rtp.c

ffmpeg源码分析

static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
    RTPDemuxContext *s = s1->priv_data;

#ifdef DEBUG
    printf("rtp_send_data size=%d\n", len);
#endif

    /* build the RTP header */
    put_byte(&s1->pb, (RTP_VERSION << 6));
    put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
    put_be16(&s1->pb, s->seq);
    put_be32(&s1->pb, s->timestamp);
    put_be32(&s1->pb, s->ssrc);

    put_buffer(&s1->pb, buf1, len);
    put_flush_packet(&s1->pb);

    s->seq++;
    s->octet_count += len;
    s->packet_count++;
}

/* send an integer number of samples and compute time stamp and fill
   the rtp send buffer before sending. */
static void rtp_send_samples(AVFormatContext *s1,
                             const uint8_t *buf1, int size, int sample_size)
{
    RTPDemuxContext *s = s1->priv_data;
    int len, max_packet_size, n;

    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
    /* not needed, but who nows */
    if ((size % sample_size) != 0)
        av_abort();
    while (size > 0) {
        len = (max_packet_size - (s->buf_ptr - s->buf));
        if (len > size)
            len = size;

        /* copy data */
        memcpy(s->buf_ptr, buf1, len);
        s->buf_ptr += len;
        buf1 += len;
        size -= len;
        n = (s->buf_ptr - s->buf);
        /* if buffer full, then send it */
        if (n >= max_packet_size) {
            rtp_send_data(s1, s->buf, n, 0);
            s->buf_ptr = s->buf;
            /* update timestamp */
            s->timestamp += n / sample_size;
        }
    }
}

/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
static void rtp_send_mpegaudio(AVFormatContext *s1,
                               const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int len, count, max_packet_size;

    max_packet_size = s->max_payload_size;

    /* test if we must flush because not enough space */
    len = (s->buf_ptr - s->buf);
    if ((len + size) > max_packet_size) {
        if (len > 4) {
            rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
            s->buf_ptr = s->buf + 4;
            /* 90 KHz time stamp */
            s->timestamp = s->base_timestamp +
                (s->cur_timestamp * 90000LL) / st->codec->sample_rate;
        }
    }

    /* add the packet */
    if (size > max_packet_size) {
        /* big packet: fragment */
        count = 0;
        while (size > 0) {
            len = max_packet_size - 4;
            if (len > size)
                len = size;
            /* build fragmented packet */
            s->buf[0] = 0;
            s->buf[1] = 0;
            s->buf[2] = count >> 8;
            s->buf[3] = count;
            memcpy(s->buf + 4, buf1, len);
            rtp_send_data(s1, s->buf, len + 4, 0);
            size -= len;
            buf1 += len;
            count += len;
        }
    } else {
        if (s->buf_ptr == s->buf + 4) {
            /* no fragmentation possible */
            s->buf[0] = 0;
            s->buf[1] = 0;
            s->buf[2] = 0;
            s->buf[3] = 0;
        }
        memcpy(s->buf_ptr, buf1, size);
        s->buf_ptr += size;
    }
    s->cur_timestamp += st->codec->frame_size;
}

/* NOTE: a single frame must be passed with sequence header if
   needed. XXX: use slices. */
static void rtp_send_mpegvideo(AVFormatContext *s1,
                               const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int len, h, max_packet_size;
    uint8_t *q;

    max_packet_size = s->max_payload_size;

    while (size > 0) {
        /* XXX: more correct headers */
        h = 0;
        if (st->codec->sub_id == 2)
            h |= 1 << 26; /* mpeg 2 indicator */
        q = s->buf;
        *q++ = h >> 24;
        *q++ = h >> 16;
        *q++ = h >> 8;
        *q++ = h;

        if (st->codec->sub_id == 2) {
            h = 0;
            *q++ = h >> 24;
            *q++ = h >> 16;
            *q++ = h >> 8;
            *q++ = h;
        }

        len = max_packet_size - (q - s->buf);
        if (len > size)
            len = size;

        memcpy(q, buf1, len);
        q += len;

        /* 90 KHz time stamp */
        s->timestamp = s->base_timestamp +
            av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
        rtp_send_data(s1, s->buf, q - s->buf, (len == size));

        buf1 += len;
        size -= len;
    }
    s->cur_timestamp++;
}

static void rtp_send_raw(AVFormatContext *s1,
                         const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int len, max_packet_size;

    max_packet_size = s->max_payload_size;

    while (size > 0) {
        len = max_packet_size;
        if (len > size)
            len = size;

        /* 90 KHz time stamp */
        s->timestamp = s->base_timestamp +
            av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
        rtp_send_data(s1, buf1, len, (len == size));

        buf1 += len;
        size -= len;
    }
    s->cur_timestamp++;
}

/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
static void rtp_send_mpegts_raw(AVFormatContext *s1,
                                const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    int len, out_len;

    while (size >= TS_PACKET_SIZE) {
        len = s->max_payload_size - (s->buf_ptr - s->buf);
        if (len > size)
            len = size;
        memcpy(s->buf_ptr, buf1, len);
        buf1 += len;
        size -= len;
        s->buf_ptr += len;

        out_len = s->buf_ptr - s->buf;
        if (out_len >= s->max_payload_size) {
            rtp_send_data(s1, s->buf, out_len, 0);
            s->buf_ptr = s->buf;
        }
    }
}

/* write an RTP packet. 'buf1' must contain a single specific frame. */
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
    RTPDemuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int rtcp_bytes;
    int64_t ntp_time;
    int size= pkt->size;
    uint8_t *buf1= pkt->data;

#ifdef DEBUG
    printf("%d: write len=%d\n", pkt->stream_index, size);
#endif

    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
        RTCP_TX_RATIO_DEN;
    if (s->first_packet || rtcp_bytes >= 28) {
        /* compute NTP time */
        /* XXX: 90 kHz timestamp hardcoded */
        ntp_time = (pkt->pts << 28) / 5625;
        rtcp_send_sr(s1, ntp_time);
        s->last_octet_count = s->octet_count;
        s->first_packet = 0;
    }

    switch(st->codec->codec_id) {
    case CODEC_ID_PCM_MULAW:
    case CODEC_ID_PCM_ALAW:
    case CODEC_ID_PCM_U8:
    case CODEC_ID_PCM_S8:
        rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
        break;
    case CODEC_ID_PCM_U16BE:
    case CODEC_ID_PCM_U16LE:
    case CODEC_ID_PCM_S16BE:
    case CODEC_ID_PCM_S16LE:
        rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
        break;
    case CODEC_ID_MP2:
    case CODEC_ID_MP3:
        rtp_send_mpegaudio(s1, buf1, size);
        break;
    case CODEC_ID_MPEG1VIDEO:
        rtp_send_mpegvideo(s1, buf1, size);
        break;
    case CODEC_ID_MPEG2TS:
        rtp_send_mpegts_raw(s1, buf1, size);
        break;
    default:
        /* better than nothing : send the codec raw data */
        rtp_send_raw(s1, buf1, size);
        break;
    }
    return 0;
}

static int rtp_write_trailer(AVFormatContext *s1)
{
    //    RTPDemuxContext *s = s1->priv_data;
    return 0;
}

AVOutputFormat rtp_mux = {
    "rtp",
    "RTP output format",
    NULL,
    NULL,
    sizeof(RTPDemuxContext),
    CODEC_ID_PCM_MULAW,
    CODEC_ID_NONE,
    rtp_write_header,
    rtp_write_packet,
    rtp_write_trailer,
};

int rtp_init(void)
{
    av_register_output_format(&rtp_mux);
    return 0;
}