[FFmpeg-cvslog] avutil/samplefmt: improve doxygen
Timothy Gu
git at videolan.org
Sun Jul 6 01:16:49 CEST 2014
ffmpeg | branch: master | Timothy Gu <timothygu99 at gmail.com> | Fri Jul 4 11:06:41 2014 -0700| [b27555a3a05f8a39c58ac98561411555da3f4a6f] | committer: Michael Niedermayer
avutil/samplefmt: improve doxygen
Categorize the enum and funcs to "Audio related".
Signed-off-by: Timothy Gu <timothygu99 at gmail.com>
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b27555a3a05f8a39c58ac98561411555da3f4a6f
---
libavutil/samplefmt.h | 45 ++++++++++++++++++++++++++++++++++-----------
1 file changed, 34 insertions(+), 11 deletions(-)
diff --git a/libavutil/samplefmt.h b/libavutil/samplefmt.h
index db17d43..2c346a3 100644
--- a/libavutil/samplefmt.h
+++ b/libavutil/samplefmt.h
@@ -25,26 +25,36 @@
#include "attributes.h"
/**
- * Audio Sample Formats
+ * @addtogroup lavu_audio
+ * @{
*
- * @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.
+ * @defgroup lavu_sampfmts Audio sample formats
*
- * @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.
+ * Audio sample format enumeration and related convenience functions.
+ * @{
*
- * @par
- * The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg
- * (such as AVFrame in libavcodec) is as follows:
+ */
+
+/**
+ * Audio sample formats
+ *
+ * - 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.
+ *
+ * - 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.
+ *
+ * - The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg
+ * (such as AVFrame in libavcodec) is as follows:
*
+ * @par
* 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,
@@ -158,6 +168,15 @@ int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
enum AVSampleFormat sample_fmt, int align);
/**
+ * @}
+ *
+ * @defgroup lavu_sampmanip Samples manipulation
+ *
+ * Functions that manipulate audio samples
+ * @{
+ */
+
+/**
* Fill plane data pointers and linesize for samples with sample
* format sample_fmt.
*
@@ -253,4 +272,8 @@ int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
int nb_channels, enum AVSampleFormat sample_fmt);
+/**
+ * @}
+ * @}
+ */
#endif /* AVUTIL_SAMPLEFMT_H */
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