Add an extra field to a buffer for checking purposes.

diff --git a/player.c b/player.c
index 4f652802a64e3d228f5fc63f7cc79249695d2104..340b40d30f0420c042c09416826551c5ae892634 100644 (file)
--- a/player.c
+++ b/player.c
@@ -462,10 +462,11 @@ static void free_audio_buffers(rtsp_conn_info *conn) {
     free(conn->audio_buffer[i].data);
 }
 
-void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len,
+void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp, uint8_t *data, int len,
                        rtsp_conn_info *conn) {
 
   // all timestamps are done at the output rate
+  // the "actual_timestamp" is the one that comes in the packet, and is carried over for debugging and checking only.
 
   int64_t ltimestamp = timestamp * conn->output_sample_ratio;
 
@@ -518,6 +519,7 @@ void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len,
           abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->ab_write, i));
           abuf->ready = 0; // to be sure, to be sure
           abuf->timestamp = 0;
+          abuf->given_timestamp = 0;
           abuf->sequence_number = 0;
         }
         // debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write);
@@ -550,11 +552,13 @@ void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len,
           abuf->ready = 1;
           abuf->length = datalen;
           abuf->timestamp = ltimestamp;
+          abuf->given_timestamp = actual_timestamp;
           abuf->sequence_number = seqno;
         } else {
           debug(1, "Bad audio packet detected and discarded.");
           abuf->ready = 0;
           abuf->timestamp = 0;
+          abuf->given_timestamp = 0;
           abuf->sequence_number = 0;
         }
       }
@@ -1890,8 +1894,8 @@ static void *player_thread_func(void *arg) {
               if (abs_sync_error > 3 * config.output_rate) {
                 warn("Very large sync error: %" PRId64 " frames, with delay: %" PRId64
                      ", td_in_frames: %" PRId64 ", rt: %" PRId64 ", nt: %" PRId64
-                     ", current_delay: %" PRId64 ", frames.",
-                     sync_error, delay, td_in_frames, rt, nt, current_delay);
+                     ", current_delay: %" PRId64 ", seqno: %u, given timestamp: %" PRIu32 ".",
+                     sync_error, delay, td_in_frames, rt, nt, current_delay, inframe->sequence_number, inframe->given_timestamp);
               }
               sync_error_out_of_bounds++;
             } else {

diff --git a/player.h b/player.h
index 6f1a8b5ba8c94840360af41a821d037917c6216a..56ea1f9105b7247e1b72d1b5e7bd4dc0b8a35115 100644 (file)
--- a/player.h
+++ b/player.h
@@ -39,6 +39,7 @@ typedef struct audio_buffer_entry { // decoded audio packets
   int ready;
   int64_t timestamp;
   seq_t sequence_number;
+  uint32_t given_timestamp; // for debugging and checking
   signed short *data;
   int length; // the length of the decoded data
 } abuf_t;
@@ -193,7 +194,7 @@ void player_stop(rtsp_conn_info *conn);
 void player_volume(double f, rtsp_conn_info *conn);
 void player_volume_without_notification(double f, rtsp_conn_info *conn);
 void player_flush(int64_t timestamp, rtsp_conn_info *conn);
-void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len,
+void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp, uint8_t *data, int len,
                        rtsp_conn_info *conn);
 
 int64_t monotonic_timestamp(uint32_t timestamp,

diff --git a/rtp.c b/rtp.c
index e8dc5d7d5c9b6b44c0542a221047e02b5f2e8e44..e4c83a1d62e999105754df96efa48f77e356d701 100644 (file)
--- a/rtp.c
+++ b/rtp.c
@@ -147,7 +147,8 @@ void *rtp_audio_receiver(void *arg) {
         //  debug(3, "RTP: Packets out of sequence: expected: %d, got %d.", last_seqno, seqno);
         last_seqno = seqno; // reset warning...
       }
-      int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
+      uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));      
+      int64_t timestamp = monotonic_timestamp(actual_timestamp, conn);
 
       // if (packet[1]&0x10)
       //       debug(1,"Audio packet Extension bit set.");
@@ -157,7 +158,7 @@ void *rtp_audio_receiver(void *arg) {
 
       // check if packet contains enough content to be reasonable
       if (plen >= 16) {
-        player_put_packet(seqno, timestamp, pktp, plen, conn);
+        player_put_packet(seqno, actual_timestamp, timestamp, pktp, plen, conn);
         continue;
       }
       if (type == 0x56 && seqno == 0) {
@@ -361,8 +362,9 @@ void *rtp_control_receiver(void *arg) {
           debug(3,"Resent audio packet %u received in the control port.",seqno);
         else
           debug(1,"Audio packet %u received in the control port.",seqno);
+        uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));
         
-        int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
+        int64_t timestamp = monotonic_timestamp(actual_timestamp, conn);
 
         pktp += 12;
         plen -= 12;
@@ -370,7 +372,7 @@ void *rtp_control_receiver(void *arg) {
         // check if packet contains enough content to be reasonable
 
         if (plen >= 16) {
-          player_put_packet(seqno, timestamp, pktp, plen, conn);
+          player_put_packet(seqno, actual_timestamp, timestamp, pktp, plen, conn);
           continue;
         } else {
           debug(3, "Too-short retransmitted audio packet received in control port, ignored.");