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https://github.com/FFmpeg/FFmpeg.git
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rtsp: Add support for depacketizing RTP data via custom IO
To use this, set sdpflags=custom_io to the sdp demuxer. During the avformat_open_input call, the SDP is read from the AVFormatContext AVIOContext (ctx->pb) - after the avformat_open_input call, during the av_read_frame() calls, the same ctx->pb is used for reading packets (and sending back RTCP RR packets). Normally, one would use this with a read-only AVIOContext for the SDP during the avformat_open_input call, then close that one and replace it with a read-write one for the packets after the avformat_open_input call has returned. This allows using the RTP depacketizers as "pure" demuxers, without having them tied to the libavformat network IO. Signed-off-by: Martin Storsjö <martin@martin.st>
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3f95f0dda5
commit
e96406eda4
@ -226,7 +226,8 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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return 1;
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}
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int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count)
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int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
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AVIOContext *avio, int count)
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{
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AVIOContext *pb;
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uint8_t *buf;
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@ -242,7 +243,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count)
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uint32_t fraction;
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uint64_t ntp_time = s->last_rtcp_ntp_time; // TODO: Get local ntp time?
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if (!fd || (count < 1))
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if ((!fd && !avio) || (count < 1))
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return -1;
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/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
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@ -255,7 +256,9 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count)
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return -1;
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s->last_octet_count = s->octet_count;
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if (avio_open_dyn_buf(&pb) < 0)
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if (!fd)
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pb = avio;
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else if (avio_open_dyn_buf(&pb) < 0)
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return -1;
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// Receiver Report
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@ -312,6 +315,8 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count)
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avio_w8(pb, 0);
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avio_flush(pb);
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if (!fd)
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return 0;
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len = avio_close_dyn_buf(pb, &buf);
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if ((len > 0) && buf) {
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int av_unused result;
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@ -68,10 +68,11 @@ void ff_rtp_send_punch_packets(URLContext* rtp_handle);
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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 URLContext
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* so we send a Receiver Report to the provided URLContext or AVIOContext
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* (we don't have access to the rtcp handle from here)
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*/
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int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count);
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int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
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AVIOContext *avio, int count);
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// these statistics are used for rtcp receiver reports...
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typedef struct RTPStatistics {
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@ -96,6 +96,7 @@ const AVOption ff_rtsp_options[] = {
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static const AVOption sdp_options[] = {
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RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
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{ "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
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RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
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RTSP_REORDERING_OPTS(),
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{ NULL },
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@ -1789,6 +1790,50 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
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}
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}
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static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
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const uint8_t *buf, int len)
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{
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RTSPState *rt = s->priv_data;
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int i;
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if (len < 0)
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return len;
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if (rt->nb_rtsp_streams == 1) {
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*rtsp_st = rt->rtsp_streams[0];
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return len;
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}
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if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
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if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
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int no_ssrc = 0;
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for (i = 0; i < rt->nb_rtsp_streams; i++) {
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RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
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if (!rtpctx)
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continue;
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if (rtpctx->ssrc == AV_RB32(&buf[4])) {
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*rtsp_st = rt->rtsp_streams[i];
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return len;
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}
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if (!rtpctx->ssrc)
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no_ssrc = 1;
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}
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if (no_ssrc) {
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av_log(s, AV_LOG_WARNING,
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"Unable to pick stream for packet - SSRC not known for "
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"all streams\n");
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return AVERROR(EAGAIN);
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}
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} else {
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for (i = 0; i < rt->nb_rtsp_streams; i++) {
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if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
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*rtsp_st = rt->rtsp_streams[i];
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return len;
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}
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}
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}
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}
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av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
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return AVERROR(EAGAIN);
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}
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int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
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{
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RTSPState *rt = s->priv_data;
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@ -1860,7 +1905,13 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
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case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
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len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
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if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
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ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, len);
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ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
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break;
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case RTSP_LOWER_TRANSPORT_CUSTOM:
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len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
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len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
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if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
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ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
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break;
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}
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if (len == AVERROR(EAGAIN) && first_queue_st &&
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@ -1973,6 +2024,8 @@ static int sdp_read_header(AVFormatContext *s)
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if (s->max_delay < 0) /* Not set by the caller */
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s->max_delay = DEFAULT_REORDERING_DELAY;
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if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
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rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
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/* read the whole sdp file */
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/* XXX: better loading */
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@ -1993,17 +2046,19 @@ static int sdp_read_header(AVFormatContext *s)
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char namebuf[50];
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rtsp_st = rt->rtsp_streams[i];
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getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
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namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
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ff_url_join(url, sizeof(url), "rtp", NULL,
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namebuf, rtsp_st->sdp_port,
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"?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
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rtsp_st->sdp_ttl,
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rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
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if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
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&s->interrupt_callback, NULL) < 0) {
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err = AVERROR_INVALIDDATA;
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goto fail;
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if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
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getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
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namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
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ff_url_join(url, sizeof(url), "rtp", NULL,
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namebuf, rtsp_st->sdp_port,
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"?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
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rtsp_st->sdp_ttl,
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rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
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if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
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&s->interrupt_callback, NULL) < 0) {
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err = AVERROR_INVALIDDATA;
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goto fail;
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}
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}
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if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
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goto fail;
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@ -42,6 +42,10 @@ enum RTSPLowerTransport {
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RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper
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transport mode as such,
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only for use via AVOptions */
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RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public
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option for lower_transport_mask,
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but set in the SDP demuxer based
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on a flag. */
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};
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/**
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@ -396,6 +400,7 @@ typedef struct RTSPState {
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receive packets only from the right
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source address and port. */
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#define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
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#define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
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/**
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* Describe a single stream, as identified by a single m= line block in the
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@ -31,7 +31,7 @@
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#define LIBAVFORMAT_VERSION_MAJOR 54
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#define LIBAVFORMAT_VERSION_MINOR 20
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#define LIBAVFORMAT_VERSION_MICRO 2
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#define LIBAVFORMAT_VERSION_MICRO 3
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#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
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LIBAVFORMAT_VERSION_MINOR, \
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