doc: move ffmpeg-resampler.texi content to separated file

This should simplify inclusion in monolithic tool manuals.

2 files changed, 223 insertions, 222 deletions

diff --git a/doc/ffmpeg-resampler.texi b/doc/ffmpeg-resampler.texi
index 525907a79f..da3d033f0e 100644
--- a/doc/ffmpeg-resampler.texi
+++ b/doc/ffmpeg-resampler.texi
@@ -19,228 +19,7 @@ and convert audio format and packing layout.
 
 @c man end DESCRIPTION
 
-@chapter Resampler Options
-@c man begin RESAMPLER OPTIONS
-
-The audio resampler supports the following named options.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, @var{option}=@var{value} for the aresample filter,
-by setting the value explicitly in the
-@code{SwrContext} options or using the @file{libavutil/opt.h} API for
-programmatic use.
-
-@table @option
-
-@item ich, in_channel_count
-Set the number of input channels. Default value is 0. Setting this
-value is not mandatory if the corresponding channel layout
-@option{in_channel_layout} is set.
-
-@item och, out_channel_count
-Set the number of output channels. Default value is 0. Setting this
-value is not mandatory if the corresponding channel layout
-@option{out_channel_layout} is set.
-
-@item uch, used_channel_count
-Set the number of used input channels. Default value is 0. This option is
-only used for special remapping.
-
-@item isr, in_sample_rate
-Set the input sample rate. Default value is 0.
-
-@item osr, out_sample_rate
-Set the output sample rate. Default value is 0.
-
-@item isf, in_sample_fmt
-Specify the input sample format. It is set by default to @code{none}.
-
-@item osf, out_sample_fmt
-Specify the output sample format. It is set by default to @code{none}.
-
-@item tsf, internal_sample_fmt
-Set the internal sample format. Default value is @code{none}.
-This will automatically be chosen when it is not explicitly set.
-
-@item icl, in_channel_layout
-Set the input channel layout.
-
-@item ocl, out_channel_layout
-Set the output channel layout.
-
-@item clev, center_mix_level
-Set the center mix level. It is a value expressed in deciBel, and must be
-in the interval [-32,32].
-
-@item slev, surround_mix_level
-Set the surround mix level. It is a value expressed in deciBel, and must
-be in the interval [-32,32].
-
-@item lfe_mix_level
-Set LFE mix into non LFE level. It is used when there is a LFE input but no
-LFE output. It is a value expressed in deciBel, and must
-be in the interval [-32,32].
-
-@item rmvol, rematrix_volume
-Set rematrix volume. Default value is 1.0.
-
-@item flags, swr_flags
-Set flags used by the converter. Default value is 0.
-
-It supports the following individual flags:
-@table @option
-@item res
-force resampling, this flag forces resampling to be used even when the
-input and output sample rates match.
-@end table
-
-@item dither_scale
-Set the dither scale. Default value is 1.
-
-@item dither_method
-Set dither method. Default value is 0.
-
-Supported values:
-@table @samp
-@item rectangular
-select rectangular dither
-@item triangular
-select triangular dither
-@item triangular_hp
-select triangular dither with high pass
-@item lipshitz
-select lipshitz noise shaping dither
-@item shibata
-select shibata noise shaping dither
-@item low_shibata
-select low shibata noise shaping dither
-@item high_shibata
-select high shibata noise shaping dither
-@item f_weighted
-select f-weighted noise shaping dither
-@item modified_e_weighted
-select modified-e-weighted noise shaping dither
-@item improved_e_weighted
-select improved-e-weighted noise shaping dither
-
-@end table
-
-@item resampler
-Set resampling engine. Default value is swr.
-
-Supported values:
-@table @samp
-@item swr
-select the native SW Resampler; filter options precision and cheby are not
-applicable in this case.
-@item soxr
-select the SoX Resampler (where available); compensation, and filter options
-filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
-case.
-@end table
-
-@item filter_size
-For swr only, set resampling filter size, default value is 32.
-
-@item phase_shift
-For swr only, set resampling phase shift, default value is 10, and must be in
-the interval [0,30].
-
-@item linear_interp
-Use Linear Interpolation if set to 1, default value is 0.
-
-@item cutoff
-Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
-value between 0 and 1.  Default value is 0.97 with swr, and 0.91 with soxr
-(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
-
-@item precision
-For soxr only, the precision in bits to which the resampled signal will be
-calculated.  The default value of 20 (which, with suitable dithering, is
-appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
-value of 28 gives SoX's 'Very High Quality'.
-
-@item cheby
-For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
-approximation for 'irrational' ratios. Default value is 0.
-
-@item async
-For swr only, simple 1 parameter audio sync to timestamps using stretching,
-squeezing, filling and trimming. Setting this to 1 will enable filling and
-trimming, larger values represent the maximum amount in samples that the data
-may be stretched or squeezed for each second.
-Default value is 0, thus no compensation is applied to make the samples match
-the audio timestamps.
-
-@item first_pts
-For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
-This allows for padding/trimming at the start of stream. By default, no
-assumption is made about the first frame's expected pts, so no padding or
-trimming is done. For example, this could be set to 0 to pad the beginning with
-silence if an audio stream starts after the video stream or to trim any samples
-with a negative pts due to encoder delay.
-
-@item min_comp
-For swr only, set the 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
-(@option{min_comp} = @code{FLT_MAX}).
-
-@item min_hard_comp
-For swr only, set the 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 @option{min_comp}.
-The default is 0.1.
-
-@item comp_duration
-For swr only, set duration (in seconds) over which data is stretched/squeezed
-to make it match the timestamps. Must be a non-negative double float value,
-default value is 1.0.
-
-@item max_soft_comp
-For swr only, set maximum factor by which data is stretched/squeezed to make it
-match the timestamps. Must be a non-negative double float value, default value
-is 0.
-
-@item matrix_encoding
-Select matrixed stereo encoding.
-
-It accepts the following values:
-@table @samp
-@item none
-select none
-@item dolby
-select Dolby
-@item dplii
-select Dolby Pro Logic II
-@end table
-
-Default value is @code{none}.
-
-@item filter_type
-For swr only, select resampling filter type. This only affects resampling
-operations.
-
-It accepts the following values:
-@table @samp
-@item cubic
-select cubic
-@item blackman_nuttall
-select Blackman Nuttall Windowed Sinc
-@item kaiser
-select Kaiser Windowed Sinc
-@end table
-
-@item kaiser_beta
-For swr only, set Kaiser Window Beta value. Must be an integer in the
-interval [2,16], default value is 9.
-
-@end table
-
-@c man end RESAMPLER OPTIONS
+@include resampler.texi
 
 @chapter See Also
 
diff --git a/doc/resampler.texi b/doc/resampler.texi
new file mode 100644
index 0000000000..d37d53d0f1
--- /dev/null
+++ b/doc/resampler.texi
@@ -0,0 +1,222 @@
+@chapter Resampler Options
+@c man begin RESAMPLER OPTIONS
+
+The audio resampler supports the following named options.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools, @var{option}=@var{value} for the aresample filter,
+by setting the value explicitly in the
+@code{SwrContext} options or using the @file{libavutil/opt.h} API for
+programmatic use.
+
+@table @option
+
+@item ich, in_channel_count
+Set the number of input channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{in_channel_layout} is set.
+
+@item och, out_channel_count
+Set the number of output channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{out_channel_layout} is set.
+
+@item uch, used_channel_count
+Set the number of used input channels. Default value is 0. This option is
+only used for special remapping.
+
+@item isr, in_sample_rate
+Set the input sample rate. Default value is 0.
+
+@item osr, out_sample_rate
+Set the output sample rate. Default value is 0.
+
+@item isf, in_sample_fmt
+Specify the input sample format. It is set by default to @code{none}.
+
+@item osf, out_sample_fmt
+Specify the output sample format. It is set by default to @code{none}.
+
+@item tsf, internal_sample_fmt
+Set the internal sample format. Default value is @code{none}.
+This will automatically be chosen when it is not explicitly set.
+
+@item icl, in_channel_layout
+Set the input channel layout.
+
+@item ocl, out_channel_layout
+Set the output channel layout.
+
+@item clev, center_mix_level
+Set the center mix level. It is a value expressed in deciBel, and must be
+in the interval [-32,32].
+
+@item slev, surround_mix_level
+Set the surround mix level. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
+
+@item lfe_mix_level
+Set LFE mix into non LFE level. It is used when there is a LFE input but no
+LFE output. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
+
+@item rmvol, rematrix_volume
+Set rematrix volume. Default value is 1.0.
+
+@item flags, swr_flags
+Set flags used by the converter. Default value is 0.
+
+It supports the following individual flags:
+@table @option
+@item res
+force resampling, this flag forces resampling to be used even when the
+input and output sample rates match.
+@end table
+
+@item dither_scale
+Set the dither scale. Default value is 1.
+
+@item dither_method
+Set dither method. Default value is 0.
+
+Supported values:
+@table @samp
+@item rectangular
+select rectangular dither
+@item triangular
+select triangular dither
+@item triangular_hp
+select triangular dither with high pass
+@item lipshitz
+select lipshitz noise shaping dither
+@item shibata
+select shibata noise shaping dither
+@item low_shibata
+select low shibata noise shaping dither
+@item high_shibata
+select high shibata noise shaping dither
+@item f_weighted
+select f-weighted noise shaping dither
+@item modified_e_weighted
+select modified-e-weighted noise shaping dither
+@item improved_e_weighted
+select improved-e-weighted noise shaping dither
+
+@end table
+
+@item resampler
+Set resampling engine. Default value is swr.
+
+Supported values:
+@table @samp
+@item swr
+select the native SW Resampler; filter options precision and cheby are not
+applicable in this case.
+@item soxr
+select the SoX Resampler (where available); compensation, and filter options
+filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
+case.
+@end table
+
+@item filter_size
+For swr only, set resampling filter size, default value is 32.
+
+@item phase_shift
+For swr only, set resampling phase shift, default value is 10, and must be in
+the interval [0,30].
+
+@item linear_interp
+Use Linear Interpolation if set to 1, default value is 0.
+
+@item cutoff
+Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
+value between 0 and 1.  Default value is 0.97 with swr, and 0.91 with soxr
+(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
+
+@item precision
+For soxr only, the precision in bits to which the resampled signal will be
+calculated.  The default value of 20 (which, with suitable dithering, is
+appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
+value of 28 gives SoX's 'Very High Quality'.
+
+@item cheby
+For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
+approximation for 'irrational' ratios. Default value is 0.
+
+@item async
+For swr only, simple 1 parameter audio sync to timestamps using stretching,
+squeezing, filling and trimming. Setting this to 1 will enable filling and
+trimming, larger values represent the maximum amount in samples that the data
+may be stretched or squeezed for each second.
+Default value is 0, thus no compensation is applied to make the samples match
+the audio timestamps.
+
+@item first_pts
+For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
+This allows for padding/trimming at the start of stream. By default, no
+assumption is made about the first frame's expected pts, so no padding or
+trimming is done. For example, this could be set to 0 to pad the beginning with
+silence if an audio stream starts after the video stream or to trim any samples
+with a negative pts due to encoder delay.
+
+@item min_comp
+For swr only, set the 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
+(@option{min_comp} = @code{FLT_MAX}).
+
+@item min_hard_comp
+For swr only, set the 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 @option{min_comp}.
+The default is 0.1.
+
+@item comp_duration
+For swr only, set duration (in seconds) over which data is stretched/squeezed
+to make it match the timestamps. Must be a non-negative double float value,
+default value is 1.0.
+
+@item max_soft_comp
+For swr only, set maximum factor by which data is stretched/squeezed to make it
+match the timestamps. Must be a non-negative double float value, default value
+is 0.
+
+@item matrix_encoding
+Select matrixed stereo encoding.
+
+It accepts the following values:
+@table @samp
+@item none
+select none
+@item dolby
+select Dolby
+@item dplii
+select Dolby Pro Logic II
+@end table
+
+Default value is @code{none}.
+
+@item filter_type
+For swr only, select resampling filter type. This only affects resampling
+operations.
+
+It accepts the following values:
+@table @samp
+@item cubic
+select cubic
+@item blackman_nuttall
+select Blackman Nuttall Windowed Sinc
+@item kaiser
+select Kaiser Windowed Sinc
+@end table
+
+@item kaiser_beta
+For swr only, set Kaiser Window Beta value. Must be an integer in the
+interval [2,16], default value is 9.
+
+@end table
+
+@c man end RESAMPLER OPTIONS