From f5c4671bfbad96bf346bd7e9a21fc4317b4959df Mon Sep 17 00:00:00 2001 From: Indrajith K L Date: Sat, 3 Dec 2022 17:00:20 +0530 Subject: Adds most of the tools --- ffmpeg/doc/ffmpeg-resampler.html | 364 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 364 insertions(+) create mode 100644 ffmpeg/doc/ffmpeg-resampler.html (limited to 'ffmpeg/doc/ffmpeg-resampler.html') diff --git a/ffmpeg/doc/ffmpeg-resampler.html b/ffmpeg/doc/ffmpeg-resampler.html new file mode 100644 index 0000000..8a11b08 --- /dev/null +++ b/ffmpeg/doc/ffmpeg-resampler.html @@ -0,0 +1,364 @@ + + + +
+ +The FFmpeg resampler provides a high-level interface to the +libswresample library audio resampling utilities. In particular it +allows one to perform audio resampling, audio channel layout rematrixing, +and convert audio format and packing layout. +
+ + +The audio resampler supports the following named options. +
+Options may be set by specifying -option value in the
+FFmpeg tools, option=value for the aresample filter,
+by setting the value explicitly in the
+SwrContext
options or using the libavutil/opt.h API for
+programmatic use.
+
Set the number of input channels. Default value is 0. Setting this +value is not mandatory if the corresponding channel layout +in_channel_layout is set. +
+Set the number of output channels. Default value is 0. Setting this +value is not mandatory if the corresponding channel layout +out_channel_layout is set. +
+Set the number of used input channels. Default value is 0. This option is +only used for special remapping. +
+Set the input sample rate. Default value is 0. +
+Set the output sample rate. Default value is 0. +
+Specify the input sample format. It is set by default to none
.
+
Specify the output sample format. It is set by default to none
.
+
Set the internal sample format. Default value is none
.
+This will automatically be chosen when it is not explicitly set.
+
Set the input/output channel layout. +
+See (ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual +for the required syntax. +
+Set the center mix level. It is a value expressed in deciBel, and must be +in the interval [-32,32]. +
+Set the surround mix level. It is a value expressed in deciBel, and must +be in the interval [-32,32]. +
+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]. +
+Set rematrix volume. Default value is 1.0. +
+Set maximum output value for rematrixing. +This can be used to prevent clipping vs. preventing volume reduction. +A value of 1.0 prevents clipping. +
+Set flags used by the converter. Default value is 0. +
+It supports the following individual flags: +
force resampling, this flag forces resampling to be used even when the +input and output sample rates match. +
Set the dither scale. Default value is 1. +
+Set dither method. Default value is 0. +
+Supported values: +
select rectangular dither +
select triangular dither +
select triangular dither with high pass +
select Lipshitz noise shaping dither. +
select Shibata noise shaping dither. +
select low Shibata noise shaping dither. +
select high Shibata noise shaping dither. +
select f-weighted noise shaping dither +
select modified-e-weighted noise shaping dither +
select improved-e-weighted noise shaping dither +
+Set resampling engine. Default value is swr. +
+Supported values: +
select the native SW Resampler; filter options precision and cheby are not +applicable in this case. +
select the SoX Resampler (where available); compensation, and filter options +filter_size, phase_shift, exact_rational, filter_type & kaiser_beta, are not +applicable in this case. +
For swr only, set resampling filter size, default value is 32. +
+For swr only, set resampling phase shift, default value is 10, and must be in +the interval [0,30]. +
+Use linear interpolation when enabled (the default). Disable it if you want +to preserve speed instead of quality when exact_rational fails. +
+For swr only, when enabled, try to use exact phase_count based on input and
+output sample rate. However, if it is larger than 1 << phase_shift
,
+the phase_count will be 1 << phase_shift
as fallback. Default is enabled.
+
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). +
+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’. +
+For soxr only, selects passband rolloff none (Chebyshev) & higher-precision +approximation for ’irrational’ ratios. Default value is 0. +
+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. +
+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. +
+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
+(min_comp = FLT_MAX
).
+
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 min_comp. +The default is 0.1. +
+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. +
+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. +
+Select matrixed stereo encoding. +
+It accepts the following values: +
select none +
select Dolby +
select Dolby Pro Logic II +
Default value is none
.
+
For swr only, select resampling filter type. This only affects resampling +operations. +
+It accepts the following values: +
select cubic +
select Blackman Nuttall windowed sinc +
select Kaiser windowed sinc +
For swr only, set Kaiser window beta value. Must be a double float value in the +interval [2,16], default value is 9. +
+For swr only, set number of used output sample bits for dithering. Must be an integer in the +interval [0,64], default value is 0, which means it’s not used. +
+ffmpeg, ffplay, ffprobe, +libswresample +
+ + +The FFmpeg developers. +
+For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+git log
in the FFmpeg source directory, or browsing the
+online repository at http://source.ffmpeg.org.
+
Maintainers for the specific components are listed in the file +MAINTAINERS in the source code tree. +
+ ++ This document was generated using makeinfo. +
+