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/resampler.texi | 222 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 222 insertions(+) create mode 100644 doc/resampler.texi (limited to 'doc/resampler.texi') 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 -- cgit v1.2.3