[Cast] Repair receiver playout time calculations and frame skip logic.

media/cast/audio_receiver/audio_receiver_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 <deque>
+#include <utility>
+
 #include "base/bind.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/audio_receiver/audio_receiver.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/logging/simple_event_subscriber.h"
 #include "media/cast/rtcp/test_rtcp_packet_builder.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/transport/pacing/mock_paced_packet_sender.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
+using ::testing::_;
+
 namespace media {
 namespace cast {
 
-using ::testing::_;
-
 namespace {
 
-const int64 kStartMillisecond = INT64_C(12345678900000);
 const uint32 kFirstFrameId = 1234;
+const int kPlayoutDelayMillis = 300;
 
 class FakeAudioClient {
  public:
   FakeAudioClient() : num_called_(0) {}
   virtual ~FakeAudioClient() {}
 
-  void SetNextExpectedResult(uint32 expected_frame_id,
-                             const base::TimeTicks& expected_playout_time) {
-    expected_frame_id_ = expected_frame_id;
-    expected_playout_time_ = expected_playout_time;
+  void AddExpectedResult(uint32 expected_frame_id,
+                         const base::TimeTicks& expected_playout_time) {
+    expected_results_.push_back(
+        std::make_pair(expected_frame_id, expected_playout_time));
   }
 
   void DeliverEncodedAudioFrame(
       scoped_ptr<transport::EncodedAudioFrame> audio_frame,
       const base::TimeTicks& playout_time) {
+    SCOPED_TRACE(::testing::Message() << "num_called_ is " << num_called_);
     ASSERT_FALSE(!audio_frame)
         << "If at shutdown: There were unsatisfied requests enqueued.";
-    EXPECT_EQ(expected_frame_id_, audio_frame->frame_id);
+    ASSERT_FALSE(expected_results_.empty());
+    EXPECT_EQ(expected_results_.front().first, audio_frame->frame_id);
     EXPECT_EQ(transport::kPcm16, audio_frame->codec);
-    EXPECT_EQ(expected_playout_time_, playout_time);
+    EXPECT_EQ(expected_results_.front().second, playout_time);
+    expected_results_.pop_front();
     num_called_++;
   }
 
   int number_times_called() const { return num_called_; }
 
  private:
+  std::deque<std::pair<uint32, base::TimeTicks> > expected_results_;
   int num_called_;
-  uint32 expected_frame_id_;
-  base::TimeTicks expected_playout_time_;
 
   DISALLOW_COPY_AND_ASSIGN(FakeAudioClient);
 };
 
 }  // namespace
 
 class AudioReceiverTest : public ::testing::Test {
  protected:
   AudioReceiverTest() {
     // Configure the audio receiver to use PCM16.
     audio_config_.rtp_payload_type = 127;
     audio_config_.frequency = 16000;
     audio_config_.channels = 1;
     audio_config_.codec = transport::kPcm16;
     audio_config_.use_external_decoder = true;
     audio_config_.feedback_ssrc = 1234;
+    audio_config_.incoming_ssrc = 5678;
+    audio_config_.rtp_max_delay_ms = kPlayoutDelayMillis;
     testing_clock_ = new base::SimpleTestTickClock();
-    testing_clock_->Advance(
-        base::TimeDelta::FromMilliseconds(kStartMillisecond));
+    testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
+    start_time_ = testing_clock_->NowTicks();
     task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
 
     cast_environment_ = new CastEnvironment(
         scoped_ptr<base::TickClock>(testing_clock_).Pass(),
         task_runner_,
         task_runner_,
         task_runner_);
 
     receiver_.reset(new AudioReceiver(cast_environment_, audio_config_,
                                       &mock_transport_));
   }
 
   virtual ~AudioReceiverTest() {}
 
   virtual void SetUp() {
     payload_.assign(kMaxIpPacketSize, 0);
     rtp_header_.is_key_frame = true;
     rtp_header_.frame_id = kFirstFrameId;
     rtp_header_.packet_id = 0;
     rtp_header_.max_packet_id = 0;
     rtp_header_.reference_frame_id = 0;
     rtp_header_.rtp_timestamp = 0;
   }
 
   void FeedOneFrameIntoReceiver() {
     receiver_->OnReceivedPayloadData(
         payload_.data(), payload_.size(), rtp_header_);
   }
 
+  void FeedLipSyncInfoIntoReceiver() {
+    const base::TimeTicks now = testing_clock_->NowTicks();
+    const int64 rtp_timestamp = (now - start_time_) *
+        audio_config_.frequency / base::TimeDelta::FromSeconds(1);
+    CHECK_LE(0, rtp_timestamp);
+    uint32 ntp_seconds;
+    uint32 ntp_fraction;
+    ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction);
+    TestRtcpPacketBuilder rtcp_packet;
+    rtcp_packet.AddSrWithNtp(audio_config_.incoming_ssrc,
+                             ntp_seconds, ntp_fraction,
+                             static_cast<uint32>(rtp_timestamp));
+    receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass());
+  }
+
   AudioReceiverConfig audio_config_;
   std::vector<uint8> payload_;
   RtpCastHeader rtp_header_;
   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
+  base::TimeTicks start_time_;
   transport::MockPacedPacketSender mock_transport_;
   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
   scoped_refptr<CastEnvironment> cast_environment_;
   FakeAudioClient fake_audio_client_;
 
   // Important for the AudioReceiver to be declared last, since its dependencies
   // must remain alive until after its destruction.
   scoped_ptr<AudioReceiver> receiver_;
 };
 
-TEST_F(AudioReceiverTest, GetOnePacketEncodedFrame) {
+TEST_F(AudioReceiverTest, ReceivesOneFrame) {
   SimpleEventSubscriber event_subscriber;
   cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
 
-  EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _)).Times(1);
+  EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
+      .WillRepeatedly(testing::Return(true));
+
+  FeedLipSyncInfoIntoReceiver();
+  task_runner_->RunTasks();
 
   // Enqueue a request for an audio frame.
   receiver_->GetEncodedAudioFrame(
       base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame,
                  base::Unretained(&fake_audio_client_)));
 
   // The request should not be satisfied since no packets have been received.
   task_runner_->RunTasks();
   EXPECT_EQ(0, fake_audio_client_.number_times_called());
 
   // Deliver one audio frame to the receiver and expect to get one frame back.
-  fake_audio_client_.SetNextExpectedResult(kFirstFrameId,
-                                           testing_clock_->NowTicks());
+  const base::TimeDelta target_playout_delay =
+      base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
+  fake_audio_client_.AddExpectedResult(
+      kFirstFrameId, testing_clock_->NowTicks() + target_playout_delay);
   FeedOneFrameIntoReceiver();
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
   std::vector<FrameEvent> frame_events;
   event_subscriber.GetFrameEventsAndReset(&frame_events);
 
   ASSERT_TRUE(!frame_events.empty());
   EXPECT_EQ(FRAME_ACK_SENT, frame_events.begin()->type);
   EXPECT_EQ(AUDIO_EVENT, frame_events.begin()->media_type);
   EXPECT_EQ(rtp_header_.frame_id, frame_events.begin()->frame_id);
   EXPECT_EQ(rtp_header_.rtp_timestamp, frame_events.begin()->rtp_timestamp);
 
   cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
 }
 
-TEST_F(AudioReceiverTest, MultiplePendingGetCalls) {
+TEST_F(AudioReceiverTest, ReceivesFramesSkippingWhenAppropriate) {
   EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
       .WillRepeatedly(testing::Return(true));
 
+  const uint32 rtp_advance_per_frame = audio_config_.frequency / 100;
+  const base::TimeDelta time_advance_per_frame =
+      base::TimeDelta::FromMilliseconds(10);
+
+  FeedLipSyncInfoIntoReceiver();
+  task_runner_->RunTasks();
+  const base::TimeTicks first_frame_capture_time = testing_clock_->NowTicks();
+
   // Enqueue a request for an audio frame.
   const AudioFrameEncodedCallback frame_encoded_callback =
       base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame,
                  base::Unretained(&fake_audio_client_));
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
   EXPECT_EQ(0, fake_audio_client_.number_times_called());
 
   // Receive one audio frame and expect to see the first request satisfied.
-  fake_audio_client_.SetNextExpectedResult(kFirstFrameId,
-                                           testing_clock_->NowTicks());
+  const base::TimeDelta target_playout_delay =
+      base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
+  fake_audio_client_.AddExpectedResult(
+      kFirstFrameId, first_frame_capture_time + target_playout_delay);
+  rtp_header_.rtp_timestamp = 0;
   FeedOneFrameIntoReceiver();
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
-  TestRtcpPacketBuilder rtcp_packet;
-
-  uint32 ntp_high;
-  uint32 ntp_low;
-  ConvertTimeTicksToNtp(testing_clock_->NowTicks(), &ntp_high, &ntp_low);
-  rtcp_packet.AddSrWithNtp(audio_config_.feedback_ssrc, ntp_high, ntp_low,
-                           rtp_header_.rtp_timestamp);
-
-  testing_clock_->Advance(base::TimeDelta::FromMilliseconds(20));
-
-  receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass());
-
   // Enqueue a second request for an audio frame, but it should not be
   // fulfilled yet.
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
   // Receive one audio frame out-of-order: Make sure that we are not continuous
   // and that the RTP timestamp represents a time in the future.
   rtp_header_.is_key_frame = false;
   rtp_header_.frame_id = kFirstFrameId + 2;
   rtp_header_.reference_frame_id = 0;
-  rtp_header_.rtp_timestamp = 960;
-  fake_audio_client_.SetNextExpectedResult(
+  rtp_header_.rtp_timestamp += 2 * rtp_advance_per_frame;
+  fake_audio_client_.AddExpectedResult(
       kFirstFrameId + 2,
-      testing_clock_->NowTicks() + base::TimeDelta::FromMilliseconds(100));
+      first_frame_capture_time + 2 * time_advance_per_frame +
+          target_playout_delay);
   FeedOneFrameIntoReceiver();
 
   // Frame 2 should not come out at this point in time.
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
   // Enqueue a third request for an audio frame.
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
-  // After 100 ms has elapsed, Frame 2 is emitted (to satisfy the second
-  // request) because a decision was made to skip over the no-show Frame 1.
-  testing_clock_->Advance(base::TimeDelta::FromMilliseconds(100));
+  // Now, advance time forward such that the receiver is convinced it should
+  // skip Frame 2.  Frame 3 is emitted (to satisfy the second request) because a
+  // decision was made to skip over the no-show Frame 2.
+  testing_clock_->Advance(2 * time_advance_per_frame + target_playout_delay);
   task_runner_->RunTasks();
   EXPECT_EQ(2, fake_audio_client_.number_times_called());
 
-  // Receive Frame 3 and expect it to fulfill the third request immediately.
+  // Receive Frame 4 and expect it to fulfill the third request immediately.
   rtp_header_.frame_id = kFirstFrameId + 3;
   rtp_header_.reference_frame_id = rtp_header_.frame_id - 1;
-  rtp_header_.rtp_timestamp = 1280;
-  fake_audio_client_.SetNextExpectedResult(kFirstFrameId + 3,
-                                           testing_clock_->NowTicks());
+  rtp_header_.rtp_timestamp += rtp_advance_per_frame;
+  fake_audio_client_.AddExpectedResult(
+      kFirstFrameId + 3, first_frame_capture_time + 3 * time_advance_per_frame +
+          target_playout_delay);
   FeedOneFrameIntoReceiver();
   task_runner_->RunTasks();
   EXPECT_EQ(3, fake_audio_client_.number_times_called());
 
-  // Move forward another 100 ms and run any pending tasks (there should be
-  // none).  Expect no additional frames where emitted.
-  testing_clock_->Advance(base::TimeDelta::FromMilliseconds(100));
+  // Move forward to the playout time of an unreceived Frame 5.  Expect no
+  // additional frames were emitted.
+  testing_clock_->Advance(3 * time_advance_per_frame);
   task_runner_->RunTasks();
   EXPECT_EQ(3, fake_audio_client_.number_times_called());
 }
 
 }  // namespace cast
 }  // namespace media