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docs: add soxr documentation
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Rob Sykes
Michael Niedermayer
parent
03d38ee207
commit
8d9a503313
@ -106,29 +106,54 @@ select triangular dither
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select triangular dither with high pass
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@end table
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@item resampler
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Set resampling engine. Default value is swr.
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Supported values:
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@table @samp
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@item swr
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select the native SW Resampler; filter options precision and cheby are not
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applicable in this case.
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@item soxr
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select the SoX Resampler (where available); compensation, and filter options
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filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
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case.
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@end table
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@item filter_size
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Set resampling filter size, default value is 16.
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For swr only, set resampling filter size, default value is 16.
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@item phase_shift
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Set resampling phase shift, default value is 10, must be included
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For swr only, set resampling phase shift, default value is 10, must be included
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between 0 and 30.
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@item linear_interp
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Use Linear Interpolation if set to 1, default value is 0.
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@item cutoff
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Set cutoff frequency ratio. Must be a float value between 0 and 1,
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default value is 0.8.
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Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
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value between 0 and 1. Default value is 0.8 with swr, and 0.91 with soxr
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(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
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@item precision
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For soxr only, the precision in bits to which the resampled signal will be
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calculated. The default value of 20 (which, with suitable dithering, is
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appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
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value of 28 gives SoX's 'Very High Quality'.
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@item cheby
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For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
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approximation for 'irrational' ratios. Default value is 0.
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@item min_comp
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Set the minimum difference between timestamps and audio data (in
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For swr only, set the minimum difference between timestamps and audio data (in
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seconds) to trigger stretching/squeezing/filling or trimming of the
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data to make it match the timestamps. The default is that
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stretching/squeezing/filling and trimming is disabled
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(@option{min_comp} = @code{FLT_MAX}).
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@item min_hard_comp
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Set the minimum difference between timestamps and audio data (in
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For swr only, set the minimum difference between timestamps and audio data (in
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seconds) to trigger adding/dropping samples to make it match the
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timestamps. This option effectively is a threshold to select between
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hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
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@ -136,14 +161,14 @@ all compensation is by default disabled through @option{min_comp}.
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The default is 0.1.
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@item comp_duration
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Set duration (in seconds) over which data is stretched/squeezed to
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make it match the timestamps. Must be a non-negative double float
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value, default value is 1.0.
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For swr only, set duration (in seconds) over which data is stretched/squeezed
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to make it match the timestamps. Must be a non-negative double float value,
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default value is 1.0.
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@item max_soft_comp
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Set maximum factor by which data is stretched/squeezed to make it
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match the timestamps. Must be a non-negative double float value,
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default value is 0.
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For swr only, set maximum factor by which data is stretched/squeezed to make it
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match the timestamps. Must be a non-negative double float value, default value
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is 0.
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@item matrix_encoding
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Select matrixed stereo encoding.
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@ -161,7 +186,7 @@ select Dolby Pro Logic II
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Default value is @code{none}.
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@item filter_type
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Select resampling filter type. This only affects resampling
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For swr only, select resampling filter type. This only affects resampling
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operations.
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It accepts the following values:
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@ -175,8 +200,8 @@ select Kaiser Windowed Sinc
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@end table
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@item kaiser_beta
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Set Kaiser Window Beta value. Must be an integer included between 2
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and 16, default value is 9.
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For swr only, set Kaiser Window Beta value. Must be an integer included between
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2 and 16, default value is 9.
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@end table
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@ -80,15 +80,17 @@ static const AVOption options[]={
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{"triangular" , "select triangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
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{"triangular_hp" , "select triangular dither with high pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
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{"filter_size" , "set resampling filter size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=16 }, 0 , INT_MAX , PARAM },
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{"phase_shift" , "set resampling phase shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
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{"filter_size" , "set swr resampling filter size", OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=16 }, 0 , INT_MAX , PARAM },
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{"phase_shift" , "set swr resampling phase shift", OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
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{"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
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{"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM },
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{"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"},
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{"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"},
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{"soxr" , "select SoX Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR }, INT_MIN, INT_MAX , PARAM, "resampler"},
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{"precision" , "set resampling precision" , OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM },
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{"cheby" , "enable Chebyshev passband" , OFFSET(cheby) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
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{"precision" , "set soxr resampling precision (in bits)"
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, OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM },
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{"cheby" , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
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, OFFSET(cheby) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
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{"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
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, OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
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{"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
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@ -105,12 +107,12 @@ static const AVOption options[]={
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{ "dolby", "select Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
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{ "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
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{ "filter_type" , "select filter type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
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{ "filter_type" , "select swr filter type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
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{ "cubic" , "select cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
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{ "blackman_nuttall", "select Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
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{ "kaiser" , "select Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
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{ "kaiser_beta" , "set Kaiser Window Beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },
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{ "kaiser_beta" , "set swr Kaiser Window Beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },
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{0}
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};
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@ -74,17 +74,17 @@ struct SwrContext {
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int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
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int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
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int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
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double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
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enum SwrFilterType filter_type; /**< resampling filter type */
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int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
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double precision; /**< resampling precision (in bits) */
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int cheby; /**< if 1 then the resampling FIR filter will be configured for maximal passband flatness */
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double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
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enum SwrFilterType filter_type; /**< swr resampling filter type */
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int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
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double precision; /**< soxr resampling precision (in bits) */
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int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
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float min_compensation; ///< minimum below which no compensation will happen
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float min_hard_compensation; ///< minimum below which no silence inject / sample drop will happen
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float soft_compensation_duration; ///< duration over which soft compensation is applied
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float max_soft_compensation; ///< maximum soft compensation in seconds over soft_compensation_duration
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float async; ///< simple 1 parameter async, similar to ffmpegs -async
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float min_compensation; ///< swr minimum below which no compensation will happen
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float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
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float soft_compensation_duration; ///< swr duration over which soft compensation is applied
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float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
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float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
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int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
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int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
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