[FFmpeg-devel] [RFC] Add a demuxer directly handling rtp:// URLs by inspecting their content

Martin Storsjo martin
Sat Oct 9 16:55:05 CEST 2010


---
This is an initial attempt at creating a demuxer that is able to play back
some RTP streams without needing a SDP description of them. It works for
RTP payloads with a static, officially registered payload type, that need
no extra parameters passed in the SDP.

With this in place, one is simply able to do "ffplay rtp://224.0.0.255:10000"
to playback a RTP stream on that multicast address, if it happens to be
of such a type.

This proof of concept piggybacks on the SDP demuxer by wrapping its
read_header function in a function that receives one RTP packet, inspects it,
creates a minimal fake SDP description, and feeds that to sdp_read_header.
This way, it is not necessary to create a full chained demuxer and try to
keep the state of the chained demuxer synced with the main one.

This was requested in roundup issues 2277 and 2280.

 configure                |    1 +
 doc/general.texi         |    2 +-
 libavformat/allformats.c |    2 +-
 libavformat/rtsp.c       |   98 ++++++++++++++++++++++++++++++++++++++++++++++
 4 files changed, 101 insertions(+), 2 deletions(-)

diff --git a/configure b/configure
index 400990a..ab2fbd7 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_muxer_select="rtp_muxer rtp_protocol"
diff --git a/doc/general.texi b/doc/general.texi
index aa4e480..8a646bf 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
- at item RTP                       @tab X @tab
+ at item RTP                       @tab X @tab X
 @item RTSP                      @tab X @tab X
 @item SAP                       @tab X @tab
 @item SDP                       @tab   @tab X
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 66de933..7defd87 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_MUXER    (SAP, sap);
     REGISTER_DEMUXER  (SDP, sdp);
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 6570c38..c75bf60 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -2104,3 +2104,101 @@ 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_WARNING, "Unrecognized payload type %d\n",
+                                   payload_type);
+        goto fail;
+    }
+
+    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,
+};
+
-- 
1.7.3.1




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