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diff --git a/third_party/libwebrtc/modules/audio_device/include/mock_audio_transport.h b/third_party/libwebrtc/modules/audio_device/include/mock_audio_transport.h
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+++ b/third_party/libwebrtc/modules/audio_device/include/mock_audio_transport.h
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+/*
+ * Copyright (c) 2016 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 MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
+
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockAudioTransport : public AudioTransport {
+ public:
+ MockAudioTransport() {}
+ ~MockAudioTransport() {}
+
+ MOCK_METHOD(int32_t,
+ RecordedDataIsAvailable,
+ (const void* audioSamples,
+ size_t nSamples,
+ size_t nBytesPerSample,
+ size_t nChannels,
+ uint32_t samplesPerSec,
+ uint32_t totalDelayMS,
+ int32_t clockDrift,
+ uint32_t currentMicLevel,
+ bool keyPressed,
+ uint32_t& newMicLevel),
+ (override));
+
+ MOCK_METHOD(int32_t,
+ RecordedDataIsAvailable,
+ (const void* audioSamples,
+ size_t nSamples,
+ size_t nBytesPerSample,
+ size_t nChannels,
+ uint32_t samplesPerSec,
+ uint32_t totalDelayMS,
+ int32_t clockDrift,
+ uint32_t currentMicLevel,
+ bool keyPressed,
+ uint32_t& newMicLevel,
+ absl::optional<int64_t> estimated_capture_time_ns),
+ (override));
+
+ MOCK_METHOD(int32_t,
+ NeedMorePlayData,
+ (size_t nSamples,
+ size_t nBytesPerSample,
+ size_t nChannels,
+ uint32_t samplesPerSec,
+ void* audioSamples,
+ size_t& nSamplesOut,
+ int64_t* elapsed_time_ms,
+ int64_t* ntp_time_ms),
+ (override));
+
+ MOCK_METHOD(void,
+ PullRenderData,
+ (int bits_per_sample,
+ int sample_rate,
+ size_t number_of_channels,
+ size_t number_of_frames,
+ void* audio_data,
+ int64_t* elapsed_time_ms,
+ int64_t* ntp_time_ms),
+ (override));
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_