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FFMPEG-RESAMPLER(1) FFMPEG-RESAMPLER(1) |
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NAME |
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ffmpeg-resampler - FFmpeg Resampler |
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DESCRIPTION |
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The FFmpeg resampler provides a high-level interface to the |
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libswresample library audio resampling utilities. In particular it |
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allows one to perform audio resampling, audio channel layout |
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rematrixing, and convert audio format and packing layout. |
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RESAMPLER OPTIONS |
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The audio resampler supports the following named options. |
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Options may be set by specifying -option value in the FFmpeg tools, |
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option=value for the aresample filter, by setting the value explicitly |
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in the "SwrContext" options or using the libavutil/opt.h API for |
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programmatic use. |
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uchl, used_chlayout |
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Set used input channel layout. Default is unset. This option is |
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only used for special remapping. |
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isr, in_sample_rate |
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Set the input sample rate. Default value is 0. |
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osr, out_sample_rate |
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Set the output sample rate. Default value is 0. |
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isf, in_sample_fmt |
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Specify the input sample format. It is set by default to "none". |
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osf, out_sample_fmt |
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Specify the output sample format. It is set by default to "none". |
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tsf, internal_sample_fmt |
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Set the internal sample format. Default value is "none". This will |
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automatically be chosen when it is not explicitly set. |
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ichl, in_chlayout |
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ochl, out_chlayout |
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Set the input/output channel layout. |
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See the Channel Layout section in the ffmpeg-utils(1) manual for |
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the required syntax. |
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clev, center_mix_level |
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Set the center mix level. It is a value expressed in deciBel, and |
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must be in the interval [-32,32]. |
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slev, surround_mix_level |
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Set the surround mix level. It is a value expressed in deciBel, and |
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must be in the interval [-32,32]. |
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lfe_mix_level |
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Set LFE mix into non LFE level. It is used when there is a LFE |
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input but no LFE output. It is a value expressed in deciBel, and |
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must be in the interval [-32,32]. |
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rmvol, rematrix_volume |
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Set rematrix volume. Default value is 1.0. |
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rematrix_maxval |
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Set maximum output value for rematrixing. This can be used to |
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prevent clipping vs. preventing volume reduction. A value of 1.0 |
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prevents clipping. |
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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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res force resampling, this flag forces resampling to be used even |
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when the input and output sample rates match. |
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dither_scale |
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Set the dither scale. Default value is 1. |
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dither_method |
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Set dither method. Default value is 0. |
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Supported values: |
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rectangular |
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select rectangular dither |
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triangular |
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select triangular dither |
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triangular_hp |
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select triangular dither with high pass |
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lipshitz |
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select Lipshitz noise shaping dither. |
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shibata |
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select Shibata noise shaping dither. |
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low_shibata |
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select low Shibata noise shaping dither. |
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high_shibata |
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select high Shibata noise shaping dither. |
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f_weighted |
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select f-weighted noise shaping dither |
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modified_e_weighted |
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select modified-e-weighted noise shaping dither |
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improved_e_weighted |
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select improved-e-weighted noise shaping dither |
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resampler |
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Set resampling engine. Default value is swr. |
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Supported values: |
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swr select the native SW Resampler; filter options precision and |
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cheby are not applicable in this case. |
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soxr |
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select the SoX Resampler (where available); compensation, and |
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filter options filter_size, phase_shift, exact_rational, |
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filter_type & kaiser_beta, are not applicable in this case. |
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filter_size |
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For swr only, set resampling filter size, default value is 32. |
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phase_shift |
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For swr only, set resampling phase shift, default value is 10, and |
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must be in the interval [0,30]. |
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linear_interp |
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Use linear interpolation when enabled (the default). Disable it if |
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you want to preserve speed instead of quality when exact_rational |
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fails. |
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exact_rational |
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For swr only, when enabled, try to use exact phase_count based on |
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input and output sample rate. However, if it is larger than "1 << |
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phase_shift", the phase_count will be "1 << phase_shift" as |
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fallback. Default is enabled. |
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cutoff |
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Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must |
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be a float value between 0 and 1. Default value is 0.97 with swr, |
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and 0.91 with soxr (which, with a sample-rate of 44100, preserves |
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the entire audio band to 20kHz). |
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precision |
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For soxr only, the precision in bits to which the resampled signal |
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will be calculated. The default value of 20 (which, with suitable |
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dithering, is appropriate for a destination bit-depth of 16) gives |
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SoX's 'High Quality'; a value of 28 gives SoX's 'Very High |
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Quality'. |
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cheby |
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For soxr only, selects passband rolloff none (Chebyshev) & higher- |
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precision approximation for 'irrational' ratios. Default value is |
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0. |
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async |
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For swr only, simple 1 parameter audio sync to timestamps using |
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stretching, squeezing, filling and trimming. Setting this to 1 will |
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enable filling and trimming, larger values represent the maximum |
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amount in samples that the data may be stretched or squeezed for |
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each second. Default value is 0, thus no compensation is applied |
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to make the samples match the audio timestamps. |
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first_pts |
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For swr only, assume the first pts should be this value. The time |
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unit is 1 / sample rate. This allows for padding/trimming at the |
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start of stream. By default, no assumption is made about the first |
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frame's expected pts, so no padding or trimming is done. For |
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example, this could be set to 0 to pad the beginning with silence |
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if an audio stream starts after the video stream or to trim any |
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samples with a negative pts due to encoder delay. |
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min_comp |
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For swr only, set the minimum difference between timestamps and |
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audio data (in seconds) to trigger stretching/squeezing/filling or |
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trimming of the data to make it match the timestamps. The default |
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is that stretching/squeezing/filling and trimming is disabled |
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(min_comp = "FLT_MAX"). |
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min_hard_comp |
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For swr only, set the minimum difference between timestamps and |
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audio data (in seconds) to trigger adding/dropping samples to make |
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it match the timestamps. This option effectively is a threshold to |
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select between hard (trim/fill) and soft (squeeze/stretch) |
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compensation. Note that all compensation is by default disabled |
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through min_comp. The default is 0.1. |
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comp_duration |
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For swr only, set duration (in seconds) over which data is |
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stretched/squeezed to make it match the timestamps. Must be a non- |
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negative double float value, default value is 1.0. |
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max_soft_comp |
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For swr only, set maximum factor by which data is |
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stretched/squeezed to make it match the timestamps. Must be a non- |
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negative double float value, default value is 0. |
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matrix_encoding |
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Select matrixed stereo encoding. |
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It accepts the following values: |
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none |
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select none |
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dolby |
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select Dolby |
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dplii |
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select Dolby Pro Logic II |
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Default value is "none". |
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filter_type |
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For swr only, select resampling filter type. This only affects |
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resampling operations. |
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It accepts the following values: |
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cubic |
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select cubic |
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blackman_nuttall |
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select Blackman Nuttall windowed sinc |
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kaiser |
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select Kaiser windowed sinc |
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kaiser_beta |
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For swr only, set Kaiser window beta value. Must be a double float |
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value in the interval [2,16], default value is 9. |
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output_sample_bits |
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For swr only, set number of used output sample bits for dithering. |
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Must be an integer in the interval [0,64], default value is 0, |
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which means it's not used. |
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SEE ALSO |
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ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3) |
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AUTHORS |
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The FFmpeg developers. |
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For details about the authorship, see the Git history of the project |
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(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in |
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the FFmpeg source directory, or browsing the online repository at |
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<https://git.ffmpeg.org/ffmpeg>. |
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Maintainers for the specific components are listed in the file |
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MAINTAINERS in the source code tree. |
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FFMPEG-RESAMPLER(1) |
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