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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: Use the presence of send/receive streams to infer which media types are active 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(
280 return network_state_; 281 webrtc::MediaType media) const {
282 switch (media) {
283 case webrtc::MediaType::AUDIO:
284 return audio_network_state_;
285 case webrtc::MediaType::VIDEO:
286 return video_network_state_;
287 default:
288 ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
289 return webrtc::kNetworkDown;
290 }
281 } 291 }
282 292
283 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( 293 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
284 const webrtc::AudioSendStream::Config& config) { 294 const webrtc::AudioSendStream::Config& config) {
285 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(config); 295 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(config);
286 audio_send_streams_.push_back(fake_stream); 296 audio_send_streams_.push_back(fake_stream);
287 ++num_created_send_streams_; 297 ++num_created_send_streams_;
288 return fake_stream; 298 return fake_stream;
289 } 299 }
290 300
291 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { 301 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
292 auto it = std::find(audio_send_streams_.begin(), 302 auto it = std::find(audio_send_streams_.begin(),
293 audio_send_streams_.end(), 303 audio_send_streams_.end(),
294 static_cast<FakeAudioSendStream*>(send_stream)); 304 static_cast<FakeAudioSendStream*>(send_stream));
295 if (it == audio_send_streams_.end()) { 305 if (it == audio_send_streams_.end()) {
296 ADD_FAILURE() << "DestroyAudioSendStream called with unknown paramter."; 306 ADD_FAILURE() << "DestroyAudioSendStream called with unknown parameter.";
297 } else { 307 } else {
298 delete *it; 308 delete *it;
299 audio_send_streams_.erase(it); 309 audio_send_streams_.erase(it);
300 } 310 }
301 } 311 }
302 312
303 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream( 313 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(
304 const webrtc::AudioReceiveStream::Config& config) { 314 const webrtc::AudioReceiveStream::Config& config) {
305 audio_receive_streams_.push_back(new FakeAudioReceiveStream(config)); 315 audio_receive_streams_.push_back(new FakeAudioReceiveStream(config));
306 ++num_created_receive_streams_; 316 ++num_created_receive_streams_;
307 return audio_receive_streams_.back(); 317 return audio_receive_streams_.back();
308 } 318 }
309 319
310 void FakeCall::DestroyAudioReceiveStream( 320 void FakeCall::DestroyAudioReceiveStream(
311 webrtc::AudioReceiveStream* receive_stream) { 321 webrtc::AudioReceiveStream* receive_stream) {
312 auto it = std::find(audio_receive_streams_.begin(), 322 auto it = std::find(audio_receive_streams_.begin(),
313 audio_receive_streams_.end(), 323 audio_receive_streams_.end(),
314 static_cast<FakeAudioReceiveStream*>(receive_stream)); 324 static_cast<FakeAudioReceiveStream*>(receive_stream));
315 if (it == audio_receive_streams_.end()) { 325 if (it == audio_receive_streams_.end()) {
316 ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown paramter."; 326 ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown parameter.";
317 } else { 327 } else {
318 delete *it; 328 delete *it;
319 audio_receive_streams_.erase(it); 329 audio_receive_streams_.erase(it);
320 } 330 }
321 } 331 }
322 332
323 webrtc::VideoSendStream* FakeCall::CreateVideoSendStream( 333 webrtc::VideoSendStream* FakeCall::CreateVideoSendStream(
324 const webrtc::VideoSendStream::Config& config, 334 const webrtc::VideoSendStream::Config& config,
325 const webrtc::VideoEncoderConfig& encoder_config) { 335 const webrtc::VideoEncoderConfig& encoder_config) {
326 FakeVideoSendStream* fake_stream = 336 FakeVideoSendStream* fake_stream =
327 new FakeVideoSendStream(config, encoder_config); 337 new FakeVideoSendStream(config, encoder_config);
328 video_send_streams_.push_back(fake_stream); 338 video_send_streams_.push_back(fake_stream);
329 ++num_created_send_streams_; 339 ++num_created_send_streams_;
330 return fake_stream; 340 return fake_stream;
331 } 341 }
332 342
333 void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { 343 void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
334 auto it = std::find(video_send_streams_.begin(), 344 auto it = std::find(video_send_streams_.begin(),
335 video_send_streams_.end(), 345 video_send_streams_.end(),
336 static_cast<FakeVideoSendStream*>(send_stream)); 346 static_cast<FakeVideoSendStream*>(send_stream));
337 if (it == video_send_streams_.end()) { 347 if (it == video_send_streams_.end()) {
338 ADD_FAILURE() << "DestroyVideoSendStream called with unknown paramter."; 348 ADD_FAILURE() << "DestroyVideoSendStream called with unknown parameter.";
339 } else { 349 } else {
340 delete *it; 350 delete *it;
341 video_send_streams_.erase(it); 351 video_send_streams_.erase(it);
342 } 352 }
343 } 353 }
344 354
345 webrtc::VideoReceiveStream* FakeCall::CreateVideoReceiveStream( 355 webrtc::VideoReceiveStream* FakeCall::CreateVideoReceiveStream(
346 const webrtc::VideoReceiveStream::Config& config) { 356 const webrtc::VideoReceiveStream::Config& config) {
347 video_receive_streams_.push_back(new FakeVideoReceiveStream(config)); 357 video_receive_streams_.push_back(new FakeVideoReceiveStream(config));
348 ++num_created_receive_streams_; 358 ++num_created_receive_streams_;
349 return video_receive_streams_.back(); 359 return video_receive_streams_.back();
350 } 360 }
351 361
352 void FakeCall::DestroyVideoReceiveStream( 362 void FakeCall::DestroyVideoReceiveStream(
353 webrtc::VideoReceiveStream* receive_stream) { 363 webrtc::VideoReceiveStream* receive_stream) {
354 auto it = std::find(video_receive_streams_.begin(), 364 auto it = std::find(video_receive_streams_.begin(),
355 video_receive_streams_.end(), 365 video_receive_streams_.end(),
356 static_cast<FakeVideoReceiveStream*>(receive_stream)); 366 static_cast<FakeVideoReceiveStream*>(receive_stream));
357 if (it == video_receive_streams_.end()) { 367 if (it == video_receive_streams_.end()) {
358 ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown paramter."; 368 ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown parameter.";
359 } else { 369 } else {
360 delete *it; 370 delete *it;
361 video_receive_streams_.erase(it); 371 video_receive_streams_.erase(it);
362 } 372 }
363 } 373 }
364 374
365 webrtc::PacketReceiver* FakeCall::Receiver() { 375 webrtc::PacketReceiver* FakeCall::Receiver() {
366 return this; 376 return this;
367 } 377 }
368 378
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409 419
410 webrtc::Call::Stats FakeCall::GetStats() const { 420 webrtc::Call::Stats FakeCall::GetStats() const {
411 return stats_; 421 return stats_;
412 } 422 }
413 423
414 void FakeCall::SetBitrateConfig( 424 void FakeCall::SetBitrateConfig(
415 const webrtc::Call::Config::BitrateConfig& bitrate_config) { 425 const webrtc::Call::Config::BitrateConfig& bitrate_config) {
416 config_.bitrate_config = bitrate_config; 426 config_.bitrate_config = bitrate_config;
417 } 427 }
418 428
419 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { 429 void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
420 network_state_ = state; 430 webrtc::NetworkState state) {
431 switch (media) {
432 case webrtc::MediaType::AUDIO:
433 audio_network_state_ = state;
434 break;
435 case webrtc::MediaType::VIDEO:
436 video_network_state_ = state;
437 break;
438 default:
439 ADD_FAILURE() << "SignalChannelNetworkState called with unknown parameter.";
the sun 2016/03/07 14:16:43 nit: indent
skvlad 2016/03/07 19:20:56 Done.
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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