FFmpeg 0.9.4
Since* 0.9
#

Buffer audio frames, and make them available to the filter chain.

This source is mainly intended for a programmatic use, in particular through the interface defined in libavfilter/asrc_abuffer.h.

It accepts the following mandatory parameters: sample_rate:sample_fmt:channel_layout:packing

sample_rate

The sample rate of the incoming audio buffers.

sample_fmt

The sample format of the incoming audio buffers. Either a sample format name or its corresponging integer representation from the enum AVSampleFormat in libavutil/samplefmt.h

channel_layout

The channel layout of the incoming audio buffers. Either a channel layout name from channel_layout_map in libavutil/audioconvert.c or its corresponding integer representation from the AV_CH_LAYOUT_* macros in libavutil/audioconvert.h

packing

Either "packed" or "planar", or their integer representation: 0 or 1 respectively.

For example:

abuffer=44100:s16:stereo:planar

will instruct the source to accept planar 16bit signed stereo at 44100Hz. Since the sample format with name "s16" corresponds to the number 1 and the "stereo" channel layout corresponds to the value 3, this is equivalent to:

abuffer=44100:1:3:1