    if (p->chmaps.num_chmaps) {
+        if (!mp_chmap_sel_adjust(&p->chmaps, &out_chmap))
+            out_chmap = p->chmaps.chmaps[0]; // violently force fallback
+    }
+
+    if (out_afmt == p->in_afmt && out_srate == p->in_srate &&
+        mp_chmap_equals(&out_chmap, &p->in_chmap) && !p->resampling_forced)
+    {
+        goto cont;
+    }
+
+    MP_VERBOSE(p, "inserting resampler\n");
+
+    struct mp_swresample *s = mp_swresample_create(f, NULL);
+    if (!s)
+        abort();
+
+    s->out_format = out_afmt;
+    s->out_rate = out_srate;
+    s->out_channels = out_chmap;
+
+    p->sub.filter = s->f;
+
+cont:
+
+    if (p->sub.filter) {
+        struct mp_filter_command cmd = {
+            .type = MP_FILTER_COMMAND_SET_SPEED_RESAMPLE,
+            .speed = p->audio_speed,
+        };
+        mp_filter_command(p->sub.filter, &cmd);
+    }
+
+    mp_subfilter_continue(&p->sub);
+}
+
+static void process(struct mp_filter *f)
+{
+    struct priv *p = f->priv;
+
+    if (!mp_subfilter_read(&p->sub))
+        return;
+
+    if (p->sub.frame.type == MP_FRAME_VIDEO) {
+        handle_video_frame(f);
+        return;
+    }
+
+    if (p->sub.frame.type == MP_FRAME_AUDIO) {
+        handle_audio_frame(f);
+        return;
+    }
+
+    mp_subfilter_continue(&p->sub);
+}
+
+void mp_autoconvert_format_change_continue(struct mp_autoconvert *c)
+{
+    struct priv *p = c->f->priv;
+
+    if (p->format_change_blocked) {
+        p->format_change_cont = true;
+        p->format_change_blocked = false;
+        mp_filter_wakeup(c->f);
+    }
+}
+
+static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
+{
+    struct priv *p = f->priv;
+
+    if (cmd->type == MP_FILTER_COMMAND_SET_SPEED_RESAMPLE) {
+        p->audio_speed = cmd->speed;
+        // If we needed resampling once, keep forcing resampling, as it might be
+        // quickly changing between 1.0 and other values for A/V compensation.
+        if (p->audio_speed != 1.0)
+            p->resampling_forced = true;
+        return true;
+    }
+
+    if (cmd->type == MP_FILTER_COMMAND_IS_ACTIVE) {
+        cmd->is_active = !!p->sub.filter;
+        return true;
+    }
+
+    return false;
+}
+
+static void reset(struct mp_filter *f)
+{
+    struct priv *p = f->priv;
+
+    mp_subfilter_reset(&p->sub);
+
+    p->format_change_cont = false;
+    p->format_change_blocked = false;
+}
+
+static void destroy(struct mp_filter *f)
+{
+    struct priv *p = f->priv;
+
+    mp_subfilter_reset(&p->sub);
+    TA_FREEP(&p->sub.filter);
+}
+
+static const struct mp_filter_info autoconvert_filter = {
+    .name = "autoconvert",
+    .priv_size = sizeof(struct priv),
+    .process = process,
+    .command = command,
+    .reset = reset,
+    .destroy = destroy,
+};
+
+struct mp_autoconvert *mp_autoconvert_create(struct mp_filter *parent)
+{
+    struct mp_filter *f = mp_filter_create(parent, &autoconvert_filter);
+    if (!f)
+        return NULL;
+
+    mp_filter_add_pin(f, MP_PIN_IN, "in");
+    mp_filter_add_pin(f, MP_PIN_OUT, "out");
+
+    struct priv *p = f->priv;
+    p->public.f = f;
+    p->log = f->log;
+    p->audio_speed = 1.0;
+    p->sub.in = f->ppins[0];
+    p->sub.out = f->ppins[1];
+
+    return &p->public;
+}
@@ -0,0 +1,60 @@
+#pragma once
+
+#include "filter.h"
+
+// A filter which automatically creates and uses a conversion filter based on
+// the filter settings, or passes through data unchanged if no conversion is
+// required.
+struct mp_autoconvert {
+    // f->pins[0] is input, f->pins[1] is output
+    struct mp_filter *f;
+
+    // If this is set, the callback is invoked (from the process function), and
+    // further data flow is blocked until mp_autoconvert_format_change_continue()
+    // is called. The idea is that you can reselect the output parameters on
+    // format changes and continue filtering when ready.
+    void (*on_audio_format_change)(void *opaque);
+    void *on_audio_format_change_opaque;
+};
+
+// (to free this, free the filter itself, mp_autoconvert.f)
+struct mp_autoconvert *mp_autoconvert_create(struct mp_filter *parent);
+
+// Add the imgfmt as allowed video image format, and error on non-video frames.
+// Each call adds to the list of allowed formats. Before the first call, all
+// formats are allowed (even non-video).
+// subfmt can be used to specify underlying surface formats for hardware formats,
+// otherwise must be 0.
+void mp_autoconvert_add_imgfmt(struct mp_autoconvert *c, int imgfmt, int subfmt);
+
+// Add the formats supported by the hwdec interops (or essentially refine them),
+// and trigger conversion if hw_subfmts mismatch. This is mostly a hack for
+// D3D11/ANGLE (which supports NV12 only).
+// Must be called mp_autoconvert_add_imgfmt(), and overrides them where formats
+// collide.
+struct mp_hwdec_devices;
+void mp_autoconvert_add_vo_hwdec_subfmts(struct mp_autoconvert *c,
+                                         struct mp_hwdec_devices *devs);
+
+// Add afmt (an AF_FORMAT_* value) as allowed audio format.
+// See mp_autoconvert_add_imgfmt() for other remarks.
+void mp_autoconvert_add_afmt(struct mp_autoconvert *c, int afmt);
+
+// Add allowed audio channel configuration.
+struct mp_chmap;
+void mp_autoconvert_add_chmap(struct mp_autoconvert *c, struct mp_chmap *chmap);
+
+// Add allowed audio sample rate.
+void mp_autoconvert_add_srate(struct mp_autoconvert *c, int rate);
+
+// Reset set of allowed formats back to initial state. (This does not flush
+// any frames or remove currently active filters, although to get reasonable
+// behavior, you need to readd all previously allowed formats, or reset the
+// filter.)
+void mp_autoconvert_clear(struct mp_autoconvert *c);
+
+// See mp_autoconvert.on_audio_format_change.
+void mp_autoconvert_format_change_continue(struct mp_autoconvert *c);
+
+// vf_d3d11vpp.c
+struct mp_filter *vf_d3d11_create_outconv(struct mp_filter *parent);
@@ -0,0 +1,723 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <stdbool.h>
+#include <math.h>
+#include <assert.h>
+
+#include <libavutil/buffer.h>
+#include <libavutil/rational.h>
+
+#include "config.h"
+#include "options/options.h"
+#include "common/msg.h"
+
+#include "osdep/timer.h"
+
+#include "demux/demux.h"
+#include "demux/packet.h"
+
+#include "common/codecs.h"
+#include "common/global.h"
+#include "common/recorder.h"
+
+#include "audio/aframe.h"
+#include "video/out/vo.h"
+#include "video/csputils.h"
+
+#include "demux/stheader.h"
+
+#include "f_decoder_wrapper.h"
+#include "f_demux_in.h"
+#include "filter_internal.h"
+
+struct priv {
+    struct mp_filter *f;
+    struct mp_log *log;
+    struct MPOpts *opts;
+
+    struct sh_stream *header;
+    struct mp_codec_params *codec;
+
+    struct mp_decoder *decoder;
+
+    // Demuxer output.
+    struct mp_pin *demux;
+
+    // Last PTS from decoder (set with each vd_driver->decode() call)
+    double codec_pts;
+    int num_codec_pts_problems;
+
+    // Last packet DTS from decoder (passed through from source packets)
+    double codec_dts;
+    int num_codec_dts_problems;
+
+    // PTS or DTS of packet first read
+    double first_packet_pdts;
+
+    // There was at least one packet with nonsense timestamps.
+    int has_broken_packet_pts; // <0: uninitialized, 0: no problems, 1: broken
+
+    int has_broken_decoded_pts;
+
+    int packets_without_output; // number packets sent without frame received
+
+    // Final PTS of previously decoded frame
+    double pts;
+
+    struct mp_image_params dec_format, last_format, fixed_format;
+
+    double start_pts;
+    double start, end;
+    struct demux_packet *new_segment;
+    struct mp_frame packet;
+
+    struct mp_frame decoded_coverart;
+    int coverart_returned; // 0: no, 1: coverart frame itself, 2: EOF returned
+
+    struct mp_decoder_wrapper public;
+};
+
+static void reset_decoder(struct priv *p)
+{
+    p->first_packet_pdts = MP_NOPTS_VALUE;
+    p->start_pts = MP_NOPTS_VALUE;
+    p->pts = MP_NOPTS_VALUE;
+    p->codec_pts = MP_NOPTS_VALUE;
+    p->codec_dts = MP_NOPTS_VALUE;
+    p->has_broken_decoded_pts = 0;
+    p->last_format = p->fixed_format = (struct mp_image_params){0};
+    p->public.dropped_frames = 0;
+    p->public.attempt_framedrops = 0;
+    p->public.pts_reset = false;
+    p->packets_without_output = 0;
+    mp_frame_unref(&p->packet);
+    talloc_free(p->new_segment);
+    p->new_segment = NULL;
+    p->start = p->end = MP_NOPTS_VALUE;
+    p->coverart_returned = 0;
+
+    if (p->decoder)
+        mp_filter_reset(p->decoder->f);
+}
+
+static void reset(struct mp_filter *f)
+{
+    struct priv *p = f->priv;
+
+    reset_decoder(p);
+}
+
+int mp_decoder_wrapper_control(struct mp_decoder_wrapper *d,
+                               enum dec_ctrl cmd, void *arg)
+{
+    struct priv *p = d->f->priv;
+    if (p->decoder && p->decoder->control)
+        return p->decoder->control(p->decoder->f, cmd, arg);
+    return CONTROL_UNKNOWN;
+}
+
+static void destroy(struct mp_filter *f)
+{
+    struct priv *p = f->priv;
+    if (p->decoder) {
+        MP_VERBOSE(f, "Uninit decoder.\n");
+        talloc_free(p->decoder->f);
+        p->decoder = NULL;
+    }
+    reset_decoder(p);
+    mp_frame_unref(&p->decoded_coverart);
+}
+
+struct mp_decoder_list *video_decoder_list(void)
+{
+    struct mp_decoder_list *list = talloc_zero(NULL, struct mp_decoder_list);
+    vd_lavc.add_decoders(list);
+    return list;
+}
+
+struct mp_decoder_list *audio_decoder_list(void)
+{
+    struct mp_decoder_list *list = talloc_zero(NULL, struct mp_decoder_list);
+    ad_lavc.add_decoders(list);
+    return list;
+}
+
+bool mp_decoder_wrapper_reinit(struct mp_decoder_wrapper *d)
+{
+    struct priv *p = d->f->priv;
+    struct MPOpts *opts = p->opts;
+
+    if (p->decoder)
+        talloc_free(p->decoder->f);
+    p->decoder = NULL;
+
+    reset_decoder(p);
+    p->has_broken_packet_pts = -10; // needs 10 packets to reach decision
+
+    const struct mp_decoder_fns *driver = NULL;
+    struct mp_decoder_list *list = NULL;
+    char *user_list = NULL;
+
+    if (p->codec->type == STREAM_VIDEO) {
+        driver = &vd_lavc;
+        user_list = opts->video_decoders;
+    } else if (p->codec->type == STREAM_AUDIO) {
+        driver = &ad_lavc;
+        user_list = opts->audio_decoders;
+
+        if (p->public.try_spdif && p->codec->codec) {
+            struct mp_decoder_list *spdif =
+                select_spdif_codec(p->codec->codec, opts->audio_spdif);
+            if (spdif->num_entries) {
+                driver = &ad_spdif;
+                list = spdif;
+            } else {
+                talloc_free(spdif);
+            }
+        }
+    }
+
+    if (!list) {
+        struct mp_decoder_list *full = talloc_zero(NULL, struct mp_decoder_list);
+        if (driver)
+            driver->add_decoders(full);
+        list = mp_select_decoders(p->log, full, p->codec->codec, user_list);
+        talloc_free(full);
+    }
+
+    mp_print_decoders(p->log, MSGL_V, "Codec list:", list);
+
+    for (int n = 0; n < list->num_entries; n++) {
+        struct mp_decoder_entry *sel = &list->entries[n];
+        MP_VERBOSE(p, "Opening decoder %s\n", sel->decoder);
+
+        p->decoder = driver->create(p->f, p->codec, sel->decoder);
+        if (p->decoder) {
+            p->public.decoder_desc =
+                talloc_asprintf(p, "%s (%s)", sel->decoder, sel->desc);
+            MP_VERBOSE(p, "Selected codec: %s\n", p->public.decoder_desc);
+            break;
+        }
+
+        MP_WARN(p, "Decoder init failed for %s\n", sel->decoder);
+    }
+
+    if (!p->decoder) {
+        MP_ERR(p, "Failed to initialize a decoder for codec '%s'.\n",
+               p->codec->codec ? p->codec->codec : "<?>");
+    }
+
+    talloc_free(list);
+    return !!p->decoder;
+}
+
+static bool is_valid_peak(float sig_peak)
+{
+    return !sig_peak || (sig_peak >= 1 && sig_peak <= 100);
+}
+
+static void fix_image_params(struct priv *p,
+                             struct mp_image_params *params)
+{
+    struct MPOpts *opts = p->opts;
+    struct mp_image_params m = *params;
+    struct mp_codec_params *c = p->codec;
+
+    MP_VERBOSE(p, "Decoder format: %s\n", mp_image_params_to_str(params));
+    p->dec_format = *params;
+
+    // While mp_image_params normally always have to have d_w/d_h set, the
+    // decoder signals unknown bitstream aspect ratio with both set to 0.
+    bool use_container = true;
+    if (opts->aspect_method == 1 && m.p_w > 0 && m.p_h > 0) {
+        MP_VERBOSE(p, "Using bitstream aspect ratio.\n");
+        use_container = false;
+    }
+
+    if (use_container && c->par_w > 0 && c->par_h) {
+        MP_VERBOSE(p, "Using container aspect ratio.\n");
+        m.p_w = c->par_w;
+        m.p_h = c->par_h;
+    }
+
+    if (opts->movie_aspect >= 0) {
+        MP_VERBOSE(p, "Forcing user-set aspect ratio.\n");
+        if (opts->movie_aspect == 0) {
+            m.p_w = m.p_h = 1;
+        } else {
+            AVRational a = av_d2q(opts->movie_aspect, INT_MAX);
+            mp_image_params_set_dsize(&m, a.num, a.den);
+        }
+    }
+
+    // Assume square pixels if no aspect ratio is set at all.
+    if (m.p_w <= 0 || m.p_h <= 0)
+        m.p_w = m.p_h = 1;
+
+    m.rotate = p->codec->rotate;
+    m.stereo3d = p->codec->stereo_mode;
+
+    if (opts->video_rotate < 0) {
+        m.rotate = 0;
+    } else {
+        m.rotate = (m.rotate + opts->video_rotate) % 360;
+    }
+
+    mp_colorspace_merge(&m.color, &c->color);
+
+    // Sanitize the HDR peak. Sadly necessary
+    if (!is_valid_peak(m.color.sig_peak)) {
+        MP_WARN(p, "Invalid HDR peak in stream: %f\n", m.color.sig_peak);
+        m.color.sig_peak = 0.0;
+    }
+
+    m.spherical = c->spherical;
+    if (m.spherical.type == MP_SPHERICAL_AUTO)
+        m.spherical.type = MP_SPHERICAL_NONE;
+
+    // Guess missing colorspace fields from metadata. This guarantees all
+    // fields are at least set to legal values afterwards.
+    mp_image_params_guess_csp(&m);
+
+    p->last_format = *params;
+    p->fixed_format = m;
+}
+
+static void process_video_frame(struct priv *p, struct mp_image *mpi)
+{
+    struct MPOpts *opts = p->opts;
+
+    // Note: the PTS is reordered, but the DTS is not. Both should be monotonic.
+    double pts = mpi->pts;
+    double dts = mpi->dts;
+
+    if (pts != MP_NOPTS_VALUE) {
+        if (pts < p->codec_pts)
+            p->num_codec_pts_problems++;
+        p->codec_pts = mpi->pts;
+    }
+
+    if (dts != MP_NOPTS_VALUE) {
+        if (dts <= p->codec_dts)
+            p->num_codec_dts_problems++;
+        p->codec_dts = mpi->dts;
+    }
+
+    if (p->has_broken_packet_pts < 0)
+        p->has_broken_packet_pts++;
+    if (p->num_codec_pts_problems)
+        p->has_broken_packet_pts = 1;
+
+    // If PTS is unset, or non-monotonic, fall back to DTS.
+    if ((p->num_codec_pts_problems > p->num_codec_dts_problems ||
+         pts == MP_NOPTS_VALUE) && dts != MP_NOPTS_VALUE)
+        pts = dts;
+
+    if (!opts->correct_pts || pts == MP_NOPTS_VALUE) {
+        double fps = p->public.fps > 0 ? p->public.fps : 25;
+
+        if (opts->correct_pts) {
+            if (p->has_broken_decoded_pts <= 1) {
+                MP_WARN(p, "No video PTS! Making something up. Using "
+                        "%f FPS.\n", fps);
+                if (p->has_broken_decoded_pts == 1)
+                    MP_WARN(p, "Ignoring further missing PTS warnings.\n");
+                p->has_broken_decoded_pts++;
+            }
+        }
+
+        double frame_time = 1.0f / fps;
+        double base = p->first_packet_pdts;
+        pts = p->pts;
+        if (pts == MP_NOPTS_VALUE) {
+            pts = base == MP_NOPTS_VALUE ? 0 : base;
+        } else {
+            pts += frame_time;
+        }
+    }
+
+    if (!mp_image_params_equal(&p->last_format, &mpi->params))
+        fix_image_params(p, &mpi->params);
+
+    mpi->params = p->fixed_format;
+    mpi->nominal_fps = p->public.fps;
+
+    mpi->pts = pts;
+    p->pts = pts;
+
+    // Compensate for incorrectly using mpeg-style DTS for avi timestamps.
+    if (p->decoder && p->decoder->control && p->codec->avi_dts &&
+        opts->correct_pts && mpi->pts != MP_NOPTS_VALUE && p->public.fps > 0)
+    {
+        int delay = -1;
+        p->decoder->control(p->decoder->f, VDCTRL_GET_BFRAMES, &delay);
+        mpi->pts -= MPMAX(delay, 0) / p->public.fps;
+    }
+
+    struct demux_packet *ccpkt = new_demux_packet_from_buf(mpi->a53_cc);
+    if (ccpkt) {
+        av_buffer_unref(&mpi->a53_cc);
+        ccpkt->pts = mpi->pts;
+        ccpkt->dts = mpi->dts;
+        demuxer_feed_caption(p->header, ccpkt);
+    }
+
+    if (mpi->pts == MP_NOPTS_VALUE || mpi->pts >= p->start_pts)
+        p->start_pts = MP_NOPTS_VALUE;
+}
+
+void mp_decoder_wrapper_reset_params(struct mp_decoder_wrapper *d)
+{
+    struct priv *p = d->f->priv;
+    p->last_format = (struct mp_image_params){0};
+}
+
+void mp_decoder_wrapper_get_video_dec_params(struct mp_decoder_wrapper *d,
+                                             struct mp_image_params *m)
+{
+    struct priv *p = d->f->priv;
+    *m = p->dec_format;
+}
+
+static void process_audio_frame(struct priv *p, struct mp_aframe *aframe)
+{
+    double frame_pts = mp_aframe_get_pts(aframe);
+    if (frame_pts != MP_NOPTS_VALUE) {
+        if (p->pts != MP_NOPTS_VALUE)
+            MP_STATS(p, "value %f audio-pts-err", p->pts - frame_pts);
+
+        double diff = fabs(p->pts - frame_pts);
+
+        // 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 (p->pts != MP_NOPTS_VALUE && diff > 0.1) {
+            MP_WARN(p, "Invalid audio PTS: %f -> %f\n", p->pts, frame_pts);
+            if (diff >= 5)
+                p->public.pts_reset = true;
+        }
+
+        // Keep the interpolated timestamp if it doesn't deviate more
+        // than 1 ms from the real one. (MKV rounded timestamps.)
+        if (p->pts == MP_NOPTS_VALUE || diff > 0.001)
+            p->pts = frame_pts;
+    }
+
+    if (p->pts == MP_NOPTS_VALUE && p->header->missing_timestamps)
+        p->pts = 0;
+
+    mp_aframe_set_pts(aframe, p->pts);
+
+    if (p->pts != MP_NOPTS_VALUE)
+        p->pts += mp_aframe_duration(aframe);
+}
+
+
+// Frames before the start timestamp can be dropped. (Used for hr-seek.)
+void mp_decoder_wrapper_set_start_pts(struct mp_decoder_wrapper *d, double pts)
+{
+    struct priv *p = d->f->priv;
+    p->start_pts = pts;
+}
+
+static bool is_new_segment(struct priv *p, struct mp_frame frame)
+{
+    if (frame.type != MP_FRAME_PACKET)
+        return false;
+    struct demux_packet *pkt = frame.data;
+    return pkt->segmented && (pkt->start != p->start || pkt->end != p->end ||
+                              pkt->codec != p->codec);
+}
+
+static void feed_packet(struct priv *p)
+{
+    if (!p->decoder || !mp_pin_in_needs_data(p->decoder->f->pins[0]))
+        return;
+
+    if (!p->packet.type && !p->new_segment) {
+        p->packet = mp_pin_out_read(p->demux);
+        if (!p->packet.type)
+            return;
+        if (p->packet.type != MP_FRAME_EOF && p->packet.type != MP_FRAME_PACKET) {
+            MP_ERR(p, "invalid frame type from demuxer\n");
+            mp_frame_unref(&p->packet);
+            mp_filter_internal_mark_failed(p->f);
+            return;
+        }
+    }
+
+    // Flush current data if the packet is a new segment.
+    if (is_new_segment(p, p->packet)) {
+        assert(!p->new_segment);
+        p->new_segment = p->packet.data;
+        p->packet = MP_EOF_FRAME;
+    }
+
+    assert(p->packet.type == MP_FRAME_PACKET || p->packet.type == MP_FRAME_EOF);
+    struct demux_packet *packet = p->packet.data;
+
+    // For video framedropping, including parts of the hr-seek logic.
+    if (p->decoder->control) {
+        double start_pts = p->start_pts;
+        if (p->start != MP_NOPTS_VALUE && (start_pts == MP_NOPTS_VALUE ||
+                                           p->start > start_pts))
+            start_pts = p->start;
+
+        int framedrop_type = 0;
+
+        if (p->public.attempt_framedrops)
+            framedrop_type = 1;
+
+        if (start_pts != MP_NOPTS_VALUE && packet &&
+            packet->pts < start_pts - .005 && !p->has_broken_packet_pts)
+            framedrop_type = 2;
+
+        p->decoder->control(p->decoder->f, VDCTRL_SET_FRAMEDROP, &framedrop_type);
+    }
+
+    if (p->public.recorder_sink)
+        mp_recorder_feed_packet(p->public.recorder_sink, packet);
+
+    double pkt_pts = packet ? packet->pts : MP_NOPTS_VALUE;
+    double pkt_dts = packet ? packet->dts : MP_NOPTS_VALUE;
+
+    if (pkt_pts == MP_NOPTS_VALUE)
+        p->has_broken_packet_pts = 1;
+
+    if (packet && packet->dts == MP_NOPTS_VALUE && !p->codec->avi_dts)
+        packet->dts = packet->pts;
+
+    double pkt_pdts = pkt_pts == MP_NOPTS_VALUE ? pkt_dts : pkt_pts;
+    if (p->first_packet_pdts == MP_NOPTS_VALUE)
+        p->first_packet_pdts = pkt_pdts;
+
+    mp_pin_in_write(p->decoder->f->pins[0], p->packet);
+    p->packet = MP_NO_FRAME;
+
+    p->packets_without_output += 1;
+}
+
+// Return true if the current frame is outside segment range.
+static bool process_decoded_frame(struct priv *p, struct mp_frame *frame)
+{
+    if (frame->type == MP_FRAME_EOF) {
+        // if we were just draining current segment, don't propagate EOF
+        if (p->new_segment)
+            mp_frame_unref(frame);
+        return true;
+    }
+
+    bool segment_ended = false;
+
+    if (frame->type == MP_FRAME_VIDEO) {
+        struct mp_image *mpi = frame->data;
+
+        process_video_frame(p, mpi);
+
+        if (mpi->pts != MP_NOPTS_VALUE) {
+            double vpts = mpi->pts;
+            segment_ended = p->end != MP_NOPTS_VALUE && vpts >= p->end;
+            if ((p->start != MP_NOPTS_VALUE && vpts < p->start) || segment_ended)
+                mp_frame_unref(frame);
+        }
+    } else if (frame->type == MP_FRAME_AUDIO) {
+        struct mp_aframe *aframe = frame->data;
+
+        process_audio_frame(p, aframe);
+
+        mp_aframe_clip_timestamps(aframe, p->start, p->end);
+        double pts = mp_aframe_get_pts(aframe);
+        if (pts != MP_NOPTS_VALUE && p->start != MP_NOPTS_VALUE)
+            segment_ended = pts >= p->end;
+        if (mp_aframe_get_size(aframe) == 0)
+            mp_frame_unref(frame);
+    } else {
+        MP_ERR(p, "unknown frame type from decoder\n");
+    }
+
+    return segment_ended;
+}
+
+static void read_frame(struct priv *p)
+{
+    struct mp_pin *pin = p->f->ppins[0];
+
+    if (!p->decoder || !mp_pin_in_needs_data(pin))
+        return;
+
+    if (p->decoded_coverart.type) {
+        if (p->coverart_returned == 0) {
+            mp_pin_in_write(pin, mp_frame_ref(p->decoded_coverart));
+            p->coverart_returned = 1;
+        } else if (p->coverart_returned == 1) {
+            mp_pin_in_write(pin, MP_EOF_FRAME);
+            p->coverart_returned = 2;
+        }
+        return;
+    }
+
+    struct mp_frame frame = mp_pin_out_read(p->decoder->f->pins[1]);
+    if (!frame.type)
+        return;
+
+    if (p->public.attempt_framedrops) {
+        int dropped = MPMAX(0, p->packets_without_output - 1);
+        p->public.attempt_framedrops =
+            MPMAX(0, p->public.attempt_framedrops - dropped);
+        p->public.dropped_frames += dropped;
+    }
+    p->packets_without_output = 0;
+
+    bool segment_ended = process_decoded_frame(p, &frame);
+
+    // If there's a new segment, start it as soon as we're drained/finished.
+    if (segment_ended && p->new_segment) {
+        struct demux_packet *new_segment = p->new_segment;
+        p->new_segment = NULL;
+
+        reset_decoder(p);
+
+        if (p->codec != new_segment->codec) {
+            p->codec = new_segment->codec;
+            if (!mp_decoder_wrapper_reinit(&p->public))
+                mp_filter_internal_mark_failed(p->f);
+        }
+
+        p->start = new_segment->start;
+        p->end = new_segment->end;
+
+        p->packet = MAKE_FRAME(MP_FRAME_PACKET, new_segment);
+        mp_filter_internal_mark_progress(p->f);
+    }
+
+    if (!frame.type) {
+        mp_filter_internal_mark_progress(p->f); // make it retry
+        return;
+    }
+
+    if (p->header->attached_picture && frame.type == MP_FRAME_VIDEO) {
+        p->decoded_coverart = mp_frame_ref(frame);
+        p->coverart_returned = 1;
+    }
+
+    mp_pin_in_write(pin, frame);
+}
+
+static void process(struct mp_filter *f)
+{
+    struct priv *p = f->priv;
+
+    feed_packet(p);
+    read_frame(p);
+}
+
+static const struct mp_filter_info decode_wrapper_filter = {
+    .name = "decode",
+    .priv_size = sizeof(struct priv),
+    .process = process,
+    .reset = reset,
+    .destroy = destroy,
+};
+
+struct mp_decoder_wrapper *mp_decoder_wrapper_create(struct mp_filter *parent,
+                                                     struct sh_stream *src)
+{
+    struct mp_filter *f = mp_filter_create(parent, &decode_wrapper_filter);
+    if (!f)
+        return NULL;
+
+    struct priv *p = f->priv;
+    struct mp_decoder_wrapper *w = &p->public;
+    p->opts = f->global->opts;
+    p->log = f->log;
+    p->f = f;
+    p->header = src;
+    p->codec = p->header->codec;
+    w->f = f;
+
+    mp_filter_add_pin(f, MP_PIN_OUT, "out");
+
+    if (p->header->type == STREAM_VIDEO) {
+        p->log = f->log = mp_log_new(f, parent->log, "!vd");
+
+        p->public.fps = src->codec->fps;
+
+        MP_VERBOSE(p, "Container reported FPS: %f\n", p->public.fps);
+
+        if (p->opts->force_fps) {
+            p->public.fps = p->opts->force_fps;
+            MP_INFO(p, "FPS forced to %5.3f.\n", p->public.fps);
+            MP_INFO(p, "Use --no-correct-pts to force FPS based timing.\n");
+        }
+    } else if (p->header->type == STREAM_AUDIO) {
+        p->log = f->log = mp_log_new(f, parent->log, "!ad");
+    }
+
+    struct mp_filter *demux = mp_demux_in_create(f, p->header);
+    if (!demux)
+        goto error;
+    p->demux = demux->pins[0];
+
+    return w;
+error:
+    talloc_free(f);
+    return NULL;
+}
+
+void lavc_process(struct mp_filter *f, bool *eof_flag,
+                  bool (*send)(struct mp_filter *f, struct demux_packet *pkt),
+                  bool (*receive)(struct mp_filter *f, struct mp_frame *res))
+{
+    if (!mp_pin_in_needs_data(f->ppins[1]))
+        return;
+
+    struct mp_frame frame = {0};
+    if (!receive(f, &frame)) {
+        if (!*eof_flag)
+            mp_pin_in_write(f->ppins[1], MP_EOF_FRAME);
+        *eof_flag = true;
+    } else if (frame.type) {
+        *eof_flag = false;
+        mp_pin_in_write(f->ppins[1], frame);
+    } else {
+        // Need to feed a packet.
+        frame = mp_pin_out_read(f->ppins[0]);
+        struct demux_packet *pkt = NULL;
+        if (frame.type == MP_FRAME_PACKET) {
+            pkt = frame.data;
+        } else if (frame.type != MP_FRAME_EOF) {
+            if (frame.type) {
+                MP_ERR(f, "unexpected frame type\n");
+                mp_frame_unref(&frame);
+                mp_filter_internal_mark_failed(f);
+            }
+            return;
+        }
+        if (!send(f, pkt)) {
+            // Should never happen, but can happen with broken decoders.
+            MP_WARN(f, "could not consume packet\n");
+            mp_pin_out_unread(f->ppins[0], frame);
+            mp_filter_wakeup(f);
+            return;
+        }
+        talloc_free(pkt);
+        mp_filter_internal_mark_progress(f);
+    }
+}
@@ -1 +1 @@
 .28.0
 .29.1