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  • \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