From dbf30963f3599717e8ff90c0820bb7bfca94a38b Mon Sep 17 00:00:00 2001 From: Thijs Date: Fri, 27 Oct 2006 18:19:29 +0000 Subject: [PATCH] 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 --- libavformat/rtp.c | 74 ++++++++++++++++++++++++++++++++++++-- libavformat/rtp.h | 2 +- libavformat/rtp_internal.h | 3 ++ libavformat/rtsp.c | 12 ++++--- 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 @@ -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) 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;