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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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