Commit 20f01548 authored by Philip Gladstone's avatar Philip Gladstone
Browse files

* 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
parent 7ac13f0c
......@@ -24,7 +24,7 @@
#define PACKET_ID 0x666d
/* each packet contains frames (which can span several packets */
#define FRAME_HEADER_SIZE 5
#define FRAME_HEADER_SIZE 8
#define FLAG_KEY_FRAME 0x01
typedef struct FFMStream {
......@@ -159,16 +159,20 @@ static int ffm_write_header(AVFormatContext *s)
put_be32(pb, (codec->frame_rate * 1000) / FRAME_RATE_BASE);
put_be16(pb, codec->width);
put_be16(pb, codec->height);
put_byte(pb, codec->qmin);
put_byte(pb, codec->qmax);
put_byte(pb, codec->max_qdiff);
put_be16(pb, (int) (codec->qcompress * 10000.0));
put_be16(pb, (int) (codec->qblur * 10000.0));
put_be16(pb, codec->gop_size);
put_byte(pb, codec->qmin);
put_byte(pb, codec->qmax);
put_byte(pb, codec->max_qdiff);
put_be16(pb, (int) (codec->qcompress * 10000.0));
put_be16(pb, (int) (codec->qblur * 10000.0));
break;
case CODEC_TYPE_AUDIO:
put_be32(pb, codec->sample_rate);
put_le16(pb, codec->channels);
put_le16(pb, codec->frame_size);
break;
default:
abort();
}
/* hack to have real time */
fst->pts = gettime();
......@@ -206,6 +210,13 @@ static int ffm_write_packet(AVFormatContext *s, int stream_index,
FFMStream *fst = st->priv_data;
INT64 pts;
UINT8 header[FRAME_HEADER_SIZE];
int duration;
if (st->codec.codec_type == CODEC_TYPE_AUDIO) {
duration = ((float)st->codec.frame_size / st->codec.sample_rate * 1000000.0);
} else {
duration = (1000000.0 * FRAME_RATE_BASE / (float)st->codec.frame_rate);
}
pts = fst->pts;
/* packet size & key_frame */
......@@ -216,14 +227,13 @@ static int ffm_write_packet(AVFormatContext *s, int stream_index,
header[2] = (size >> 16) & 0xff;
header[3] = (size >> 8) & 0xff;
header[4] = size & 0xff;
header[5] = (duration >> 16) & 0xff;
header[6] = (duration >> 8) & 0xff;
header[7] = duration & 0xff;
ffm_write_data(s, header, FRAME_HEADER_SIZE, pts, 1);
ffm_write_data(s, buf, size, pts, 0);
if (st->codec.codec_type == CODEC_TYPE_AUDIO) {
fst->pts += (INT64)((float)st->codec.frame_size / st->codec.sample_rate * 1000000.0);
} else {
fst->pts += (INT64)(1000000.0 * FRAME_RATE_BASE / (float)st->codec.frame_rate);
}
fst->pts += duration;
return 0;
}
......@@ -372,16 +382,20 @@ static int ffm_read_header(AVFormatContext *s, AVFormatParameters *ap)
codec->frame_rate = ((INT64)get_be32(pb) * FRAME_RATE_BASE) / 1000;
codec->width = get_be16(pb);
codec->height = get_be16(pb);
codec->qmin = get_byte(pb);
codec->qmax = get_byte(pb);
codec->max_qdiff = get_byte(pb);
codec->qcompress = get_be16(pb) / 10000.0;
codec->qblur = get_be16(pb) / 10000.0;
codec->gop_size = get_be16(pb);
codec->qmin = get_byte(pb);
codec->qmax = get_byte(pb);
codec->max_qdiff = get_byte(pb);
codec->qcompress = get_be16(pb) / 10000.0;
codec->qblur = get_be16(pb) / 10000.0;
break;
case CODEC_TYPE_AUDIO:
codec->sample_rate = get_be32(pb);
codec->channels = get_le16(pb);
codec->frame_size = get_le16(pb);
break;
default:
abort();
}
}
......@@ -418,6 +432,7 @@ static int ffm_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int size;
FFMContext *ffm = s->priv_data;
int duration;
switch(ffm->read_state) {
case READ_HEADER:
......@@ -446,6 +461,8 @@ static int ffm_read_packet(AVFormatContext *s, AVPacket *pkt)
return -EAGAIN;
}
duration = (ffm->header[5] << 16) | (ffm->header[6] << 8) | ffm->header[7];
av_new_packet(pkt, size);
pkt->stream_index = ffm->header[0];
if (ffm->header[1] & FLAG_KEY_FRAME)
......@@ -457,6 +474,8 @@ static int ffm_read_packet(AVFormatContext *s, AVPacket *pkt)
av_free_packet(pkt);
return -EAGAIN;
}
pkt->pts = ffm->pts;
pkt->duration = duration;
break;
}
return 0;
......
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