2f55370634
This reverts commit 18c869bc36b342cd4a79947067e52a93a04a7808. Reason for revert: Added a field trial that allows landing the code without affecting performance in prod. This CL also incorporates subsequent CLs that also had to be reverted. Original change's description: > Revert "Use two MediaChannels for 2 directions." > > This reverts commit 8981a6fac3d665beac4a58b9453e6c39988a024f. > > Reason for revert: Quality regression detected. > > Original change's description: > > Use two MediaChannels for 2 directions. > > > > This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other. > > > > The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it. > > > > Bug: webrtc:13931 > > Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39340} > > No-Try: true > Bug: webrtc:13931 > Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39445} Bug: webrtc:13931 Change-Id: I1318910a685188e2b846c9040e1efc04c2c894ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296080 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39494}
130 lines
4.5 KiB
C++
130 lines
4.5 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 <atomic>
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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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#include "test/run_loop.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 double kVolumeMuted = 0.0;
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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_(cricket::MediaChannel::Role::kReceive,
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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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EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolumeMuted));
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receiver_->SetMediaChannel(nullptr);
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}
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rtc::AutoThread main_thread_;
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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_.AsVoiceReceiveChannel());
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EXPECT_CALL(media_channel_, SetDefaultRawAudioSink(_)).Times(0);
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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_.AsVoiceReceiveChannel());
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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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// Tests that OnChanged notifications are processed correctly on the worker
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// thread when a media channel pointer is passed to the receiver via the
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// constructor.
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TEST(AudioRtpReceiver, OnChangedNotificationsAfterConstruction) {
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webrtc::test::RunLoop loop;
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auto* thread = rtc::Thread::Current(); // Points to loop's thread.
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cricket::MockVoiceMediaChannel media_channel(
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cricket::MediaChannel::Role::kReceive, thread);
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auto receiver = rtc::make_ref_counted<AudioRtpReceiver>(
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thread, std::string(), std::vector<std::string>(), true, &media_channel);
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EXPECT_CALL(media_channel, SetDefaultRawAudioSink(_)).Times(1);
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EXPECT_CALL(media_channel, SetDefaultOutputVolume(kDefaultVolume)).Times(1);
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receiver->SetupUnsignaledMediaChannel();
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loop.Flush();
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// Mark the track as disabled.
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receiver->track()->set_enabled(false);
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// When the track was marked as disabled, an async notification was queued
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// for the worker thread. This notification should trigger the volume
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// of the media channel to be set to kVolumeMuted.
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// Flush the worker thread, but set the expectation first for the call.
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EXPECT_CALL(media_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
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loop.Flush();
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EXPECT_CALL(media_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
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receiver->SetMediaChannel(nullptr);
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}
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} // namespace webrtc
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