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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/audio/test/audio_end_to_end_test.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/test/audio_end_to_end_test.cc')
-rw-r--r-- | third_party/libwebrtc/audio/test/audio_end_to_end_test.cc | 91 |
1 files changed, 91 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc b/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc new file mode 100644 index 0000000000..b7dfdb89f6 --- /dev/null +++ b/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc @@ -0,0 +1,91 @@ +/* + * Copyright (c) 2017 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/test/audio_end_to_end_test.h" + +#include <algorithm> +#include <memory> + +#include "api/task_queue/task_queue_base.h" +#include "call/fake_network_pipe.h" +#include "call/simulated_network.h" +#include "system_wrappers/include/sleep.h" +#include "test/gtest.h" + +namespace webrtc { +namespace test { +namespace { +// Wait half a second between stopping sending and stopping receiving audio. +constexpr int kExtraRecordTimeMs = 500; + +constexpr int kSampleRate = 48000; +} // namespace + +AudioEndToEndTest::AudioEndToEndTest() + : EndToEndTest(CallTest::kDefaultTimeout) {} + +size_t AudioEndToEndTest::GetNumVideoStreams() const { + return 0; +} + +size_t AudioEndToEndTest::GetNumAudioStreams() const { + return 1; +} + +size_t AudioEndToEndTest::GetNumFlexfecStreams() const { + return 0; +} + +std::unique_ptr<TestAudioDeviceModule::Capturer> +AudioEndToEndTest::CreateCapturer() { + return TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, kSampleRate); +} + +std::unique_ptr<TestAudioDeviceModule::Renderer> +AudioEndToEndTest::CreateRenderer() { + return TestAudioDeviceModule::CreateDiscardRenderer(kSampleRate); +} + +void AudioEndToEndTest::OnFakeAudioDevicesCreated( + TestAudioDeviceModule* send_audio_device, + TestAudioDeviceModule* recv_audio_device) { + send_audio_device_ = send_audio_device; +} + +void AudioEndToEndTest::ModifyAudioConfigs( + AudioSendStream::Config* send_config, + std::vector<AudioReceiveStreamInterface::Config>* receive_configs) { + // Large bitrate by default. + const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2, + {{"stereo", "1"}}); + send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( + test::CallTest::kAudioSendPayloadType, kDefaultFormat); + send_config->min_bitrate_bps = 32000; + send_config->max_bitrate_bps = 32000; +} + +void AudioEndToEndTest::OnAudioStreamsCreated( + AudioSendStream* send_stream, + const std::vector<AudioReceiveStreamInterface*>& receive_streams) { + ASSERT_NE(nullptr, send_stream); + ASSERT_EQ(1u, receive_streams.size()); + ASSERT_NE(nullptr, receive_streams[0]); + send_stream_ = send_stream; + receive_stream_ = receive_streams[0]; +} + +void AudioEndToEndTest::PerformTest() { + // Wait until the input audio file is done... + send_audio_device_->WaitForRecordingEnd(); + // and some extra time to account for network delay. + SleepMs(GetSendTransportConfig().queue_delay_ms + kExtraRecordTimeMs); +} +} // namespace test +} // namespace webrtc |