[FFmpeg-devel] [PATCH 2/8] avutil/samplefmt: improve doxygen

[Timothy Gu]

Categorize the enum and funcs to "Audio related".

Signed-off-by: Timothy Gu <timothygu99 at gmail.com>
---
 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 */
-- 
1.9.1