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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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100 BitrateEstimatorTest() : receive_config_(nullptr) {} | 100 BitrateEstimatorTest() : receive_config_(nullptr) {} |
101 | 101 |
102 virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); } | 102 virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); } |
103 | 103 |
104 virtual void SetUp() { | 104 virtual void SetUp() { |
105 Call::Config config(event_log_.get()); | 105 Call::Config config(event_log_.get()); |
106 receiver_call_.reset(Call::Create(config)); | 106 receiver_call_.reset(Call::Create(config)); |
107 sender_call_.reset(Call::Create(config)); | 107 sender_call_.reset(Call::Create(config)); |
108 | 108 |
109 send_transport_.reset( | 109 send_transport_.reset( |
110 new test::DirectTransport(sender_call_.get(), MediaType::VIDEO)); | 110 new test::DirectTransport(sender_call_.get(), payload_type_map_)); |
111 send_transport_->SetReceiver(receiver_call_->Receiver()); | 111 send_transport_->SetReceiver(receiver_call_->Receiver()); |
112 receive_transport_.reset( | 112 receive_transport_.reset( |
113 new test::DirectTransport(receiver_call_.get(), MediaType::VIDEO)); | 113 new test::DirectTransport(receiver_call_.get(), payload_type_map_)); |
114 receive_transport_->SetReceiver(sender_call_->Receiver()); | 114 receive_transport_->SetReceiver(sender_call_->Receiver()); |
115 | 115 |
116 video_send_config_ = VideoSendStream::Config(send_transport_.get()); | 116 video_send_config_ = VideoSendStream::Config(send_transport_.get()); |
117 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]); | 117 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]); |
118 // Encoders will be set separately per stream. | 118 // Encoders will be set separately per stream. |
119 video_send_config_.encoder_settings.encoder = nullptr; | 119 video_send_config_.encoder_settings.encoder = nullptr; |
120 video_send_config_.encoder_settings.payload_name = "FAKE"; | 120 video_send_config_.encoder_settings.payload_name = "FAKE"; |
121 video_send_config_.encoder_settings.payload_type = | 121 video_send_config_.encoder_settings.payload_type = |
122 kFakeVideoSendPayloadType; | 122 kFakeVideoSendPayloadType; |
123 test::FillEncoderConfiguration(1, &video_encoder_config_); | 123 test::FillEncoderConfiguration(1, &video_encoder_config_); |
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297 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId); | 297 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId); |
298 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); | 298 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); |
299 receiver_log_.PushExpectedLogLine( | 299 receiver_log_.PushExpectedLogLine( |
300 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); | 300 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); |
301 streams_.push_back(new Stream(this)); | 301 streams_.push_back(new Stream(this)); |
302 streams_[0]->StopSending(); | 302 streams_[0]->StopSending(); |
303 streams_[1]->StopSending(); | 303 streams_[1]->StopSending(); |
304 EXPECT_TRUE(receiver_log_.Wait()); | 304 EXPECT_TRUE(receiver_log_.Wait()); |
305 } | 305 } |
306 } // namespace webrtc | 306 } // namespace webrtc |
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