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