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synced 2024-12-14 11:14:44 +00:00
use new audio interleaving generic code
Originally committed as revision 17039 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
parent
f1544e79f2
commit
63601677fe
@ -62,7 +62,7 @@ OBJS-$(CONFIG_FRAMECRC_MUXER) += framecrcenc.o
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OBJS-$(CONFIG_GIF_MUXER) += gif.o
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OBJS-$(CONFIG_GSM_DEMUXER) += raw.o
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OBJS-$(CONFIG_GXF_DEMUXER) += gxf.o
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OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o
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OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o audiointerleave.o
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OBJS-$(CONFIG_H261_DEMUXER) += raw.o
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OBJS-$(CONFIG_H261_MUXER) += raw.o
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OBJS-$(CONFIG_H263_DEMUXER) += raw.o
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@ -23,12 +23,13 @@
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#include "avformat.h"
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#include "gxf.h"
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#include "riff.h"
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#include "audiointerleave.h"
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#define GXF_AUDIO_PACKET_SIZE 65536
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typedef struct GXFStreamContext {
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AudioInterleaveContext aic;
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AVCodecContext *codec;
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AVFifoBuffer audio_buffer;
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uint32_t track_type;
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uint32_t sample_size;
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uint32_t sample_rate;
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@ -587,6 +588,8 @@ static int gxf_write_umf_packet(ByteIOContext *pb, GXFContext *ctx)
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#define GXF_NODELAY -5000
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static const int GXF_samples_per_frame[] = { 32768, 0 };
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static int gxf_write_header(AVFormatContext *s)
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{
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ByteIOContext *pb = s->pb;
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@ -627,7 +630,6 @@ static int gxf_write_header(AVFormatContext *s)
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sc->fields = -2;
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gxf->audio_tracks++;
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gxf->flags |= 0x04000000; /* audio is 16 bit pcm */
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av_fifo_init(&sc->audio_buffer, 3*GXF_AUDIO_PACKET_SIZE);
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} else if (sc->codec->codec_type == CODEC_TYPE_VIDEO) {
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/* FIXME check from time_base ? */
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if (sc->codec->height == 480 || sc->codec->height == 512) { /* NTSC or NTSC+VBI */
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@ -670,6 +672,10 @@ static int gxf_write_header(AVFormatContext *s)
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}
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}
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}
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if (ff_audio_interleave_init(s, GXF_samples_per_frame, (AVRational){ 1, 48000 }) < 0)
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return -1;
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gxf_write_map_packet(pb, gxf);
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//gxf_write_flt_packet(pb, gxf);
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gxf_write_umf_packet(pb, gxf);
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@ -690,13 +696,8 @@ static int gxf_write_trailer(AVFormatContext *s)
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ByteIOContext *pb = s->pb;
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GXFContext *gxf = s->priv_data;
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int64_t end;
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int i;
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for (i = 0; i < s->nb_streams; ++i) {
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AVStream *st = s->streams[i];
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if (st->codec->codec_type == CODEC_TYPE_AUDIO)
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av_fifo_free(&((GXFStreamContext*)st->priv_data)->audio_buffer);
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}
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ff_audio_interleave_close(s);
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gxf_write_eos_packet(pb, gxf);
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end = url_ftell(pb);
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@ -786,47 +787,10 @@ static int gxf_write_packet(AVFormatContext *s, AVPacket *pkt)
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return 0;
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}
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static int gxf_new_audio_packet(GXFContext *gxf, GXFStreamContext *sc, AVPacket *pkt, int flush)
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{
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int size = flush ? av_fifo_size(&sc->audio_buffer) : GXF_AUDIO_PACKET_SIZE;
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if (!size)
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return 0;
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av_new_packet(pkt, size);
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av_fifo_read(&sc->audio_buffer, pkt->data, size);
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pkt->stream_index = sc->index;
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pkt->dts = sc->current_dts;
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sc->current_dts += size / 2; /* we only support 16 bit pcm mono for now */
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return size;
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}
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static int gxf_interleave_packet(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
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{
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GXFContext *gxf = s->priv_data;
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AVPacket new_pkt;
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int i;
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for (i = 0; i < s->nb_streams; i++) {
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AVStream *st = s->streams[i];
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GXFStreamContext *sc = st->priv_data;
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if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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if (pkt && pkt->stream_index == i) {
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av_fifo_generic_write(&sc->audio_buffer, pkt->data, pkt->size, NULL);
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pkt = NULL;
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}
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if (flush || av_fifo_size(&sc->audio_buffer) >= GXF_AUDIO_PACKET_SIZE) {
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if (!pkt && gxf_new_audio_packet(gxf, sc, &new_pkt, flush) > 0) {
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pkt = &new_pkt;
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break; /* add pkt right now into list */
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}
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}
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} else if (pkt && pkt->stream_index == i) {
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if (sc->dts_delay == GXF_NODELAY) /* adjust dts if needed */
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sc->dts_delay = pkt->dts;
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pkt->dts -= sc->dts_delay;
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}
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}
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return av_interleave_packet_per_dts(s, out, pkt, flush);
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return ff_audio_interleave(s, out, pkt, flush,
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av_interleave_packet_per_dts, ff_interleave_compare_dts);
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}
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AVOutputFormat gxf_muxer = {
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