Wire up PacketTime to ReceiveStreams.

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 2930 matching lines @@
 
   return true;
 }
 
 void WebRtcVoiceMediaChannel::OnPacketReceived(
     rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   // If hooked up to a "Call", forward packet there too.
   if (call_) {
-    call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
-        reinterpret_cast<const uint8_t*>(packet->data()), packet->size());
+    const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
+                                                packet_time.not_before);
+    call_->Receiver()->DeliverPacket(
+        webrtc::MediaType::AUDIO,
+        reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+        webrtc_packet_time);
   }
 
   // Pick which channel to send this packet to. If this packet doesn't match
   // any multiplexed streams, just send it to the default channel. Otherwise,
   // send it to the specific decoder instance for that stream.
   int which_channel =
       GetReceiveChannelNum(ParseSsrc(packet->data(), packet->size(), false));
   if (which_channel == -1) {
     which_channel = voe_channel();
   }
@@ skipping 19 matching lines @@
       which_channel, packet->data(), packet->size(),
       webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
 }
 
 void WebRtcVoiceMediaChannel::OnRtcpReceived(
     rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   // If hooked up to a "Call", forward packet there too.
   if (call_) {
-    call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
-        reinterpret_cast<const uint8_t*>(packet->data()), packet->size());
+    const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
+                                                packet_time.not_before);
+    call_->Receiver()->DeliverPacket(
+        webrtc::MediaType::AUDIO,
+        reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+        webrtc_packet_time);
   }
 
   // Sending channels need all RTCP packets with feedback information.
   // Even sender reports can contain attached report blocks.
   // Receiving channels need sender reports in order to create
   // correct receiver reports.
   int type = 0;
   if (!GetRtcpType(packet->data(), packet->size(), &type)) {
     LOG(LS_WARNING) << "Failed to parse type from received RTCP packet";
     return;
@@ skipping 609 matching lines @@
 
 int WebRtcSoundclipStream::Rewind() {
   mem_.Rewind();
   // Return -1 to keep VoiceEngine from looping.
   return (loop_) ? 0 : -1;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE