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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_receive_stream.h" | 14 #include "webrtc/audio/audio_receive_stream.h" |
15 #include "webrtc/audio/conversion.h" | 15 #include "webrtc/audio/conversion.h" |
16 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" | 16 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
17 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" | 17 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" |
| 18 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
18 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller
.h" | 19 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller
.h" |
19 #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" |
20 #include "webrtc/modules/pacing/packet_router.h" | 21 #include "webrtc/modules/pacing/packet_router.h" |
21 #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" |
22 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 23 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
23 #include "webrtc/system_wrappers/include/clock.h" | 24 #include "webrtc/system_wrappers/include/clock.h" |
24 #include "webrtc/test/gtest.h" | 25 #include "webrtc/test/gtest.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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77 EXPECT_CALL(voice_engine_, | 78 EXPECT_CALL(voice_engine_, |
78 RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); | 79 RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); |
79 EXPECT_CALL(voice_engine_, | 80 EXPECT_CALL(voice_engine_, |
80 DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); | 81 DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); |
81 EXPECT_CALL(voice_engine_, audio_processing()); | 82 EXPECT_CALL(voice_engine_, audio_processing()); |
82 EXPECT_CALL(voice_engine_, audio_device_module()); | 83 EXPECT_CALL(voice_engine_, audio_device_module()); |
83 EXPECT_CALL(voice_engine_, audio_transport()); | 84 EXPECT_CALL(voice_engine_, audio_transport()); |
84 | 85 |
85 AudioState::Config config; | 86 AudioState::Config config; |
86 config.voice_engine = &voice_engine_; | 87 config.voice_engine = &voice_engine_; |
| 88 config.audio_mixer = AudioMixerImpl::Create(); |
87 audio_state_ = AudioState::Create(config); | 89 audio_state_ = AudioState::Create(config); |
88 | 90 |
89 EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) | 91 EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) |
90 .WillOnce(Invoke([this](int channel_id) { | 92 .WillOnce(Invoke([this](int channel_id) { |
91 EXPECT_FALSE(channel_proxy_); | 93 EXPECT_FALSE(channel_proxy_); |
92 channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); | 94 channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); |
93 EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1); | 95 EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1); |
94 EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1); | 96 EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1); |
95 EXPECT_CALL(*channel_proxy_, | 97 EXPECT_CALL(*channel_proxy_, |
96 SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId)) | 98 SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId)) |
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360 ConfigHelper helper; | 362 ConfigHelper helper; |
361 internal::AudioReceiveStream recv_stream( | 363 internal::AudioReceiveStream recv_stream( |
362 helper.congestion_controller(), helper.config(), helper.audio_state(), | 364 helper.congestion_controller(), helper.config(), helper.audio_state(), |
363 helper.event_log()); | 365 helper.event_log()); |
364 EXPECT_CALL(*helper.channel_proxy(), | 366 EXPECT_CALL(*helper.channel_proxy(), |
365 SetChannelOutputVolumeScaling(FloatEq(0.765f))); | 367 SetChannelOutputVolumeScaling(FloatEq(0.765f))); |
366 recv_stream.SetGain(0.765f); | 368 recv_stream.SetGain(0.765f); |
367 } | 369 } |
368 } // namespace test | 370 } // namespace test |
369 } // namespace webrtc | 371 } // namespace webrtc |
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