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| 1 // Copyright 2013 The Chromium Authors. All rights reserved. | |
| 2 // Use of this source code is governed by a BSD-style license that can be | |
| 3 // found in the LICENSE file. | |
| 4 | |
| 5 #include <deque> | |
| 6 #include <utility> | |
| 7 | |
| 8 #include "base/bind.h" | |
| 9 #include "base/memory/ref_counted.h" | |
| 10 #include "base/memory/scoped_ptr.h" | |
| 11 #include "base/test/simple_test_tick_clock.h" | |
| 12 #include "media/cast/audio_receiver/audio_receiver.h" | |
| 13 #include "media/cast/cast_defines.h" | |
| 14 #include "media/cast/cast_environment.h" | |
| 15 #include "media/cast/logging/simple_event_subscriber.h" | |
| 16 #include "media/cast/rtcp/test_rtcp_packet_builder.h" | |
| 17 #include "media/cast/test/fake_single_thread_task_runner.h" | |
| 18 #include "media/cast/test/utility/default_config.h" | |
| 19 #include "media/cast/transport/pacing/mock_paced_packet_sender.h" | |
| 20 #include "testing/gmock/include/gmock/gmock.h" | |
| 21 | |
| 22 using ::testing::_; | |
| 23 | |
| 24 namespace media { | |
| 25 namespace cast { | |
| 26 | |
| 27 namespace { | |
| 28 | |
| 29 const uint32 kFirstFrameId = 1234; | |
| 30 const int kPlayoutDelayMillis = 300; | |
| 31 | |
| 32 class FakeAudioClient { | |
| 33 public: | |
| 34 FakeAudioClient() : num_called_(0) {} | |
| 35 virtual ~FakeAudioClient() {} | |
| 36 | |
| 37 void AddExpectedResult(uint32 expected_frame_id, | |
| 38 const base::TimeTicks& expected_playout_time) { | |
| 39 expected_results_.push_back( | |
| 40 std::make_pair(expected_frame_id, expected_playout_time)); | |
| 41 } | |
| 42 | |
| 43 void DeliverEncodedAudioFrame( | |
| 44 scoped_ptr<transport::EncodedFrame> audio_frame) { | |
| 45 SCOPED_TRACE(::testing::Message() << "num_called_ is " << num_called_); | |
| 46 ASSERT_FALSE(!audio_frame) | |
| 47 << "If at shutdown: There were unsatisfied requests enqueued."; | |
| 48 ASSERT_FALSE(expected_results_.empty()); | |
| 49 EXPECT_EQ(expected_results_.front().first, audio_frame->frame_id); | |
| 50 EXPECT_EQ(expected_results_.front().second, audio_frame->reference_time); | |
| 51 expected_results_.pop_front(); | |
| 52 num_called_++; | |
| 53 } | |
| 54 | |
| 55 int number_times_called() const { return num_called_; } | |
| 56 | |
| 57 private: | |
| 58 std::deque<std::pair<uint32, base::TimeTicks> > expected_results_; | |
| 59 int num_called_; | |
| 60 | |
| 61 DISALLOW_COPY_AND_ASSIGN(FakeAudioClient); | |
| 62 }; | |
| 63 | |
| 64 } // namespace | |
| 65 | |
| 66 class AudioReceiverTest : public ::testing::Test { | |
| 67 protected: | |
| 68 AudioReceiverTest() { | |
| 69 // Configure the audio receiver to use PCM16. | |
| 70 audio_config_ = GetDefaultAudioReceiverConfig(); | |
| 71 audio_config_.rtp_max_delay_ms = kPlayoutDelayMillis; | |
| 72 audio_config_.frequency = 16000; | |
| 73 audio_config_.channels = 1; | |
| 74 audio_config_.codec.audio = transport::kPcm16; | |
| 75 testing_clock_ = new base::SimpleTestTickClock(); | |
| 76 testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks()); | |
| 77 start_time_ = testing_clock_->NowTicks(); | |
| 78 task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_); | |
| 79 | |
| 80 cast_environment_ = new CastEnvironment( | |
| 81 scoped_ptr<base::TickClock>(testing_clock_).Pass(), | |
| 82 task_runner_, | |
| 83 task_runner_, | |
| 84 task_runner_); | |
| 85 | |
| 86 receiver_.reset(new AudioReceiver(cast_environment_, audio_config_, | |
| 87 &mock_transport_)); | |
| 88 } | |
| 89 | |
| 90 virtual ~AudioReceiverTest() {} | |
| 91 | |
| 92 virtual void SetUp() { | |
| 93 payload_.assign(kMaxIpPacketSize, 0); | |
| 94 rtp_header_.is_key_frame = true; | |
| 95 rtp_header_.frame_id = kFirstFrameId; | |
| 96 rtp_header_.packet_id = 0; | |
| 97 rtp_header_.max_packet_id = 0; | |
| 98 rtp_header_.reference_frame_id = rtp_header_.frame_id; | |
| 99 rtp_header_.rtp_timestamp = 0; | |
| 100 } | |
| 101 | |
| 102 void FeedOneFrameIntoReceiver() { | |
| 103 receiver_->OnReceivedPayloadData( | |
| 104 payload_.data(), payload_.size(), rtp_header_); | |
| 105 } | |
| 106 | |
| 107 void FeedLipSyncInfoIntoReceiver() { | |
| 108 const base::TimeTicks now = testing_clock_->NowTicks(); | |
| 109 const int64 rtp_timestamp = (now - start_time_) * | |
| 110 audio_config_.frequency / base::TimeDelta::FromSeconds(1); | |
| 111 CHECK_LE(0, rtp_timestamp); | |
| 112 uint32 ntp_seconds; | |
| 113 uint32 ntp_fraction; | |
| 114 ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction); | |
| 115 TestRtcpPacketBuilder rtcp_packet; | |
| 116 rtcp_packet.AddSrWithNtp(audio_config_.incoming_ssrc, | |
| 117 ntp_seconds, ntp_fraction, | |
| 118 static_cast<uint32>(rtp_timestamp)); | |
| 119 receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass()); | |
| 120 } | |
| 121 | |
| 122 FrameReceiverConfig audio_config_; | |
| 123 std::vector<uint8> payload_; | |
| 124 RtpCastHeader rtp_header_; | |
| 125 base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment. | |
| 126 base::TimeTicks start_time_; | |
| 127 transport::MockPacedPacketSender mock_transport_; | |
| 128 scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_; | |
| 129 scoped_refptr<CastEnvironment> cast_environment_; | |
| 130 FakeAudioClient fake_audio_client_; | |
| 131 | |
| 132 // Important for the AudioReceiver to be declared last, since its dependencies | |
| 133 // must remain alive until after its destruction. | |
| 134 scoped_ptr<AudioReceiver> receiver_; | |
| 135 }; | |
| 136 | |
| 137 TEST_F(AudioReceiverTest, ReceivesOneFrame) { | |
| 138 SimpleEventSubscriber event_subscriber; | |
| 139 cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber); | |
| 140 | |
| 141 EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _)) | |
| 142 .WillRepeatedly(testing::Return(true)); | |
| 143 | |
| 144 FeedLipSyncInfoIntoReceiver(); | |
| 145 task_runner_->RunTasks(); | |
| 146 | |
| 147 // Enqueue a request for an audio frame. | |
| 148 receiver_->GetEncodedAudioFrame( | |
| 149 base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame, | |
| 150 base::Unretained(&fake_audio_client_))); | |
| 151 | |
| 152 // The request should not be satisfied since no packets have been received. | |
| 153 task_runner_->RunTasks(); | |
| 154 EXPECT_EQ(0, fake_audio_client_.number_times_called()); | |
| 155 | |
| 156 // Deliver one audio frame to the receiver and expect to get one frame back. | |
| 157 const base::TimeDelta target_playout_delay = | |
| 158 base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis); | |
| 159 fake_audio_client_.AddExpectedResult( | |
| 160 kFirstFrameId, testing_clock_->NowTicks() + target_playout_delay); | |
| 161 FeedOneFrameIntoReceiver(); | |
| 162 task_runner_->RunTasks(); | |
| 163 EXPECT_EQ(1, fake_audio_client_.number_times_called()); | |
| 164 | |
| 165 std::vector<FrameEvent> frame_events; | |
| 166 event_subscriber.GetFrameEventsAndReset(&frame_events); | |
| 167 | |
| 168 ASSERT_TRUE(!frame_events.empty()); | |
| 169 EXPECT_EQ(FRAME_ACK_SENT, frame_events.begin()->type); | |
| 170 EXPECT_EQ(AUDIO_EVENT, frame_events.begin()->media_type); | |
| 171 EXPECT_EQ(rtp_header_.frame_id, frame_events.begin()->frame_id); | |
| 172 EXPECT_EQ(rtp_header_.rtp_timestamp, frame_events.begin()->rtp_timestamp); | |
| 173 | |
| 174 cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber); | |
| 175 } | |
| 176 | |
| 177 TEST_F(AudioReceiverTest, ReceivesFramesSkippingWhenAppropriate) { | |
| 178 EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _)) | |
| 179 .WillRepeatedly(testing::Return(true)); | |
| 180 | |
| 181 const uint32 rtp_advance_per_frame = | |
| 182 audio_config_.frequency / audio_config_.max_frame_rate; | |
| 183 const base::TimeDelta time_advance_per_frame = | |
| 184 base::TimeDelta::FromSeconds(1) / audio_config_.max_frame_rate; | |
| 185 | |
| 186 FeedLipSyncInfoIntoReceiver(); | |
| 187 task_runner_->RunTasks(); | |
| 188 const base::TimeTicks first_frame_capture_time = testing_clock_->NowTicks(); | |
| 189 | |
| 190 // Enqueue a request for an audio frame. | |
| 191 const FrameEncodedCallback frame_encoded_callback = | |
| 192 base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame, | |
| 193 base::Unretained(&fake_audio_client_)); | |
| 194 receiver_->GetEncodedAudioFrame(frame_encoded_callback); | |
| 195 task_runner_->RunTasks(); | |
| 196 EXPECT_EQ(0, fake_audio_client_.number_times_called()); | |
| 197 | |
| 198 // Receive one audio frame and expect to see the first request satisfied. | |
| 199 const base::TimeDelta target_playout_delay = | |
| 200 base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis); | |
| 201 fake_audio_client_.AddExpectedResult( | |
| 202 kFirstFrameId, first_frame_capture_time + target_playout_delay); | |
| 203 rtp_header_.rtp_timestamp = 0; | |
| 204 FeedOneFrameIntoReceiver(); | |
| 205 task_runner_->RunTasks(); | |
| 206 EXPECT_EQ(1, fake_audio_client_.number_times_called()); | |
| 207 | |
| 208 // Enqueue a second request for an audio frame, but it should not be | |
| 209 // fulfilled yet. | |
| 210 receiver_->GetEncodedAudioFrame(frame_encoded_callback); | |
| 211 task_runner_->RunTasks(); | |
| 212 EXPECT_EQ(1, fake_audio_client_.number_times_called()); | |
| 213 | |
| 214 // Receive one audio frame out-of-order: Make sure that we are not continuous | |
| 215 // and that the RTP timestamp represents a time in the future. | |
| 216 rtp_header_.is_key_frame = false; | |
| 217 rtp_header_.frame_id = kFirstFrameId + 2; | |
| 218 rtp_header_.reference_frame_id = 0; | |
| 219 rtp_header_.rtp_timestamp += 2 * rtp_advance_per_frame; | |
| 220 fake_audio_client_.AddExpectedResult( | |
| 221 kFirstFrameId + 2, | |
| 222 first_frame_capture_time + 2 * time_advance_per_frame + | |
| 223 target_playout_delay); | |
| 224 FeedOneFrameIntoReceiver(); | |
| 225 | |
| 226 // Frame 2 should not come out at this point in time. | |
| 227 task_runner_->RunTasks(); | |
| 228 EXPECT_EQ(1, fake_audio_client_.number_times_called()); | |
| 229 | |
| 230 // Enqueue a third request for an audio frame. | |
| 231 receiver_->GetEncodedAudioFrame(frame_encoded_callback); | |
| 232 task_runner_->RunTasks(); | |
| 233 EXPECT_EQ(1, fake_audio_client_.number_times_called()); | |
| 234 | |
| 235 // Now, advance time forward such that the receiver is convinced it should | |
| 236 // skip Frame 2. Frame 3 is emitted (to satisfy the second request) because a | |
| 237 // decision was made to skip over the no-show Frame 2. | |
| 238 testing_clock_->Advance(2 * time_advance_per_frame + target_playout_delay); | |
| 239 task_runner_->RunTasks(); | |
| 240 EXPECT_EQ(2, fake_audio_client_.number_times_called()); | |
| 241 | |
| 242 // Receive Frame 4 and expect it to fulfill the third request immediately. | |
| 243 rtp_header_.frame_id = kFirstFrameId + 3; | |
| 244 rtp_header_.reference_frame_id = rtp_header_.frame_id - 1; | |
| 245 rtp_header_.rtp_timestamp += rtp_advance_per_frame; | |
| 246 fake_audio_client_.AddExpectedResult( | |
| 247 kFirstFrameId + 3, first_frame_capture_time + 3 * time_advance_per_frame + | |
| 248 target_playout_delay); | |
| 249 FeedOneFrameIntoReceiver(); | |
| 250 task_runner_->RunTasks(); | |
| 251 EXPECT_EQ(3, fake_audio_client_.number_times_called()); | |
| 252 | |
| 253 // Move forward to the playout time of an unreceived Frame 5. Expect no | |
| 254 // additional frames were emitted. | |
| 255 testing_clock_->Advance(3 * time_advance_per_frame); | |
| 256 task_runner_->RunTasks(); | |
| 257 EXPECT_EQ(3, fake_audio_client_.number_times_called()); | |
| 258 } | |
| 259 | |
| 260 } // namespace cast | |
| 261 } // namespace media | |
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Chromium Code Reviews
Issue 308043006:
[Cast] Clean-up: Merge RtpReceiver+AudioReceiver+VideoReceiver-->FrameReceiver. (Closed)
Base URL: svn://svn.chromium.org/chrome/trunk/src