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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 "webrtc/audio/audio_send_stream.h" | 14 #include "webrtc/audio/audio_send_stream.h" |
15 #include "webrtc/audio/audio_state.h" | 15 #include "webrtc/audio/audio_state.h" |
16 #include "webrtc/audio/conversion.h" | 16 #include "webrtc/audio/conversion.h" |
| 17 #include "webrtc/base/ptr_util.h" |
17 #include "webrtc/base/task_queue.h" | 18 #include "webrtc/base/task_queue.h" |
| 19 #include "webrtc/call/fake_rtp_transport_controller_send.h" |
18 #include "webrtc/call/rtp_transport_controller_send_interface.h" | 20 #include "webrtc/call/rtp_transport_controller_send_interface.h" |
19 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" | 21 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
20 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" | 22 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
21 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" | 23 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" |
22 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_obse
rver.h" | 24 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_obse
rver.h" |
23 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" | 25 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" |
24 #include "webrtc/modules/pacing/paced_sender.h" | 26 #include "webrtc/modules/pacing/paced_sender.h" |
25 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" | 27 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" |
26 #include "webrtc/test/gtest.h" | 28 #include "webrtc/test/gtest.h" |
27 #include "webrtc/test/mock_voe_channel_proxy.h" | 29 #include "webrtc/test/mock_voe_channel_proxy.h" |
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63 void(uint32_t min_send_bitrate_bps, | 65 void(uint32_t min_send_bitrate_bps, |
64 uint32_t max_padding_bitrate_bps)); | 66 uint32_t max_padding_bitrate_bps)); |
65 }; | 67 }; |
66 | 68 |
67 class MockTransmitMixer : public voe::TransmitMixer { | 69 class MockTransmitMixer : public voe::TransmitMixer { |
68 public: | 70 public: |
69 MOCK_CONST_METHOD0(AudioLevelFullRange, int16_t()); | 71 MOCK_CONST_METHOD0(AudioLevelFullRange, int16_t()); |
70 }; | 72 }; |
71 | 73 |
72 struct ConfigHelper { | 74 struct ConfigHelper { |
73 class FakeRtpTransportController | |
74 : public RtpTransportControllerSendInterface { | |
75 public: | |
76 explicit FakeRtpTransportController(RtcEventLog* event_log) | |
77 : simulated_clock_(123456), | |
78 send_side_cc_(&simulated_clock_, | |
79 &bitrate_observer_, | |
80 event_log, | |
81 &packet_router_) {} | |
82 PacketRouter* packet_router() override { return &packet_router_; } | |
83 | |
84 SendSideCongestionController* send_side_cc() override { | |
85 return &send_side_cc_; | |
86 } | |
87 TransportFeedbackObserver* transport_feedback_observer() override { | |
88 return &send_side_cc_; | |
89 } | |
90 | |
91 RtpPacketSender* packet_sender() override { return send_side_cc_.pacer(); } | |
92 | |
93 private: | |
94 SimulatedClock simulated_clock_; | |
95 testing::NiceMock<MockCongestionObserver> bitrate_observer_; | |
96 PacketRouter packet_router_; | |
97 SendSideCongestionController send_side_cc_; | |
98 }; | |
99 | |
100 explicit ConfigHelper(bool audio_bwe_enabled) | 75 explicit ConfigHelper(bool audio_bwe_enabled) |
101 : stream_config_(nullptr), | 76 : stream_config_(nullptr), |
102 fake_transport_(&event_log_), | 77 simulated_clock_(123456), |
| 78 send_side_cc_(rtc::MakeUnique<SendSideCongestionController>( |
| 79 &simulated_clock_, |
| 80 nullptr /* observer */, |
| 81 &event_log_, |
| 82 &packet_router_)), |
| 83 fake_transport_(send_side_cc_.get()), |
103 bitrate_allocator_(&limit_observer_), | 84 bitrate_allocator_(&limit_observer_), |
104 worker_queue_("ConfigHelper_worker_queue") { | 85 worker_queue_("ConfigHelper_worker_queue") { |
105 using testing::Invoke; | 86 using testing::Invoke; |
106 | 87 |
107 EXPECT_CALL(voice_engine_, | 88 EXPECT_CALL(voice_engine_, |
108 RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); | 89 RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); |
109 EXPECT_CALL(voice_engine_, | 90 EXPECT_CALL(voice_engine_, |
110 DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); | 91 DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); |
111 EXPECT_CALL(voice_engine_, audio_device_module()); | 92 EXPECT_CALL(voice_engine_, audio_device_module()); |
112 EXPECT_CALL(voice_engine_, audio_processing()); | 93 EXPECT_CALL(voice_engine_, audio_processing()); |
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252 | 233 |
253 EXPECT_CALL(audio_processing_, GetStatistics()) | 234 EXPECT_CALL(audio_processing_, GetStatistics()) |
254 .WillRepeatedly(Return(audio_processing_stats_)); | 235 .WillRepeatedly(Return(audio_processing_stats_)); |
255 } | 236 } |
256 | 237 |
257 private: | 238 private: |
258 testing::StrictMock<MockVoiceEngine> voice_engine_; | 239 testing::StrictMock<MockVoiceEngine> voice_engine_; |
259 rtc::scoped_refptr<AudioState> audio_state_; | 240 rtc::scoped_refptr<AudioState> audio_state_; |
260 AudioSendStream::Config stream_config_; | 241 AudioSendStream::Config stream_config_; |
261 testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; | 242 testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; |
262 testing::NiceMock<MockCongestionObserver> bitrate_observer_; | |
263 MockAudioProcessing audio_processing_; | 243 MockAudioProcessing audio_processing_; |
264 MockTransmitMixer transmit_mixer_; | 244 MockTransmitMixer transmit_mixer_; |
265 AudioProcessing::AudioProcessingStatistics audio_processing_stats_; | 245 AudioProcessing::AudioProcessingStatistics audio_processing_stats_; |
| 246 SimulatedClock simulated_clock_; |
| 247 PacketRouter packet_router_; |
| 248 std::unique_ptr<SendSideCongestionController> send_side_cc_; |
266 FakeRtpTransportController fake_transport_; | 249 FakeRtpTransportController fake_transport_; |
267 MockRtcEventLog event_log_; | 250 MockRtcEventLog event_log_; |
268 MockRtcpRttStats rtcp_rtt_stats_; | 251 MockRtcpRttStats rtcp_rtt_stats_; |
269 testing::NiceMock<MockLimitObserver> limit_observer_; | 252 testing::NiceMock<MockLimitObserver> limit_observer_; |
270 BitrateAllocator bitrate_allocator_; | 253 BitrateAllocator bitrate_allocator_; |
271 // |worker_queue| is defined last to ensure all pending tasks are cancelled | 254 // |worker_queue| is defined last to ensure all pending tasks are cancelled |
272 // and deleted before any other members. | 255 // and deleted before any other members. |
273 rtc::TaskQueue worker_queue_; | 256 rtc::TaskQueue worker_queue_; |
274 }; | 257 }; |
275 } // namespace | 258 } // namespace |
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479 internal::AudioSendStream send_stream( | 462 internal::AudioSendStream send_stream( |
480 helper.config(), helper.audio_state(), helper.worker_queue(), | 463 helper.config(), helper.audio_state(), helper.worker_queue(), |
481 helper.transport(), helper.bitrate_allocator(), helper.event_log(), | 464 helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
482 helper.rtcp_rtt_stats()); | 465 helper.rtcp_rtt_stats()); |
483 EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000)); | 466 EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000)); |
484 send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000); | 467 send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000); |
485 } | 468 } |
486 | 469 |
487 } // namespace test | 470 } // namespace test |
488 } // namespace webrtc | 471 } // namespace webrtc |
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