AecDump interface

webrtc/modules/audio_processing/include/mock_audio_processing.h

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 156 matching lines @@
   MOCK_METHOD4(ProcessReverseStream, int(const float* const* src,
                                          const StreamConfig& input_config,
                                          const StreamConfig& output_config,
                                          float* const* dest));
   MOCK_METHOD1(set_stream_delay_ms, int(int delay));
   MOCK_CONST_METHOD0(stream_delay_ms, int());
   MOCK_CONST_METHOD0(was_stream_delay_set, bool());
   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_METHOD3(StartDebugRecording,
+               int(const char filename[kMaxFilenameSize],
+                   int64_t max_log_size_bytes,
+                   rtc::TaskQueue* worker_queue));
+  MOCK_METHOD3(StartDebugRecording,
+               int(FILE* handle,
+                   int64_t max_log_size_bytes,
+                   rtc::TaskQueue* worker_queue));
+  MOCK_METHOD2(StartDebugRecording,
+               int(FILE* handle, rtc::TaskQueue* worker_queue));
+  MOCK_METHOD2(StartDebugRecordingForPlatformFile,
+               int(rtc::PlatformFile handle, rtc::TaskQueue* worker_queue));
   MOCK_METHOD0(StopDebugRecording, int());
   MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
   MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics());
   virtual MockEchoCancellation* echo_cancellation() const {
     return echo_cancellation_.get();
   }
   virtual MockEchoControlMobile* echo_control_mobile() const {
     return echo_control_mobile_.get();
   }
   virtual MockGainControl* gain_control() const {
@@ skipping 19 matching lines @@
   std::unique_ptr<MockHighPassFilter> high_pass_filter_;
   std::unique_ptr<MockLevelEstimator> level_estimator_;
   std::unique_ptr<MockNoiseSuppression> noise_suppression_;
   std::unique_ptr<MockVoiceDetection> voice_detection_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_