From 73a28ee066d62c238e8c669ab781bdf289b720ee Mon Sep 17 00:00:00 2001
From: peah <peah@webrtc.org>
Date: Wed, 12 Oct 2016 03:01:49 -0700
Subject: [PATCH] The AudioProcessing class is used as an interface to the
 functionality in the audio processing module. Therefore, it should be a pure
 interface. This CL ensures that is the case.

BUG=webrtc:6515

Review-Url: https://codereview.webrtc.org/2406193002
Cr-Commit-Position: refs/heads/master@{#14608}
---
 webrtc/media/engine/fakewebrtcvoiceengine.h                 | 2 ++
 webrtc/modules/audio_processing/audio_processing_impl.cc    | 4 ++++
 webrtc/modules/audio_processing/audio_processing_impl.h     | 2 +-
 webrtc/modules/audio_processing/include/audio_processing.cc | 4 ----
 webrtc/modules/audio_processing/include/audio_processing.h  | 6 ++----
 .../audio_processing/include/mock_audio_processing.h        | 4 ++++
 6 files changed, 13 insertions(+), 9 deletions(-)

diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 0526b3995e..e0e71fb800 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -116,6 +116,8 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
   WEBRTC_STUB(StartDebugRecording,
               (const char filename[kMaxFilenameSize], int64_t max_size_bytes));
   WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes));
+  WEBRTC_STUB(StartDebugRecording, (FILE * handle));
+  WEBRTC_STUB(StartDebugRecordingForPlatformFile, (rtc::PlatformFile handle));
   WEBRTC_STUB(StopDebugRecording, ());
   WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
   webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 85f159716c..c892b7886c 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1251,6 +1251,10 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle,
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
+int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
+  return StartDebugRecording(handle, -1);
+}
+
 int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
     rtc::PlatformFile handle) {
   // Run in a single-threaded manner.
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index be0e9518c3..21bc588198 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -65,7 +65,7 @@ class AudioProcessingImpl : public AudioProcessing {
   int StartDebugRecording(const char filename[kMaxFilenameSize],
                           int64_t max_log_size_bytes) override;
   int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) override;
-
+  int StartDebugRecording(FILE* handle) override;
   int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
   int StopDebugRecording() override;
 
diff --git a/webrtc/modules/audio_processing/include/audio_processing.cc b/webrtc/modules/audio_processing/include/audio_processing.cc
index 37d4a4bc92..43b2c271fc 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.cc
+++ b/webrtc/modules/audio_processing/include/audio_processing.cc
@@ -31,9 +31,5 @@ Beamforming::Beamforming(bool enabled,
 
 Beamforming::~Beamforming() {}
 
-int AudioProcessing::StartDebugRecordingForPlatformFile(
-    rtc::PlatformFile handle) {
-  return -1;
-}
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index b5d2aa22c0..0fdbfd2d75 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -454,14 +454,12 @@ class AudioProcessing {
   virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0;
 
   // TODO(ivoc): Remove this function after Chrome stops using it.
-  int StartDebugRecording(FILE* handle) {
-    return StartDebugRecording(handle, -1);
-  }
+  virtual int StartDebugRecording(FILE* handle) = 0;
 
   // Same as above but uses an existing PlatformFile handle. Takes ownership
   // of |handle| and closes it at StopDebugRecording().
   // TODO(xians): Make this interface pure virtual.
-  virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle);
+  virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) = 0;
 
   // Stops recording debugging information, and closes the file. Recording
   // cannot be resumed in the same file (without overwriting it).
diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h
index 8bb9e08ead..998153deaa 100644
--- a/webrtc/modules/audio_processing/include/mock_audio_processing.h
+++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -259,6 +259,10 @@ class MockAudioProcessing : public AudioProcessing {
                    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(StartDebugRecording,
+               int(rtc::PlatformFile handle));
   MOCK_METHOD0(StopDebugRecording,
       int());
   MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());