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* Add the gop_size to the video parameters. Also the audio framesize.
* Copy the duration over as well, though I'm not 100% certain that that is still needed. Originally committed as revision 462 to svn://svn.ffmpeg.org/ffmpeg/trunk
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parent
7ac13f0c7e
commit
20f0154831
51
libav/ffm.c
51
libav/ffm.c
@ -24,7 +24,7 @@
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#define PACKET_ID 0x666d
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/* each packet contains frames (which can span several packets */
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#define FRAME_HEADER_SIZE 5
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#define FRAME_HEADER_SIZE 8
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#define FLAG_KEY_FRAME 0x01
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typedef struct FFMStream {
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@ -159,16 +159,20 @@ static int ffm_write_header(AVFormatContext *s)
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put_be32(pb, (codec->frame_rate * 1000) / FRAME_RATE_BASE);
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put_be16(pb, codec->width);
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put_be16(pb, codec->height);
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put_byte(pb, codec->qmin);
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put_byte(pb, codec->qmax);
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put_byte(pb, codec->max_qdiff);
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put_be16(pb, (int) (codec->qcompress * 10000.0));
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put_be16(pb, (int) (codec->qblur * 10000.0));
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put_be16(pb, codec->gop_size);
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put_byte(pb, codec->qmin);
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put_byte(pb, codec->qmax);
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put_byte(pb, codec->max_qdiff);
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put_be16(pb, (int) (codec->qcompress * 10000.0));
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put_be16(pb, (int) (codec->qblur * 10000.0));
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break;
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case CODEC_TYPE_AUDIO:
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put_be32(pb, codec->sample_rate);
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put_le16(pb, codec->channels);
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put_le16(pb, codec->frame_size);
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break;
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default:
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abort();
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}
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/* hack to have real time */
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fst->pts = gettime();
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@ -206,6 +210,13 @@ static int ffm_write_packet(AVFormatContext *s, int stream_index,
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FFMStream *fst = st->priv_data;
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INT64 pts;
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UINT8 header[FRAME_HEADER_SIZE];
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int duration;
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if (st->codec.codec_type == CODEC_TYPE_AUDIO) {
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duration = ((float)st->codec.frame_size / st->codec.sample_rate * 1000000.0);
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} else {
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duration = (1000000.0 * FRAME_RATE_BASE / (float)st->codec.frame_rate);
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}
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pts = fst->pts;
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/* packet size & key_frame */
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@ -216,14 +227,13 @@ static int ffm_write_packet(AVFormatContext *s, int stream_index,
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header[2] = (size >> 16) & 0xff;
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header[3] = (size >> 8) & 0xff;
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header[4] = size & 0xff;
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header[5] = (duration >> 16) & 0xff;
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header[6] = (duration >> 8) & 0xff;
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header[7] = duration & 0xff;
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ffm_write_data(s, header, FRAME_HEADER_SIZE, pts, 1);
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ffm_write_data(s, buf, size, pts, 0);
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if (st->codec.codec_type == CODEC_TYPE_AUDIO) {
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fst->pts += (INT64)((float)st->codec.frame_size / st->codec.sample_rate * 1000000.0);
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} else {
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fst->pts += (INT64)(1000000.0 * FRAME_RATE_BASE / (float)st->codec.frame_rate);
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}
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fst->pts += duration;
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return 0;
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}
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@ -372,16 +382,20 @@ static int ffm_read_header(AVFormatContext *s, AVFormatParameters *ap)
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codec->frame_rate = ((INT64)get_be32(pb) * FRAME_RATE_BASE) / 1000;
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codec->width = get_be16(pb);
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codec->height = get_be16(pb);
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codec->qmin = get_byte(pb);
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codec->qmax = get_byte(pb);
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codec->max_qdiff = get_byte(pb);
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codec->qcompress = get_be16(pb) / 10000.0;
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codec->qblur = get_be16(pb) / 10000.0;
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codec->gop_size = get_be16(pb);
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codec->qmin = get_byte(pb);
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codec->qmax = get_byte(pb);
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codec->max_qdiff = get_byte(pb);
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codec->qcompress = get_be16(pb) / 10000.0;
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codec->qblur = get_be16(pb) / 10000.0;
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break;
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case CODEC_TYPE_AUDIO:
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codec->sample_rate = get_be32(pb);
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codec->channels = get_le16(pb);
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codec->frame_size = get_le16(pb);
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break;
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default:
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abort();
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}
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}
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@ -418,6 +432,7 @@ static int ffm_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int size;
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FFMContext *ffm = s->priv_data;
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int duration;
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switch(ffm->read_state) {
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case READ_HEADER:
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@ -446,6 +461,8 @@ static int ffm_read_packet(AVFormatContext *s, AVPacket *pkt)
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return -EAGAIN;
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}
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duration = (ffm->header[5] << 16) | (ffm->header[6] << 8) | ffm->header[7];
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av_new_packet(pkt, size);
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pkt->stream_index = ffm->header[0];
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if (ffm->header[1] & FLAG_KEY_FRAME)
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@ -457,6 +474,8 @@ static int ffm_read_packet(AVFormatContext *s, AVPacket *pkt)
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av_free_packet(pkt);
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return -EAGAIN;
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}
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pkt->pts = ffm->pts;
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pkt->duration = duration;
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break;
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}
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return 0;
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