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

1 files changed, 72 insertions, 2 deletions

diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index 4969acbf5b..e075ba6e6d 100644
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -259,12 +259,77 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
 }
 
 /**
+ * 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)