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doc: add libswresample.texi and ffmpeg-resampler.texi files
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Stefano Sabatini
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DOCLIBS-$(CONFIG_AVUTIL) += libavutil
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DOCLIBS-$(CONFIG_SWRESAMPLE) += libswresample
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DOCLIBS-$(CONFIG_AVCODEC) += libavcodec
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DOCLIBS-$(CONFIG_AVFILTER) += libavfilter
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COMPONENTS=$(PROGS-yes) ffmpeg-codecs ffmpeg-filters
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COMPONENTS=$(PROGS-yes) ffmpeg-codecs ffmpeg-filters ffmpeg-resampler
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MANPAGES = $(COMPONENTS:%=doc/%.1) $(DOCLIBS-yes:%=doc/%.3)
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PODPAGES = $(COMPONENTS:%=doc/%.pod) $(DOCLIBS-yes:%=doc/%.pod)
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198
doc/ffmpeg-resampler.texi
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198
doc/ffmpeg-resampler.texi
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\input texinfo @c -*- texinfo -*-
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@settitle FFmpeg Resampler Documentation
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@titlepage
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@center @titlefont{FFmpeg Resampler Documentation}
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@end titlepage
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@top
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@contents
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@chapter Description
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@c man begin DESCRIPTION
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The FFmpeg resampler provides an high-level interface to the
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libswresample library audio resampling utilities. In particular it
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allows to perform audio resampling, audio channel layout rematrixing,
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and convert audio format and packing layout.
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@c man end DESCRIPTION
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@chapter Resampler Options
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@c man begin RESAMPLER OPTIONS
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The audio resampler supports the following named options.
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Options may be set by specifying -@var{option} @var{value} in the
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FFmpeg tools, or by setting the value explicitly in the
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@code{SwrContext} options or using the @file{libavutil/opt.h} API for
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programmatic use.
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@table @option
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@item ich, in_channel_count
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Set the number of input channels. Default value is 0. Setting this
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value is not mandatory if the corresponding channel layout
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@option{in_channel_layout} is set.
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@item och, out_channel_count
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Set the number of output channels. Default value is 0. Setting this
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value is not mandatory if the corresponding channel layout
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@option{out_channel_layout} is set.
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@item uch, used_channel_count
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Set the number of used channels. Default value is 0. This option is
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only used for special remapping.
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@item isr, in_sample_rate
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Set the input sample rate. Default value is 0.
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@item osr, out_sample_rate
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Set the output sample rate. Default value is 0.
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@item isf, in_sample_fmt
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Specify the input sample format. Must be an integer representing the
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corresponding sample format specified in
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@file{libavutil/samplefmt.h} header. Default value is -1
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(corresponding to @code{AV_SAMPLE_FMT_NONE}).
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@item osf, out_sample_fmt
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Specify the output sample format. Must be an integer representing the
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corresponding sample format specified in
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@file{libavutil/samplefmt.h} header. Default value is -1
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(corresponding to @code{AV_SAMPLE_FMT_NONE}).
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@item tsf, internal_sample_fmt
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Set the internal sample format. Default value is -1.
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@item icl, in_channel_layout
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Set the input channel layout.
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@item ocl, out_channel_layout
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Set the output channel layout.
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@item clev, center_mix_level
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Set center mix level. It is a value expressed in deciBel, and must be
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inclusively included between -32 and +32.
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@item slev, surround_mix_level
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Set surround mix level. It is a value expressed in deciBel, and must
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be inclusively included between -32 and +32.
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@item lfe_mix_evel
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Set LFE mix level.
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@item rmvol, rematrix_volume
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Set rematrix volume. Default value is 1.0.
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@item flags, swr_flags
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Set flags used by the converter. Default value is 0.
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It supports the following individual flags:
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@table @option
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@item res
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force resampling
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@end table
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@item dither_scale
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Set the dither scale. Default value is 1.
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@item dither_method
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Set dither method. Default value is 0.
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Supported values:
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@table @samp
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@item rectangular
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select rectangular dither
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@item triangular
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select triangular dither
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@item triangular_hp
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select triangular dither with high pass
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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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@item phase_shift
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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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@item min_comp
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Set minimum difference between timestamps and audio data (in seconds)
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below which no timestamp compensation of either kind is applied.
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Default value is @code{FLT_MAX}.
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@item min_hard_comp
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Set minimum difference between timestamps and audio data (in seconds)
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to trigger padding/trimming the data. Must be a non-negative double,
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default value 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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@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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@item matrix_encoding
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Select matrixed stereo encoding.
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It accepts the following values:
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@table @samp
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@item none
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select none
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@item dolby
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select Dolby
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@item dplii
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select Dolby Pro Logic II
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@end table
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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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operations.
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It accepts the following values:
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@table @samp
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@item cubic
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select cubic
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@item blackman_nuttall
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select Blackman Nuttall Windowed Sinc
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@item kaiser
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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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@end table
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@c man end RESAMPLER OPTIONS
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@ignore
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@setfilename ffmpeg-resampler
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@settitle FFmpeg Resampler
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@c man begin SEEALSO
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ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
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@c man end
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@c man begin AUTHORS
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See Git history (git://source.ffmpeg.org/ffmpeg)
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@c man end
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@end ignore
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62
doc/libswresample.texi
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62
doc/libswresample.texi
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\input texinfo @c -*- texinfo -*-
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@settitle Libswresample Documentation
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@titlepage
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@center @titlefont{Libswresample Documentation}
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@end titlepage
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@top
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@contents
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@chapter Description
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@c man begin DESCRIPTION
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The libswresample library performs highly optimized audio resampling,
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rematrixing and sample format conversion operations.
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Specifically, this library performs the following conversions:
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@itemize
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@item
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@emph{Resampling}: is the process of changing the audio rate, for
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example from an high sample rate of 44100Hz to 8000Hz. Audio
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conversion from high to low sample rate is a lossy process. Several
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resampling options and algorithms are available.
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@item
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@emph{Format conversion}: is the process of converting the type of
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samples, for example from 16-bit signed samples to unsigned 8-bit or
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float samples. It also handles packing conversion, when passing from
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packed layout (all samples belonging to distinct channels interleaved
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in the same buffer), to planar layout (all samples belonging to the
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same channel stored in a dedicated buffer or "plane").
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@item
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@emph{Rematrixing}: is the process of changing the channel layout, for
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example from stereo to mono. When the input channels cannot be mapped
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to the output streams, the process is lossy, since it involves
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different gain factors and mixing.
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@end itemize
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Various other audio conversions (e.g. stretching and padding) are
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enabled through dedicated options.
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@c man end DESCRIPTION
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@ignore
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@setfilename libswresample
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@settitle audio resampling library
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@c man begin SEEALSO
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ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), ffmpeg-resampler(1), libavutil(3)
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@c man end
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@c man begin AUTHORS
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See Git history (git://source.ffmpeg.org/ffmpeg)
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@c man end
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@end ignore
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@bye
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