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Diffstat (limited to 'DOCS/man/en/af.rst')
| -rw-r--r-- | DOCS/man/en/af.rst | 58 |
1 files changed, 49 insertions, 9 deletions
diff --git a/DOCS/man/en/af.rst b/DOCS/man/en/af.rst index 2eaeabed0c..08e7853990 100644 --- a/DOCS/man/en/af.rst +++ b/DOCS/man/en/af.rst @@ -30,15 +30,15 @@ filter list. Available filters are: lavrresample[=option1:option2:...] - Changes the sample rate of the audio stream to an integer <srate> in Hz. - Can be used if you have a fixed frequency sound card or if you are stuck - with an old sound card that is only capable of max 44.1kHz. + This filter uses libavresample (or libswresample, depending on the build) + to change sample rate, sample format, or channel layout of the audio stream. + This filter is automatically enabled if the audio output doesn't support + the audio configuration of the file being played. - This filter is automatically enabled if necessary. It only supports the - 16-bit integer native-endian format. + It supports only the following sample formats: u8, s16ne, s32ne, floatne. srate=<srate> - the output sample rate (defaut: 44100) + the output sample rate length=<length> length of the filter with respect to the lower sampling rate (default: 16) @@ -50,6 +50,11 @@ lavrresample[=option1:option2:...] linear if set then filters will be linearly interpolated between polyphase entries (default: no) + no-detach + don't detach if input and output audio format/rate/channels are the + same. You should add this option if you specify additional parameters, + as automatically inserted lavrresample instances will use the + default settings. lavcac3enc[=tospdif[:bitrate[:minchn]]] Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports @@ -205,6 +210,34 @@ channels=nch[:nr:from1:to1:from2:to2:from3:to3:...] Would change the number of channels to 6 and set up 4 routes that copy channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence. +force=in-format:in-srate:in-channels:out-format:out-srate:out-channels + Force a specific audio format/configuration without actually changing the + audio data. Keep in mind that the filter system might auto-insert actual + conversion filters before or after this filter if needed. + + All parameters are optional. The ``in-`` variants restrict what the filter + accepts as input. The ``out-`` variants change the audio format, without + actually doing a conversion. The data will be 'reinterpreted' by the + filters or audio outputs following this filter. + + <in-format> + Force conversion to this format. See ``format`` filter for valid audio + format values. + + <in-srate> + Force conversion to a specific sample rate. The rate is an integer, + 48000 for example. + + <in-channels> + Force mixing to a specific channel layout. See ``--channels`` option + for possible values. + + <out-format> + + <out-srate> + + <out-channels> + format[=format] Convert between different sample formats. Automatically enabled when needed by the sound card or another filter. See also ``--format``. @@ -219,7 +252,7 @@ format[=format] rule that are also valid format specifiers: u8, s8, floatle, floatbe, floatne, mpeg2, and ac3. -volume[=v[:sc]] +volume[=v[:sc[:fast]]] Implements software volume control. Use this filter with caution since it can reduce the signal to noise ratio of the sound. In most cases it is best to set the level for the PCM sound to max, leave this filter out and @@ -233,8 +266,7 @@ volume[=v[:sc]] This filter has a second feature: It measures the overall maximum sound level and prints out that level when mpv exits. This feature currently - only works with floating-point data, use e.g. ``--af-adv=force=5``, or use - ``--af=stats``. + only works with floating-point data. *NOTE*: This filter is not reentrant and can therefore only be enabled once for every audio stream. @@ -250,6 +282,9 @@ volume[=v[:sc]] *WARNING*: This feature creates distortion and should be considered a last resort. + <fast> + Force S16 sample format if set to 1. Lower quality, but might be faster + in some situations. *EXAMPLE*: @@ -286,6 +321,11 @@ pan=n[:L00:L01:L02:...L10:L11:L12:...Ln0:Ln1:Ln2:...] channels 0 and 1 into output channel 2 (which could be sent to a subwoofer for example). + *NOTE*: if you just want to force remixing to a certain output channel + layout, it's easier to use the ``force`` filter. For example, + ``mpv '--af=force=channels=5.1' '--channels=5.1'`` would always force + remixing audio to 5.1 and output it like this. + sub[=fc:ch] Adds a subwoofer channel to the audio stream. The audio data used for creating the subwoofer channel is an average of the sound in channel 0 and |