webrtc_m130/pc/audio_rtp_receiver_unittest.cc

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/*
* 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 <atomic>
#include "pc/test/mock_voice_media_channel.h"
#include "rtc_base/gunit.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/run_loop.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 double kVolumeMuted = 0.0;
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() {
EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolumeMuted));
receiver_->SetMediaChannel(nullptr);
}
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_);
Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver." This is a reland of 3ed36c0521546881656c73984456485dcab16205 Original change's description: > Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver." > > This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f > > 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} Bug: webrtc:13540 Change-Id: I6d7d67fddb1ddfc69a302f0f69a9b815f2fd82f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251386 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35967}
2022-02-08 21:12:15 +01:00
EXPECT_CALL(media_channel_, SetDefaultRawAudioSink(_)).Times(0);
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);
}
// Tests that OnChanged notifications are processed correctly on the worker
// thread when a media channel pointer is passed to the receiver via the
// constructor.
TEST(AudioRtpReceiver, OnChangedNotificationsAfterConstruction) {
webrtc::test::RunLoop loop;
auto* thread = rtc::Thread::Current(); // Points to loop's thread.
cricket::MockVoiceMediaChannel media_channel(thread);
auto receiver = rtc::make_ref_counted<AudioRtpReceiver>(
thread, std::string(), std::vector<std::string>(), true, &media_channel);
EXPECT_CALL(media_channel, SetDefaultRawAudioSink(_)).Times(1);
EXPECT_CALL(media_channel, SetDefaultOutputVolume(kDefaultVolume)).Times(1);
receiver->SetupUnsignaledMediaChannel();
loop.Flush();
// Mark the track as disabled.
receiver->track()->set_enabled(false);
// When the track was marked as disabled, an async notification was queued
// for the worker thread. This notification should trigger the volume
// of the media channel to be set to kVolumeMuted.
// Flush the worker thread, but set the expectation first for the call.
EXPECT_CALL(media_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
loop.Flush();
EXPECT_CALL(media_channel, SetDefaultOutputVolume(kVolumeMuted)).Times(1);
receiver->SetMediaChannel(nullptr);
}
} // namespace webrtc