Examples


Decode, display and encode video input with ffmpeg at the same time: 

ffmpeg -i INPUT OUTPUT -f xv display

Decode and display the input video to multiple X11 windows: 

ffmpeg -i INPUT -f xv normal -vf negate -f xv negated

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.

'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 'in_channel_layout' is set.

'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 'out_channel_layout' is set.

'uch, used_channel_count'

Set the number of used input channels. Default value is 0. This option is only used for special remapping.

'isr, in_sample_rate'

Set the input sample rate. Default value is 0.

'osr, out_sample_rate'

Set the output sample rate. Default value is 0.

'isf, in_sample_fmt'

Specify the input sample format. It is set by default to none.

'osf, out_sample_fmt'

Specify the output sample format. It is set by default to none.

'tsf, internal_sample_fmt'

Set the internal sample format. Default value is none. This will automatically be chosen when it is not explicitly set.

'icl, in_channel_layout'
'ocl, out_channel_layout'

Set the input/output channel layout.

See for the required syntax.

'clev, center_mix_level'

Set the center mix level. It is a value expressed in deciBel, and must be in the interval [-32,32].

'slev, surround_mix_level'

Set the surround mix level. It is a value expressed in deciBel, and must be in the interval [-32,32].

'lfe_mix_level'

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].

'rmvol, rematrix_volume'

Set rematrix volume. Default value is 1.0.

'rematrix_maxval'

Set maximum output value for rematrixing. This can be used to prevent clipping vs. preventing volume reduction. A value of 1.0 prevents clipping.

'flags, swr_flags'

Set flags used by the converter. Default value is 0.

It supports the following individual flags:

'res'

force resampling, this flag forces resampling to be used even when the input and output sample rates match.

'dither_scale'

Set the dither scale. Default value is 1.

'dither_method'

Set dither method. Default value is 0.

Supported values:

'rectangular'

select rectangular dither

'triangular'

select triangular dither

'triangular_hp'

select triangular dither with high pass

'lipshitz'

select Lipshitz noise shaping dither.

'shibata'

select Shibata noise shaping dither.

'low_shibata'

select low Shibata noise shaping dither.

'high_shibata'

select high Shibata noise shaping dither.

'f_weighted'

select f-weighted noise shaping dither

'modified_e_weighted'

select modified-e-weighted noise shaping dither

'improved_e_weighted'

select improved-e-weighted noise shaping dither

'resampler'

Set resampling engine. Default value is swr.

Supported values:

'swr'

select the native SW Resampler; filter options precision and cheby are not applicable in this case.

'soxr'

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.

'filter_size'

For swr only, set resampling filter size, default value is 32.

'phase_shift'

For swr only, set resampling phase shift, default value is 10, and must be in the interval [0,30].

'linear_interp'

Use linear interpolation when enabled (the default). Disable it if you want to preserve speed instead of quality when exact_rational fails.

'exact_rational'

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.

'cutoff'

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).

'precision'

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'.

'cheby'

For soxr only, selects passband rolloff none (Chebyshev) & higher-precision approximation for 'irrational' ratios. Default value is 0.

'async'

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.

'first_pts'

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.

'min_comp'

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).

'min_hard_comp'

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.

'comp_duration'

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.

'max_soft_comp'

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.

'matrix_encoding'

Select matrixed stereo encoding.

It accepts the following values:

'none'

select none

'dolby'

select Dolby

'dplii'

select Dolby Pro Logic II

Default value is none.

'filter_type'

For swr only, select resampling filter type. This only affects resampling operations.

It accepts the following values:

'cubic'

select cubic

'blackman_nuttall'

select Blackman Nuttall windowed sinc

'kaiser'

select Kaiser windowed sinc

'kaiser_beta'

For swr only, set Kaiser window beta value. Must be a double float value in the interval [2,16], default value is 9.

'output_sample_bits'

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.

The video scaler supports the following named options.

Options may be set by specifying -option value in the FFmpeg tools, with a few API-only exceptions noted below. For programmatic use, they can be set explicitly in the SwsContext options or through the 'libavutil/opt.h' API.

'sws_flags'

Set the scaler flags. This is also used to set the scaling algorithm. Only a single algorithm should be selected. Default value is 'bicubic'.

It accepts the following values:

'fast_bilinear'

Select fast bilinear scaling algorithm.

'bilinear'

Select bilinear scaling algorithm.

'bicubic'

Select bicubic scaling algorithm.

'experimental'

Select experimental scaling algorithm.

'neighbor'

Select nearest neighbor rescaling algorithm.

'area'

Select averaging area rescaling algorithm.

'bicublin'

Select bicubic scaling algorithm for the luma component, bilinear for chroma components.

'gauss'

Select Gaussian rescaling algorithm.

'sinc'

Select sinc rescaling algorithm.

'lanczos'

Select Lanczos rescaling algorithm.

'spline'

Select natural bicubic spline rescaling algorithm.

'print_info'

Enable printing/debug logging.

'accurate_rnd'

Enable accurate rounding.

'full_chroma_int'

Enable full chroma interpolation.

'full_chroma_inp'

Select full chroma input.

'bitexact'

Enable bitexact output.

'srcw (API only)'

Set source width.

'srch (API only)'

Set source height.

'dstw (API only)'

Set destination width.

'dsth (API only)'

Set destination height.

'src_format (API only)'

Set source pixel format (must be expressed as an integer).

'dst_format (API only)'

Set destination pixel format (must be expressed as an integer).

'src_range'

Select source range.

'dst_range'

Select destination range.

'param0, param1'

Set scaling algorithm parameters. The specified values are specific of some scaling algorithms and ignored by others. The specified values are floating point number values.

'sws_dither'

Set the dithering algorithm. Accepts one of the following values. Default value is 'auto'.

'auto'

automatic choice

'none'

no dithering

'bayer'

bayer dither

'ed'

error diffusion dither

'a_dither'

arithmetic dither, based using addition

'x_dither'

arithmetic dither, based using xor (more random/less apparent patterning that a_dither).

'alphablend'

Set the alpha blending to use when the input has alpha but the output does not. Default value is 'none'.

'uniform_color'

Blend onto a uniform background color

'checkerboard'

Blend onto a checkerboard

'none'

No blending

Filtering in FFmpeg is enabled through the libavfilter library.

In libavfilter, a filter can have multiple inputs and multiple outputs. To illustrate the sorts of things that are possible, we consider the following filtergraph. 

                [main]
input --> split ---------------------> overlay --> output
            |                             ^
            |[tmp]                  [flip]|
            +-----> crop --> vflip -------+

This filtergraph splits the input stream in two streams, then sends one stream through the crop filter and the vflip filter, before merging it back with the other stream by overlaying it on top. You can use the following command to achieve this: 

ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT

The result will be that the top half of the video is mirrored onto the bottom half of the output video.

Filters in the same linear chain are separated by commas, and distinct linear chains of filters are separated by semicolons. In our example, crop,vflip are in one linear chain, split and overlay are separately in another. The points where the linear chains join are labelled by names enclosed in square brackets. In the example, the split filter generates two outputs that are associated to the labels [main] and [tmp].

The stream sent to the second output of split, labelled as [tmp], is processed through the crop filter, which crops away the lower half part of the video, and then vertically flipped. The overlay filter takes in input the first unchanged output of the split filter (which was labelled as [main]), and overlay on its lower half the output generated by the crop,vflip filterchain.

Some filters take in input a list of parameters: they are specified after the filter name and an equal sign, and are separated from each other by a colon.

There exist so-called source filters that do not have an audio/video input, and sink filters that will not have audio/video output.

The 'graph2dot' program included in the FFmpeg 'tools' directory can be used to parse a filtergraph description and issue a corresponding textual representation in the dot language.

Invoke the command: 

graph2dot -h

to see how to use 'graph2dot'.

You can then pass the dot description to the 'dot' program (from the graphviz suite of programs) and obtain a graphical representation of the filtergraph.

For example the sequence of commands: 

echo GRAPH_DESCRIPTION | \
tools/graph2dot -o graph.tmp && \
dot -Tpng graph.tmp -o graph.png && \
display graph.png

can be used to create and display an image representing the graph described by the GRAPH_DESCRIPTION string. Note that this string must be a complete self-contained graph, with its inputs and outputs explicitly defined. For example if your command line is of the form: 

ffmpeg -i infile -vf scale=640:360 outfile

your GRAPH_DESCRIPTION string will need to be of the form: 

nullsrc,scale=640:360,nullsink

you may also need to set the nullsrc parameters and add a format filter in order to simulate a specific input file.

A filtergraph is a directed graph of connected filters. It can contain cycles, and there can be multiple links between a pair of filters. Each link has one input pad on one side connecting it to one filter from which it takes its input, and one output pad on the other side connecting it to one filter accepting its output.

Each filter in a filtergraph is an instance of a filter class registered in the application, which defines the features and the number of input and output pads of the filter.

A filter with no input pads is called a "source", and a filter with no output pads is called a "sink".