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

4 files changed, 83 insertions, 8 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)
diff --git a/libavformat/rtp.h b/libavformat/rtp.h
index 9ce6f90d49..60ccc50ee4 100644
--- a/libavformat/rtp.h
+++ b/libavformat/rtp.h
@@ -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);
diff --git a/libavformat/rtp_internal.h b/libavformat/rtp_internal.h
index 5c178f3811..3930966ad8 100644
--- a/libavformat/rtp_internal.h
+++ b/libavformat/rtp_internal.h
@@ -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
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index d340819bdf..08c71c534b 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -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;