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Issue 2469743002: Passed AudioMixer to AudioState::Config. (Closed)
Patch Set: Rebase. GYP removed! Created 4 years, 1 month ago
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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 <list> 11 #include <list>
12 #include <memory> 12 #include <memory>
13 13
14 #include "webrtc/api/call/audio_state.h" 14 #include "webrtc/api/call/audio_state.h"
15 #include "webrtc/call.h" 15 #include "webrtc/call.h"
16 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" 16 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
17 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
17 #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"
18 #include "webrtc/test/gtest.h" 19 #include "webrtc/test/gtest.h"
19 #include "webrtc/test/mock_voice_engine.h" 20 #include "webrtc/test/mock_voice_engine.h"
20 21
21 namespace { 22 namespace {
22 23
23 struct CallHelper { 24 struct CallHelper {
24 explicit CallHelper( 25 explicit CallHelper(
25 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) 26 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
26 : voice_engine_(decoder_factory) { 27 : voice_engine_(decoder_factory) {
27 webrtc::AudioState::Config audio_state_config; 28 webrtc::AudioState::Config audio_state_config;
28 audio_state_config.voice_engine = &voice_engine_; 29 audio_state_config.voice_engine = &voice_engine_;
30 audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
29 EXPECT_CALL(voice_engine_, audio_device_module()); 31 EXPECT_CALL(voice_engine_, audio_device_module());
30 EXPECT_CALL(voice_engine_, audio_processing()); 32 EXPECT_CALL(voice_engine_, audio_processing());
31 EXPECT_CALL(voice_engine_, audio_transport()); 33 EXPECT_CALL(voice_engine_, audio_transport());
32 webrtc::Call::Config config(&event_log_); 34 webrtc::Call::Config config(&event_log_);
33 config.audio_state = webrtc::AudioState::Create(audio_state_config); 35 config.audio_state = webrtc::AudioState::Create(audio_state_config);
34 call_.reset(webrtc::Call::Create(config)); 36 call_.reset(webrtc::Call::Create(config));
35 } 37 }
36 38
37 webrtc::Call* operator->() { return call_.get(); } 39 webrtc::Call* operator->() { return call_.get(); }
38 webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; } 40 webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; }
(...skipping 243 matching lines...) Expand 10 before | Expand all | Expand 10 after
282 stream = call->CreateFlexfecReceiveStream(config); 284 stream = call->CreateFlexfecReceiveStream(config);
283 EXPECT_NE(stream, nullptr); 285 EXPECT_NE(stream, nullptr);
284 streams.push_back(stream); 286 streams.push_back(stream);
285 287
286 for (auto s : streams) { 288 for (auto s : streams) {
287 call->DestroyFlexfecReceiveStream(s); 289 call->DestroyFlexfecReceiveStream(s);
288 } 290 }
289 } 291 }
290 292
291 } // namespace webrtc 293 } // namespace webrtc
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