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Side by Side Diff: media/cast/net/cast_transport_impl_unittest.cc

Issue 2113783002: Refactoring: Merge VideoSenderConfig and AudioSenderConfig. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 5 months ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "media/cast/net/cast_transport_impl.h" 5 #include "media/cast/net/cast_transport_impl.h"
6 6
7 #include <gtest/gtest.h> 7 #include <gtest/gtest.h>
8 #include <stdint.h> 8 #include <stdint.h>
9 9
10 #include <memory> 10 #include <memory>
(...skipping 88 matching lines...) Expand 10 before | Expand all | Expand 10 after
99 ~CastTransportImplTest() override {} 99 ~CastTransportImplTest() override {}
100 100
101 void InitWithoutLogging(); 101 void InitWithoutLogging();
102 void InitWithOptions(); 102 void InitWithOptions();
103 void InitWithLogging(); 103 void InitWithLogging();
104 104
105 void InitializeVideo() { 105 void InitializeVideo() {
106 CastTransportRtpConfig rtp_config; 106 CastTransportRtpConfig rtp_config;
107 rtp_config.ssrc = kVideoSsrc; 107 rtp_config.ssrc = kVideoSsrc;
108 rtp_config.feedback_ssrc = 2; 108 rtp_config.feedback_ssrc = 2;
109 rtp_config.rtp_payload_type = 3; 109 rtp_config.rtp_payload_type = RtpPayloadType::VIDEO_VP8;
110 transport_sender_->InitializeVideo( 110 transport_sender_->InitializeVideo(
111 rtp_config, base::WrapUnique(new StubRtcpObserver())); 111 rtp_config, base::WrapUnique(new StubRtcpObserver()));
112 } 112 }
113 113
114 void InitializeAudio() { 114 void InitializeAudio() {
115 CastTransportRtpConfig rtp_config; 115 CastTransportRtpConfig rtp_config;
116 rtp_config.ssrc = kAudioSsrc; 116 rtp_config.ssrc = kAudioSsrc;
117 rtp_config.feedback_ssrc = 3; 117 rtp_config.feedback_ssrc = 3;
118 rtp_config.rtp_payload_type = 4; 118 rtp_config.rtp_payload_type = RtpPayloadType::AUDIO_OPUS;
119 transport_sender_->InitializeAudio( 119 transport_sender_->InitializeAudio(
120 rtp_config, base::WrapUnique(new StubRtcpObserver())); 120 rtp_config, base::WrapUnique(new StubRtcpObserver()));
121 } 121 }
122 122
123 base::SimpleTestTickClock testing_clock_; 123 base::SimpleTestTickClock testing_clock_;
124 scoped_refptr<FakeSingleThreadTaskRunner> task_runner_; 124 scoped_refptr<FakeSingleThreadTaskRunner> task_runner_;
125 std::unique_ptr<CastTransportImpl> transport_sender_; 125 std::unique_ptr<CastTransportImpl> transport_sender_;
126 FakePacketSender* transport_; // Owned by CastTransport. 126 FakePacketSender* transport_; // Owned by CastTransport.
127 int num_times_logging_callback_called_; 127 int num_times_logging_callback_called_;
128 }; 128 };
(...skipping 272 matching lines...) Expand 10 before | Expand all | Expand 10 after
401 task_runner_->RunTasks(); 401 task_runner_->RunTasks();
402 EXPECT_EQ(7, transport_->packets_sent()); 402 EXPECT_EQ(7, transport_->packets_sent());
403 EXPECT_EQ(1, num_times_logging_callback_called_); // Only 8 ms since last. 403 EXPECT_EQ(1, num_times_logging_callback_called_); // Only 8 ms since last.
404 404
405 task_runner_->Sleep(base::TimeDelta::FromMilliseconds(2)); 405 task_runner_->Sleep(base::TimeDelta::FromMilliseconds(2));
406 EXPECT_EQ(2, num_times_logging_callback_called_); 406 EXPECT_EQ(2, num_times_logging_callback_called_);
407 } 407 }
408 408
409 } // namespace cast 409 } // namespace cast
410 } // namespace media 410 } // namespace media
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