Changeset 2224

Timestamp:

Aug 19, 2008 12:11:39 PM (16 years ago)

Author:

nanang

Message:

Ticket #593: Fixed RTP timestamp calculation in RX & TX for multichannel audio.

File:

• pjproject/trunk/pjmedia/src/pjmedia/stream.c (modified) (8 diffs)

pjproject/trunk/pjmedia/src/pjmedia/stream.c

@@ -144 +144 @@
                                                  normalization process 
                                                  is needed                  */
-    unsigned                 rtp_tx_samples_per_pkt;
-                                            /**< Normalized samples per packet 
+    unsigned                 rtp_tx_ts_len_per_pkt;
+                                            /**< Normalized ts length per packet
                                                  transmitted according to 
                                                  'erroneous' definition     */
-    unsigned                 rtp_rx_samples_per_frame;
-                                            /**< Normalized samples per frame 
+    unsigned                 rtp_rx_ts_len_per_frame;
+                                            /**< Normalized ts length per frame
                                                  received according to 
                                                  'erroneous' definition     */
@@ -687 +687 @@
     /* Number of samples in the frame */
     if (frame->type == PJMEDIA_FRAME_TYPE_AUDIO)
-        ts_len = (frame->size >> 1);
+        ts_len = (frame->size >> 1) / stream->codec_param.info.channel_cnt;
     else
         ts_len = 0;
@@ -699 +699 @@
      */
     if (stream->has_g722_mpeg_bug)
-        rtp_ts_len = stream->rtp_tx_samples_per_pkt;
+        rtp_ts_len = stream->rtp_tx_ts_len_per_pkt;
     else
         rtp_ts_len = ts_len;
@@ -1247 +1247 @@
                                     stream->port.info.samples_per_frame >> 1))
                     {
-                        if (peer_frm_ts_diff < stream->rtp_rx_samples_per_frame)
-                            stream->rtp_rx_samples_per_frame = peer_frm_ts_diff;
+                        if (peer_frm_ts_diff < stream->rtp_rx_ts_len_per_frame)
+                            stream->rtp_rx_ts_len_per_frame = peer_frm_ts_diff;
 
                         if (--stream->rtp_rx_check_cnt == 0) {
                             PJ_LOG(4, (THIS_FILE, "G722 codec used, remote"
                                        " samples per frame detected = %d", 
-                                       stream->rtp_rx_samples_per_frame));
+                                       stream->rtp_rx_ts_len_per_frame));
                             
                             /* Reset jitter buffer once detection done */
@@ -1265 +1265 @@
             }
 
-            ts_span = stream->rtp_rx_samples_per_frame;
+            ts_span = stream->rtp_rx_ts_len_per_frame;
 
         } else {
             ts_span = stream->codec_param.info.frm_ptime * 
-                      stream->codec_param.info.clock_rate *
-                      stream->codec_param.info.channel_cnt /
+                      stream->codec_param.info.clock_rate /
                       1000;
         }
 #else
         ts_span = stream->codec_param.info.frm_ptime * 
-                  stream->codec_param.info.clock_rate *
-                  stream->codec_param.info.channel_cnt /
+                  stream->codec_param.info.clock_rate /
                   1000;
 #endif
@@ -1563 +1561 @@
 
         stream->enc_samples_per_pkt = stream->codec_param.info.enc_ptime *
+                                      stream->codec_param.info.channel_cnt *
                                       stream->port.info.clock_rate / 1000;
 
@@ -1613 +1612 @@
     stream->rtp_rx_last_ts = 0;
     stream->rtp_rx_last_cnt = 0;
-    stream->rtp_tx_samples_per_pkt = stream->enc_samples_per_pkt;
-    stream->rtp_rx_samples_per_frame = stream->port.info.samples_per_frame;
+    stream->rtp_tx_ts_len_per_pkt = stream->enc_samples_per_pkt /
+                                     stream->codec_param.info.channel_cnt;
+    stream->rtp_rx_ts_len_per_frame = stream->port.info.samples_per_frame / 
+                                       stream->codec_param.info.channel_cnt;
 
     /* Init RTCP session: */
@@ -1626 +1627 @@
         stream->has_g722_mpeg_bug = PJ_TRUE;
         /* RTP clock rate = 1/2 real clock rate */
-        stream->rtp_tx_samples_per_pkt >>= 1;
+        stream->rtp_tx_ts_len_per_pkt >>= 1;
     } else {
         pjmedia_rtcp_init(&stream->rtcp, stream->port.info.name.ptr,