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Issue 1226123005: Define Stream base classes (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Code review follow-up 3 Created 5 years, 5 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2015 Google Inc. 3 * Copyright 2015 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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187 receiving_ = false; 187 receiving_ = false;
188 } 188 }
189 189
190 void FakeVideoReceiveStream::SetStats( 190 void FakeVideoReceiveStream::SetStats(
191 const webrtc::VideoReceiveStream::Stats& stats) { 191 const webrtc::VideoReceiveStream::Stats& stats) {
192 stats_ = stats; 192 stats_ = stats;
193 } 193 }
194 194
195 FakeCall::FakeCall(const webrtc::Call::Config& config) 195 FakeCall::FakeCall(const webrtc::Call::Config& config)
196 : config_(config), 196 : config_(config),
197 network_state_(kNetworkUp), 197 network_state_(webrtc::kNetworkUp),
198 num_created_send_streams_(0), 198 num_created_send_streams_(0),
199 num_created_receive_streams_(0) { 199 num_created_receive_streams_(0) {
200 } 200 }
201 201
202 FakeCall::~FakeCall() { 202 FakeCall::~FakeCall() {
203 EXPECT_EQ(0u, video_send_streams_.size()); 203 EXPECT_EQ(0u, video_send_streams_.size());
204 EXPECT_EQ(0u, video_receive_streams_.size()); 204 EXPECT_EQ(0u, video_receive_streams_.size());
205 EXPECT_EQ(0u, audio_receive_streams_.size()); 205 EXPECT_EQ(0u, audio_receive_streams_.size());
206 } 206 }
207 207
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223 223
224 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) { 224 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
225 for (const auto p : GetAudioReceiveStreams()) { 225 for (const auto p : GetAudioReceiveStreams()) {
226 if (p->GetConfig().rtp.remote_ssrc == ssrc) { 226 if (p->GetConfig().rtp.remote_ssrc == ssrc) {
227 return p; 227 return p;
228 } 228 }
229 } 229 }
230 return nullptr; 230 return nullptr;
231 } 231 }
232 232
233 webrtc::Call::NetworkState FakeCall::GetNetworkState() const { 233 webrtc::NetworkState FakeCall::GetNetworkState() const {
234 return network_state_; 234 return network_state_;
235 } 235 }
236 236
237 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( 237 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
238 const webrtc::AudioSendStream::Config& config) { 238 const webrtc::AudioSendStream::Config& config) {
239 return nullptr; 239 return nullptr;
240 } 240 }
241 241
242 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { 242 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
243 } 243 }
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349 349
350 webrtc::Call::Stats FakeCall::GetStats() const { 350 webrtc::Call::Stats FakeCall::GetStats() const {
351 return stats_; 351 return stats_;
352 } 352 }
353 353
354 void FakeCall::SetBitrateConfig( 354 void FakeCall::SetBitrateConfig(
355 const webrtc::Call::Config::BitrateConfig& bitrate_config) { 355 const webrtc::Call::Config::BitrateConfig& bitrate_config) {
356 config_.bitrate_config = bitrate_config; 356 config_.bitrate_config = bitrate_config;
357 } 357 }
358 358
359 void FakeCall::SignalNetworkState(webrtc::Call::NetworkState state) { 359 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
360 network_state_ = state; 360 network_state_ = state;
361 } 361 }
362 } // namespace cricket 362 } // namespace cricket
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