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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
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ed78754216
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dbf30963f3
@ -258,13 +258,78 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
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return 0;
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}
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/**
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* some rtp servers assume client is dead if they don't hear from them...
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* so we send a Receiver Report to the provided ByteIO context
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* (we don't have access to the rtcp handle from here)
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*/
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int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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{
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ByteIOContext pb;
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uint8_t *buf;
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int len;
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int rtcp_bytes;
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if (!s->rtp_ctx || (count < 1))
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return -1;
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/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
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s->octet_count += count;
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rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
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RTCP_TX_RATIO_DEN;
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rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
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if (rtcp_bytes < 28)
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return -1;
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s->last_octet_count = s->octet_count;
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if (url_open_dyn_buf(&pb) < 0)
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return -1;
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// Receiver Report
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put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
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put_byte(&pb, 201);
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put_be16(&pb, 7); /* length in words - 1 */
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put_be32(&pb, s->ssrc); // our own SSRC
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put_be32(&pb, s->ssrc); // XXX: should be the server's here!
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// some placeholders we should really fill...
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put_be32(&pb, ((0 << 24) | (0 & 0x0ffffff))); /* 0% lost, total 0 lost */
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put_be32(&pb, (0 << 16) | s->seq);
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put_be32(&pb, 0x68); /* jitter */
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put_be32(&pb, -1); /* last SR timestamp */
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put_be32(&pb, 1); /* delay since last SR */
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// CNAME
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put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
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put_byte(&pb, 202);
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len = strlen(s->hostname);
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put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
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put_be32(&pb, s->ssrc);
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put_byte(&pb, 0x01);
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put_byte(&pb, len);
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put_buffer(&pb, s->hostname, len);
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// padding
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for (len = (6 + len) % 4; len % 4; len++) {
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put_byte(&pb, 0);
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}
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put_flush_packet(&pb);
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len = url_close_dyn_buf(&pb, &buf);
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if ((len > 0) && buf) {
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#if defined(DEBUG)
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printf("sending %d bytes of RR\n", len);
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#endif
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url_write(s->rtp_ctx, buf, len);
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av_free(buf);
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}
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return 0;
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}
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/**
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* open a new RTP parse context for stream 'st'. 'st' can be NULL for
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* MPEG2TS streams to indicate that they should be demuxed inside the
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* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
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* TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
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*/
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_t *rtp_payload_data)
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
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{
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RTPDemuxContext *s;
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@ -299,6 +364,9 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_t
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break;
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}
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}
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// needed to send back RTCP RR in RTSP sessions
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s->rtp_ctx = rtpc;
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gethostname(s->hostname, sizeof(s->hostname));
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return s;
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}
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@ -399,6 +467,8 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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seq = (buf[2] << 8) | buf[3];
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timestamp = decode_be32(buf + 4);
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ssrc = decode_be32(buf + 8);
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/* store the ssrc in the RTPDemuxContext */
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s->ssrc = ssrc;
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/* NOTE: we can handle only one payload type */
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if (s->payload_type != payload_type)
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@ -30,7 +30,7 @@ int rtp_get_payload_type(AVCodecContext *codec);
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typedef struct RTPDemuxContext RTPDemuxContext;
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typedef struct rtp_payload_data_s rtp_payload_data_s;
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_s *rtp_payload_data);
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data);
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int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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const uint8_t *buf, int len);
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void rtp_parse_close(RTPDemuxContext *s);
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@ -60,6 +60,9 @@ struct RTPDemuxContext {
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struct MpegTSContext *ts; /* only used for MP2T payloads */
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int read_buf_index;
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int read_buf_size;
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/* used to send back RTCP RR */
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URLContext *rtp_ctx;
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char hostname[256];
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/* rtcp sender statistics receive */
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int64_t last_rtcp_ntp_time; // TODO: move into statistics
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@ -884,7 +884,7 @@ static int rtsp_read_header(AVFormatContext *s,
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if (RTSP_RTP_PORT_MIN != 0) {
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while(j <= RTSP_RTP_PORT_MAX) {
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snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
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if (url_open(&rtsp_st->rtp_handle, buf, URL_RDONLY) == 0) {
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if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) {
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j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
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goto rtp_opened;
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}
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@ -981,7 +981,7 @@ static int rtsp_read_header(AVFormatContext *s,
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host,
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reply->transports[0].server_port_min,
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ttl);
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if (url_open(&rtsp_st->rtp_handle, url, URL_RDONLY) < 0) {
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if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
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err = AVERROR_INVALIDDATA;
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goto fail;
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}
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@ -994,7 +994,7 @@ static int rtsp_read_header(AVFormatContext *s,
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st = s->streams[rtsp_st->stream_index];
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if (!st)
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s->ctx_flags |= AVFMTCTX_NOHEADER;
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rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
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rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
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if (!rtsp_st->rtp_ctx) {
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err = AVERROR_NOMEM;
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@ -1157,6 +1157,8 @@ static int rtsp_read_packet(AVFormatContext *s,
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case RTSP_PROTOCOL_RTP_UDP:
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case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
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len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
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if (rtsp_st->rtp_ctx)
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rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len);
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break;
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}
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if (len < 0)
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@ -1336,7 +1338,7 @@ static int sdp_read_header(AVFormatContext *s,
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inet_ntoa(rtsp_st->sdp_ip),
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rtsp_st->sdp_port,
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rtsp_st->sdp_ttl);
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if (url_open(&rtsp_st->rtp_handle, url, URL_RDONLY) < 0) {
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if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
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err = AVERROR_INVALIDDATA;
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goto fail;
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}
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@ -1346,7 +1348,7 @@ static int sdp_read_header(AVFormatContext *s,
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st = s->streams[rtsp_st->stream_index];
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if (!st)
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s->ctx_flags |= AVFMTCTX_NOHEADER;
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rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
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rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
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if (!rtsp_st->rtp_ctx) {
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err = AVERROR_NOMEM;
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goto fail;
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