| /* |
| * Copyright 2021 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/audio_rtp_receiver.h" |
| |
| #include "media/base/media_channel.h" |
| #include "pc/test/mock_voice_media_channel.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/thread.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| using ::testing::_; |
| using ::testing::InvokeWithoutArgs; |
| using ::testing::Mock; |
| |
| static const int kTimeOut = 100; |
| static const double kDefaultVolume = 1; |
| static const double kVolume = 3.7; |
| static const uint32_t kSsrc = 3; |
| |
| namespace webrtc { |
| class AudioRtpReceiverTest : public ::testing::Test { |
| protected: |
| AudioRtpReceiverTest() |
| : worker_(rtc::Thread::Current()), |
| receiver_( |
| rtc::make_ref_counted<AudioRtpReceiver>(worker_, |
| std::string(), |
| std::vector<std::string>(), |
| false)), |
| media_channel_(rtc::Thread::Current()) { |
| EXPECT_CALL(media_channel_, SetRawAudioSink(kSsrc, _)); |
| EXPECT_CALL(media_channel_, SetBaseMinimumPlayoutDelayMs(kSsrc, _)); |
| } |
| |
| ~AudioRtpReceiverTest() { |
| receiver_->SetMediaChannel(nullptr); |
| receiver_->Stop(); |
| } |
| |
| rtc::Thread* worker_; |
| rtc::scoped_refptr<AudioRtpReceiver> receiver_; |
| cricket::MockVoiceMediaChannel media_channel_; |
| }; |
| |
| TEST_F(AudioRtpReceiverTest, SetOutputVolumeIsCalled) { |
| std::atomic_int set_volume_calls(0); |
| |
| EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kDefaultVolume)) |
| .WillOnce(InvokeWithoutArgs([&] { |
| set_volume_calls++; |
| return true; |
| })); |
| |
| receiver_->track(); |
| receiver_->track()->set_enabled(true); |
| receiver_->SetMediaChannel(&media_channel_); |
| receiver_->SetupMediaChannel(kSsrc); |
| |
| EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolume)) |
| .WillOnce(InvokeWithoutArgs([&] { |
| set_volume_calls++; |
| return true; |
| })); |
| |
| receiver_->OnSetVolume(kVolume); |
| EXPECT_TRUE_WAIT(set_volume_calls == 2, kTimeOut); |
| } |
| |
| TEST_F(AudioRtpReceiverTest, VolumesSetBeforeStartingAreRespected) { |
| // Set the volume before setting the media channel. It should still be used |
| // as the initial volume. |
| receiver_->OnSetVolume(kVolume); |
| |
| receiver_->track()->set_enabled(true); |
| receiver_->SetMediaChannel(&media_channel_); |
| |
| // The previosly set initial volume should be propagated to the provided |
| // media_channel_ as soon as SetupMediaChannel is called. |
| EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolume)); |
| |
| receiver_->SetupMediaChannel(kSsrc); |
| } |
| } // namespace webrtc |