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use new audio interleaving generic code

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Originally committed as revision 17039 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier 2009-02-08 04:33:53 +00:00
parent f1544e79f2
commit 63601677fe
2 changed files with 12 additions and 48 deletions
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2
libavformat/Makefile
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@ -62,7 +62,7 @@ OBJS-$(CONFIG_FRAMECRC_MUXER) += framecrcenc.o
OBJS-$(CONFIG_GIF_MUXER) += gif.o
OBJS-$(CONFIG_GSM_DEMUXER) += raw.o
OBJS-$(CONFIG_GXF_DEMUXER) += gxf.o
OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o
OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o audiointerleave.o
OBJS-$(CONFIG_H261_DEMUXER) += raw.o
OBJS-$(CONFIG_H261_MUXER) += raw.o
OBJS-$(CONFIG_H263_DEMUXER) += raw.o

58
libavformat/gxfenc.c
View File

@ -23,12 +23,13 @@
#include "avformat.h"
#include "gxf.h"
#include "riff.h"
#include "audiointerleave.h"
#define GXF_AUDIO_PACKET_SIZE 65536
typedef struct GXFStreamContext {
AudioInterleaveContext aic;
AVCodecContext *codec;
AVFifoBuffer audio_buffer;
uint32_t track_type;
uint32_t sample_size;
uint32_t sample_rate;
@ -587,6 +588,8 @@ static int gxf_write_umf_packet(ByteIOContext *pb, GXFContext *ctx)
#define GXF_NODELAY -5000
static const int GXF_samples_per_frame[] = { 32768, 0 };
static int gxf_write_header(AVFormatContext *s)
{
ByteIOContext *pb = s->pb;
@ -627,7 +630,6 @@ static int gxf_write_header(AVFormatContext *s)
sc->fields = -2;
gxf->audio_tracks++;
gxf->flags |= 0x04000000; /* audio is 16 bit pcm */
av_fifo_init(&sc->audio_buffer, 3*GXF_AUDIO_PACKET_SIZE);
} else if (sc->codec->codec_type == CODEC_TYPE_VIDEO) {
/* FIXME check from time_base ? */
if (sc->codec->height == 480 || sc->codec->height == 512) { /* NTSC or NTSC+VBI */
@ -670,6 +672,10 @@ static int gxf_write_header(AVFormatContext *s)
}
}
}
if (ff_audio_interleave_init(s, GXF_samples_per_frame, (AVRational){ 1, 48000 }) < 0)
return -1;
gxf_write_map_packet(pb, gxf);
//gxf_write_flt_packet(pb, gxf);
gxf_write_umf_packet(pb, gxf);
@ -690,13 +696,8 @@ static int gxf_write_trailer(AVFormatContext *s)
ByteIOContext *pb = s->pb;
GXFContext *gxf = s->priv_data;
int64_t end;
int i;
for (i = 0; i < s->nb_streams; ++i) {
AVStream *st = s->streams[i];
if (st->codec->codec_type == CODEC_TYPE_AUDIO)
av_fifo_free(&((GXFStreamContext*)st->priv_data)->audio_buffer);
}
ff_audio_interleave_close(s);
gxf_write_eos_packet(pb, gxf);
end = url_ftell(pb);
@ -786,47 +787,10 @@ 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 ? av_fifo_size(&sc->audio_buffer) : GXF_AUDIO_PACKET_SIZE;
if (!size)
return 0;
av_new_packet(pkt, size);
av_fifo_read(&sc->audio_buffer, pkt->data, size);
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)
{
GXFContext *gxf = s->priv_data;
AVPacket new_pkt;
int i;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
GXFStreamContext *sc = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
if (pkt && pkt->stream_index == i) {
av_fifo_generic_write(&sc->audio_buffer, pkt->data, pkt->size, NULL);
pkt = NULL;
}
if (flush || av_fifo_size(&sc->audio_buffer) >= GXF_AUDIO_PACKET_SIZE) {
if (!pkt && gxf_new_audio_packet(gxf, sc, &new_pkt, flush) > 0) {
pkt = &new_pkt;
break; /* add pkt right now into list */
}
}
} else if (pkt && pkt->stream_index == i) {
if (sc->dts_delay == GXF_NODELAY) /* adjust dts if needed */
sc->dts_delay = pkt->dts;
pkt->dts -= sc->dts_delay;
}
}
return av_interleave_packet_per_dts(s, out, pkt, flush);
return ff_audio_interleave(s, out, pkt, flush,
av_interleave_packet_per_dts, ff_interleave_compare_dts);
}
AVOutputFormat gxf_muxer = {