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Revert "lavf: eliminate ff_get_audio_frame_size()"

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This reverts commit 30e50c50274f88f0f5ae829f401cd3c7f5266719.

The original commit broke the ability to stream AAC over HTTP/Icecast. It looks
like avformat_find_stream_info() gets stuck in an infinite loop, never hitting
AVFormatContext.max_analyze_duration since duration is never set for any of
the packets.

Example stream: http://listen.classicrocklounge.com:8000/aac64

Signed-off-by: Anton Khirnov <anton@khirnov.net>
This commit is contained in:
Andrew Stone 2014-08-18 17:28:23 -04:00 committed by Anton Khirnov
parent 67a7695c14
commit 04361427e6
2 changed files with 24 additions and 1 deletions
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2
libavformat/internal.h
View File

@ -326,6 +326,8 @@ int ff_interleave_packet_per_dts(AVFormatContext *s, AVPacket *out,
void ff_compute_frame_duration(int *pnum, int *pden, AVStream *st,
AVCodecParserContext *pc, AVPacket *pkt);
int ff_get_audio_frame_size(AVCodecContext *enc, int size, int mux);
unsigned int ff_codec_get_tag(const AVCodecTag *tags, enum AVCodecID id);
enum AVCodecID ff_codec_get_id(const AVCodecTag *tags, unsigned int tag);

23
libavformat/utils.c
View File

@ -453,6 +453,27 @@ 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.
*/
@ -488,7 +509,7 @@ void ff_compute_frame_duration(int *pnum, int *pden, AVStream *st,
}
break;
case AVMEDIA_TYPE_AUDIO:
frame_size = av_get_audio_frame_duration(st->codec, pkt->size);
frame_size = ff_get_audio_frame_size(st->codec, pkt->size, 0);
if (frame_size <= 0 || st->codec->sample_rate <= 0)
break;
*pnum = frame_size;