[FFmpeg-devel] a bug about RTSP_PROTOCOL_RTP_TCP

[连海东]

Hi:
      When we use ffserver to transer video stream over
RTP(RTSP_PROTOCOL_RTP_TCP), we found that rtcp packet don't transfer
correctly.  Please have look at the code below.




/* should convert the format at the same time */
/* send data starting at c->buffer_ptr to the output connection
   (either UDP or TCP connection) */
static int http_send_data(HTTPContext *c)
{
    int len, ret;

    for(;;) {
        if (c->buffer_ptr >= c->buffer_end) {
            ret = http_prepare_data(c);
            if (ret < 0)
                return -1;
            else if (ret != 0)
                /* state change requested */
                break;
        } else {
            if (c->is_packetized) {
                /* RTP data output */
                len = c->buffer_end - c->buffer_ptr;
                if (len < 4) {
                    /* fail safe - should never happen */
                fail1:
                    c->buffer_ptr = c->buffer_end;
                    return 0;
                }
                len = (c->buffer_ptr[0] << 24) |
                    (c->buffer_ptr[1] << 16) |
                    (c->buffer_ptr[2] << 8) |
                    (c->buffer_ptr[3]);
                if (len > (c->buffer_end - c->buffer_ptr))
                    goto fail1;
                if ((get_packet_send_clock(c) - get_server_clock(c)) > 0) {
                    /* nothing to send yet: we can wait */
                    return 0;
                }

                c->data_count += len;
                update_datarate(&c->datarate, c->data_count);
                if (c->stream)
                    c->stream->bytes_served += len;

                if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP) {
                    /* RTP packets are sent inside the RTSP TCP connection
*/
                    ByteIOContext *pb;
                    int interleaved_index, size;
                    uint8_t header[4];
                    HTTPContext *rtsp_c;

                    rtsp_c = c->rtsp_c;
                    /* if no RTSP connection left, error */
                    if (!rtsp_c)
                        return -1;
                    /* if already sending something, then wait. */
                    if (rtsp_c->state != RTSPSTATE_WAIT_REQUEST)
                        break;
                    if (url_open_dyn_buf(&pb) < 0)
                        goto fail1;
                    interleaved_index = c->packet_stream_index * 2;
                    /* RTCP packets are sent at odd indexes */
                    if (c->buffer_ptr[1] == 200)
------------------------------------------>  should be  c->buffer_ptr[5] ==
200
                        interleaved_index++;
                    /* write RTSP TCP header */
                    header[0] = '$';
                    header[1] = interleaved_index;
                    header[2] = len >> 8;
                    header[3] = len;
                    put_buffer(pb, header, 4);
                    /* write RTP packet data */
                    c->buffer_ptr += 4;
                    put_buffer(pb, c->buffer_ptr, len);
                    size = url_close_dyn_buf(pb, &c->packet_buffer);
                    /* prepare asynchronous TCP sending */
                    rtsp_c->packet_buffer_ptr = c->packet_buffer;
                    rtsp_c->packet_buffer_end = c->packet_buffer + size;
                    c->buffer_ptr += len;

                    /* send everything we can NOW */
                    len = send(rtsp_c->fd, rtsp_c->packet_buffer_ptr,
                                rtsp_c->packet_buffer_end -
rtsp_c->packet_buffer_ptr, 0);
                    if (len > 0)
                        rtsp_c->packet_buffer_ptr += len;
                    if (rtsp_c->packet_buffer_ptr <
rtsp_c->packet_buffer_end) {
                        /* if we could not send all the data, we will
                           send it later, so a new state is needed to
                           "lock" the RTSP TCP connection */
                        rtsp_c->state = RTSPSTATE_SEND_PACKET;
                        break;
                    } else
                        /* all data has been sent */
                        av_freep(&c->packet_buffer);
                } else {
                    /* send RTP packet directly in UDP */
                    c->buffer_ptr += 4;
                    url_write(c->rtp_handles[c->packet_stream_index],
                              c->buffer_ptr, len);
                    c->buffer_ptr += len;
                    /* here we continue as we can send several packets per
10 ms slot */
                }
            } else {
                /* TCP data output */
                len = send(c->fd, c->buffer_ptr, c->buffer_end -
c->buffer_ptr, 0);
                if (len < 0) {
                    if (ff_neterrno() != FF_NETERROR(EAGAIN) &&
                        ff_neterrno() != FF_NETERROR(EINTR))
                        /* error : close connection */
                        return -1;
                    else
                        return 0;
                } else
                    c->buffer_ptr += len;

                c->data_count += len;
                update_datarate(&c->datarate, c->data_count);
                if (c->stream)
                    c->stream->bytes_served += len;
                break;
            }
        }
    } /* for(;;) */
    return 0;
}