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>

1 files changed, 26 insertions, 1 deletions

diff --git a/libavutil/samplefmt.h b/libavutil/samplefmt.h
index af015d0cc3..99fde99e8c 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