mirror of https://git.ffmpeg.org/ffmpeg.git
doc: add libswresample.texi and ffmpeg-resampler.texi files
This commit is contained in:
Stefano Sabatini
parent
f62abbf3b7
commit
e903cb4194
|
|
@ -1,8 +1,9 @@
|
|||
DOCLIBS-$(CONFIG_AVUTIL) += libavutil
|
||||
DOCLIBS-$(CONFIG_SWRESAMPLE) += libswresample
|
||||
DOCLIBS-$(CONFIG_AVCODEC) += libavcodec
|
||||
DOCLIBS-$(CONFIG_AVFILTER) += libavfilter
|
||||
|
||||
COMPONENTS=$(PROGS-yes) ffmpeg-codecs ffmpeg-filters
|
||||
COMPONENTS=$(PROGS-yes) ffmpeg-codecs ffmpeg-filters ffmpeg-resampler
|
||||
|
||||
MANPAGES = $(COMPONENTS:%=doc/%.1) $(DOCLIBS-yes:%=doc/%.3)
|
||||
PODPAGES = $(COMPONENTS:%=doc/%.pod) $(DOCLIBS-yes:%=doc/%.pod)
|
||||
|
|
|
|||
|
|
@ -0,0 +1,198 @@
|
|||
\input texinfo @c -*- texinfo -*-
|
||||
|
||||
@settitle FFmpeg Resampler Documentation
|
||||
@titlepage
|
||||
@center @titlefont{FFmpeg Resampler Documentation}
|
||||
@end titlepage
|
||||
|
||||
@top
|
||||
|
||||
@contents
|
||||
|
||||
@chapter Description
|
||||
@c man begin DESCRIPTION
|
||||
|
||||
The FFmpeg resampler provides an high-level interface to the
|
||||
libswresample library audio resampling utilities. In particular it
|
||||
allows to perform audio resampling, audio channel layout rematrixing,
|
||||
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, or 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 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. Must be an integer representing the
|
||||
corresponding sample format specified in
|
||||
@file{libavutil/samplefmt.h} header. Default value is -1
|
||||
(corresponding to @code{AV_SAMPLE_FMT_NONE}).
|
||||
|
||||
@item osf, out_sample_fmt
|
||||
Specify the output sample format. Must be an integer representing the
|
||||
corresponding sample format specified in
|
||||
@file{libavutil/samplefmt.h} header. Default value is -1
|
||||
(corresponding to @code{AV_SAMPLE_FMT_NONE}).
|
||||
|
||||
@item tsf, internal_sample_fmt
|
||||
Set the internal sample format. Default value is -1.
|
||||
|
||||
@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 center mix level. It is a value expressed in deciBel, and must be
|
||||
inclusively included between -32 and +32.
|
||||
|
||||
@item slev, surround_mix_level
|
||||
Set surround mix level. It is a value expressed in deciBel, and must
|
||||
be inclusively included between -32 and +32.
|
||||
|
||||
@item lfe_mix_evel
|
||||
Set LFE mix level.
|
||||
|
||||
@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
|
||||
@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
|
||||
@end table
|
||||
|
||||
@item filter_size
|
||||
Set resampling filter size, default value is 16.
|
||||
|
||||
@item phase_shift
|
||||
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.
|
||||
|
||||
@item min_comp
|
||||
Set minimum difference between timestamps and audio data (in seconds)
|
||||
below which no timestamp compensation of either kind is applied.
|
||||
Default value is @code{FLT_MAX}.
|
||||
|
||||
@item min_hard_comp
|
||||
Set minimum difference between timestamps and audio data (in seconds)
|
||||
to trigger padding/trimming the data. Must be a non-negative double,
|
||||
default value 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.
|
||||
|
||||
@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.
|
||||
|
||||
@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
|
||||
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
|
||||
Set Kaiser Window Beta value. Must be an integer included between 2
|
||||
and 16, default value is 9.
|
||||
|
||||
@end table
|
||||
|
||||
@c man end RESAMPLER OPTIONS
|
||||
|
||||
@ignore
|
||||
|
||||
@setfilename ffmpeg-resampler
|
||||
@settitle FFmpeg Resampler
|
||||
|
||||
@c man begin SEEALSO
|
||||
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
|
||||
@c man end
|
||||
|
||||
@c man begin AUTHORS
|
||||
See Git history (git://source.ffmpeg.org/ffmpeg)
|
||||
@c man end
|
||||
|
||||
@end ignore
|
||||
|
|
@ -0,0 +1,62 @@
|
|||
\input texinfo @c -*- texinfo -*-
|
||||
|
||||
@settitle Libswresample Documentation
|
||||
@titlepage
|
||||
@center @titlefont{Libswresample Documentation}
|
||||
@end titlepage
|
||||
|
||||
@top
|
||||
|
||||
@contents
|
||||
|
||||
@chapter Description
|
||||
@c man begin DESCRIPTION
|
||||
|
||||
The libswresample library performs highly optimized audio resampling,
|
||||
rematrixing and sample format conversion operations.
|
||||
|
||||
Specifically, this library performs the following conversions:
|
||||
|
||||
@itemize
|
||||
@item
|
||||
@emph{Resampling}: is the process of changing the audio rate, for
|
||||
example from an high sample rate of 44100Hz to 8000Hz. Audio
|
||||
conversion from high to low sample rate is a lossy process. Several
|
||||
resampling options and algorithms are available.
|
||||
|
||||
@item
|
||||
@emph{Format conversion}: is the process of converting the type of
|
||||
samples, for example from 16-bit signed samples to unsigned 8-bit or
|
||||
float samples. It also handles packing conversion, when passing from
|
||||
packed layout (all samples belonging to distinct channels interleaved
|
||||
in the same buffer), to planar layout (all samples belonging to the
|
||||
same channel stored in a dedicated buffer or "plane").
|
||||
|
||||
@item
|
||||
@emph{Rematrixing}: is the process of changing the channel layout, for
|
||||
example from stereo to mono. When the input channels cannot be mapped
|
||||
to the output streams, the process is lossy, since it involves
|
||||
different gain factors and mixing.
|
||||
@end itemize
|
||||
|
||||
Various other audio conversions (e.g. stretching and padding) are
|
||||
enabled through dedicated options.
|
||||
|
||||
@c man end DESCRIPTION
|
||||
|
||||
@ignore
|
||||
|
||||
@setfilename libswresample
|
||||
@settitle audio resampling library
|
||||
|
||||
@c man begin SEEALSO
|
||||
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), ffmpeg-resampler(1), libavutil(3)
|
||||
@c man end
|
||||
|
||||
@c man begin AUTHORS
|
||||
See Git history (git://source.ffmpeg.org/ffmpeg)
|
||||
@c man end
|
||||
|
||||
@end ignore
|
||||
|
||||
@bye
|
||||
Loading…
Reference in New Issue