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docs: add soxr documentation

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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Rob Sykes 2012-12-27 12:07:15 +01:00 committed by Michael Niedermayer
parent 03d38ee207
commit 8d9a503313
3 changed files with 58 additions and 31 deletions
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55
doc/ffmpeg-resampler.texi
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@ -106,29 +106,54 @@ select triangular dither
select triangular dither with high pass
@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
Set resampling filter size, default value is 16.
For swr only, set resampling filter size, default value is 16.
@item phase_shift
Set resampling phase shift, default value is 10, must be included
For swr only, set resampling phase shift, default value is 10, must be included
between 0 and 30.
@item linear_interp
Use Linear Interpolation if set to 1, default value is 0.
@item cutoff
Set cutoff frequency ratio. Must be a float value between 0 and 1,
default value is 0.8.
Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
value between 0 and 1. Default value is 0.8 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 min_comp
Set the minimum difference between timestamps and audio data (in
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
Set the minimum difference between timestamps and audio data (in
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
@ -136,14 +161,14 @@ all compensation is by default disabled through @option{min_comp}.
The default is 0.1.
@item comp_duration
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.
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
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.
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.
@ -161,7 +186,7 @@ select Dolby Pro Logic II
Default value is @code{none}.
@item filter_type
Select resampling filter type. This only affects resampling
For swr only, select resampling filter type. This only affects resampling
operations.
It accepts the following values:
@ -175,8 +200,8 @@ select Kaiser Windowed Sinc
@end table
@item kaiser_beta
Set Kaiser Window Beta value. Must be an integer included between 2
and 16, default value is 9.
For swr only, set Kaiser Window Beta value. Must be an integer included between
2 and 16, default value is 9.
@end table

14
libswresample/swresample.c
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@ -80,15 +80,17 @@ static const AVOption options[]={
{"triangular" , "select triangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
{"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"},
{"filter_size" , "set resampling filter size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=16 }, 0 , INT_MAX , PARAM },
{"phase_shift" , "set resampling phase shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
{"filter_size" , "set swr resampling filter size", OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=16 }, 0 , INT_MAX , PARAM },
{"phase_shift" , "set swr resampling phase shift", OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
{"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
{"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM },
{"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"},
{"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"},
{"soxr" , "select SoX Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR }, INT_MIN, INT_MAX , PARAM, "resampler"},
{"precision" , "set resampling precision" , OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM },
{"cheby" , "enable Chebyshev passband" , OFFSET(cheby) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
{"precision" , "set soxr resampling precision (in bits)"
, OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM },
{"cheby" , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
, OFFSET(cheby) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
{"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
, OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
{"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
@ -105,12 +107,12 @@ static const AVOption options[]={
{ "dolby", "select Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "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" },
{ "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" },
{ "cubic" , "select cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "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" },
{ "kaiser" , "select Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "kaiser_beta" , "set Kaiser Window Beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },
{ "kaiser_beta" , "set swr Kaiser Window Beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },
{0}
};

20
libswresample/swresample_internal.h
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@ -74,17 +74,17 @@ struct SwrContext {
int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
enum SwrFilterType filter_type; /**< resampling filter type */
int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
double precision; /**< resampling precision (in bits) */
int cheby; /**< if 1 then the resampling FIR filter will be configured for maximal passband flatness */
double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
enum SwrFilterType filter_type; /**< swr resampling filter type */
int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
double precision; /**< soxr resampling precision (in bits) */
int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
float min_compensation; ///< minimum below which no compensation will happen
float min_hard_compensation; ///< minimum below which no silence inject / sample drop will happen
float soft_compensation_duration; ///< duration over which soft compensation is applied
float max_soft_compensation; ///< maximum soft compensation in seconds over soft_compensation_duration
float async; ///< simple 1 parameter async, similar to ffmpegs -async
float min_compensation; ///< swr minimum below which no compensation will happen
float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
float soft_compensation_duration; ///< swr duration over which soft compensation is applied
float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)