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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 |
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30 | 30 |
31 namespace { | 31 namespace { |
32 | 32 |
33 struct CallHelper { | 33 struct CallHelper { |
34 explicit CallHelper( | 34 explicit CallHelper( |
35 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) | 35 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) |
36 : voice_engine_(decoder_factory) { | 36 : voice_engine_(decoder_factory) { |
37 webrtc::AudioState::Config audio_state_config; | 37 webrtc::AudioState::Config audio_state_config; |
38 audio_state_config.voice_engine = &voice_engine_; | 38 audio_state_config.voice_engine = &voice_engine_; |
39 audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create(); | 39 audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create(); |
| 40 audio_state_config.audio_processing = webrtc::AudioProcessing::Create(); |
40 EXPECT_CALL(voice_engine_, audio_device_module()); | 41 EXPECT_CALL(voice_engine_, audio_device_module()); |
41 EXPECT_CALL(voice_engine_, audio_processing()); | |
42 EXPECT_CALL(voice_engine_, audio_transport()); | 42 EXPECT_CALL(voice_engine_, audio_transport()); |
43 webrtc::Call::Config config(&event_log_); | 43 webrtc::Call::Config config(&event_log_); |
44 config.audio_state = webrtc::AudioState::Create(audio_state_config); | 44 config.audio_state = webrtc::AudioState::Create(audio_state_config); |
45 call_.reset(webrtc::Call::Create(config)); | 45 call_.reset(webrtc::Call::Create(config)); |
46 } | 46 } |
47 | 47 |
48 webrtc::Call* operator->() { return call_.get(); } | 48 webrtc::Call* operator->() { return call_.get(); } |
49 webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; } | 49 webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; } |
50 | 50 |
51 private: | 51 private: |
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446 ~ScopedVoiceEngine() { | 446 ~ScopedVoiceEngine() { |
447 base->Release(); | 447 base->Release(); |
448 EXPECT_TRUE(VoiceEngine::Delete(voe)); | 448 EXPECT_TRUE(VoiceEngine::Delete(voe)); |
449 } | 449 } |
450 | 450 |
451 VoiceEngine* voe; | 451 VoiceEngine* voe; |
452 VoEBase* base; | 452 VoEBase* base; |
453 }; | 453 }; |
454 ScopedVoiceEngine voice_engine; | 454 ScopedVoiceEngine voice_engine; |
455 | 455 |
456 voice_engine.base->Init(&mock_adm); | |
457 AudioState::Config audio_state_config; | 456 AudioState::Config audio_state_config; |
458 audio_state_config.voice_engine = voice_engine.voe; | 457 audio_state_config.voice_engine = voice_engine.voe; |
459 audio_state_config.audio_mixer = mock_mixer; | 458 audio_state_config.audio_mixer = mock_mixer; |
| 459 audio_state_config.audio_processing = AudioProcessing::Create(); |
| 460 voice_engine.base->Init(&mock_adm, audio_state_config.audio_processing.get()); |
460 auto audio_state = AudioState::Create(audio_state_config); | 461 auto audio_state = AudioState::Create(audio_state_config); |
| 462 |
461 RtcEventLogNullImpl event_log; | 463 RtcEventLogNullImpl event_log; |
462 Call::Config call_config(&event_log); | 464 Call::Config call_config(&event_log); |
463 call_config.audio_state = audio_state; | 465 call_config.audio_state = audio_state; |
464 std::unique_ptr<Call> call(Call::Create(call_config)); | 466 std::unique_ptr<Call> call(Call::Create(call_config)); |
465 | 467 |
466 auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) { | 468 auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) { |
467 AudioSendStream::Config config(nullptr); | 469 AudioSendStream::Config config(nullptr); |
468 config.rtp.ssrc = ssrc; | 470 config.rtp.ssrc = ssrc; |
469 config.voe_channel_id = voice_engine.base->CreateChannel(); | 471 config.voe_channel_id = voice_engine.base->CreateChannel(); |
470 AudioSendStream* stream = call->CreateAudioSendStream(config); | 472 AudioSendStream* stream = call->CreateAudioSendStream(config); |
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704 mask.min_bitrate_bps = rtc::Optional<int>(2000); | 706 mask.min_bitrate_bps = rtc::Optional<int>(2000); |
705 EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, -1, 1000)); | 707 EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, -1, 1000)); |
706 call->SetBitrateConfigMask(mask); | 708 call->SetBitrateConfigMask(mask); |
707 | 709 |
708 // Set min to 3000; the clamped value stays the same so nothing happens. | 710 // Set min to 3000; the clamped value stays the same so nothing happens. |
709 mask.min_bitrate_bps = rtc::Optional<int>(3000); | 711 mask.min_bitrate_bps = rtc::Optional<int>(3000); |
710 call->SetBitrateConfigMask(mask); | 712 call->SetBitrateConfigMask(mask); |
711 } | 713 } |
712 | 714 |
713 } // namespace webrtc | 715 } // namespace webrtc |
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