[FFmpeg-devel] [PATCH] doc/filters: drop duplicated description of resampler options in aresample docs

[Stefano Sabatini]

Avoid duplication, and point to the complete description of options.
Also provide a more complete specification of the supported syntax.
---
 doc/filters.texi |   68 +++++++-----------------------------------------------
 1 file changed, 8 insertions(+), 60 deletions(-)

diff --git a/doc/filters.texi b/doc/filters.texi
index 765a9b4..34b216d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -410,71 +410,19 @@ Pad the end of a audio stream with silence, this can be used together with
 @anchor{aresample}
 @section aresample
 
-Resample the input audio to the specified parameters. If none are specified
-then the filter will automatically convert between its input
-and output.
+Resample the input audio to the specified parameters, using the
+libswresample library. If none are specified then the filter will
+automatically convert between its input and output.
 
 This filter is also able to stretch/squeeze the audio data to make it match
 the timestamps or to inject silence / cut out audio to make it match the
 timestamps, do a combination of both or do neither.
 
-The filter accepts the following named parameters:
- at table @option
-
- at item min_comp
-Minimum difference between timestamps and audio data (in seconds) to trigger
-stretching/squeezing/filling or trimming of the data to make it match the
-timestamps. The default is that stretching/squeezing/filling and
-trimming is disabled (min_comp = infinite).
-
- at item min_hard_comp
-Minimum difference between timestamps and audio data (in seconds) to trigger
-adding/dropping samples to make it match the timestamps.
-This option effectively is a threshold to select between hard (trim/fill) and
-soft (squeeze/stretch) compensation. Note that all compensation is by default
-disabled through min_comp.
-The default is 0.1 seconds.
-
- at item max_soft_comp
-Maximum stretch/squeeze factor.
-Default value 0.
-
- at item tsf, internal_sample_fmt
-Internal sampling format.
-Default is automatic selection
-
- at item clev, center_mix_level
-center mix level, for rematrixing
-Default is 3.0dB
-
- at item slev, surround_mix_level
-surround mix level, for rematrixing
-Default is 3.0dB
-
- at item rmvol, rematrix_volume
-rematrix volume
-Default is 1.0
-
- at item lfe_mix_level
-Low frequency effects mix level.
-Default is 0
-
- at item matrix_encoding
-matrixed stereo encoding
- at table @option
- at item none
-No matrixed stereo encoding
-
- at item dolby
-Dolby matrixed stereo encoding
-
- at item dplii
-Dolby Pro Logic II matrixed stereo encoding
- at end table
-
-Default value is @code{none}.
-
- at end table
+The filter accepts the syntax
+[@var{sample_rate}:]@var{resampler_options}, where @var{sample_rate}
+express a sample rate and @var{resampler_options} is a list of
+ at var{key}=@var{value} pairs, separated by ":". See the
+ffmpeg-resampler manual for the complete list of supported options.
 
 For example, to resample the input audio to 44100Hz:
 @example
-- 
1.7.9.5