[Ffmpeg-cvslog] r5987 - trunk/libavformat/gxfenc.c
bcoudurier
subversion
Sat Aug 12 02:18:59 CEST 2006
Author: bcoudurier
Date: Sat Aug 12 02:18:58 2006
New Revision: 5987
Modified:
trunk/libavformat/gxfenc.c
Log:
use packet dts as correct media field number and use av_interleave_pkt_per_dts
Modified: trunk/libavformat/gxfenc.c
==============================================================================
--- trunk/libavformat/gxfenc.c (original)
+++ trunk/libavformat/gxfenc.c Sat Aug 12 02:18:58 2006
@@ -41,6 +41,7 @@
int p_per_gop;
int b_per_gop;
int closed_gop;
+ int64_t current_dts;
} GXFStreamContext;
typedef struct GXFContext {
@@ -59,7 +60,6 @@
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 @@
}
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 @@
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_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 @@
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 @@
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 @@
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 = {
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