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authorMartin Storsjö <martin@martin.st>2012-12-11 15:59:24 +0200
committerMartin Storsjö <martin@martin.st>2013-01-08 17:48:14 +0200
commit86d9181cf41edc3382bf2481f95a2fb321058689 (patch)
tree5f83f288f2ee9da82e789bc870e8410ca9b77d74
parent42805eda554a7fc44341282771531e7837ac72b7 (diff)
rtpdec: Support sending RTCP feedback packets
This sends NACK for missed packets and PLI (picture loss indication) if a depacketizer indicates that it needs a new keyframe, according to RFC 4585. This is only enabled if the SDP indicated that feedback is supported (via the AVPF or SAVPF profile names). The feedback packets are throttled to a certain maximum interval (currently 250 ms) to make sure the feedback packets don't eat up too much bandwidth (which might be counterproductive). The RFC specifies a more elaborate feedback packet scheduling. The feedback packets are currently sent independently from normal RTCP RR packets, which is not totally spec compliant, but works fine in the environments I've tested it in. (RFC 5506 allows this, but requires a SDP attribute for enabling it.) Signed-off-by: Martin Storsjö <martin@martin.st>
-rw-r--r--libavformat/rtpdec.c96
-rw-r--r--libavformat/rtpdec.h5
-rw-r--r--libavformat/rtsp.c8
-rw-r--r--libavformat/rtsp.h3
-rw-r--r--libavformat/version.h2
5 files changed, 113 insertions, 1 deletions
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 348b796d83..4064e70192 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -30,6 +30,8 @@
#include "rtpdec.h"
#include "rtpdec_formats.h"
+#define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
+
static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
.enc_name = "X-MP3-draft-00",
.codec_type = AVMEDIA_TYPE_AUDIO,
@@ -366,6 +368,100 @@ void ff_rtp_send_punch_packets(URLContext *rtp_handle)
av_free(buf);
}
+static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
+ uint16_t *missing_mask)
+{
+ int i;
+ uint16_t next_seq = s->seq + 1;
+ RTPPacket *pkt = s->queue;
+
+ if (!pkt || pkt->seq == next_seq)
+ return 0;
+
+ *missing_mask = 0;
+ for (i = 1; i <= 16; i++) {
+ uint16_t missing_seq = next_seq + i;
+ while (pkt) {
+ int16_t diff = pkt->seq - missing_seq;
+ if (diff >= 0)
+ break;
+ pkt = pkt->next;
+ }
+ if (!pkt)
+ break;
+ if (pkt->seq == missing_seq)
+ continue;
+ *missing_mask |= 1 << (i - 1);
+ }
+
+ *first_missing = next_seq;
+ return 1;
+}
+
+int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
+ AVIOContext *avio)
+{
+ int len, need_keyframe, missing_packets;
+ AVIOContext *pb;
+ uint8_t *buf;
+ int64_t now;
+ uint16_t first_missing, missing_mask;
+
+ if (!fd && !avio)
+ return -1;
+
+ need_keyframe = s->handler && s->handler->need_keyframe &&
+ s->handler->need_keyframe(s->dynamic_protocol_context);
+ missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
+
+ if (!need_keyframe && !missing_packets)
+ return 0;
+
+ /* Send new feedback if enough time has elapsed since the last
+ * feedback packet. */
+
+ now = av_gettime();
+ if (s->last_feedback_time &&
+ (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
+ return 0;
+ s->last_feedback_time = now;
+
+ if (!fd)
+ pb = avio;
+ else if (avio_open_dyn_buf(&pb) < 0)
+ return -1;
+
+ if (need_keyframe) {
+ avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
+ avio_w8(pb, RTCP_PSFB);
+ avio_wb16(pb, 2); /* length in words - 1 */
+ // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
+ avio_wb32(pb, s->ssrc + 1);
+ avio_wb32(pb, s->ssrc); // server SSRC
+ }
+
+ if (missing_packets) {
+ avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
+ avio_w8(pb, RTCP_RTPFB);
+ avio_wb16(pb, 3); /* length in words - 1 */
+ avio_wb32(pb, s->ssrc + 1);
+ avio_wb32(pb, s->ssrc); // server SSRC
+
+ avio_wb16(pb, first_missing);
+ avio_wb16(pb, missing_mask);
+ }
+
+ avio_flush(pb);
+ if (!fd)
+ return 0;
+ len = avio_close_dyn_buf(pb, &buf);
+ if (len > 0 && buf) {
+ ffurl_write(fd, buf, len);
+ av_free(buf);
+ }
+ return 0;
+}
+
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG2-TS streams to indicate that they should be demuxed inside the
diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h
index 75863b8ccb..ee085b1843 100644
--- a/libavformat/rtpdec.h
+++ b/libavformat/rtpdec.h
@@ -73,6 +73,8 @@ void ff_rtp_send_punch_packets(URLContext* rtp_handle);
*/
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
AVIOContext *avio, int count);
+int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
+ AVIOContext *avio);
// these statistics are used for rtcp receiver reports...
typedef struct RTPStatistics {
@@ -130,6 +132,7 @@ struct RTPDynamicProtocolHandler {
void (*free)(PayloadContext *protocol_data);
/** Parse handler for this dynamic packet */
DynamicPayloadPacketHandlerProc parse_packet;
+ int (*need_keyframe)(PayloadContext *context);
struct RTPDynamicProtocolHandler *next;
};
@@ -180,6 +183,8 @@ struct RTPDemuxContext {
unsigned int packet_count;
unsigned int octet_count;
unsigned int last_octet_count;
+ int64_t last_feedback_time;
+
/* buffer for partially parsed packets */
uint8_t buf[RTP_MAX_PACKET_LENGTH];
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index ba9b756879..88e0cbf2db 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -379,6 +379,8 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
get_word(buf1, sizeof(buf1), &p); /* protocol */
if (!strcmp(buf1, "udp"))
rt->transport = RTSP_TRANSPORT_RAW;
+ else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
+ rtsp_st->feedback = 1;
/* XXX: handle list of formats */
get_word(buf1, sizeof(buf1), &p); /* format list */
@@ -1936,6 +1938,12 @@ redo:
ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
} else if (rt->transport == RTSP_TRANSPORT_RTP) {
ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
+ if (rtsp_st->feedback) {
+ AVIOContext *pb = NULL;
+ if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
+ pb = s->pb;
+ ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
+ }
if (ret < 0) {
/* Either bad packet, or a RTCP packet. Check if the
* first_rtcp_ntp_time field was initialized. */
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index 3260f6c92d..e8d57f283a 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -437,6 +437,9 @@ typedef struct RTSPStream {
/** private data associated with the dynamic protocol */
PayloadContext *dynamic_protocol_context;
//@}
+
+ /** Enable sending RTCP feedback messages according to RFC 4585 */
+ int feedback;
} RTSPStream;
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
diff --git a/libavformat/version.h b/libavformat/version.h
index c2c1e3a8d6..2944d5e1f8 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 3
+#define LIBAVFORMAT_VERSION_MICRO 4
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, \