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2da85916ab
This reverts commit3ed36c0521. Reason for revert: Breaks downstream project. Original change's description: > Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver." > > This is a reland ofbb57e2d7aa> > The difference from the original CL is that a check for > `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed. > This caused a side effect that registering the sink while the source > was in an "initializing" state, failed. The last remaining state > however, is `kEnded` - but since there's no logic in the class around > the expected value of the states, the check inside of AddSink() > doesn't provide an additional value - it's rather a surprise for > developers if it doesn't succeed. So, now removed. > > Original change's description: > > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver. > > > > This simplifies the logic in these classes a bit, which makes upcoming > > change easier. The `stopped_` flag in these classes was essentially > > the same thing as `media_channel_ == nullptr`, which is what's > > consistently used now for the same checks. > > > > Bug: webrtc:13540 > > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35907} > > Bug: webrtc:13540 > Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326 > Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35958} TBR=ilnik@webrtc.org,tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com Change-Id: Ieb7235d88c808c78ad0847403be991d4dce1ace6 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:13540 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251383 Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35963}
92 lines
2.9 KiB
C++
92 lines
2.9 KiB
C++
/*
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* Copyright 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/audio_rtp_receiver.h"
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#include "media/base/media_channel.h"
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#include "pc/test/mock_voice_media_channel.h"
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#include "rtc_base/gunit.h"
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#include "rtc_base/thread.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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using ::testing::_;
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using ::testing::InvokeWithoutArgs;
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using ::testing::Mock;
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static const int kTimeOut = 100;
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static const double kDefaultVolume = 1;
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static const double kVolume = 3.7;
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static const uint32_t kSsrc = 3;
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namespace webrtc {
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class AudioRtpReceiverTest : public ::testing::Test {
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protected:
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AudioRtpReceiverTest()
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: worker_(rtc::Thread::Current()),
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receiver_(
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rtc::make_ref_counted<AudioRtpReceiver>(worker_,
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std::string(),
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std::vector<std::string>(),
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false)),
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media_channel_(rtc::Thread::Current()) {
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EXPECT_CALL(media_channel_, SetRawAudioSink(kSsrc, _));
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EXPECT_CALL(media_channel_, SetBaseMinimumPlayoutDelayMs(kSsrc, _));
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}
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~AudioRtpReceiverTest() {
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receiver_->SetMediaChannel(nullptr);
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receiver_->Stop();
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}
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rtc::Thread* worker_;
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rtc::scoped_refptr<AudioRtpReceiver> receiver_;
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cricket::MockVoiceMediaChannel media_channel_;
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};
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TEST_F(AudioRtpReceiverTest, SetOutputVolumeIsCalled) {
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std::atomic_int set_volume_calls(0);
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EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kDefaultVolume))
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.WillOnce(InvokeWithoutArgs([&] {
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set_volume_calls++;
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return true;
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}));
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receiver_->track();
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receiver_->track()->set_enabled(true);
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receiver_->SetMediaChannel(&media_channel_);
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receiver_->SetupMediaChannel(kSsrc);
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EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolume))
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.WillOnce(InvokeWithoutArgs([&] {
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set_volume_calls++;
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return true;
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}));
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receiver_->OnSetVolume(kVolume);
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EXPECT_TRUE_WAIT(set_volume_calls == 2, kTimeOut);
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}
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TEST_F(AudioRtpReceiverTest, VolumesSetBeforeStartingAreRespected) {
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// Set the volume before setting the media channel. It should still be used
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// as the initial volume.
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receiver_->OnSetVolume(kVolume);
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receiver_->track()->set_enabled(true);
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receiver_->SetMediaChannel(&media_channel_);
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// The previosly set initial volume should be propagated to the provided
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// media_channel_ as soon as SetupMediaChannel is called.
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EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolume));
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receiver_->SetupMediaChannel(kSsrc);
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}
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} // namespace webrtc
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