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authorMartin Storsjö <martin@martin.st>2010-10-19 07:38:53 +0000
committerMartin Storsjö <martin@martin.st>2010-10-19 07:38:53 +0000
commit44594cc7980a4651aba8ac40c59192c283634748 (patch)
tree2c7b16f9124a68bc1b09429f5d122475553b3613
parent4ad08021e8e81057d89c75fd63b97cd1f0757a23 (diff)
Add a demuxer for receiving raw rtp:// URLs without an SDP description
The demuxer inspects the payload type of a received RTP packet and handles the cases where the content is fully described by the payload type. Originally committed as revision 25527 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--Changelog1
-rwxr-xr-xconfigure1
-rw-r--r--doc/general.texi2
-rw-r--r--libavformat/allformats.c2
-rw-r--r--libavformat/avformat.h2
-rw-r--r--libavformat/rtsp.c104
6 files changed, 109 insertions, 3 deletions
diff --git a/Changelog b/Changelog
index c1bf118b95..1289036a55 100644
--- a/Changelog
+++ b/Changelog
@@ -49,6 +49,7 @@ version <next>:
- ffmpeg -crop* options removed
- transpose filter added
- ffmpeg -force_key_frames option added
+- demuxer for receiving raw rtp:// URLs without an SDP description
version 0.6:
diff --git a/configure b/configure
index 0e6e4394bf..d725942ee8 100755
--- a/configure
+++ b/configure
@@ -1361,6 +1361,7 @@ mpegtsraw_demuxer_select="mpegts_demuxer"
mxf_d10_muxer_select="mxf_muxer"
ogg_demuxer_select="golomb"
psp_muxer_select="mov_muxer"
+rtp_demuxer_select="sdp_demuxer"
rtsp_demuxer_select="http_protocol sdp_demuxer"
rtsp_muxer_select="rtp_muxer http_protocol sdp_demuxer"
sap_demuxer_select="sdp_demuxer"
diff --git a/doc/general.texi b/doc/general.texi
index 4f61b2bac3..f59258c8e1 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -215,7 +215,7 @@ library:
@item Lego Mindstorms RSO @tab X @tab X
@item RTMP @tab X @tab X
@tab Output is performed by publishing stream to RTMP server
-@item RTP @tab X @tab
+@item RTP @tab X @tab X
@item RTSP @tab X @tab X
@item SAP @tab X @tab X
@item SDP @tab @tab X
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 9769de9f0f..9f33e31822 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -179,7 +179,7 @@ void av_register_all(void)
REGISTER_MUXDEMUX (ROQ, roq);
REGISTER_DEMUXER (RPL, rpl);
REGISTER_MUXDEMUX (RSO, rso);
- REGISTER_MUXER (RTP, rtp);
+ REGISTER_MUXDEMUX (RTP, rtp);
REGISTER_MUXDEMUX (RTSP, rtsp);
REGISTER_MUXDEMUX (SAP, sap);
REGISTER_DEMUXER (SDP, sdp);
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index 62bf42b968..a9cf174af9 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -22,7 +22,7 @@
#define AVFORMAT_AVFORMAT_H
#define LIBAVFORMAT_VERSION_MAJOR 52
-#define LIBAVFORMAT_VERSION_MINOR 83
+#define LIBAVFORMAT_VERSION_MINOR 84
#define LIBAVFORMAT_VERSION_MICRO 0
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 6570c3880b..aff745a370 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -2104,3 +2104,107 @@ AVInputFormat sdp_demuxer = {
rtsp_fetch_packet,
sdp_read_close,
};
+
+static int rtp_probe(AVProbeData *p)
+{
+ if (av_strstart(p->filename, "rtp:", NULL))
+ return AVPROBE_SCORE_MAX;
+ return 0;
+}
+
+static int rtp_read_header(AVFormatContext *s,
+ AVFormatParameters *ap)
+{
+ uint8_t recvbuf[1500];
+ char host[500], sdp[500];
+ int ret, port;
+ URLContext* in = NULL;
+ int payload_type;
+ AVCodecContext codec;
+ struct sockaddr_storage addr;
+ ByteIOContext pb;
+ socklen_t addrlen = sizeof(addr);
+
+ if (!ff_network_init())
+ return AVERROR(EIO);
+
+ ret = url_open(&in, s->filename, URL_RDONLY);
+ if (ret)
+ goto fail;
+
+ while (1) {
+ ret = url_read(in, recvbuf, sizeof(recvbuf));
+ if (ret == AVERROR(EAGAIN))
+ continue;
+ if (ret < 0)
+ goto fail;
+ if (ret < 12) {
+ av_log(s, AV_LOG_WARNING, "Received too short packet\n");
+ continue;
+ }
+
+ if ((recvbuf[0] & 0xc0) != 0x80) {
+ av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
+ "received\n");
+ continue;
+ }
+
+ payload_type = recvbuf[1] & 0x7f;
+ break;
+ }
+ getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
+ url_close(in);
+ in = NULL;
+
+ memset(&codec, 0, sizeof(codec));
+ if (ff_rtp_get_codec_info(&codec, payload_type)) {
+ av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
+ "without an SDP file describing it\n",
+ payload_type);
+ goto fail;
+ }
+ if (codec.codec_type != AVMEDIA_TYPE_DATA) {
+ av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
+ "properly you need an SDP file "
+ "describing it\n");
+ }
+
+ av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
+ NULL, 0, s->filename);
+
+ snprintf(sdp, sizeof(sdp),
+ "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
+ addr.ss_family == AF_INET ? 4 : 6, host,
+ codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
+ codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
+ port, payload_type);
+ av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
+
+ init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
+ s->pb = &pb;
+
+ /* sdp_read_header initializes this again */
+ ff_network_close();
+
+ ret = sdp_read_header(s, ap);
+ s->pb = NULL;
+ return ret;
+
+fail:
+ if (in)
+ url_close(in);
+ ff_network_close();
+ return ret;
+}
+
+AVInputFormat rtp_demuxer = {
+ "rtp",
+ NULL_IF_CONFIG_SMALL("RTP input format"),
+ sizeof(RTSPState),
+ rtp_probe,
+ rtp_read_header,
+ rtsp_fetch_packet,
+ sdp_read_close,
+ .flags = AVFMT_NOFILE,
+};
+