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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 <functional> | 10 #include <functional> |
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104 public: | 104 public: |
105 BitrateEstimatorTest() : mock_voice_engine_(decoder_factory_), | 105 BitrateEstimatorTest() : mock_voice_engine_(decoder_factory_), |
106 receive_config_(nullptr) {} | 106 receive_config_(nullptr) {} |
107 | 107 |
108 virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); } | 108 virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); } |
109 | 109 |
110 virtual void SetUp() { | 110 virtual void SetUp() { |
111 AudioState::Config audio_state_config; | 111 AudioState::Config audio_state_config; |
112 audio_state_config.voice_engine = &mock_voice_engine_; | 112 audio_state_config.voice_engine = &mock_voice_engine_; |
113 Call::Config config; | 113 Call::Config config; |
114 config.audio_state = AudioState::Create(audio_state_config); | 114 config.audio_state = AudioState::Create(audio_state_config, nullptr); |
115 receiver_call_.reset(Call::Create(config)); | 115 receiver_call_.reset(Call::Create(config)); |
116 sender_call_.reset(Call::Create(config)); | 116 sender_call_.reset(Call::Create(config)); |
117 | 117 |
118 send_transport_.reset(new test::DirectTransport(sender_call_.get())); | 118 send_transport_.reset(new test::DirectTransport(sender_call_.get())); |
119 send_transport_->SetReceiver(receiver_call_->Receiver()); | 119 send_transport_->SetReceiver(receiver_call_->Receiver()); |
120 receive_transport_.reset(new test::DirectTransport(receiver_call_.get())); | 120 receive_transport_.reset(new test::DirectTransport(receiver_call_.get())); |
121 receive_transport_->SetReceiver(sender_call_->Receiver()); | 121 receive_transport_->SetReceiver(sender_call_->Receiver()); |
122 | 122 |
123 video_send_config_ = VideoSendStream::Config(send_transport_.get()); | 123 video_send_config_ = VideoSendStream::Config(send_transport_.get()); |
124 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]); | 124 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]); |
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324 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId); | 324 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId); |
325 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); | 325 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); |
326 receiver_log_.PushExpectedLogLine( | 326 receiver_log_.PushExpectedLogLine( |
327 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); | 327 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); |
328 streams_.push_back(new Stream(this, false)); | 328 streams_.push_back(new Stream(this, false)); |
329 streams_[0]->StopSending(); | 329 streams_[0]->StopSending(); |
330 streams_[1]->StopSending(); | 330 streams_[1]->StopSending(); |
331 EXPECT_TRUE(receiver_log_.Wait()); | 331 EXPECT_TRUE(receiver_log_.Wait()); |
332 } | 332 } |
333 } // namespace webrtc | 333 } // namespace webrtc |
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