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https://github.com/librempeg/librempeg
synced 2024-11-23 11:39:47 +00:00
flv fixes
Originally committed as revision 2863 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
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c7516a000c
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@ -57,6 +57,8 @@ static int flv_read_header(AVFormatContext *s,
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return AVERROR_NOMEM;
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st->codec.codec_type = CODEC_TYPE_VIDEO;
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st->codec.codec_id = CODEC_ID_FLV1;
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st->codec.frame_rate= ap->frame_rate;
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st->codec.frame_rate_base= ap->frame_rate_base;
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}
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if ((flags & 4)) {
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@ -83,6 +85,7 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
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type = get_byte(&s->pb);
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size = get_be24(&s->pb);
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pts = get_be24(&s->pb);
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// av_log(s, AV_LOG_DEBUG, "type:%d, size:%d, pts:%d\n", type, size, pts);
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if (url_feof(&s->pb))
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return -EIO;
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url_fskip(&s->pb, 4); /* reserved */
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@ -37,7 +37,7 @@ typedef struct FLVContext {
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int audioInPos;
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int audioOutPos;
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int audioSize;
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int audioRate;
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// int audioRate;
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int initDelay;
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int soundDelay;
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uint8_t *audioFifo;
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@ -98,31 +98,32 @@ static int mp3info(void *data, int *byteSize, int *samplesPerFrame, int *sampleR
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}
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if ( !isPadded ) {
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printf("Fatal error: mp3 data is not padded!\n");
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exit(0);
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// printf("Fatal error: mp3 data is not padded!\n");
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// exit(0);
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}
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*isMono = ((header >> 6) & 0x03) == 0x03;
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if ( (header >> 19 ) & 0x01 ) {
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//MPEG1
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*sampleRate = sSampleRates[0][sampleRateID];
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bitRate = sBitRates[0][layerID][bitRateID] * 1000;
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*samplesPerFrame = sSamplesPerFrame[0][layerID];
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} else {
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if ( (header >> 20) & 0x01 ) {
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//MPEG2
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*sampleRate = sSampleRates[1][sampleRateID];
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bitRate = sBitRates[1][layerID][bitRateID] * 1000;
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*samplesPerFrame = sSamplesPerFrame[1][layerID];
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} else {
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//MPEG2.5
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*sampleRate = sSampleRates[2][sampleRateID];
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bitRate = sBitRates[1][layerID][bitRateID] * 1000;
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*samplesPerFrame = sSamplesPerFrame[2][layerID];
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}
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}
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*byteSize = ( ( ( ( *samplesPerFrame * (bitRate / bitsPerSlot) ) / *sampleRate ) + isPadded ) );
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return 1;
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}
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#endif // CONFIG_MP3LAME
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@ -143,7 +144,7 @@ static int flv_write_header(AVFormatContext *s)
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flv->audioInPos = 0;
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flv->audioOutPos = 0;
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flv->audioSize = 0;
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flv->audioRate = 44100;
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// flv->audioRate = 44100;
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flv->initDelay = -1;
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flv->soundDelay = 0;
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#endif // CONFIG_MP3LAME
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@ -270,6 +271,7 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
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#ifdef CONFIG_MP3LAME
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if (enc->codec_id == CODEC_ID_MP3 ) {
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int c=0;
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for (;c<size;c++) {
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flv->audioFifo[(flv->audioOutPos+c)%AUDIO_FIFO_SIZE] = buf[c];
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}
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@ -281,13 +283,13 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
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flv->initDelay = timestamp;
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}
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if ( flv->audioTime == -1 ) {
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// if ( flv->audioTime == -1 ) {
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flv->audioTime = timestamp;
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// flv->audioTime = ( ( ( flv->sampleCount - enc->delay ) * 8000 ) / flv->audioRate ) - flv->initDelay - 250;
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// if ( flv->audioTime < 0 ) {
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// flv->audioTime = 0;
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// }
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}
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// }
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}
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for ( ; flv->audioSize >= 4 ; ) {
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@ -310,7 +312,7 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
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int c=0;
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FLVFrame *frame = av_malloc(sizeof(FLVFrame));
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flv->audioRate = mp3SampleRate;
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// flv->audioRate = mp3SampleRate;
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switch (mp3SampleRate) {
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case 44100:
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@ -332,6 +334,7 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
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frame->type = 8;
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frame->flags = soundFormat;
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frame->timestamp = flv->audioTime;
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frame->timestamp = (1000*flv->sampleCount + mp3SampleRate/2)/(mp3SampleRate);
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frame->size = mp3FrameSize;
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frame->data = av_malloc(mp3FrameSize);
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@ -344,15 +347,18 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
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flv->audioInPos %= AUDIO_FIFO_SIZE;
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flv->sampleCount += mp3SamplesPerFrame;
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// Reset audio for next round
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flv->audioTime = -1;
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flv->audioTime += 1000*mp3SamplesPerFrame/mp3SampleRate;
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// We got audio! Make sure we set this to the global flags on closure
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flv->hasAudio = 1;
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InsertSorted(flv,frame);
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// av_log(NULL,AV_LOG_DEBUG, "insert sound\n");
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continue;
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}
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// av_log(NULL,AV_LOG_DEBUG, "insuficent data\n");
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break;
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}
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av_log(NULL,AV_LOG_DEBUG, "head trashed\n");
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flv->audioInPos ++;
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flv->audioSize --;
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flv->audioInPos %= AUDIO_FIFO_SIZE;
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