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Side by Side Diff: webrtc/media/engine/fakewebrtccall.cc

Issue 1757683002: Make the audio channel communicate network state changes to the call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing code review issues Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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 10
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220 receiving_ = false; 220 receiving_ = false;
221 } 221 }
222 222
223 void FakeVideoReceiveStream::SetStats( 223 void FakeVideoReceiveStream::SetStats(
224 const webrtc::VideoReceiveStream::Stats& stats) { 224 const webrtc::VideoReceiveStream::Stats& stats) {
225 stats_ = stats; 225 stats_ = stats;
226 } 226 }
227 227
228 FakeCall::FakeCall(const webrtc::Call::Config& config) 228 FakeCall::FakeCall(const webrtc::Call::Config& config)
229 : config_(config), 229 : config_(config),
230 network_state_(webrtc::kNetworkUp), 230 audio_network_state_(webrtc::kNetworkUp),
231 video_network_state_(webrtc::kNetworkUp),
231 num_created_send_streams_(0), 232 num_created_send_streams_(0),
232 num_created_receive_streams_(0) {} 233 num_created_receive_streams_(0) {}
233 234
234 FakeCall::~FakeCall() { 235 FakeCall::~FakeCall() {
235 EXPECT_EQ(0u, video_send_streams_.size()); 236 EXPECT_EQ(0u, video_send_streams_.size());
236 EXPECT_EQ(0u, audio_send_streams_.size()); 237 EXPECT_EQ(0u, audio_send_streams_.size());
237 EXPECT_EQ(0u, video_receive_streams_.size()); 238 EXPECT_EQ(0u, video_receive_streams_.size());
238 EXPECT_EQ(0u, audio_receive_streams_.size()); 239 EXPECT_EQ(0u, audio_receive_streams_.size());
239 } 240 }
240 241
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269 270
270 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) { 271 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
271 for (const auto* p : GetAudioReceiveStreams()) { 272 for (const auto* p : GetAudioReceiveStreams()) {
272 if (p->GetConfig().rtp.remote_ssrc == ssrc) { 273 if (p->GetConfig().rtp.remote_ssrc == ssrc) {
273 return p; 274 return p;
274 } 275 }
275 } 276 }
276 return nullptr; 277 return nullptr;
277 } 278 }
278 279
279 webrtc::NetworkState FakeCall::GetNetworkState() const { 280 webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const {
280 return network_state_; 281 switch (media) {
282 case webrtc::MediaType::AUDIO:
283 return audio_network_state_;
284 case webrtc::MediaType::VIDEO:
285 return video_network_state_;
286 default:
287 ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
288 return webrtc::kNetworkDown;
289 }
281 } 290 }
282 291
283 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( 292 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
284 const webrtc::AudioSendStream::Config& config) { 293 const webrtc::AudioSendStream::Config& config) {
285 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(config); 294 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(config);
286 audio_send_streams_.push_back(fake_stream); 295 audio_send_streams_.push_back(fake_stream);
287 ++num_created_send_streams_; 296 ++num_created_send_streams_;
288 return fake_stream; 297 return fake_stream;
289 } 298 }
290 299
291 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { 300 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
292 auto it = std::find(audio_send_streams_.begin(), 301 auto it = std::find(audio_send_streams_.begin(),
293 audio_send_streams_.end(), 302 audio_send_streams_.end(),
294 static_cast<FakeAudioSendStream*>(send_stream)); 303 static_cast<FakeAudioSendStream*>(send_stream));
295 if (it == audio_send_streams_.end()) { 304 if (it == audio_send_streams_.end()) {
296 ADD_FAILURE() << "DestroyAudioSendStream called with unknown paramter."; 305 ADD_FAILURE() << "DestroyAudioSendStream called with unknown parameter.";
297 } else { 306 } else {
298 delete *it; 307 delete *it;
299 audio_send_streams_.erase(it); 308 audio_send_streams_.erase(it);
300 } 309 }
301 } 310 }
302 311
303 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream( 312 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(
304 const webrtc::AudioReceiveStream::Config& config) { 313 const webrtc::AudioReceiveStream::Config& config) {
305 audio_receive_streams_.push_back(new FakeAudioReceiveStream(config)); 314 audio_receive_streams_.push_back(new FakeAudioReceiveStream(config));
306 ++num_created_receive_streams_; 315 ++num_created_receive_streams_;
307 return audio_receive_streams_.back(); 316 return audio_receive_streams_.back();
308 } 317 }
309 318
310 void FakeCall::DestroyAudioReceiveStream( 319 void FakeCall::DestroyAudioReceiveStream(
311 webrtc::AudioReceiveStream* receive_stream) { 320 webrtc::AudioReceiveStream* receive_stream) {
312 auto it = std::find(audio_receive_streams_.begin(), 321 auto it = std::find(audio_receive_streams_.begin(),
313 audio_receive_streams_.end(), 322 audio_receive_streams_.end(),
314 static_cast<FakeAudioReceiveStream*>(receive_stream)); 323 static_cast<FakeAudioReceiveStream*>(receive_stream));
315 if (it == audio_receive_streams_.end()) { 324 if (it == audio_receive_streams_.end()) {
316 ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown paramter."; 325 ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown parameter.";
317 } else { 326 } else {
318 delete *it; 327 delete *it;
319 audio_receive_streams_.erase(it); 328 audio_receive_streams_.erase(it);
320 } 329 }
321 } 330 }
322 331
323 webrtc::VideoSendStream* FakeCall::CreateVideoSendStream( 332 webrtc::VideoSendStream* FakeCall::CreateVideoSendStream(
324 const webrtc::VideoSendStream::Config& config, 333 const webrtc::VideoSendStream::Config& config,
325 const webrtc::VideoEncoderConfig& encoder_config) { 334 const webrtc::VideoEncoderConfig& encoder_config) {
326 FakeVideoSendStream* fake_stream = 335 FakeVideoSendStream* fake_stream =
327 new FakeVideoSendStream(config, encoder_config); 336 new FakeVideoSendStream(config, encoder_config);
328 video_send_streams_.push_back(fake_stream); 337 video_send_streams_.push_back(fake_stream);
329 ++num_created_send_streams_; 338 ++num_created_send_streams_;
330 return fake_stream; 339 return fake_stream;
331 } 340 }
332 341
333 void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { 342 void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
334 auto it = std::find(video_send_streams_.begin(), 343 auto it = std::find(video_send_streams_.begin(),
335 video_send_streams_.end(), 344 video_send_streams_.end(),
336 static_cast<FakeVideoSendStream*>(send_stream)); 345 static_cast<FakeVideoSendStream*>(send_stream));
337 if (it == video_send_streams_.end()) { 346 if (it == video_send_streams_.end()) {
338 ADD_FAILURE() << "DestroyVideoSendStream called with unknown paramter."; 347 ADD_FAILURE() << "DestroyVideoSendStream called with unknown parameter.";
339 } else { 348 } else {
340 delete *it; 349 delete *it;
341 video_send_streams_.erase(it); 350 video_send_streams_.erase(it);
342 } 351 }
343 } 352 }
344 353
345 webrtc::VideoReceiveStream* FakeCall::CreateVideoReceiveStream( 354 webrtc::VideoReceiveStream* FakeCall::CreateVideoReceiveStream(
346 const webrtc::VideoReceiveStream::Config& config) { 355 const webrtc::VideoReceiveStream::Config& config) {
347 video_receive_streams_.push_back(new FakeVideoReceiveStream(config)); 356 video_receive_streams_.push_back(new FakeVideoReceiveStream(config));
348 ++num_created_receive_streams_; 357 ++num_created_receive_streams_;
349 return video_receive_streams_.back(); 358 return video_receive_streams_.back();
350 } 359 }
351 360
352 void FakeCall::DestroyVideoReceiveStream( 361 void FakeCall::DestroyVideoReceiveStream(
353 webrtc::VideoReceiveStream* receive_stream) { 362 webrtc::VideoReceiveStream* receive_stream) {
354 auto it = std::find(video_receive_streams_.begin(), 363 auto it = std::find(video_receive_streams_.begin(),
355 video_receive_streams_.end(), 364 video_receive_streams_.end(),
356 static_cast<FakeVideoReceiveStream*>(receive_stream)); 365 static_cast<FakeVideoReceiveStream*>(receive_stream));
357 if (it == video_receive_streams_.end()) { 366 if (it == video_receive_streams_.end()) {
358 ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown paramter."; 367 ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown parameter.";
359 } else { 368 } else {
360 delete *it; 369 delete *it;
361 video_receive_streams_.erase(it); 370 video_receive_streams_.erase(it);
362 } 371 }
363 } 372 }
364 373
365 webrtc::PacketReceiver* FakeCall::Receiver() { 374 webrtc::PacketReceiver* FakeCall::Receiver() {
366 return this; 375 return this;
367 } 376 }
368 377
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409 418
410 webrtc::Call::Stats FakeCall::GetStats() const { 419 webrtc::Call::Stats FakeCall::GetStats() const {
411 return stats_; 420 return stats_;
412 } 421 }
413 422
414 void FakeCall::SetBitrateConfig( 423 void FakeCall::SetBitrateConfig(
415 const webrtc::Call::Config::BitrateConfig& bitrate_config) { 424 const webrtc::Call::Config::BitrateConfig& bitrate_config) {
416 config_.bitrate_config = bitrate_config; 425 config_.bitrate_config = bitrate_config;
417 } 426 }
418 427
419 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { 428 void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
420 network_state_ = state; 429 webrtc::NetworkState state) {
430 switch (media) {
431 case webrtc::MediaType::AUDIO:
432 audio_network_state_ = state;
433 break;
434 case webrtc::MediaType::VIDEO:
435 video_network_state_ = state;
436 break;
437 default:
438 ADD_FAILURE()
439 << "SignalChannelNetworkState called with unknown parameter.";
440 }
421 } 441 }
422 442
423 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { 443 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
424 last_sent_packet_ = sent_packet; 444 last_sent_packet_ = sent_packet;
425 } 445 }
426 } // namespace cricket 446 } // namespace cricket
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