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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include <string> | 11 #include <string> |
12 #include <vector> | 12 #include <vector> |
13 | 13 |
14 #include "testing/gtest/include/gtest/gtest.h" | 14 #include "testing/gtest/include/gtest/gtest.h" |
15 | 15 |
16 #include "webrtc/audio/audio_receive_stream.h" | 16 #include "webrtc/audio/audio_receive_stream.h" |
17 #include "webrtc/audio/conversion.h" | 17 #include "webrtc/audio/conversion.h" |
| 18 #include "webrtc/call/mock/mock_rtc_event_log.h" |
18 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" | 19 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" |
19 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller
.h" | 20 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller
.h" |
20 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_cont
roller.h" | 21 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_cont
roller.h" |
21 #include "webrtc/modules/pacing/packet_router.h" | 22 #include "webrtc/modules/pacing/packet_router.h" |
22 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitra
te_estimator.h" | 23 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitra
te_estimator.h" |
23 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 24 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
24 #include "webrtc/system_wrappers/include/clock.h" | 25 #include "webrtc/system_wrappers/include/clock.h" |
25 #include "webrtc/test/mock_voe_channel_proxy.h" | 26 #include "webrtc/test/mock_voe_channel_proxy.h" |
26 #include "webrtc/test/mock_voice_engine.h" | 27 #include "webrtc/test/mock_voice_engine.h" |
27 | 28 |
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63 const NetworkStatistics kNetworkStats = { | 64 const NetworkStatistics kNetworkStats = { |
64 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0}; | 65 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0}; |
65 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest(); | 66 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest(); |
66 | 67 |
67 struct ConfigHelper { | 68 struct ConfigHelper { |
68 ConfigHelper() | 69 ConfigHelper() |
69 : simulated_clock_(123456), | 70 : simulated_clock_(123456), |
70 decoder_factory_(new rtc::RefCountedObject<MockAudioDecoderFactory>), | 71 decoder_factory_(new rtc::RefCountedObject<MockAudioDecoderFactory>), |
71 congestion_controller_(&simulated_clock_, | 72 congestion_controller_(&simulated_clock_, |
72 &bitrate_observer_, | 73 &bitrate_observer_, |
73 &remote_bitrate_observer_) { | 74 &remote_bitrate_observer_, |
| 75 &event_log_) { |
74 using testing::Invoke; | 76 using testing::Invoke; |
75 | 77 |
76 EXPECT_CALL(voice_engine_, | 78 EXPECT_CALL(voice_engine_, |
77 RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); | 79 RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); |
78 EXPECT_CALL(voice_engine_, | 80 EXPECT_CALL(voice_engine_, |
79 DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); | 81 DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); |
80 AudioState::Config config; | 82 AudioState::Config config; |
81 config.voice_engine = &voice_engine_; | 83 config.voice_engine = &voice_engine_; |
82 audio_state_ = AudioState::Create(config); | 84 audio_state_ = AudioState::Create(config); |
83 | 85 |
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102 EXPECT_CALL(congestion_controller_, packet_router()) | 104 EXPECT_CALL(congestion_controller_, packet_router()) |
103 .WillOnce(Return(&packet_router_)); | 105 .WillOnce(Return(&packet_router_)); |
104 EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()) | 106 EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()) |
105 .Times(1); | 107 .Times(1); |
106 EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)) | 108 EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)) |
107 .Times(1); | 109 .Times(1); |
108 EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()) | 110 EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()) |
109 .Times(1); | 111 .Times(1); |
110 EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory()) | 112 EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory()) |
111 .WillOnce(ReturnRef(decoder_factory_)); | 113 .WillOnce(ReturnRef(decoder_factory_)); |
| 114 testing::Expectation expect_set = |
| 115 EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_)) |
| 116 .Times(1); |
| 117 EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull())) |
| 118 .Times(1) |
| 119 .After(expect_set); |
112 return channel_proxy_; | 120 return channel_proxy_; |
113 })); | 121 })); |
114 stream_config_.voe_channel_id = kChannelId; | 122 stream_config_.voe_channel_id = kChannelId; |
115 stream_config_.rtp.local_ssrc = kLocalSsrc; | 123 stream_config_.rtp.local_ssrc = kLocalSsrc; |
116 stream_config_.rtp.remote_ssrc = kRemoteSsrc; | 124 stream_config_.rtp.remote_ssrc = kRemoteSsrc; |
117 stream_config_.rtp.nack.rtp_history_ms = 300; | 125 stream_config_.rtp.nack.rtp_history_ms = 300; |
118 stream_config_.rtp.extensions.push_back( | 126 stream_config_.rtp.extensions.push_back( |
119 RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); | 127 RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); |
120 stream_config_.rtp.extensions.push_back( | 128 stream_config_.rtp.extensions.push_back( |
121 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); | 129 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); |
122 stream_config_.rtp.extensions.push_back(RtpExtension( | 130 stream_config_.rtp.extensions.push_back(RtpExtension( |
123 RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); | 131 RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); |
124 stream_config_.decoder_factory = decoder_factory_; | 132 stream_config_.decoder_factory = decoder_factory_; |
125 } | 133 } |
126 | 134 |
127 MockCongestionController* congestion_controller() { | 135 MockCongestionController* congestion_controller() { |
128 return &congestion_controller_; | 136 return &congestion_controller_; |
129 } | 137 } |
130 MockRemoteBitrateEstimator* remote_bitrate_estimator() { | 138 MockRemoteBitrateEstimator* remote_bitrate_estimator() { |
131 return &remote_bitrate_estimator_; | 139 return &remote_bitrate_estimator_; |
132 } | 140 } |
| 141 MockRtcEventLog* event_log() { return &event_log_; } |
133 AudioReceiveStream::Config& config() { return stream_config_; } | 142 AudioReceiveStream::Config& config() { return stream_config_; } |
134 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } | 143 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } |
135 MockVoiceEngine& voice_engine() { return voice_engine_; } | 144 MockVoiceEngine& voice_engine() { return voice_engine_; } |
136 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } | 145 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } |
137 | 146 |
138 void SetupMockForBweFeedback(bool send_side_bwe) { | 147 void SetupMockForBweFeedback(bool send_side_bwe) { |
139 EXPECT_CALL(congestion_controller_, | 148 EXPECT_CALL(congestion_controller_, |
140 GetRemoteBitrateEstimator(send_side_bwe)) | 149 GetRemoteBitrateEstimator(send_side_bwe)) |
141 .WillOnce(Return(&remote_bitrate_estimator_)); | 150 .WillOnce(Return(&remote_bitrate_estimator_)); |
142 EXPECT_CALL(remote_bitrate_estimator_, | 151 EXPECT_CALL(remote_bitrate_estimator_, |
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164 } | 173 } |
165 | 174 |
166 private: | 175 private: |
167 SimulatedClock simulated_clock_; | 176 SimulatedClock simulated_clock_; |
168 PacketRouter packet_router_; | 177 PacketRouter packet_router_; |
169 testing::NiceMock<MockCongestionObserver> bitrate_observer_; | 178 testing::NiceMock<MockCongestionObserver> bitrate_observer_; |
170 testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_; | 179 testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_; |
171 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 180 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
172 MockCongestionController congestion_controller_; | 181 MockCongestionController congestion_controller_; |
173 MockRemoteBitrateEstimator remote_bitrate_estimator_; | 182 MockRemoteBitrateEstimator remote_bitrate_estimator_; |
| 183 MockRtcEventLog event_log_; |
174 testing::StrictMock<MockVoiceEngine> voice_engine_; | 184 testing::StrictMock<MockVoiceEngine> voice_engine_; |
175 rtc::scoped_refptr<AudioState> audio_state_; | 185 rtc::scoped_refptr<AudioState> audio_state_; |
176 AudioReceiveStream::Config stream_config_; | 186 AudioReceiveStream::Config stream_config_; |
177 testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; | 187 testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; |
178 }; | 188 }; |
179 | 189 |
180 void BuildOneByteExtension(std::vector<uint8_t>::iterator it, | 190 void BuildOneByteExtension(std::vector<uint8_t>::iterator it, |
181 int id, | 191 int id, |
182 uint32_t extension_value, | 192 uint32_t extension_value, |
183 size_t value_length) { | 193 size_t value_length) { |
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241 "nack: {rtp_history_ms: 0}, extensions: [{uri: " | 251 "nack: {rtp_history_ms: 0}, extensions: [{uri: " |
242 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, " | 252 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, " |
243 "rtcp_send_transport: nullptr, " | 253 "rtcp_send_transport: nullptr, " |
244 "voe_channel_id: 2}", | 254 "voe_channel_id: 2}", |
245 config.ToString()); | 255 config.ToString()); |
246 } | 256 } |
247 | 257 |
248 TEST(AudioReceiveStreamTest, ConstructDestruct) { | 258 TEST(AudioReceiveStreamTest, ConstructDestruct) { |
249 ConfigHelper helper; | 259 ConfigHelper helper; |
250 internal::AudioReceiveStream recv_stream( | 260 internal::AudioReceiveStream recv_stream( |
251 helper.congestion_controller(), helper.config(), helper.audio_state()); | 261 helper.congestion_controller(), helper.config(), helper.audio_state(), |
| 262 helper.event_log()); |
252 } | 263 } |
253 | 264 |
254 MATCHER_P(VerifyHeaderExtension, expected_extension, "") { | 265 MATCHER_P(VerifyHeaderExtension, expected_extension, "") { |
255 return arg.extension.hasAbsoluteSendTime == | 266 return arg.extension.hasAbsoluteSendTime == |
256 expected_extension.hasAbsoluteSendTime && | 267 expected_extension.hasAbsoluteSendTime && |
257 arg.extension.absoluteSendTime == | 268 arg.extension.absoluteSendTime == |
258 expected_extension.absoluteSendTime && | 269 expected_extension.absoluteSendTime && |
259 arg.extension.hasTransportSequenceNumber == | 270 arg.extension.hasTransportSequenceNumber == |
260 expected_extension.hasTransportSequenceNumber && | 271 expected_extension.hasTransportSequenceNumber && |
261 arg.extension.transportSequenceNumber == | 272 arg.extension.transportSequenceNumber == |
262 expected_extension.transportSequenceNumber; | 273 expected_extension.transportSequenceNumber; |
263 } | 274 } |
264 | 275 |
265 TEST(AudioReceiveStreamTest, ReceiveRtpPacket) { | 276 TEST(AudioReceiveStreamTest, ReceiveRtpPacket) { |
266 ConfigHelper helper; | 277 ConfigHelper helper; |
267 helper.config().rtp.transport_cc = true; | 278 helper.config().rtp.transport_cc = true; |
268 helper.SetupMockForBweFeedback(true); | 279 helper.SetupMockForBweFeedback(true); |
269 internal::AudioReceiveStream recv_stream( | 280 internal::AudioReceiveStream recv_stream( |
270 helper.congestion_controller(), helper.config(), helper.audio_state()); | 281 helper.congestion_controller(), helper.config(), helper.audio_state(), |
| 282 helper.event_log()); |
271 const int kTransportSequenceNumberValue = 1234; | 283 const int kTransportSequenceNumberValue = 1234; |
272 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( | 284 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( |
273 kTransportSequenceNumberId, kTransportSequenceNumberValue, 2); | 285 kTransportSequenceNumberId, kTransportSequenceNumberValue, 2); |
274 PacketTime packet_time(5678000, 0); | 286 PacketTime packet_time(5678000, 0); |
275 const size_t kExpectedHeaderLength = 20; | 287 const size_t kExpectedHeaderLength = 20; |
276 RTPHeaderExtension expected_extension; | 288 RTPHeaderExtension expected_extension; |
277 expected_extension.hasTransportSequenceNumber = true; | 289 expected_extension.hasTransportSequenceNumber = true; |
278 expected_extension.transportSequenceNumber = kTransportSequenceNumberValue; | 290 expected_extension.transportSequenceNumber = kTransportSequenceNumberValue; |
279 EXPECT_CALL(*helper.remote_bitrate_estimator(), | 291 EXPECT_CALL(*helper.remote_bitrate_estimator(), |
280 IncomingPacket(packet_time.timestamp / 1000, | 292 IncomingPacket(packet_time.timestamp / 1000, |
281 rtp_packet.size() - kExpectedHeaderLength, | 293 rtp_packet.size() - kExpectedHeaderLength, |
282 VerifyHeaderExtension(expected_extension))) | 294 VerifyHeaderExtension(expected_extension))) |
283 .Times(1); | 295 .Times(1); |
284 EXPECT_CALL(*helper.channel_proxy(), | 296 EXPECT_CALL(*helper.channel_proxy(), |
285 ReceivedRTPPacket(&rtp_packet[0], | 297 ReceivedRTPPacket(&rtp_packet[0], |
286 rtp_packet.size(), | 298 rtp_packet.size(), |
287 _)) | 299 _)) |
288 .WillOnce(Return(true)); | 300 .WillOnce(Return(true)); |
289 EXPECT_TRUE( | 301 EXPECT_TRUE( |
290 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); | 302 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); |
291 } | 303 } |
292 | 304 |
293 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) { | 305 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) { |
294 ConfigHelper helper; | 306 ConfigHelper helper; |
295 helper.config().rtp.transport_cc = true; | 307 helper.config().rtp.transport_cc = true; |
296 helper.SetupMockForBweFeedback(true); | 308 helper.SetupMockForBweFeedback(true); |
297 internal::AudioReceiveStream recv_stream( | 309 internal::AudioReceiveStream recv_stream( |
298 helper.congestion_controller(), helper.config(), helper.audio_state()); | 310 helper.congestion_controller(), helper.config(), helper.audio_state(), |
| 311 helper.event_log()); |
299 | 312 |
300 std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport(); | 313 std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport(); |
301 EXPECT_CALL(*helper.channel_proxy(), | 314 EXPECT_CALL(*helper.channel_proxy(), |
302 ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size())) | 315 ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size())) |
303 .WillOnce(Return(true)); | 316 .WillOnce(Return(true)); |
304 EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size())); | 317 EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size())); |
305 } | 318 } |
306 | 319 |
307 TEST(AudioReceiveStreamTest, GetStats) { | 320 TEST(AudioReceiveStreamTest, GetStats) { |
308 ConfigHelper helper; | 321 ConfigHelper helper; |
309 internal::AudioReceiveStream recv_stream( | 322 internal::AudioReceiveStream recv_stream( |
310 helper.congestion_controller(), helper.config(), helper.audio_state()); | 323 helper.congestion_controller(), helper.config(), helper.audio_state(), |
| 324 helper.event_log()); |
311 helper.SetupMockForGetStats(); | 325 helper.SetupMockForGetStats(); |
312 AudioReceiveStream::Stats stats = recv_stream.GetStats(); | 326 AudioReceiveStream::Stats stats = recv_stream.GetStats(); |
313 EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc); | 327 EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc); |
314 EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd); | 328 EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd); |
315 EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived), | 329 EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived), |
316 stats.packets_rcvd); | 330 stats.packets_rcvd); |
317 EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost); | 331 EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost); |
318 EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost); | 332 EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost); |
319 EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name); | 333 EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name); |
320 EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum); | 334 EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum); |
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342 EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc); | 356 EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc); |
343 EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng); | 357 EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng); |
344 EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng); | 358 EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng); |
345 EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_, | 359 EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_, |
346 stats.capture_start_ntp_time_ms); | 360 stats.capture_start_ntp_time_ms); |
347 } | 361 } |
348 | 362 |
349 TEST(AudioReceiveStreamTest, SetGain) { | 363 TEST(AudioReceiveStreamTest, SetGain) { |
350 ConfigHelper helper; | 364 ConfigHelper helper; |
351 internal::AudioReceiveStream recv_stream( | 365 internal::AudioReceiveStream recv_stream( |
352 helper.congestion_controller(), helper.config(), helper.audio_state()); | 366 helper.congestion_controller(), helper.config(), helper.audio_state(), |
| 367 helper.event_log()); |
353 EXPECT_CALL(*helper.channel_proxy(), | 368 EXPECT_CALL(*helper.channel_proxy(), |
354 SetChannelOutputVolumeScaling(FloatEq(0.765f))); | 369 SetChannelOutputVolumeScaling(FloatEq(0.765f))); |
355 recv_stream.SetGain(0.765f); | 370 recv_stream.SetGain(0.765f); |
356 } | 371 } |
357 } // namespace test | 372 } // namespace test |
358 } // namespace webrtc | 373 } // namespace webrtc |
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