| Index: webrtc/modules/audio_processing/include/mock_audio_processing.h
|
| diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h
|
| index b5a1fa4a8851329111db743bbacf77d1c33a0f89..79787c9d2065a3e0917bd24d95ef5d1b3c9e6b82 100644
|
| --- a/webrtc/modules/audio_processing/include/mock_audio_processing.h
|
| +++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h
|
| @@ -13,6 +13,7 @@
|
|
|
| #include <memory>
|
|
|
| +#include "webrtc/modules/audio_processing/include/aec_dump.h"
|
| #include "webrtc/modules/audio_processing/include/audio_processing.h"
|
| #include "webrtc/test/gmock.h"
|
|
|
| @@ -174,14 +175,9 @@ class MockAudioProcessing : public AudioProcessing {
|
| MOCK_METHOD1(set_stream_key_pressed, void(bool key_pressed));
|
| MOCK_METHOD1(set_delay_offset_ms, void(int offset));
|
| MOCK_CONST_METHOD0(delay_offset_ms, int());
|
| - MOCK_METHOD2(StartDebugRecording, int(const char filename[kMaxFilenameSize],
|
| - int64_t max_log_size_bytes));
|
| - MOCK_METHOD2(StartDebugRecording, int(FILE* handle,
|
| - int64_t max_log_size_bytes));
|
| - MOCK_METHOD1(StartDebugRecording, int (FILE* handle));
|
| - MOCK_METHOD1(StartDebugRecordingForPlatformFile,
|
| - int(rtc::PlatformFile handle));
|
| - MOCK_METHOD0(StopDebugRecording, int());
|
| + // MOCK_METHOD1(StartDebugRecording,
|
| + // void(std::unique_ptr<AecDump> aec_dump));
|
| + MOCK_METHOD0(StopDebugRecording, void());
|
| MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
|
| MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics());
|
| virtual MockEchoCancellation* echo_cancellation() const {
|
| @@ -206,6 +202,8 @@ class MockAudioProcessing : public AudioProcessing {
|
| return voice_detection_.get();
|
| }
|
|
|
| + virtual void StartDebugRecording(std::unique_ptr<AecDump> aec_dump) {}
|
| +
|
| private:
|
| std::unique_ptr<MockEchoCancellation> echo_cancellation_;
|
| std::unique_ptr<MockEchoControlMobile> echo_control_mobile_;
|
|
|