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authorAnton Khirnov <anton@khirnov.net>2014-07-28 13:27:57 +0000
committerAnton Khirnov <anton@khirnov.net>2014-08-13 17:41:11 +0000
commit30e50c50274f88f0f5ae829f401cd3c7f5266719 (patch)
tree6279a000642b3c6e929709be96a1a3818e34993f /libavformat/utils.c
parent481a3667495425db9fdffb653292b6460fb68208 (diff)
lavf: eliminate ff_get_audio_frame_size()
It is basically a wrapper around av_get_audio_frame_duration(), with a fallback to AVCodecContext.frame_size. However, that field is set only when the stream codec context is actually used for encoding or decoding, which is discouraged. For muxing, it is generally the responsibility of the caller to set the packet duration. For demuxing, if the duration is not stored at the container level, it should be set by the parser. Therefore, removing the frame_size fallback should not break any important case.
Diffstat (limited to 'libavformat/utils.c')
-rw-r--r--libavformat/utils.c23
1 files changed, 1 insertions, 22 deletions
diff --git a/libavformat/utils.c b/libavformat/utils.c
index 973ab94d6f..4cc246d9ee 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -454,27 +454,6 @@ int ff_read_packet(AVFormatContext *s, AVPacket *pkt)
/**********************************************************/
/**
- * Get the number of samples of an audio frame. Return -1 on error.
- */
-int ff_get_audio_frame_size(AVCodecContext *enc, int size, int mux)
-{
- int frame_size;
-
- /* give frame_size priority if demuxing */
- if (!mux && enc->frame_size > 1)
- return enc->frame_size;
-
- if ((frame_size = av_get_audio_frame_duration(enc, size)) > 0)
- return frame_size;
-
- /* Fall back on using frame_size if muxing. */
- if (enc->frame_size > 1)
- return enc->frame_size;
-
- return -1;
-}
-
-/**
* Return the frame duration in seconds. Return 0 if not available.
*/
void ff_compute_frame_duration(int *pnum, int *pden, AVStream *st,
@@ -509,7 +488,7 @@ void ff_compute_frame_duration(int *pnum, int *pden, AVStream *st,
}
break;
case AVMEDIA_TYPE_AUDIO:
- frame_size = ff_get_audio_frame_size(st->codec, pkt->size, 0);
+ frame_size = av_get_audio_frame_duration(st->codec, pkt->size);
if (frame_size <= 0 || st->codec->sample_rate <= 0)
break;
*pnum = frame_size;