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Side by Side Diff: webrtc/call/bitrate_estimator_tests.cc

Issue 2794243002: Making FakeNetworkPipe demux audio and video packets. (Closed)
Patch Set: fixing android Created 3 years, 8 months ago
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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>
(...skipping 89 matching lines...) Expand 10 before | Expand all | Expand 10 after
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_);
(...skipping 173 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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