[Cast] Aggressively send sender reports until first receiver report is received.

media/cast/audio_sender/audio_sender_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include <stdint.h>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
@@ skipping 15 matching lines @@
 
 class TestPacketSender : public transport::PacketSender {
  public:
   TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
 
   virtual bool SendPacket(transport::PacketRef packet,
                           const base::Closure& cb) OVERRIDE {
     if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
       ++number_of_rtcp_packets_;
     } else {
+      // Check that at least one RTCP packet was sent before the first RTP
+      // packet.  This confirms that the receiver will have the necessary lip
+      // sync info before it has to calculate the playout time of the first
+      // frame.
+      if (number_of_rtp_packets_ == 0)
+        EXPECT_LE(1, number_of_rtcp_packets_);
       ++number_of_rtp_packets_;
     }
     return true;
   }
 
   int number_of_rtp_packets() const { return number_of_rtp_packets_; }
 
   int number_of_rtcp_packets() const { return number_of_rtcp_packets_; }
 
  private:
@@ skipping 35 matching lines @@
         task_runner_,
         &transport_));
     audio_sender_.reset(new AudioSender(
         cast_environment_, audio_config_, transport_sender_.get()));
     task_runner_->RunTasks();
   }
 
   virtual ~AudioSenderTest() {}
 
   static void UpdateCastTransportStatus(transport::CastTransportStatus status) {
-    EXPECT_EQ(status, transport::TRANSPORT_AUDIO_INITIALIZED);
+    EXPECT_EQ(transport::TRANSPORT_AUDIO_INITIALIZED, status);
   }
 
   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
   TestPacketSender transport_;
   scoped_ptr<transport::CastTransportSenderImpl> transport_sender_;
   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
   scoped_ptr<AudioSender> audio_sender_;
   scoped_refptr<CastEnvironment> cast_environment_;
   AudioSenderConfig audio_config_;
 };
 
 TEST_F(AudioSenderTest, Encode20ms) {
   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
   scoped_ptr<AudioBus> bus(
       TestAudioBusFactory(audio_config_.channels,
                           audio_config_.frequency,
                           TestAudioBusFactory::kMiddleANoteFreq,
                           0.5f).NextAudioBus(kDuration));
 
   audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks());
   task_runner_->RunTasks();
-  EXPECT_GE(
-      transport_.number_of_rtp_packets() + transport_.number_of_rtcp_packets(),
-      1);
+  EXPECT_LE(1, transport_.number_of_rtp_packets());
+  EXPECT_LE(1, transport_.number_of_rtcp_packets());
 }
 
 TEST_F(AudioSenderTest, RtcpTimer) {
   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
   scoped_ptr<AudioBus> bus(
       TestAudioBusFactory(audio_config_.channels,
                           audio_config_.frequency,
                           TestAudioBusFactory::kMiddleANoteFreq,
                           0.5f).NextAudioBus(kDuration));
 
   audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks());
   task_runner_->RunTasks();
 
   // Make sure that we send at least one RTCP packet.
   base::TimeDelta max_rtcp_timeout =
       base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2);
   testing_clock_->Advance(max_rtcp_timeout);
   task_runner_->RunTasks();
-  EXPECT_GE(transport_.number_of_rtp_packets(), 1);
-  EXPECT_EQ(transport_.number_of_rtcp_packets(), 1);
+  EXPECT_LE(1, transport_.number_of_rtp_packets());
+  EXPECT_LE(1, transport_.number_of_rtcp_packets());
 }
 
 }  // namespace cast
 }  // namespace media