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ffmpeg
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use packet dts as correct media field number and use av_interleave_pkt_per_dts 

Originally committed as revision 5987 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier 2006-08-12 00:18:58 +00:00
parent 1ce83a36bf
commit 201f1d4546
1 changed files with 33 additions and 55 deletions
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88
libavformat/gxfenc.c
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@ -41,6 +41,7 @@ typedef struct GXFStreamContext {
int p_per_gop;
int b_per_gop;
int closed_gop;
int64_t current_dts;
} GXFStreamContext;
typedef struct GXFContext {
@ -59,7 +60,6 @@ typedef struct GXFContext {
uint16_t umf_media_size;
int audio_written;
int sample_rate;
int field_number;
int flags;
AVFormatContext *fc;
GXFStreamContext streams[48];
@ -596,6 +596,7 @@ static int gxf_write_header(AVFormatContext *s)
}
sc->track_type = 2;
sc->sample_rate = st->codec->sample_rate;
av_set_pts_info(st, 64, 1, sc->sample_rate);
sc->sample_size = 16;
sc->frame_rate_index = -2;
sc->lines_index = -2;
@ -616,6 +617,7 @@ static int gxf_write_header(AVFormatContext *s)
gxf->flags |= 0x00000040;
}
gxf->sample_rate = sc->sample_rate;
av_set_pts_info(st, 64, 1, sc->sample_rate);
if (gxf_find_lines_index(sc) < 0)
sc->lines_index = -1;
sc->sample_size = st->codec->bit_rate;
@ -698,10 +700,11 @@ static int gxf_parse_mpeg_frame(GXFStreamContext *sc, const uint8_t *buf, int si
static int gxf_write_media_preamble(ByteIOContext *pb, GXFContext *ctx, AVPacket *pkt, int size)
{
GXFStreamContext *sc = &ctx->streams[pkt->stream_index];
int64_t dts = av_rescale(pkt->dts, ctx->sample_rate, sc->sample_rate);
put_byte(pb, sc->media_type);
put_byte(pb, sc->index);
put_be32(pb, ctx->field_number);
put_be32(pb, dts);
if (sc->codec->codec_type == CODEC_TYPE_AUDIO) {
put_be16(pb, 0);
put_be16(pb, size / 2);
@ -723,7 +726,7 @@ static int gxf_write_media_preamble(ByteIOContext *pb, GXFContext *ctx, AVPacket
put_be24(pb, 0);
} else
put_be32(pb, size);
put_be32(pb, ctx->field_number);
put_be32(pb, dts);
put_byte(pb, 1); /* flags */
put_byte(pb, 0); /* reserved */
return 16;
@ -743,8 +746,6 @@ static int gxf_write_media_packet(ByteIOContext *pb, GXFContext *ctx, AVPacket *
gxf_write_media_preamble(pb, ctx, pkt, pkt->size + padding);
put_buffer(pb, pkt->data, pkt->size);
gxf_write_padding(pb, padding);
if (sc->codec->codec_type == CODEC_TYPE_VIDEO)
ctx->field_number += 2;
return updatePacketSize(pb, pos);
}
@ -757,65 +758,42 @@ static int gxf_write_packet(AVFormatContext *s, AVPacket *pkt)
return 0;
}
static int gxf_new_audio_packet(GXFContext *gxf, GXFStreamContext *sc, AVPacket *pkt, int flush)
{
int size = flush ? fifo_size(&sc->audio_buffer, NULL) : GXF_AUDIO_PACKET_SIZE;
if (!size)
return 0;
av_new_packet(pkt, size);
fifo_read(&sc->audio_buffer, pkt->data, size, NULL);
pkt->stream_index = sc->index;
pkt->dts = sc->current_dts;
sc->current_dts += size / 2; /* we only support 16 bit pcm mono for now */
return size;
}
static int gxf_interleave_packet(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
{
AVPacketList *pktl, **next_point, *this_pktl;
GXFContext *gxf = s->priv_data;
GXFStreamContext *sc;
AVPacket new_pkt;
int i;
if (pkt) {
sc = &gxf->streams[pkt->stream_index];
if (sc->codec->codec_type == CODEC_TYPE_AUDIO) {
fifo_write(&sc->audio_buffer, pkt->data, pkt->size, NULL);
} else {
this_pktl = av_mallocz(sizeof(AVPacketList));
this_pktl->pkt = *pkt;
if(pkt->destruct == av_destruct_packet)
pkt->destruct = NULL; // non shared -> must keep original from being freed
else
av_dup_packet(&this_pktl->pkt); //shared -> must dup
next_point = &s->packet_buffer;
while(*next_point){
AVStream *st= s->streams[ pkt->stream_index];
AVStream *st2= s->streams[ (*next_point)->pkt.stream_index];
int64_t left= st2->time_base.num * (int64_t)st ->time_base.den;
int64_t right= st ->time_base.num * (int64_t)st2->time_base.den;
if((*next_point)->pkt.dts * left > pkt->dts * right) //FIXME this can overflow
break;
next_point= &(*next_point)->next;
for (i = 0; i < s->nb_streams; i++) {
if (s->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO) {
GXFStreamContext *sc = &gxf->streams[i];
if (pkt && pkt->stream_index == i) {
fifo_write(&sc->audio_buffer, pkt->data, pkt->size, NULL);
pkt = NULL;
}
this_pktl->next = *next_point;
*next_point = this_pktl;
}
}
if (gxf->audio_written == gxf->audio_tracks) {
if (!s->packet_buffer) {
gxf->audio_written = 0;
return 0;
}
pktl = s->packet_buffer;
*out = pktl->pkt;
s->packet_buffer = pktl->next;
av_freep(&pktl);
return 1;
} else {
for (i = 0; i < s->nb_streams; i++) {
sc = &gxf->streams[i];
if (sc->codec->codec_type == CODEC_TYPE_AUDIO &&
(flush || fifo_size(&sc->audio_buffer, NULL) >= GXF_AUDIO_PACKET_SIZE)) {
int size = flush ? fifo_size(&sc->audio_buffer, NULL) : GXF_AUDIO_PACKET_SIZE;
av_new_packet(out, size);
fifo_read(&sc->audio_buffer, out->data, size, NULL);
gxf->audio_written++;
out->stream_index = i;
return 1;
if (flush || fifo_size(&sc->audio_buffer, NULL) >= GXF_AUDIO_PACKET_SIZE) {
if (gxf_new_audio_packet(gxf, sc, &new_pkt, flush) > 0) {
pkt = &new_pkt;
break; /* add pkt right now into list */
}
}
}
}
av_init_packet(out);
return 0;
return av_interleave_packet_per_dts(s, out, pkt, flush);
}
AVOutputFormat gxf_muxer = {