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>

5 files changed, 84 insertions, 18 deletions

diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 77df188cb9..08150b734a 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -226,7 +226,8 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
     return 1;
 }
 
-int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count)
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
+                                  AVIOContext *avio, int count)
 {
     AVIOContext *pb;
     uint8_t *buf;
@@ -242,7 +243,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count)
     uint32_t fraction;
     uint64_t ntp_time = s->last_rtcp_ntp_time; // TODO: Get local ntp time?
 
-    if (!fd || (count < 1))
+    if ((!fd && !avio) || (count < 1))
         return -1;
 
     /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
@@ -255,7 +256,9 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count)
         return -1;
     s->last_octet_count = s->octet_count;
 
-    if (avio_open_dyn_buf(&pb) < 0)
+    if (!fd)
+        pb = avio;
+    else if (avio_open_dyn_buf(&pb) < 0)
         return -1;
 
     // Receiver Report
@@ -312,6 +315,8 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count)
         avio_w8(pb, 0);
 
     avio_flush(pb);
+    if (!fd)
+        return 0;
     len = avio_close_dyn_buf(pb, &buf);
     if ((len > 0) && buf) {
         int av_unused result;
diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h
index 2d962c82c7..7c51f5a8d2 100644
--- a/libavformat/rtpdec.h
+++ b/libavformat/rtpdec.h
@@ -68,10 +68,11 @@ void ff_rtp_send_punch_packets(URLContext* rtp_handle);
 
 /**
  * some rtp servers assume client is dead if they don't hear from them...
- * so we send a Receiver Report to the provided URLContext
+ * so we send a Receiver Report to the provided URLContext or AVIOContext
  * (we don't have access to the rtcp handle from here)
  */
-int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count);
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
+                                  AVIOContext *avio, int count);
 
 // these statistics are used for rtcp receiver reports...
 typedef struct RTPStatistics {
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index d5b3242da4..134b3223e4 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -96,6 +96,7 @@ const AVOption ff_rtsp_options[] = {
 
 static const AVOption sdp_options[] = {
     RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
+    { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
     RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
     RTSP_REORDERING_OPTS(),
     { NULL },
@@ -1789,6 +1790,50 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
     }
 }
 
+static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
+                       const uint8_t *buf, int len)
+{
+    RTSPState *rt = s->priv_data;
+    int i;
+    if (len < 0)
+        return len;
+    if (rt->nb_rtsp_streams == 1) {
+        *rtsp_st = rt->rtsp_streams[0];
+        return len;
+    }
+    if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
+        if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
+            int no_ssrc = 0;
+            for (i = 0; i < rt->nb_rtsp_streams; i++) {
+                RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
+                if (!rtpctx)
+                    continue;
+                if (rtpctx->ssrc == AV_RB32(&buf[4])) {
+                    *rtsp_st = rt->rtsp_streams[i];
+                    return len;
+                }
+                if (!rtpctx->ssrc)
+                    no_ssrc = 1;
+            }
+            if (no_ssrc) {
+                av_log(s, AV_LOG_WARNING,
+                       "Unable to pick stream for packet - SSRC not known for "
+                       "all streams\n");
+                return AVERROR(EAGAIN);
+            }
+        } else {
+            for (i = 0; i < rt->nb_rtsp_streams; i++) {
+                if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
+                    *rtsp_st = rt->rtsp_streams[i];
+                    return len;
+                }
+            }
+        }
+    }
+    av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
+    return AVERROR(EAGAIN);
+}
+
 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
 {
     RTSPState *rt = s->priv_data;
@@ -1860,7 +1905,13 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
     case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
         len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
         if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
-            ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, len);
+            ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
+        break;
+    case RTSP_LOWER_TRANSPORT_CUSTOM:
+        len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
+        len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
+        if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
+            ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
         break;
     }
     if (len == AVERROR(EAGAIN) && first_queue_st &&
@@ -1973,6 +2024,8 @@ static int sdp_read_header(AVFormatContext *s)
 
     if (s->max_delay < 0) /* Not set by the caller */
         s->max_delay = DEFAULT_REORDERING_DELAY;
+    if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
+        rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
 
     /* read the whole sdp file */
     /* XXX: better loading */
@@ -1993,17 +2046,19 @@ static int sdp_read_header(AVFormatContext *s)
         char namebuf[50];
         rtsp_st = rt->rtsp_streams[i];
 
-        getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
-                    namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
-        ff_url_join(url, sizeof(url), "rtp", NULL,
-                    namebuf, rtsp_st->sdp_port,
-                    "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
-                    rtsp_st->sdp_ttl,
-                    rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
-        if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
-                       &s->interrupt_callback, NULL) < 0) {
-            err = AVERROR_INVALIDDATA;
-            goto fail;
+        if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
+            getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
+                        namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
+            ff_url_join(url, sizeof(url), "rtp", NULL,
+                        namebuf, rtsp_st->sdp_port,
+                        "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
+                        rtsp_st->sdp_ttl,
+                        rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
+            if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
+                           &s->interrupt_callback, NULL) < 0) {
+                err = AVERROR_INVALIDDATA;
+                goto fail;
+            }
         }
         if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
             goto fail;
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index 043b67aa6f..3260f6c92d 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -42,6 +42,10 @@ enum RTSPLowerTransport {
     RTSP_LOWER_TRANSPORT_HTTP = 8,          /**< HTTP tunneled - not a proper
                                                  transport mode as such,
                                                  only for use via AVOptions */
+    RTSP_LOWER_TRANSPORT_CUSTOM = 16,       /**< Custom IO - not a public
+                                                 option for lower_transport_mask,
+                                                 but set in the SDP demuxer based
+                                                 on a flag. */
 };
 
 /**
@@ -396,6 +400,7 @@ typedef struct RTSPState {
                                           receive packets only from the right
                                           source address and port. */
 #define RTSP_FLAG_LISTEN      0x2    /**< Wait for incoming connections. */
+#define RTSP_FLAG_CUSTOM_IO   0x4    /**< Do all IO via the AVIOContext. */
 
 /**
  * Describe a single stream, as identified by a single m= line block in the
diff --git a/libavformat/version.h b/libavformat/version.h
index 89af0bac6e..c2c1e3a8d6 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -31,7 +31,7 @@
 
 #define LIBAVFORMAT_VERSION_MAJOR 54
 #define LIBAVFORMAT_VERSION_MINOR 20
-#define LIBAVFORMAT_VERSION_MICRO  2
+#define LIBAVFORMAT_VERSION_MICRO  3
 
 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
                                                LIBAVFORMAT_VERSION_MINOR, \