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author | wm4 <wm4@nowhere> | 2013-10-22 01:20:43 +0200 |
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committer | wm4 <wm4@nowhere> | 2013-10-23 10:04:12 +0200 |
commit | e60b8f181dec744af25c3a52fb88f600cd1b63ea (patch) | |
tree | 31ab322709b207dea61d86600d5ce177b318238e /DOCS/man | |
parent | 33707c6d6394592f528fb200af2dd6e104fc6df6 (diff) | |
download | mpv-e60b8f181dec744af25c3a52fb88f600cd1b63ea.tar.bz2 mpv-e60b8f181dec744af25c3a52fb88f600cd1b63ea.tar.xz |
audio/filter: split af_format into separate filters, rename af_force
af_format is the old audio conversion filter. It could do all possible
conversions supported by the audio chain. However, ever since the
addition of af_lavrresample, most conversions are done by
libav/swresample, and af_format is used as fallback.
Separate out the fallback cases and remove af_format. af_convert24 does
24 bit <-> 32 bit conversions, while af_convertsignendian does sign and
endian conversions. Maybe the way the conversions are split sounds a bit
odd. But the former changes the size of the audio data, while the latter
is fully in-place, so there's at least different buffer management.
This requires a quite complicated algorithm to make sure all these
"partial" conversion filters can actually get from one format to
another. E.g. s24le->s32be always requires convertsignendian and
convert24, but af.c has no idea what the intermediate format should
be. So I added a graph search (trying every possible format and
filter) to determine required format and filter. When I wrote this,
it seemed this was still better than messing everything into
af_lavrresample, but maybe this is overkill and I'll change my
opinion. For now, it seems nice to get rid of af_format though.
The AC3->IEC61937 conversion isn't supported anymore, but I don't think
this is needed anywhere. Most AOs test all formats explicitly, or use
the AF_FORMAT_IS_IEC61937() macro (which includes AC3).
One positive consequence of this change is that conversions always
include dithering (done by libav/swresample), instead of possibly going
through af_format, which doesn't do anything fancy.
Rename af_force to af_format. It's essentially compatible with command
line uses of af_format. We retain a compatibility alias for af_force.
Diffstat (limited to 'DOCS/man')
-rw-r--r-- | DOCS/man/en/af.rst | 45 |
1 files changed, 27 insertions, 18 deletions
diff --git a/DOCS/man/en/af.rst b/DOCS/man/en/af.rst index 8844077781..3513b9dd0c 100644 --- a/DOCS/man/en/af.rst +++ b/DOCS/man/en/af.rst @@ -208,7 +208,7 @@ Available filters are: 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`` +``format=format:srate: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. @@ -218,15 +218,22 @@ Available filters are: 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. + ``<format>`` + Force conversion to this format. Use ``--af=format=format=help`` to get + a list of valid formats. The general form is 'sbe', where 's' denotes + the sign (either 's' for signed or 'u' for unsigned), 'b' denotes the + number of bits per sample (16, 24 or 32) and 'e' denotes the + endianness ('le' means little-endian, 'be' big-endian and 'ne' the + endianness of the computer mpv is running on). Valid values (amongst + others) are: 's16le', 'u32be' and 'u24ne'. Exceptions to this rule that + are also valid format specifiers: u8, s8, floatle, floatbe, floatne, + mpeg2, and ac3. - ``<in-srate>`` + ``<srate>`` Force conversion to a specific sample rate. The rate is an integer, 48000 for example. - ``<in-channels>`` + ``<channels>`` Force mixing to a specific channel layout. See ``--channels`` option for possible values. @@ -236,19 +243,21 @@ Available filters are: ``<out-channels>`` -``format[=format]`` - Convert between different sample formats. Automatically enabled when - needed by the audio output or another filter. See also ``--format``. + See also ``--format``, ``--srate``, and ``--channels`` for related options. + Keep in mind that ``--channels`` does not actually force the number of + channels in most cases, while this filter can do this. - ``<format>`` - Sets the desired format. The general form is 'sbe', where 's' denotes - the sign (either 's' for signed or 'u' for unsigned), 'b' denotes the - number of bits per sample (16, 24 or 32) and 'e' denotes the - endianness ('le' means little-endian, 'be' big-endian and 'ne' the - endianness of the computer mpv is running on). Valid values (amongst - others) are: 's16le', 'u32be' and 'u24ne'. Exceptions to this rule that - are also valid format specifiers: u8, s8, floatle, floatbe, floatne, - mpeg2, and ac3. + *NOTE*: this filter used to be named ``force``. Also, unlike the old + ``format`` filter, this does not do any actual conversion anymore. + Conversion is done by other, automatically inserted filters. + +``convert24`` + Filter for internal use only. Converts between 24-bit and 32-bit sample + formats. + +``convertsignendian`` + Filter for internal use only. Converts between signed/unsigned formats + and formats with different endian. ``volume[=v[:sc[:fast]]]`` Implements software volume control. Use this filter with caution since it |