From a6cd26fc9331e3793f06be3756427e23e926b61d Mon Sep 17 00:00:00 2001 From: Stefano Sabatini Date: Tue, 19 Mar 2013 20:03:57 +0100 Subject: doc: move ffmpeg-resampler.texi content to separated file This should simplify inclusion in monolithic tool manuals. --- doc/ffmpeg-resampler.texi | 223 +--------------------------------------------- 1 file changed, 1 insertion(+), 222 deletions(-) (limited to 'doc/ffmpeg-resampler.texi') 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 -- cgit v1.2.3