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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/audio/test/audio_end_to_end_test.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/test/audio_end_to_end_test.h')
-rw-r--r-- | third_party/libwebrtc/audio/test/audio_end_to_end_test.h | 64 |
1 files changed, 64 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/test/audio_end_to_end_test.h b/third_party/libwebrtc/audio/test/audio_end_to_end_test.h new file mode 100644 index 0000000000..607fe692be --- /dev/null +++ b/third_party/libwebrtc/audio/test/audio_end_to_end_test.h @@ -0,0 +1,64 @@ +/* + * 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. + */ +#ifndef AUDIO_TEST_AUDIO_END_TO_END_TEST_H_ +#define AUDIO_TEST_AUDIO_END_TO_END_TEST_H_ + +#include <memory> +#include <string> +#include <vector> + +#include "api/task_queue/task_queue_base.h" +#include "api/test/simulated_network.h" +#include "test/call_test.h" + +namespace webrtc { +namespace test { + +class AudioEndToEndTest : public test::EndToEndTest { + public: + AudioEndToEndTest(); + + protected: + TestAudioDeviceModule* send_audio_device() { return send_audio_device_; } + const AudioSendStream* send_stream() const { return send_stream_; } + const AudioReceiveStreamInterface* receive_stream() const { + return receive_stream_; + } + + size_t GetNumVideoStreams() const override; + size_t GetNumAudioStreams() const override; + size_t GetNumFlexfecStreams() const override; + + std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override; + std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override; + + void OnFakeAudioDevicesCreated( + TestAudioDeviceModule* send_audio_device, + TestAudioDeviceModule* recv_audio_device) override; + + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector<AudioReceiveStreamInterface::Config>* + receive_configs) override; + void OnAudioStreamsCreated(AudioSendStream* send_stream, + const std::vector<AudioReceiveStreamInterface*>& + receive_streams) override; + + void PerformTest() override; + + private: + TestAudioDeviceModule* send_audio_device_ = nullptr; + AudioSendStream* send_stream_ = nullptr; + AudioReceiveStreamInterface* receive_stream_ = nullptr; +}; + +} // namespace test +} // namespace webrtc + +#endif // AUDIO_TEST_AUDIO_END_TO_END_TEST_H_ |