Wire up PacketTime to ReceiveStreams.

webrtc/video/call.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 75 matching lines @@
       const VideoEncoderConfig& encoder_config) override;
   void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
 
   webrtc::VideoReceiveStream* CreateVideoReceiveStream(
       const webrtc::VideoReceiveStream::Config& config) override;
   void DestroyVideoReceiveStream(
       webrtc::VideoReceiveStream* receive_stream) override;
 
   Stats GetStats() const override;
 
-  DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
-                               size_t length) override;
+  DeliveryStatus DeliverPacket(MediaType media_type,
+                               const uint8_t* packet,
+                               size_t length,
+                               const PacketTime& packet_time) override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
   void SignalNetworkState(NetworkState state) override;
 
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
-  DeliveryStatus DeliverRtp(MediaType media_type, const uint8_t* packet,
-                            size_t length);
+  DeliveryStatus DeliverRtp(MediaType media_type,
+                            const uint8_t* packet,
+                            size_t length,
+                            const PacketTime& packet_time);
 
   void SetBitrateControllerConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config);
 
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
   const int num_cpu_cores_;
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
   const rtc::scoped_ptr<ChannelGroup> channel_group_;
@@ skipping 382 matching lines @@
     for (VideoSendStream* stream : video_send_streams_) {
       if (stream->DeliverRtcp(packet, length))
         rtcp_delivered = true;
     }
   }
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
-                                                size_t length) {
+                                                size_t length,
+                                                const PacketTime& packet_time) {
   // Minimum RTP header size.
   if (length < 12)
     return DELIVERY_PACKET_ERROR;
 
   uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
 
   ReadLockScoped read_lock(*receive_crit_);
   if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
     auto it = audio_receive_ssrcs_.find(ssrc);
     if (it != audio_receive_ssrcs_.end()) {
-      return it->second->DeliverRtp(packet, length) ? DELIVERY_OK
-                                                    : DELIVERY_PACKET_ERROR;
+      return it->second->DeliverRtp(packet, length, packet_time)
+                 ? DELIVERY_OK
+                 : DELIVERY_PACKET_ERROR;
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     auto it = video_receive_ssrcs_.find(ssrc);
     if (it != video_receive_ssrcs_.end()) {
-      return it->second->DeliverRtp(packet, length) ? DELIVERY_OK
-                                                    : DELIVERY_PACKET_ERROR;
+      return it->second->DeliverRtp(packet, length, packet_time)
+                 ? DELIVERY_OK
+                 : DELIVERY_PACKET_ERROR;
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
 
-PacketReceiver::DeliveryStatus Call::DeliverPacket(MediaType media_type,
-                                                   const uint8_t* packet,
-                                                   size_t length) {
+PacketReceiver::DeliveryStatus Call::DeliverPacket(
+    MediaType media_type,
+    const uint8_t* packet,
+    size_t length,
+    const PacketTime& packet_time) {
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
-  return DeliverRtp(media_type, packet, length);
+  return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc