diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2012-04-17 22:18:21 +0200 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2012-04-18 00:28:06 +0200 |
commit | 123272374180d8dc3ce9dff50f0c77d5b3b3341f (patch) | |
tree | 2c2add85161e74ebe6d4301a6607adc521112582 /libavutil/samplefmt.h | |
parent | a66675268f63dd6794ce946c7edbcb8b49ae0f13 (diff) | |
parent | 0f96f0d9968a767ead3aec823fcdfb78f26f7be7 (diff) | |
download | ffmpeg-streaming-123272374180d8dc3ce9dff50f0c77d5b3b3341f.zip ffmpeg-streaming-123272374180d8dc3ce9dff50f0c77d5b3b3341f.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
aacenc: Fix issues with huge values of bit_rate.
dv_tablegen: Drop unnecessary av_unused attribute from dv_vlc_map_tableinit().
proresenc: multithreaded quantiser search
riff: use bps instead of bits_per_coded_sample in the WAVEFORMATEXTENSIBLE header
avconv: only set the "channels" option when it exists for the specified input format
avplay: update get_buffer to be inline with avconv
aacdec: More robust output configuration.
faac: Fix multi-channel ordering
faac: Add .channel_layouts
rtmp: Support 'rtmp_playpath', an option which overrides the stream identifier
rtmp: Support 'rtmp_app', an option which overrides the name of application
avutil: add better documentation for AVSampleFormat
Conflicts:
libavcodec/aac.h
libavcodec/aacdec.c
libavcodec/aacenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavutil/samplefmt.h')
-rw-r--r-- | libavutil/samplefmt.h | 27 |
1 files changed, 26 insertions, 1 deletions
diff --git a/libavutil/samplefmt.h b/libavutil/samplefmt.h index af015d0..99fde99 100644 --- a/libavutil/samplefmt.h +++ b/libavutil/samplefmt.h @@ -22,7 +22,26 @@ #include "avutil.h" /** - * all in native-endian format + * Audio Sample Formats + * + * @par + * The data described by the sample format is always in native-endian order. + * Sample values can be expressed by native C types, hence the lack of a signed + * 24-bit sample format even though it is a common raw audio data format. + * + * @par + * The floating-point formats are based on full volume being in the range + * [-1.0, 1.0]. Any values outside this range are beyond full volume level. + * + * @par + * The data layout as used in av_samples_fill_arrays() and elsewhere in Libav + * (such as AVFrame in libavcodec) is as follows: + * + * For planar sample formats, each audio channel is in a separate data plane, + * and linesize is the buffer size, in bytes, for a single plane. All data + * planes must be the same size. For packed sample formats, only the first data + * plane is used, and samples for each channel are interleaved. In this case, + * linesize is the buffer size, in bytes, for the 1 plane. */ enum AVSampleFormat { AV_SAMPLE_FMT_NONE = -1, @@ -147,6 +166,9 @@ int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, * buffer for planar layout, or the aligned size of the buffer for all channels * for packed layout. * + * @see enum AVSampleFormat + * The documentation for AVSampleFormat describes the data layout. + * * @param[out] audio_data array to be filled with the pointer for each channel * @param[out] linesize calculated linesize, may be NULL * @param buf the pointer to a buffer containing the samples @@ -165,6 +187,9 @@ int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, uint8_t *buf, * linesize accordingly. * The allocated samples buffer can be freed by using av_freep(&audio_data[0]) * + * @see enum AVSampleFormat + * The documentation for AVSampleFormat describes the data layout. + * * @param[out] audio_data array to be filled with the pointer for each channel * @param[out] linesize aligned size for audio buffer(s), may be NULL * @param nb_channels number of audio channels |