[FFmpeg-cvslog] flvdec: Add param change side data if the sample rate or channels have changed
Martin Storsjö
git at videolan.org
Thu Dec 22 02:05:05 CET 2011
ffmpeg | branch: master | Martin Storsjö <martin at martin.st> | Fri Dec 16 12:48:24 2011 +0200| [2215c39e94e01f57b3191a29e0e51d7e230daf49] | committer: Martin Storsjö
flvdec: Add param change side data if the sample rate or channels have changed
Signed-off-by: Martin Storsjö <martin at martin.st>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=2215c39e94e01f57b3191a29e0e51d7e230daf49
---
libavformat/flvdec.c | 31 +++++++++++++++++++++++++------
1 files changed, 25 insertions(+), 6 deletions(-)
diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c
index 2248061..4f1c805 100644
--- a/libavformat/flvdec.c
+++ b/libavformat/flvdec.c
@@ -43,6 +43,8 @@ typedef struct {
int wrong_dts; ///< wrong dts due to negative cts
uint8_t *new_extradata[2];
int new_extradata_size[2];
+ int last_sample_rate;
+ int last_channels;
} FLVContext;
static int flv_probe(AVProbeData *p)
@@ -56,8 +58,7 @@ static int flv_probe(AVProbeData *p)
return 0;
}
-static void flv_set_audio_codec(AVFormatContext *s, AVStream *astream, int flv_codecid) {
- AVCodecContext *acodec = astream->codec;
+static void flv_set_audio_codec(AVFormatContext *s, AVStream *astream, AVCodecContext *acodec, int flv_codecid) {
switch(flv_codecid) {
//no distinction between S16 and S8 PCM codec flags
case FLV_CODECID_PCM:
@@ -440,6 +441,7 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
int ret, i, type, size, flags, is_audio;
int64_t next, pos;
int64_t dts, pts = AV_NOPTS_VALUE;
+ int sample_rate, channels;
AVStream *st = NULL;
for(;;avio_skip(s->pb, 4)){ /* pkt size is repeated at end. skip it */
@@ -524,13 +526,24 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
}
if(is_audio){
+ int bits_per_coded_sample;
+ channels = (flags & FLV_AUDIO_CHANNEL_MASK) == FLV_STEREO ? 2 : 1;
+ sample_rate = (44100 << ((flags & FLV_AUDIO_SAMPLERATE_MASK) >> FLV_AUDIO_SAMPLERATE_OFFSET) >> 3);
+ bits_per_coded_sample = (flags & FLV_AUDIO_SAMPLESIZE_MASK) ? 16 : 8;
if(!st->codec->channels || !st->codec->sample_rate || !st->codec->bits_per_coded_sample) {
- st->codec->channels = (flags & FLV_AUDIO_CHANNEL_MASK) == FLV_STEREO ? 2 : 1;
- st->codec->sample_rate = (44100 << ((flags & FLV_AUDIO_SAMPLERATE_MASK) >> FLV_AUDIO_SAMPLERATE_OFFSET) >> 3);
- st->codec->bits_per_coded_sample = (flags & FLV_AUDIO_SAMPLESIZE_MASK) ? 16 : 8;
+ st->codec->channels = channels;
+ st->codec->sample_rate = sample_rate;
+ st->codec->bits_per_coded_sample = bits_per_coded_sample;
}
if(!st->codec->codec_id){
- flv_set_audio_codec(s, st, flags & FLV_AUDIO_CODECID_MASK);
+ flv_set_audio_codec(s, st, st->codec, flags & FLV_AUDIO_CODECID_MASK);
+ flv->last_sample_rate = st->codec->sample_rate;
+ flv->last_channels = st->codec->channels;
+ } else {
+ AVCodecContext ctx;
+ ctx.sample_rate = sample_rate;
+ flv_set_audio_codec(s, st, &ctx, flags & FLV_AUDIO_CODECID_MASK);
+ sample_rate = ctx.sample_rate;
}
}else{
size -= flv_set_video_codec(s, st, flags & FLV_VIDEO_CODECID_MASK);
@@ -603,6 +616,12 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
flv->new_extradata_size[is_audio] = 0;
}
}
+ if (is_audio && (sample_rate != flv->last_sample_rate ||
+ channels != flv->last_channels)) {
+ flv->last_sample_rate = sample_rate;
+ flv->last_channels = channels;
+ ff_add_param_change(pkt, channels, 0, sample_rate, 0, 0);
+ }
if (is_audio || ((flags & FLV_VIDEO_FRAMETYPE_MASK) == FLV_FRAME_KEY))
pkt->flags |= AV_PKT_FLAG_KEY;
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