comparison mxfenc.c @ 4400:65adb9e5214f libavformat

extract audio interleaving code from mxf muxer, will be used by gxf and dv

4399:530e55405feb

#include <math.h>
#include <time.h>

#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 };

#define MXF_INDEX_CLUSTER_SIZE 4096
#define KAG_SIZE 512

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
    AVRational time_base;         ///< time base of output audio packets
} AudioInterleaveContext;

typedef struct {
    int local_tag;
    UID uid;
} MXFLocalTagPair;

    }
    sc->codec_ul = mxf_get_mpeg2_codec_ul(st->codec);
    return !!sc->codec_ul;
}

static int ff_audio_interleave_init(AVFormatContext *s,
                                    const int *samples_per_frame,
                                    AVRational time_base)
{
    int i;

    if (!samples_per_frame)
        return -1;

    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;
            aic->time_base = time_base;

            av_fifo_init(&aic->fifo, 100 * *aic->samples);
        }
    }

    return 0;
}

static void ff_audio_interleave_close(AVFormatContext *s)
{
    int i;
    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)
            av_fifo_free(&aic->fifo);
    }
}

static uint64_t mxf_parse_timestamp(time_t timestamp)
{
    struct tm *time = localtime(&timestamp);
    return (uint64_t)(time->tm_year+1900) << 48 |
           (uint64_t)(time->tm_mon+1)     << 40 |

    mxf_free(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, st->time_base, aic->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, AVPacket *pkt, int flush)
{
    AVPacketList *pktl;
    int stream_count = 0;
    int streams[MAX_STREAMS];

        (next->dts == pkt->dts && 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);
        }
        pkt = NULL;
    }

    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, pkt, flush);
}

AVOutputFormat mxf_muxer = {
    "mxf",
    NULL_IF_CONFIG_SMALL("Material eXchange Format"),

4400:65adb9e5214f

#include <math.h>
#include <time.h>

#include "libavutil/fifo.h"
#include "audiointerleave.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 };

#define MXF_INDEX_CLUSTER_SIZE 4096
#define KAG_SIZE 512

typedef struct {
    int local_tag;
    UID uid;
} MXFLocalTagPair;

    }
    sc->codec_ul = mxf_get_mpeg2_codec_ul(st->codec);
    return !!sc->codec_ul;
}

static uint64_t mxf_parse_timestamp(time_t timestamp)
{
    struct tm *time = localtime(&timestamp);
    return (uint64_t)(time->tm_year+1900) << 48 |
           (uint64_t)(time->tm_mon+1)     << 40 |

    mxf_free(s);
    return 0;
}

static int mxf_interleave_get_packet(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
{
    AVPacketList *pktl;
    int stream_count = 0;
    int streams[MAX_STREAMS];

        (next->dts == pkt->dts && sc->order < sc2->order);
}

static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
{
    return ff_audio_interleave(s, out, pkt, flush,
                               mxf_interleave_get_packet, mxf_compare_timestamps);
}

AVOutputFormat mxf_muxer = {
    "mxf",
    NULL_IF_CONFIG_SMALL("Material eXchange Format"),