[FFmpeg-devel] a bug about RTSP_PROTOCOL_RTP_TCP
连海东
lianhaidong
Sat Jan 3 02:59:58 CET 2009
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;
}
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