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| 1 /* | 1 /* |
| 2 * Copyright (c) 2008 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2008 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 |
| (...skipping 3322 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 3333 EXPECT_EQ(webrtc::kNetworkUp, | 3333 EXPECT_EQ(webrtc::kNetworkUp, |
| 3334 call_.GetNetworkState(webrtc::MediaType::VIDEO)); | 3334 call_.GetNetworkState(webrtc::MediaType::VIDEO)); |
| 3335 | 3335 |
| 3336 channel_->OnReadyToSend(true); | 3336 channel_->OnReadyToSend(true); |
| 3337 EXPECT_EQ(webrtc::kNetworkUp, | 3337 EXPECT_EQ(webrtc::kNetworkUp, |
| 3338 call_.GetNetworkState(webrtc::MediaType::AUDIO)); | 3338 call_.GetNetworkState(webrtc::MediaType::AUDIO)); |
| 3339 EXPECT_EQ(webrtc::kNetworkUp, | 3339 EXPECT_EQ(webrtc::kNetworkUp, |
| 3340 call_.GetNetworkState(webrtc::MediaType::VIDEO)); | 3340 call_.GetNetworkState(webrtc::MediaType::VIDEO)); |
| 3341 } | 3341 } |
| 3342 | 3342 |
| 3343 // Test that playout is still started after changing parameters | |
| 3344 TEST_F(WebRtcVoiceEngineTestFake, ParametersShouldPreservePlayout) { | |
|
the sun
2016/10/03 12:01:49
nit: Name is confusing - I don't see how the param
aleloi
2016/10/03 12:07:58
I agree. I went with your proposal.
| |
| 3345 SetupRecvStream(); | |
| 3346 channel_->SetPlayout(true); | |
|
the sun
2016/10/03 12:01:49
Add
EXPECT_TRUE(GetRecvStream(kSsrc1).started());
aleloi
2016/10/03 12:07:57
Done.
| |
| 3347 | |
| 3348 // Changing RTP header extensions will recreate the AudioReceiveStream. | |
| 3349 cricket::AudioRecvParameters parameters; | |
| 3350 parameters.extensions.push_back( | |
| 3351 webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); | |
| 3352 channel_->SetRecvParameters(parameters); | |
| 3353 | |
| 3354 EXPECT_TRUE(GetRecvStream(kSsrc1).started()); | |
| 3355 } | |
| 3356 | |
| 3343 // Tests that the library initializes and shuts down properly. | 3357 // Tests that the library initializes and shuts down properly. |
| 3344 TEST(WebRtcVoiceEngineTest, StartupShutdown) { | 3358 TEST(WebRtcVoiceEngineTest, StartupShutdown) { |
| 3345 // If the VoiceEngine wants to gather available codecs early, that's fine but | 3359 // If the VoiceEngine wants to gather available codecs early, that's fine but |
| 3346 // we never want it to create a decoder at this stage. | 3360 // we never want it to create a decoder at this stage. |
| 3347 cricket::WebRtcVoiceEngine engine( | 3361 cricket::WebRtcVoiceEngine engine( |
| 3348 nullptr, webrtc::MockAudioDecoderFactory::CreateUnusedFactory()); | 3362 nullptr, webrtc::MockAudioDecoderFactory::CreateUnusedFactory()); |
| 3349 std::unique_ptr<webrtc::Call> call( | 3363 std::unique_ptr<webrtc::Call> call( |
| 3350 webrtc::Call::Create(webrtc::Call::Config())); | 3364 webrtc::Call::Create(webrtc::Call::Config())); |
| 3351 cricket::VoiceMediaChannel* channel = engine.CreateChannel( | 3365 cricket::VoiceMediaChannel* channel = engine.CreateChannel( |
| 3352 call.get(), cricket::MediaConfig(), cricket::AudioOptions()); | 3366 call.get(), cricket::MediaConfig(), cricket::AudioOptions()); |
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| 3493 cricket::WebRtcVoiceEngine engine( | 3507 cricket::WebRtcVoiceEngine engine( |
| 3494 nullptr, webrtc::CreateBuiltinAudioDecoderFactory()); | 3508 nullptr, webrtc::CreateBuiltinAudioDecoderFactory()); |
| 3495 std::unique_ptr<webrtc::Call> call( | 3509 std::unique_ptr<webrtc::Call> call( |
| 3496 webrtc::Call::Create(webrtc::Call::Config())); | 3510 webrtc::Call::Create(webrtc::Call::Config())); |
| 3497 cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(), | 3511 cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(), |
| 3498 cricket::AudioOptions(), call.get()); | 3512 cricket::AudioOptions(), call.get()); |
| 3499 cricket::AudioRecvParameters parameters; | 3513 cricket::AudioRecvParameters parameters; |
| 3500 parameters.codecs = engine.recv_codecs(); | 3514 parameters.codecs = engine.recv_codecs(); |
| 3501 EXPECT_TRUE(channel.SetRecvParameters(parameters)); | 3515 EXPECT_TRUE(channel.SetRecvParameters(parameters)); |
| 3502 } | 3516 } |
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