Commit dbf30963 authored by Thijs's avatar Thijs Committed by Guillaume Poirier
Browse files

make ffmpeg able to send back a RTCP receiver report.

Patch by Thijs thijsvermeir A telenet P be
Original thread:
Date: Oct 27, 2006 12:58 PM
Subject: [Ffmpeg-devel] [PATCH proposal] RTCP receiver report

Originally committed as revision 6805 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent ed787542
......@@ -258,13 +258,78 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
return 0;
}
/**
* some rtp servers assume client is dead if they don't hear from them...
* so we send a Receiver Report to the provided ByteIO context
* (we don't have access to the rtcp handle from here)
*/
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
ByteIOContext pb;
uint8_t *buf;
int len;
int rtcp_bytes;
if (!s->rtp_ctx || (count < 1))
return -1;
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
s->octet_count += count;
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
RTCP_TX_RATIO_DEN;
rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
if (rtcp_bytes < 28)
return -1;
s->last_octet_count = s->octet_count;
if (url_open_dyn_buf(&pb) < 0)
return -1;
// Receiver Report
put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
put_byte(&pb, 201);
put_be16(&pb, 7); /* length in words - 1 */
put_be32(&pb, s->ssrc); // our own SSRC
put_be32(&pb, s->ssrc); // XXX: should be the server's here!
// some placeholders we should really fill...
put_be32(&pb, ((0 << 24) | (0 & 0x0ffffff))); /* 0% lost, total 0 lost */
put_be32(&pb, (0 << 16) | s->seq);
put_be32(&pb, 0x68); /* jitter */
put_be32(&pb, -1); /* last SR timestamp */
put_be32(&pb, 1); /* delay since last SR */
// CNAME
put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
put_byte(&pb, 202);
len = strlen(s->hostname);
put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
put_be32(&pb, s->ssrc);
put_byte(&pb, 0x01);
put_byte(&pb, len);
put_buffer(&pb, s->hostname, len);
// padding
for (len = (6 + len) % 4; len % 4; len++) {
put_byte(&pb, 0);
}
put_flush_packet(&pb);
len = url_close_dyn_buf(&pb, &buf);
if ((len > 0) && buf) {
#if defined(DEBUG)
printf("sending %d bytes of RR\n", len);
#endif
url_write(s->rtp_ctx, buf, len);
av_free(buf);
}
return 0;
}
/**
* 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)
* TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
*/
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_t *rtp_payload_data)
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
{
RTPDemuxContext *s;
......@@ -299,6 +364,9 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_t
break;
}
}
// needed to send back RTCP RR in RTSP sessions
s->rtp_ctx = rtpc;
gethostname(s->hostname, sizeof(s->hostname));
return s;
}
......@@ -399,6 +467,8 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
seq = (buf[2] << 8) | buf[3];
timestamp = decode_be32(buf + 4);
ssrc = decode_be32(buf + 8);
/* store the ssrc in the RTPDemuxContext */
s->ssrc = ssrc;
/* NOTE: we can handle only one payload type */
if (s->payload_type != payload_type)
......
......@@ -30,7 +30,7 @@ int rtp_get_payload_type(AVCodecContext *codec);
typedef struct RTPDemuxContext RTPDemuxContext;
typedef struct rtp_payload_data_s rtp_payload_data_s;
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_s *rtp_payload_data);
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data);
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len);
void rtp_parse_close(RTPDemuxContext *s);
......
......@@ -60,6 +60,9 @@ struct RTPDemuxContext {
struct MpegTSContext *ts; /* only used for MP2T payloads */
int read_buf_index;
int read_buf_size;
/* used to send back RTCP RR */
URLContext *rtp_ctx;
char hostname[256];
/* rtcp sender statistics receive */
int64_t last_rtcp_ntp_time; // TODO: move into statistics
......
......@@ -884,7 +884,7 @@ static int rtsp_read_header(AVFormatContext *s,
if (RTSP_RTP_PORT_MIN != 0) {
while(j <= RTSP_RTP_PORT_MAX) {
snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
if (url_open(&rtsp_st->rtp_handle, buf, URL_RDONLY) == 0) {
if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) {
j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
goto rtp_opened;
}
......@@ -981,7 +981,7 @@ static int rtsp_read_header(AVFormatContext *s,
host,
reply->transports[0].server_port_min,
ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDONLY) < 0) {
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
......@@ -994,7 +994,7 @@ static int rtsp_read_header(AVFormatContext *s,
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
if (!rtsp_st->rtp_ctx) {
err = AVERROR_NOMEM;
......@@ -1157,6 +1157,8 @@ static int rtsp_read_packet(AVFormatContext *s,
case RTSP_PROTOCOL_RTP_UDP:
case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
if (rtsp_st->rtp_ctx)
rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len);
break;
}
if (len < 0)
......@@ -1336,7 +1338,7 @@ static int sdp_read_header(AVFormatContext *s,
inet_ntoa(rtsp_st->sdp_ip),
rtsp_st->sdp_port,
rtsp_st->sdp_ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDONLY) < 0) {
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
......@@ -1346,7 +1348,7 @@ static int sdp_read_header(AVFormatContext *s,
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
if (!rtsp_st->rtp_ctx) {
err = AVERROR_NOMEM;
goto fail;
......
Markdown is supported
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment