Remove older AEC-dump interface.

This CL completely removes the methods
AudioProcessing::{Start,Stop}DebugDumpRecording. These methods have
been replaced with AudioProcessing::{Attach,Detach}AecDump. Their
implementation was removed in the parent CL
https://chromium-review.googlesource.com/c/589147

Bug: webrtc:7404
Change-Id: Ia3d5314985af9c74f79c94c514ded1f8afc78fb5
Reviewed-on: https://chromium-review.googlesource.com/589152
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19334}

webrtc/modules/audio_processing/audio_processing_impl.cc

@@ -1517,36 +1517,6 @@ void AudioProcessingImpl::DetachAecDump() {
   }
 }
 
-int AudioProcessingImpl::StartDebugRecording(
-    const char filename[AudioProcessing::kMaxFilenameSize],
-    int64_t max_log_size_bytes) {
-  RTC_NOTREACHED();
-  return kUnsupportedFunctionError;
-}
-
-int AudioProcessingImpl::StartDebugRecording(FILE* handle,
-                                             int64_t max_log_size_bytes) {
-  RTC_NOTREACHED();
-  return kUnsupportedFunctionError;
-}
-
-int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
-  RTC_NOTREACHED();
-  return kUnsupportedFunctionError;
-}
-
-int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
-    rtc::PlatformFile handle) {
-  RTC_NOTREACHED();
-  return kUnsupportedFunctionError;
-}
-
-int AudioProcessingImpl::StopDebugRecording() {
-  // DetachAecDump();
-  RTC_NOTREACHED();
-  return kUnsupportedFunctionError;
-}
-
 AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics() {
   residual_echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
   echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);

webrtc/modules/audio_processing/audio_processing_impl.h

@@ -56,12 +56,6 @@ class AudioProcessingImpl : public AudioProcessing {
   void UpdateHistogramsOnCallEnd() override;
   void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
   void DetachAecDump() override;
-  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;
 
   // Capture-side exclusive methods possibly running APM in a
   // multi-threaded manner. Acquire the capture lock.

webrtc/modules/audio_processing/include/audio_processing.h

@@ -471,33 +471,6 @@ class AudioProcessing : public rtc::RefCountInterface {
   // all pending logging tasks are completed.
   virtual void DetachAecDump() = 0;
 
-  // Starts recording debugging information to a file specified by |filename|,
-  // a NULL-terminated string. If there is an ongoing recording, the old file
-  // will be closed, and recording will continue in the newly specified file.
-  // An already existing file will be overwritten without warning. A maximum
-  // file size (in bytes) for the log can be specified. The logging is stopped
-  // once the limit has been reached. If max_log_size_bytes is set to a value
-  // <= 0, no limit will be used.
-  static const size_t kMaxFilenameSize = 1024;
-  virtual int StartDebugRecording(const char filename[kMaxFilenameSize],
-                                  int64_t max_log_size_bytes) = 0;
-
-  // Same as above but uses an existing file handle. Takes ownership
-  // of |handle| and closes it at StopDebugRecording().
-  virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0;
-
-  // TODO(ivoc): Remove this function after Chrome stops using it.
-  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) = 0;
-
-  // Stops recording debugging information, and closes the file. Recording
-  // cannot be resumed in the same file (without overwriting it).
-  virtual int StopDebugRecording() = 0;
-
   // Use to send UMA histograms at end of a call. Note that all histogram
   // specific member variables are reset.
   virtual void UpdateHistogramsOnCallEnd() = 0;

webrtc/modules/audio_processing/include/mock_audio_processing.h

@@ -179,14 +179,6 @@ class MockAudioProcessing : public AudioProcessing {
   virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) {}
   MOCK_METHOD0(DetachAecDump, void());
 
-  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_METHOD0(UpdateHistogramsOnCallEnd, void());
   MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics());
   virtual MockEchoCancellation* echo_cancellation() const {