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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 #include "webrtc/base/checks.h" | 10 #include "webrtc/base/checks.h" |
11 #include "webrtc/config.h" | 11 #include "webrtc/config.h" |
12 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" | 12 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" |
| 13 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
13 #include "webrtc/test/call_test.h" | 14 #include "webrtc/test/call_test.h" |
14 #include "webrtc/test/testsupport/fileutils.h" | 15 #include "webrtc/test/testsupport/fileutils.h" |
15 #include "webrtc/voice_engine/include/voe_base.h" | 16 #include "webrtc/voice_engine/include/voe_base.h" |
16 | 17 |
17 namespace webrtc { | 18 namespace webrtc { |
18 namespace test { | 19 namespace test { |
19 | 20 |
20 namespace { | 21 namespace { |
21 const int kVideoRotationRtpExtensionId = 4; | 22 const int kVideoRotationRtpExtensionId = 4; |
22 } | 23 } |
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39 | 40 |
40 void CallTest::RunBaseTest(BaseTest* test) { | 41 void CallTest::RunBaseTest(BaseTest* test) { |
41 num_video_streams_ = test->GetNumVideoStreams(); | 42 num_video_streams_ = test->GetNumVideoStreams(); |
42 num_audio_streams_ = test->GetNumAudioStreams(); | 43 num_audio_streams_ = test->GetNumAudioStreams(); |
43 RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0); | 44 RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0); |
44 Call::Config send_config(test->GetSenderCallConfig()); | 45 Call::Config send_config(test->GetSenderCallConfig()); |
45 if (num_audio_streams_ > 0) { | 46 if (num_audio_streams_ > 0) { |
46 CreateVoiceEngines(); | 47 CreateVoiceEngines(); |
47 AudioState::Config audio_state_config; | 48 AudioState::Config audio_state_config; |
48 audio_state_config.voice_engine = voe_send_.voice_engine; | 49 audio_state_config.voice_engine = voe_send_.voice_engine; |
| 50 audio_state_config.audio_mixer = AudioMixerImpl::Create(); |
49 send_config.audio_state = AudioState::Create(audio_state_config); | 51 send_config.audio_state = AudioState::Create(audio_state_config); |
50 } | 52 } |
51 CreateSenderCall(send_config); | 53 CreateSenderCall(send_config); |
52 if (test->ShouldCreateReceivers()) { | 54 if (test->ShouldCreateReceivers()) { |
53 Call::Config recv_config(test->GetReceiverCallConfig()); | 55 Call::Config recv_config(test->GetReceiverCallConfig()); |
54 if (num_audio_streams_ > 0) { | 56 if (num_audio_streams_ > 0) { |
55 AudioState::Config audio_state_config; | 57 AudioState::Config audio_state_config; |
56 audio_state_config.voice_engine = voe_recv_.voice_engine; | 58 audio_state_config.voice_engine = voe_recv_.voice_engine; |
| 59 audio_state_config.audio_mixer = AudioMixerImpl::Create(); |
57 recv_config.audio_state = AudioState::Create(audio_state_config); | 60 recv_config.audio_state = AudioState::Create(audio_state_config); |
58 } | 61 } |
59 CreateReceiverCall(recv_config); | 62 CreateReceiverCall(recv_config); |
60 } | 63 } |
61 test->OnCallsCreated(sender_call_.get(), receiver_call_.get()); | 64 test->OnCallsCreated(sender_call_.get(), receiver_call_.get()); |
62 receive_transport_.reset(test->CreateReceiveTransport()); | 65 receive_transport_.reset(test->CreateReceiveTransport()); |
63 send_transport_.reset(test->CreateSendTransport(sender_call_.get())); | 66 send_transport_.reset(test->CreateSendTransport(sender_call_.get())); |
64 | 67 |
65 if (test->ShouldCreateReceivers()) { | 68 if (test->ShouldCreateReceivers()) { |
66 send_transport_->SetReceiver(receiver_call_->Receiver()); | 69 send_transport_->SetReceiver(receiver_call_->Receiver()); |
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439 | 442 |
440 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 443 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
441 } | 444 } |
442 | 445 |
443 bool EndToEndTest::ShouldCreateReceivers() const { | 446 bool EndToEndTest::ShouldCreateReceivers() const { |
444 return true; | 447 return true; |
445 } | 448 } |
446 | 449 |
447 } // namespace test | 450 } // namespace test |
448 } // namespace webrtc | 451 } // namespace webrtc |
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