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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 <utility> | 12 #include <utility> |
13 #include <vector> | 13 #include <vector> |
14 | 14 |
15 #include "webrtc/audio/audio_send_stream.h" | 15 #include "webrtc/audio/audio_send_stream.h" |
16 #include "webrtc/audio/audio_state.h" | 16 #include "webrtc/audio/audio_state.h" |
17 #include "webrtc/audio/conversion.h" | 17 #include "webrtc/audio/conversion.h" |
18 #include "webrtc/base/task_queue.h" | 18 #include "webrtc/base/task_queue.h" |
19 #include "webrtc/call/rtp_transport_controller_send_interface.h" | 19 #include "webrtc/call/rtp_transport_controller_send_interface.h" |
20 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" | 20 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
21 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h" | 21 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h" |
22 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder_factory.h" | 22 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder_factory.h" |
23 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" | 23 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
24 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" | 24 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" |
25 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_obse
rver.h" | 25 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_obse
rver.h" |
26 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" | 26 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" |
27 #include "webrtc/modules/pacing/paced_sender.h" | 27 #include "webrtc/modules/pacing/paced_sender.h" |
| 28 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
28 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" | 29 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" |
29 #include "webrtc/test/gtest.h" | 30 #include "webrtc/test/gtest.h" |
30 #include "webrtc/test/mock_voe_channel_proxy.h" | 31 #include "webrtc/test/mock_voe_channel_proxy.h" |
31 #include "webrtc/test/mock_voice_engine.h" | 32 #include "webrtc/test/mock_voice_engine.h" |
32 #include "webrtc/voice_engine/transmit_mixer.h" | 33 #include "webrtc/voice_engine/transmit_mixer.h" |
33 | 34 |
34 namespace webrtc { | 35 namespace webrtc { |
35 namespace test { | 36 namespace test { |
36 namespace { | 37 namespace { |
37 | 38 |
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210 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } | 211 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } |
211 RtpTransportControllerSendInterface* transport() { return &fake_transport_; } | 212 RtpTransportControllerSendInterface* transport() { return &fake_transport_; } |
212 BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; } | 213 BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; } |
213 rtc::TaskQueue* worker_queue() { return &worker_queue_; } | 214 rtc::TaskQueue* worker_queue() { return &worker_queue_; } |
214 RtcEventLog* event_log() { return &event_log_; } | 215 RtcEventLog* event_log() { return &event_log_; } |
215 MockVoiceEngine* voice_engine() { return &voice_engine_; } | 216 MockVoiceEngine* voice_engine() { return &voice_engine_; } |
216 | 217 |
217 void SetupDefaultChannelProxy(bool audio_bwe_enabled) { | 218 void SetupDefaultChannelProxy(bool audio_bwe_enabled) { |
218 using testing::StrEq; | 219 using testing::StrEq; |
219 channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); | 220 channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); |
| 221 EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _)) |
| 222 .WillRepeatedly(Invoke( |
| 223 [this](RtpRtcp** rtp_rtcp_module, RtpReceiver** rtp_receiver) { |
| 224 *rtp_rtcp_module = &this->rtp_rtcp_; |
| 225 *rtp_receiver = nullptr; // Not deemed necessary for tests yet. |
| 226 })); |
220 EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1); | 227 EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1); |
221 EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1); | 228 EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1); |
222 EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1); | 229 EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1); |
223 EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1); | 230 EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1); |
224 EXPECT_CALL(*channel_proxy_, | 231 EXPECT_CALL(*channel_proxy_, |
225 SetSendAudioLevelIndicationStatus(true, kAudioLevelId)) | 232 SetSendAudioLevelIndicationStatus(true, kAudioLevelId)) |
226 .Times(1); | 233 .Times(1); |
227 if (audio_bwe_enabled) { | 234 if (audio_bwe_enabled) { |
228 EXPECT_CALL(*channel_proxy_, | 235 EXPECT_CALL(*channel_proxy_, |
229 EnableSendTransportSequenceNumber(kTransportSequenceNumberId)) | 236 EnableSendTransportSequenceNumber(kTransportSequenceNumberId)) |
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314 testing::StrictMock<MockVoiceEngine> voice_engine_; | 321 testing::StrictMock<MockVoiceEngine> voice_engine_; |
315 rtc::scoped_refptr<AudioState> audio_state_; | 322 rtc::scoped_refptr<AudioState> audio_state_; |
316 AudioSendStream::Config stream_config_; | 323 AudioSendStream::Config stream_config_; |
317 testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; | 324 testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; |
318 testing::NiceMock<MockCongestionObserver> bitrate_observer_; | 325 testing::NiceMock<MockCongestionObserver> bitrate_observer_; |
319 MockAudioProcessing audio_processing_; | 326 MockAudioProcessing audio_processing_; |
320 MockTransmitMixer transmit_mixer_; | 327 MockTransmitMixer transmit_mixer_; |
321 AudioProcessing::AudioProcessingStatistics audio_processing_stats_; | 328 AudioProcessing::AudioProcessingStatistics audio_processing_stats_; |
322 FakeRtpTransportController fake_transport_; | 329 FakeRtpTransportController fake_transport_; |
323 MockRtcEventLog event_log_; | 330 MockRtcEventLog event_log_; |
| 331 MockRtpRtcp rtp_rtcp_; |
324 MockRtcpRttStats rtcp_rtt_stats_; | 332 MockRtcpRttStats rtcp_rtt_stats_; |
325 testing::NiceMock<MockLimitObserver> limit_observer_; | 333 testing::NiceMock<MockLimitObserver> limit_observer_; |
326 BitrateAllocator bitrate_allocator_; | 334 BitrateAllocator bitrate_allocator_; |
327 // |worker_queue| is defined last to ensure all pending tasks are cancelled | 335 // |worker_queue| is defined last to ensure all pending tasks are cancelled |
328 // and deleted before any other members. | 336 // and deleted before any other members. |
329 rtc::TaskQueue worker_queue_; | 337 rtc::TaskQueue worker_queue_; |
330 }; | 338 }; |
331 } // namespace | 339 } // namespace |
332 | 340 |
333 TEST(AudioSendStreamTest, ConfigToString) { | 341 TEST(AudioSendStreamTest, ConfigToString) { |
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546 stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105); | 554 stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105); |
547 internal::AudioSendStream send_stream( | 555 internal::AudioSendStream send_stream( |
548 stream_config, helper.audio_state(), helper.worker_queue(), | 556 stream_config, helper.audio_state(), helper.worker_queue(), |
549 helper.transport(), helper.bitrate_allocator(), helper.event_log(), | 557 helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
550 helper.rtcp_rtt_stats()); | 558 helper.rtcp_rtt_stats()); |
551 send_stream.Reconfigure(stream_config); | 559 send_stream.Reconfigure(stream_config); |
552 } | 560 } |
553 | 561 |
554 } // namespace test | 562 } // namespace test |
555 } // namespace webrtc | 563 } // namespace webrtc |
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