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authorMartin Storsjö <martin@martin.st>2010-10-01 17:50:24 +0000
committerMartin Storsjö <martin@martin.st>2010-10-01 17:50:24 +0000
commit58ee09911e0dab6432b18e9a2c5a876fdf8ef078 (patch)
tree1180f080d8185600d0b538f2a54d3e43b392b4bd
parent0260741876b535c11cef43a959d1a9a970a308e9 (diff)
rtpdec: Reorder received RTP packets according to the seq number
Reordering is enabled only when receiving over UDP. Originally committed as revision 25294 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavformat/rtpdec.c120
-rw-r--r--libavformat/rtpdec.h21
-rw-r--r--libavformat/rtsp.c5
3 files changed, 139 insertions, 7 deletions
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 33497e648f..d349ce79b2 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -334,7 +334,7 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
* MPEG2TS streams to indicate that they should be demuxed inside the
* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
*/
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type)
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
{
RTPDemuxContext *s;
@@ -346,6 +346,7 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
s->ic = s1;
s->st = st;
+ s->queue_size = queue_size;
rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
s->ts = ff_mpegts_parse_open(s->ic);
@@ -504,9 +505,84 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
// now perform timestamp things....
finalize_packet(s, pkt, timestamp);
+ s->prev_ret = rv;
return rv;
}
+void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
+{
+ while (s->queue) {
+ RTPPacket *next = s->queue->next;
+ av_free(s->queue->buf);
+ av_free(s->queue);
+ s->queue = next;
+ }
+ s->seq = 0;
+ s->queue_len = 0;
+ s->prev_ret = 0;
+}
+
+static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
+{
+ uint16_t seq = AV_RB16(buf + 2);
+ RTPPacket *cur = s->queue, *prev = NULL, *packet;
+
+ /* Find the correct place in the queue to insert the packet */
+ while (cur) {
+ int16_t diff = seq - cur->seq;
+ if (diff < 0)
+ break;
+ prev = cur;
+ cur = cur->next;
+ }
+
+ packet = av_mallocz(sizeof(*packet));
+ if (!packet)
+ return;
+ packet->recvtime = av_gettime();
+ packet->seq = seq;
+ packet->len = len;
+ packet->buf = buf;
+ packet->next = cur;
+ if (prev)
+ prev->next = packet;
+ else
+ s->queue = packet;
+ s->queue_len++;
+}
+
+static int has_next_packet(RTPDemuxContext *s)
+{
+ return s->queue && s->queue->seq == s->seq + 1;
+}
+
+int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
+{
+ return s->queue ? s->queue->recvtime : 0;
+}
+
+static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
+{
+ int rv;
+ RTPPacket *next;
+
+ if (s->queue_len <= 0)
+ return -1;
+
+ if (!has_next_packet(s))
+ av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
+ "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
+
+ /* Parse the first packet in the queue, and dequeue it */
+ rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
+ next = s->queue->next;
+ av_free(s->queue->buf);
+ av_free(s->queue);
+ s->queue = next;
+ s->queue_len--;
+ return rv ? rv : has_next_packet(s);
+}
+
/**
* Parse an RTP or RTCP packet directly sent as a buffer.
* @param s RTP parse context.
@@ -525,6 +601,11 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
int rv= 0;
if (!buf) {
+ /* If parsing of the previous packet actually returned 0, there's
+ * nothing more to be parsed from that packet, but we may have
+ * indicated that we can return the next enqueued packet. */
+ if (!s->prev_ret)
+ return rtp_parse_queued_packet(s, pkt);
/* return the next packets, if any */
if(s->st && s->parse_packet) {
/* timestamp should be overwritten by parse_packet, if not,
@@ -533,7 +614,8 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
s->st, pkt, &timestamp, NULL, 0, flags);
finalize_packet(s, pkt, timestamp);
- return rv;
+ s->prev_ret = rv;
+ return rv ? rv : has_next_packet(s);
} else {
// TODO: Move to a dynamic packet handler (like above)
if (s->read_buf_index >= s->read_buf_size)
@@ -545,8 +627,10 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
s->read_buf_index += ret;
if (s->read_buf_index < s->read_buf_size)
return 1;
- else
- return 0;
+ else {
+ s->prev_ret = 0;
+ return has_next_packet(s);
+ }
}
}
@@ -559,11 +643,37 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
return rtcp_parse_packet(s, buf, len);
}
- return rtp_parse_packet_internal(s, pkt, buf, len);
+ if (s->seq == 0 || s->queue_size <= 1) {
+ /* First packet, or no reordering */
+ return rtp_parse_packet_internal(s, pkt, buf, len);
+ } else {
+ uint16_t seq = AV_RB16(buf + 2);
+ int16_t diff = seq - s->seq;
+ if (diff < 0) {
+ /* Packet older than the previously emitted one, drop */
+ av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
+ "RTP: dropping old packet received too late\n");
+ return -1;
+ } else if (diff <= 1) {
+ /* Correct packet */
+ rv = rtp_parse_packet_internal(s, pkt, buf, len);
+ return rv ? rv : has_next_packet(s);
+ } else {
+ /* Still missing some packet, enqueue this one. */
+ enqueue_packet(s, buf, len);
+ *bufptr = NULL;
+ /* Return the first enqueued packet if the queue is full,
+ * even if we're missing something */
+ if (s->queue_len >= s->queue_size)
+ return rtp_parse_queued_packet(s, pkt);
+ return -1;
+ }
+ }
}
void rtp_parse_close(RTPDemuxContext *s)
{
+ ff_rtp_reset_packet_queue(s);
if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
ff_mpegts_parse_close(s->ts);
}
diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h
index df2ec77680..84deefec68 100644
--- a/libavformat/rtpdec.h
+++ b/libavformat/rtpdec.h
@@ -32,15 +32,19 @@ typedef struct RTPDynamicProtocolHandler_s RTPDynamicProtocolHandler;
#define RTP_MIN_PACKET_LENGTH 12
#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
+#define RTP_REORDER_QUEUE_DEFAULT_SIZE 10
+
#define RTP_NOTS_VALUE ((uint32_t)-1)
typedef struct RTPDemuxContext RTPDemuxContext;
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type);
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size);
void rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
RTPDynamicProtocolHandler *handler);
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **buf, int len);
void rtp_parse_close(RTPDemuxContext *s);
+int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s);
+void ff_rtp_reset_packet_queue(RTPDemuxContext *s);
#if (LIBAVFORMAT_VERSION_MAJOR <= 53)
int rtp_get_local_port(URLContext *h);
#endif
@@ -131,6 +135,14 @@ struct RTPDynamicProtocolHandler_s {
struct RTPDynamicProtocolHandler_s *next;
};
+typedef struct RTPPacket {
+ uint16_t seq;
+ uint8_t *buf;
+ int len;
+ int64_t recvtime;
+ struct RTPPacket *next;
+} RTPPacket;
+
// moved out of rtp.c, because the h264 decoder needs to know about this structure..
struct RTPDemuxContext {
AVFormatContext *ic;
@@ -152,6 +164,13 @@ struct RTPDemuxContext {
RTPStatistics statistics; ///< Statistics for this stream (used by RTCP receiver reports)
+ /** Fields for packet reordering @{ */
+ int prev_ret; ///< The return value of the actual parsing of the previous packet
+ RTPPacket* queue; ///< A sorted queue of buffered packets not yet returned
+ int queue_len; ///< The number of packets in queue
+ int queue_size; ///< The size of queue, or 0 if reordering is disabled
+ /*@}*/
+
/* rtcp sender statistics receive */
int64_t last_rtcp_ntp_time; // TODO: move into statistics
int64_t first_rtcp_ntp_time; // TODO: move into statistics
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 92aa352703..1e4b9c1db7 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -582,7 +582,9 @@ static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
rtsp_st->dynamic_handler);
else
rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
- rtsp_st->sdp_payload_type);
+ rtsp_st->sdp_payload_type,
+ (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
+ ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
if (!rtsp_st->transport_priv) {
return AVERROR(ENOMEM);
@@ -1270,6 +1272,7 @@ static int rtsp_read_play(AVFormatContext *s)
continue;
if (rtsp_st->stream_index >= 0)
st = s->streams[rtsp_st->stream_index];
+ ff_rtp_reset_packet_queue(rtpctx);
if (reply->range_start != AV_NOPTS_VALUE) {
rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;