summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/audio/test/audio_end_to_end_test.h
diff options
context:
space:
mode:
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.h64
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_