Adding upstream version 115.8.0esr.upstream/115.8.0esr

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

1 files changed, 50 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/channel_receive_unittest.cc b/third_party/libwebrtc/audio/channel_receive_unittest.cc
new file mode 100644
index 0000000000..3d9baebe89
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_receive_unittest.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2023 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 "audio/channel_receive.h"
+
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "rtc_base/thread.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace voe {
+
+TEST(ChannelReceiveTest, CreateAndDestroy) {
+  GlobalSimulatedTimeController time_controller(Timestamp::Seconds(5555));
+  uint32_t local_ssrc = 1111;
+  uint32_t remote_ssrc = 2222;
+  webrtc::CryptoOptions crypto_options;
+  rtc::scoped_refptr<test::MockAudioDeviceModule> audio_device_module =
+      test::MockAudioDeviceModule::CreateNice();
+  MockTransport transport;
+  auto channel = CreateChannelReceive(
+      time_controller.GetClock(),
+      /* neteq_factory= */ nullptr, audio_device_module.get(), &transport,
+      /* rtc_event_log= */ nullptr, local_ssrc, remote_ssrc,
+      /* jitter_buffer_max_packets= */ 0,
+      /* jitter_buffer_fast_playout= */ false,
+      /* jitter_buffer_min_delay_ms= */ 0,
+      /* enable_non_sender_rtt= */ false,
+      /* decoder_factory= */ nullptr,
+      /* codec_pair_id= */ absl::nullopt,
+      /* frame_decryptor_interface= */ nullptr, crypto_options,
+      /* frame_transformer= */ nullptr);
+  EXPECT_TRUE(!!channel);
+}
+
+}  // namespace voe
+}  // namespace webrtc