Adding upstream version 110.0.1.upstream/110.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 127 insertions, 0 deletions

diff --git a/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc b/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc
new file mode 100644
index 0000000000..bab6b74f9f
--- /dev/null
+++ b/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc
@@ -0,0 +1,127 @@
+/*
+ *  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/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::AutoThread main_thread_;
+  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_);
+  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