Wire up PacketTime to ReceiveStreams.

webrtc/video/end_to_end_tests.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.
  */
 #include <algorithm>
@@ skipping 975 matching lines @@
   class PacketInputObserver : public PacketReceiver {
    public:
     explicit PacketInputObserver(PacketReceiver* receiver)
         : receiver_(receiver), delivered_packet_(EventWrapper::Create()) {}
 
     EventTypeWrapper Wait() {
       return delivered_packet_->Wait(kDefaultTimeoutMs);
     }
 
    private:
-    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 {
       if (RtpHeaderParser::IsRtcp(packet, length)) {
-        return receiver_->DeliverPacket(media_type, packet, length);
+        return receiver_->DeliverPacket(media_type, packet, length,
+                                        packet_time);
       } else {
         DeliveryStatus delivery_status =
-            receiver_->DeliverPacket(media_type, packet, length);
+            receiver_->DeliverPacket(media_type, packet, length, packet_time);
         EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
         delivered_packet_->Set();
         return delivery_status;
       }
     }
 
     PacketReceiver* receiver_;
     rtc::scoped_ptr<EventWrapper> delivered_packet_;
   };
 
@@ skipping 532 matching lines @@
 
 TEST_F(EndToEndTest, VerifyBandwidthStats) {
   class RtcpObserver : public test::EndToEndTest, public PacketReceiver {
    public:
     RtcpObserver()
         : EndToEndTest(kDefaultTimeoutMs),
           sender_call_(nullptr),
           receiver_call_(nullptr),
           has_seen_pacer_delay_(false) {}
 
-    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 {
       Call::Stats sender_stats = sender_call_->GetStats();
       Call::Stats receiver_stats = receiver_call_->GetStats();
       if (!has_seen_pacer_delay_)
         has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
       if (sender_stats.send_bandwidth_bps > 0 &&
           receiver_stats.recv_bandwidth_bps > 0 && has_seen_pacer_delay_) {
         observation_complete_->Set();
       }
       return receiver_call_->Receiver()->DeliverPacket(media_type, packet,
-                                                       length);
+                                                       length, packet_time);
     }
 
     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
       receiver_call_ = receiver_call;
     }
 
     void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
                                            "non-zero bandwidth stats.";
@@ skipping 135 matching lines @@
           start_runtime_ms_(-1) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (MinMetricRunTimePassed())
         observation_complete_->Set();
 
       return SEND_PACKET;
     }
 
-    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 {
       // GetStats calls GetSendChannelRtcpStatistics
       // (via VideoSendStream::GetRtt) which updates ReportBlockStats used by
       // WebRTC.Video.SentPacketsLostInPercent.
       // TODO(asapersson): Remove dependency on calling GetStats.
       sender_call_->GetStats();
       return receiver_call_->Receiver()->DeliverPacket(media_type, packet,
-                                                       length);
+                                                       length, packet_time);
     }
 
     bool MinMetricRunTimePassed() {
       int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
       if (start_runtime_ms_ == -1) {
         start_runtime_ms_ = now;
         return false;
       }
       int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
       return elapsed_sec > metrics::kMinRunTimeInSeconds * 2;
@@ skipping 1299 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc