[FFmpeg-cvslog] rtsp: Add support for depacketizing RTP data via custom IO
Martin Storsjö
git at videolan.org
Fri Jan 4 13:28:51 CET 2013
ffmpeg | branch: master | Martin Storsjö <martin at martin.st> | Tue Jan 1 23:41:29 2013 +0200| [e96406eda4f143f101bd44372f7b2d542183000a] | committer: Martin Storsjö
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 at martin.st>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e96406eda4f143f101bd44372f7b2d542183000a
---
libavformat/rtpdec.c | 11 +++++--
libavformat/rtpdec.h | 5 ++--
libavformat/rtsp.c | 79 +++++++++++++++++++++++++++++++++++++++++--------
libavformat/rtsp.h | 5 ++++
libavformat/version.h | 2 +-
5 files changed, 84 insertions(+), 18 deletions(-)
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 77df188..08150b7 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 2d962c8..7c51f5a 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 d5b3242..134b322 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 043b67a..3260f6c 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 89af0ba..c2c1e3a 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, \
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