comparison rtp.c @ 885:da1d5db0ce5c libavformat

COSMETICS: Remove all trailing whitespace.

884:2ece9c9dd94c

         - add support for h263/mpeg4 packetized output : IDEA: send a
         buffer to 'rtp_write_packet' contains all the packets for ONE
         frame. Each packet should have a four byte header containing
         the length in big endian format (same trick as
         'url_open_dyn_packet_buf') 
*/

/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
AVRtpPayloadType_t AVRtpPayloadTypes[]=
{

    uint32_t cur_timestamp;
    int max_payload_size;
    MpegTSContext *ts; /* only used for MP2T payloads */
    int read_buf_index;
    int read_buf_size;
    
    /* rtcp sender statistics receive */
    int64_t last_rtcp_ntp_time;
    int64_t first_rtcp_ntp_time;
    uint32_t last_rtcp_timestamp;
    /* rtcp sender statistics */

}

/**
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
 * MPEG2TS streams to indicate that they should be demuxed inside the
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) 
 */
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_t *rtp_payload_data)
{
    RTPDemuxContext *s;

    return 0;
}

/**
 * Parse an RTP or RTCP packet directly sent as a buffer. 
 * @param s RTP parse context.
 * @param pkt returned packet
 * @param buf input buffer or NULL to read the next packets
 * @param len buffer len
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow 
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
 */
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 
                     const uint8_t *buf, int len)
{
    unsigned int ssrc, h;
    int payload_type, seq, delta_timestamp, ret;
    AVStream *st;
    uint32_t timestamp;
    
    if (!buf) {
        /* return the next packets, if any */
        if (s->read_buf_index >= s->read_buf_size)
            return -1;
        ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, 
                                  s->read_buf_size - s->read_buf_index);
        if (ret < 0)
            return -1;
        s->read_buf_index += ret;
        if (s->read_buf_index < s->read_buf_size)

    }
    payload_type = buf[1] & 0x7f;
    seq  = (buf[2] << 8) | buf[3];
    timestamp = decode_be32(buf + 4);
    ssrc = decode_be32(buf + 8);
    
    /* NOTE: we can handle only one payload type */
    if (s->payload_type != payload_type)
        return -1;
#if defined(DEBUG) || 1
    if (seq != ((s->seq + 1) & 0xffff)) {
        av_log(s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", 
               payload_type, seq, ((s->seq + 1) & 0xffff));
    }
#endif
    s->seq = seq;
    len -= 12;

        default:
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        }
        
        switch(st->codec->codec_id) {
        case CODEC_ID_MP2:
        case CODEC_ID_MPEG1VIDEO:
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
                int64_t addend;

    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++;
}

            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)

    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 */

            *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;

        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;

            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;
        }

    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) {

885:da1d5db0ce5c

         - add support for h263/mpeg4 packetized output : IDEA: send a
         buffer to 'rtp_write_packet' contains all the packets for ONE
         frame. Each packet should have a four byte header containing
         the length in big endian format (same trick as
         'url_open_dyn_packet_buf')
*/

/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
AVRtpPayloadType_t AVRtpPayloadTypes[]=
{

    uint32_t cur_timestamp;
    int max_payload_size;
    MpegTSContext *ts; /* only used for MP2T payloads */
    int read_buf_index;
    int read_buf_size;

    /* rtcp sender statistics receive */
    int64_t last_rtcp_ntp_time;
    int64_t first_rtcp_ntp_time;
    uint32_t last_rtcp_timestamp;
    /* rtcp sender statistics */

}

/**
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
 * MPEG2TS streams to indicate that they should be demuxed inside the
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
 */
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_t *rtp_payload_data)
{
    RTPDemuxContext *s;

    return 0;
}

/**
 * Parse an RTP or RTCP packet directly sent as a buffer.
 * @param s RTP parse context.
 * @param pkt returned packet
 * @param buf input buffer or NULL to read the next packets
 * @param len buffer len
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
 */
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
                     const uint8_t *buf, int len)
{
    unsigned int ssrc, h;
    int payload_type, seq, delta_timestamp, ret;
    AVStream *st;
    uint32_t timestamp;

    if (!buf) {
        /* return the next packets, if any */
        if (s->read_buf_index >= s->read_buf_size)
            return -1;
        ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
                                  s->read_buf_size - s->read_buf_index);
        if (ret < 0)
            return -1;
        s->read_buf_index += ret;
        if (s->read_buf_index < s->read_buf_size)

    }
    payload_type = buf[1] & 0x7f;
    seq  = (buf[2] << 8) | buf[3];
    timestamp = decode_be32(buf + 4);
    ssrc = decode_be32(buf + 8);

    /* NOTE: we can handle only one payload type */
    if (s->payload_type != payload_type)
        return -1;
#if defined(DEBUG) || 1
    if (seq != ((s->seq + 1) & 0xffff)) {
        av_log(s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
               payload_type, seq, ((s->seq + 1) & 0xffff));
    }
#endif
    s->seq = seq;
    len -= 12;

        default:
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        }

        switch(st->codec->codec_id) {
        case CODEC_ID_MP2:
        case CODEC_ID_MPEG1VIDEO:
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
                int64_t addend;

    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++;
}

            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)

    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 */

            *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;

        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;

            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;
        }

    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) {