[Ffmpeg-cvslog] r6805 - in trunk/libavformat: rtp.c rtp.h rtp_internal.h rtsp.c
gpoirier
subversion
Fri Oct 27 20:19:30 CEST 2006
Author: gpoirier
Date: Fri Oct 27 20:19:29 2006
New Revision: 6805
Modified:
trunk/libavformat/rtp.c
trunk/libavformat/rtp.h
trunk/libavformat/rtp_internal.h
trunk/libavformat/rtsp.c
Log:
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
Modified: trunk/libavformat/rtp.c
==============================================================================
--- trunk/libavformat/rtp.c (original)
+++ trunk/libavformat/rtp.c Fri Oct 27 20:19:29 2006
@@ -259,12 +259,77 @@
}
/**
+ * 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 @@
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 @@
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)
Modified: trunk/libavformat/rtp.h
==============================================================================
--- trunk/libavformat/rtp.h (original)
+++ trunk/libavformat/rtp.h Fri Oct 27 20:19:29 2006
@@ -30,7 +30,7 @@
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);
Modified: trunk/libavformat/rtp_internal.h
==============================================================================
--- trunk/libavformat/rtp_internal.h (original)
+++ trunk/libavformat/rtp_internal.h Fri Oct 27 20:19:29 2006
@@ -60,6 +60,9 @@
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
Modified: trunk/libavformat/rtsp.c
==============================================================================
--- trunk/libavformat/rtsp.c (original)
+++ trunk/libavformat/rtsp.c Fri Oct 27 20:19:29 2006
@@ -884,7 +884,7 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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;
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