correct pcm in flv handling

Originally committed as revision 3968 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Alex Beregszaszi 2005-02-21 18:05:21 +00:00
parent 58aa2b1d35
commit 923bd441fe
2 changed files with 18 additions and 5 deletions

View File

@ -60,7 +60,7 @@ static int flv_read_header(AVFormatContext *s,
static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret, i, type, size, pts, flags, is_audio;
AVStream *st;
AVStream *st = NULL;
for(;;){
url_fskip(&s->pb, 4); /* size of previous packet */
@ -122,7 +122,12 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
else
st->codec.sample_rate = (44100<<((flags>>2)&3))>>3;
switch(flags >> 4){/* 0: uncompressed 1: ADPCM 2: mp3 5: Nellymoser 8kHz mono 6: Nellymoser*/
case 0: if (flags&2) st->codec.codec_id = CODEC_ID_PCM_S16BE;
else st->codec.codec_id = CODEC_ID_PCM_S8; break;
case 2: st->codec.codec_id = CODEC_ID_MP3; break;
// this is not listed at FLV but at SWF, strange...
case 3: if (flags&2) st->codec.codec_id = CODEC_ID_PCM_S16LE;
else st->codec.codec_id = CODEC_ID_PCM_S8; break;
default:
st->codec.codec_tag= (flags >> 4);
}

View File

@ -35,7 +35,7 @@ static void put_be24(ByteIOContext *pb, int value)
}
static int get_audio_flags(AVCodecContext *enc){
int flags = 0x02;
int flags = 0;
switch (enc->sample_rate) {
case 44100:
@ -61,8 +61,16 @@ static int get_audio_flags(AVCodecContext *enc){
switch(enc->codec_id){
case CODEC_ID_MP3:
flags |= 0x20;
flags |= 0x20 | 0x2;
break;
case CODEC_ID_PCM_S8:
break;
case CODEC_ID_PCM_S16BE:
flags |= 0x60 | 0x2;
break;
case CODEC_ID_PCM_S16LE:
flags |= 0x2;
break;
case 0:
flags |= enc->codec_tag<<4;
break;
@ -155,7 +163,7 @@ static int flv_write_packet(AVFormatContext *s, AVPacket *pkt)
put_be32(pb,flv->reserved);
put_byte(pb,flags);
put_buffer(pb, pkt->data, size);
put_be32(pb,size+1+11); // reserved
put_be32(pb,size+1+11); // previous tag size
put_flush_packet(pb);
return 0;