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https://github.com/mpv-player/mpv
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c8b8fe9981
This filter wasn't referenced anywhere and thus was dead code. It should have been in the audio filter list in user_filters.c. This was intended as compatibility wrapper (to avoid breaking old command lines and config files), and has no real use. Apparently I forgot to add it to the filter list (did I even test this shit?), and so it was rotting around for 1.5 years doing nothing (just like myself). Note that users can just use the libavfilter provided filter to force resampling, just that it has a different name and different options. There's also af_format to force inserting auto conversion through the internal f_swsresample filter.
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223 lines
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AUDIO FILTERS
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=============
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Audio filters allow you to modify the audio stream and its properties. The
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syntax is:
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``--af=...``
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Setup a chain of audio filters. See ``--vf`` (`VIDEO FILTERS`_) for the
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full syntax.
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.. note::
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To get a full list of available audio filters, see ``--af=help``.
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Also, keep in mind that most actual filters are available via the ``lavfi``
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wrapper, which gives you access to most of libavfilter's filters. This
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includes all filters that have been ported from MPlayer to libavfilter.
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The ``--vf`` description describes how libavfilter can be used and how to
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workaround deprecated mpv filters.
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See ``--vf`` group of options for info on how ``--af-defaults``, ``--af-add``,
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``--af-pre``, ``--af-del``, ``--af-clr``, and possibly others work.
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Available filters are:
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``lavcac3enc[=options]``
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Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports
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16-bit native-endian input format, maximum 6 channels. The output is
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big-endian when outputting a raw AC-3 stream, native-endian when
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outputting to S/PDIF. If the input sample rate is not 48 kHz, 44.1 kHz or
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32 kHz, it will be resampled to 48 kHz.
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``tospdif=<yes|no>``
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Output raw AC-3 stream if ``no``, output to S/PDIF for
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pass-through if ``yes`` (default).
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``bitrate=<rate>``
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The bitrate use for the AC-3 stream. Set it to 384 to get 384 kbps.
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The default is 640. Some receivers might not be able to handle this.
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Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128,
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160, 192, 224, 256, 320, 384, 448, 512, 576, 640.
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The special value ``auto`` selects a default bitrate based on the
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input channel number:
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:1ch: 96
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:2ch: 192
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:3ch: 224
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:4ch: 384
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:5ch: 448
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:6ch: 448
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``minch=<n>``
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If the input channel number is less than ``<minch>``, the filter will
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detach itself (default: 3).
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``encoder=<name>``
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Select the libavcodec encoder used. Currently, this should be an AC-3
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encoder, and using another codec will fail horribly.
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``format=format:srate:channels:out-srate:out-channels``
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Does not do any format conversion itself. Rather, it may cause the
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filter system to insert necessary conversion filters before or after this
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filter if needed. It is primarily useful for controlling the audio format
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going into other filters. To specify the format for audio output, see
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``--audio-format``, ``--audio-samplerate``, and ``--audio-channels``. This
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filter is able to force a particular format, whereas ``--audio-*``
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may be overridden by the ao based on output compatibility.
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All parameters are optional. The first 3 parameters restrict what the filter
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accepts as input. They will therefore cause conversion filters to be
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inserted before this one. The ``out-`` parameters tell the filters or audio
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outputs following this filter how to interpret the data without actually
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doing a conversion. Setting these will probably just break things unless you
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really know you want this for some reason, such as testing or dealing with
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broken media.
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``<format>``
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Force conversion to this format. Use ``--af=format=format=help`` to get
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a list of valid formats.
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``<srate>``
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Force conversion to a specific sample rate. The rate is an integer,
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48000 for example.
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``<channels>``
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Force mixing to a specific channel layout. See ``--audio-channels`` option
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for possible values.
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``<out-srate>``
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``<out-channels>``
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*NOTE*: this filter used to be named ``force``. The old ``format`` filter
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used to do conversion itself, unlike this one which lets the filter system
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handle the conversion.
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``scaletempo[=option1:option2:...]``
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Scales audio tempo without altering pitch, optionally synced to playback
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speed (default).
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This works by playing 'stride' ms of audio at normal speed then consuming
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'stride*scale' ms of input audio. It pieces the strides together by
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blending 'overlap'% of stride with audio following the previous stride. It
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optionally performs a short statistical analysis on the next 'search' ms
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of audio to determine the best overlap position.
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``scale=<amount>``
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Nominal amount to scale tempo. Scales this amount in addition to
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speed. (default: 1.0)
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``stride=<amount>``
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Length in milliseconds to output each stride. Too high of a value will
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cause noticeable skips at high scale amounts and an echo at low scale
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amounts. Very low values will alter pitch. Increasing improves
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performance. (default: 60)
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``overlap=<percent>``
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Percentage of stride to overlap. Decreasing improves performance.
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(default: .20)
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``search=<amount>``
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Length in milliseconds to search for best overlap position. Decreasing
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improves performance greatly. On slow systems, you will probably want
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to set this very low. (default: 14)
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``speed=<tempo|pitch|both|none>``
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Set response to speed change.
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tempo
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Scale tempo in sync with speed (default).
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pitch
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Reverses effect of filter. Scales pitch without altering tempo.
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Add this to your ``input.conf`` to step by musical semi-tones::
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[ multiply speed 0.9438743126816935
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] multiply speed 1.059463094352953
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.. warning::
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Loses sync with video.
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both
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Scale both tempo and pitch.
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none
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Ignore speed changes.
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.. admonition:: Examples
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``mpv --af=scaletempo --speed=1.2 media.ogg``
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Would play media at 1.2x normal speed, with audio at normal
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pitch. Changing playback speed would change audio tempo to match.
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``mpv --af=scaletempo=scale=1.2:speed=none --speed=1.2 media.ogg``
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Would play media at 1.2x normal speed, with audio at normal
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pitch, but changing playback speed would have no effect on audio
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tempo.
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``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
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Would tweak the quality and performance parameters.
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``mpv --af=scaletempo=scale=1.2:speed=pitch audio.ogg``
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Would play media at 1.2x normal speed, with audio at normal pitch.
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Changing playback speed would change pitch, leaving audio tempo at
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1.2x.
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``rubberband``
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High quality pitch correction with librubberband. This can be used in place
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of ``scaletempo``, and will be used to adjust audio pitch when playing
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at speed different from normal. It can also be used to adjust audio pitch
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without changing playback speed.
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``<pitch-scale>``
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Sets the pitch scaling factor. Frequencies are multiplied by this value.
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This filter has a number of additional sub-options. You can list them with
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``mpv --af=rubberband=help``. This will also show the default values
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for each option. The options are not documented here, because they are
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merely passed to librubberband. Look at the librubberband documentation
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to learn what each option does:
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http://breakfastquay.com/rubberband/code-doc/classRubberBand_1_1RubberBandStretcher.html
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(The mapping of the mpv rubberband filter sub-option names and values to
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those of librubberband follows a simple pattern: ``"Option" + Name + Value``.)
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This filter supports the following ``af-command`` commands:
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``set-pitch``
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Set the ``<pitch-scale>`` argument dynamically. This can be used to
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change the playback pitch at runtime. Note that speed is controlled
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using the standard ``speed`` property, not ``af-command``.
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``multiply-pitch <factor>``
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Multiply the current value of ``<pitch-scale>`` dynamically. For
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example: 0.5 to go down by an octave, 1.5 to go up by a perfect fifth.
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If you want to go up or down by semi-tones, use 1.059463094352953 and
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0.9438743126816935
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``lavfi=graph``
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Filter audio using FFmpeg's libavfilter.
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``<graph>``
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Libavfilter graph. See ``lavfi`` video filter for details - the graph
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syntax is the same.
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.. warning::
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Don't forget to quote libavfilter graphs as described in the lavfi
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video filter section.
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``o=<string>``
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AVOptions.
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``fix-pts=<yes|no>``
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Determine PTS based on sample count (default: no). If this is enabled,
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the player won't rely on libavfilter passing through PTS accurately.
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Instead, it pass a sample count as PTS to libavfilter, and compute the
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PTS used by mpv based on that and the input PTS. This helps with filters
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which output a recomputed PTS instead of the original PTS (including
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filters which require the PTS to start at 0). mpv normally expects
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filters to not touch the PTS (or only to the extent of changing frame
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boundaries), so this is not the default, but it will be needed to use
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broken filters. In practice, these broken filters will either cause slow
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A/V desync over time (with some files), or break playback completely if
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you seek or start playback from the middle of a file.
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