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