Chiebot-Mirror
/
FFmpeg
8
0
Fork 0
Code Issues Pull Requests Projects Releases Wiki Activity

correctly pack and interleave pcm samples in mxf

Browse Source 
Originally committed as revision 16887 to svn://svn.ffmpeg.org/ffmpeg/trunk
pull/126/head
Baptiste Coudurier 16 years ago
parent 34c65ac668
commit cc812c8c7e
1 changed files with 189 additions and 5 deletions
Whitespace
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. 194
      libavformat/mxfenc.c

194
libavformat/mxfenc.c
Unescape View File

@ -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,
};

Write Preview
Loading…
Cancel
Save