Index: webrtc/modules/audio_device/include/mock_audio_transport.h |
diff --git a/webrtc/modules/audio_device/include/mock_audio_transport.h b/webrtc/modules/audio_device/include/mock_audio_transport.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..a05e7ea839298ea6ab1acd3e14c8b173ffada042 |
--- /dev/null |
+++ b/webrtc/modules/audio_device/include/mock_audio_transport.h |
@@ -0,0 +1,68 @@ |
+/* |
+ * 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 WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_ |
+#define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_ |
+ |
+#include "webrtc/modules/audio_device/include/audio_device_defines.h" |
+#include "webrtc/test/gmock.h" |
+ |
+namespace webrtc { |
+namespace test { |
+ |
+class MockAudioTransport : public AudioTransport { |
+ public: |
+ MockAudioTransport() {} |
+ ~MockAudioTransport() {} |
+ |
+ MOCK_METHOD10(RecordedDataIsAvailable, |
+ int32_t(const void* audioSamples, |
+ const size_t nSamples, |
+ const size_t nBytesPerSample, |
+ const size_t nChannels, |
+ const uint32_t samplesPerSec, |
+ const uint32_t totalDelayMS, |
+ const int32_t clockDrift, |
+ const uint32_t currentMicLevel, |
+ const bool keyPressed, |
+ uint32_t& newMicLevel)); |
+ |
+ MOCK_METHOD8(NeedMorePlayData, |
+ int32_t(const size_t nSamples, |
+ const size_t nBytesPerSample, |
+ const size_t nChannels, |
+ const uint32_t samplesPerSec, |
+ void* audioSamples, |
+ size_t& nSamplesOut, |
+ int64_t* elapsed_time_ms, |
+ int64_t* ntp_time_ms)); |
+ |
+ MOCK_METHOD6(PushCaptureData, |
+ void(int voe_channel, |
+ const void* audio_data, |
+ int bits_per_sample, |
+ int sample_rate, |
+ size_t number_of_channels, |
+ size_t number_of_frames)); |
+ |
+ MOCK_METHOD7(PullRenderData, |
+ void(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)); |
+}; |
+ |
+} // namespace test |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_ |