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correctly pack and interleave pcm samples in mxf

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Originally committed as revision 16887 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier 2009-01-31 06:18:25 +00:00
parent 34c65ac668
commit cc812c8c7e
1 changed files with 189 additions and 5 deletions
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194
libavformat/mxfenc.c
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@ -31,18 +31,33 @@
//#define DEBUG
#include "libavutil/fifo.h"
#include "mxf.h"
static const int NTSC_samples_per_frame[] = { 1602, 1601, 1602, 1601, 1602, 0 };
static const int PAL_samples_per_frame[] = { 1920, 0 };
typedef struct {
AVFifoBuffer fifo;
unsigned fifo_size; ///< current fifo size allocated
uint64_t dts; ///< current dts
int sample_size; ///< size of one sample all channels included
const int *samples_per_frame; ///< must be 0 terminated
const int *samples; ///< current samples per frame, pointer to samples_per_frame
} AudioInterleaveContext;
typedef struct {
int local_tag;
UID uid;
} MXFLocalTagPair;
typedef struct {
AudioInterleaveContext aic;
UID track_essence_element_key;
int index; //<<< index in mxf_essence_container_uls table
const UID *codec_ul;
int64_t duration;
int order; ///< interleaving order if dts are equal
} MXFStreamContext;
typedef struct {
@ -75,6 +90,7 @@ typedef struct MXFContext {
int64_t footer_partition_offset;
int essence_container_count;
uint8_t essence_containers_indices[FF_ARRAY_ELEMS(mxf_essence_container_uls)];
AVRational time_base;
} MXFContext;
static const uint8_t uuid_base[] = { 0xAD,0xAB,0x44,0x24,0x2f,0x25,0x4d,0xc7,0x92,0xff,0x29,0xbd };
@ -781,11 +797,40 @@ static const UID *mxf_get_mpeg2_codec_ul(AVCodecContext *avctx)
return NULL;
}
static int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame)
{
int i;
if (!samples_per_frame)
samples_per_frame = PAL_samples_per_frame;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
aic->sample_size = (st->codec->channels *
av_get_bits_per_sample(st->codec->codec_id)) / 8;
if (!aic->sample_size) {
av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
return -1;
}
aic->samples_per_frame = samples_per_frame;
aic->samples = aic->samples_per_frame;
av_fifo_init(&aic->fifo, 100 * *aic->samples);
}
}
return 0;
}
static int mxf_write_header(AVFormatContext *s)
{
MXFContext *mxf = s->priv_data;
int i;
uint8_t present[FF_ARRAY_ELEMS(mxf_essence_container_uls)] = {0};
const int *samples_per_frame = NULL;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
@ -793,11 +838,24 @@ static int mxf_write_header(AVFormatContext *s)
if (!sc)
return AVERROR(ENOMEM);
st->priv_data = sc;
// set pts information
if (st->codec->codec_type == CODEC_TYPE_VIDEO)
av_set_pts_info(st, 64, 1, st->codec->time_base.den);
else if (st->codec->codec_type == CODEC_TYPE_AUDIO)
av_set_pts_info(st, 64, 1, st->codec->sample_rate);
if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
if (!av_cmp_q(st->codec->time_base, (AVRational){ 1, 25 })) {
samples_per_frame = PAL_samples_per_frame;
mxf->time_base = (AVRational){ 1, 25 };
} else if (!av_cmp_q(st->codec->time_base, (AVRational){ 1001, 30000 })) {
samples_per_frame = NTSC_samples_per_frame;
mxf->time_base = (AVRational){ 1001, 30000 };
} else {
av_log(s, AV_LOG_ERROR, "unsupported video frame rate\n");
return -1;
}
} else if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
if (st->codec->sample_rate != 48000) {
av_log(s, AV_LOG_ERROR, "only 48khz is implemented\n");
return -1;
}
}
sc->duration = -1;
sc->index = mxf_get_essence_container_ul_index(st->codec->codec_id);
@ -832,6 +890,17 @@ static int mxf_write_header(AVFormatContext *s)
PRINT_KEY(s, "track essence element key", sc->track_essence_element_key);
}
for (i = 0; i < s->nb_streams; i++) {
MXFStreamContext *sc = s->streams[i]->priv_data;
av_set_pts_info(s->streams[i], 64, mxf->time_base.num, mxf->time_base.den);
// update element count
sc->track_essence_element_key[13] = present[sc->index];
sc->order = AV_RB32(sc->track_essence_element_key+12);
}
if (ff_audio_interleave_init(s, samples_per_frame) < 0)
return -1;
mxf_write_partition(s, 1, header_open_partition_key, 1);
return 0;
@ -868,6 +937,118 @@ static int mxf_write_footer(AVFormatContext *s)
return 0;
}
static int mxf_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
int stream_index, int flush)
{
AVStream *st = s->streams[stream_index];
AudioInterleaveContext *aic = st->priv_data;
int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
return 0;
av_new_packet(pkt, size);
av_fifo_read(&aic->fifo, pkt->data, size);
pkt->dts = pkt->pts = aic->dts;
pkt->duration = av_rescale_q(*aic->samples,
(AVRational){ 1, st->codec->sample_rate },
st->time_base);
pkt->stream_index = stream_index;
aic->dts += pkt->duration;
aic->samples++;
if (!*aic->samples)
aic->samples = aic->samples_per_frame;
return size;
}
static int mxf_interleave_get_packet(AVFormatContext *s, AVPacket *out, int flush)
{
AVPacketList *pktl;
int stream_count = 0;
int streams[MAX_STREAMS];
memset(streams, 0, sizeof(streams));
pktl = s->packet_buffer;
while (pktl) {
//av_log(s, AV_LOG_DEBUG, "show st:%d dts:%lld\n", pktl->pkt.stream_index, pktl->pkt.dts);
if (!streams[pktl->pkt.stream_index])
stream_count++;
streams[pktl->pkt.stream_index]++;
pktl = pktl->next;
}
if (stream_count && (s->nb_streams == stream_count || flush)) {
pktl = s->packet_buffer;
*out = pktl->pkt;
//av_log(s, AV_LOG_DEBUG, "out st:%d dts:%lld\n", (*out).stream_index, (*out).dts);
s->packet_buffer = pktl->next;
av_freep(&pktl);
if (flush && stream_count < s->nb_streams) {
// purge packet queue
pktl = s->packet_buffer;
while (pktl) {
AVPacketList *next = pktl->next;
av_free_packet(&pktl->pkt);
av_freep(&pktl);
pktl = next;
}
s->packet_buffer = NULL;
}
return 1;
} else {
av_init_packet(out);
return 0;
}
}
static int mxf_compare_timestamps(AVFormatContext *s, AVPacket *next, AVPacket *pkt)
{
AVStream *st = s->streams[pkt ->stream_index];
AVStream *st2 = s->streams[next->stream_index];
MXFStreamContext *sc = st ->priv_data;
MXFStreamContext *sc2 = st2->priv_data;
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;
return next->dts * left > pkt->dts * right || // FIXME this can overflow
(next->dts * left == pkt->dts * right && sc->order < sc2->order);
}
static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
{
int i;
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
} else {
// rewrite pts and dts to be decoded time line position
pkt->pts = pkt->dts = aic->dts;
aic->dts += pkt->duration;
ff_interleave_add_packet(s, pkt, mxf_compare_timestamps);
}
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
AVPacket new_pkt;
while (mxf_interleave_new_audio_packet(s, &new_pkt, i, flush))
ff_interleave_add_packet(s, &new_pkt, mxf_compare_timestamps);
}
}
return mxf_interleave_get_packet(s, out, flush);
}
AVOutputFormat mxf_muxer = {
"mxf",
NULL_IF_CONFIG_SMALL("Material eXchange Format"),
@ -879,6 +1060,9 @@ AVOutputFormat mxf_muxer = {
mxf_write_header,
mxf_write_packet,
mxf_write_footer,
0,
NULL,
mxf_interleave,
};