Test submission of complete AEC-dump refactoring.

webrtc/modules/audio_processing/audio_processing_impl.h

Index: webrtc/modules/audio_processing/audio_processing_impl.h
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 01b640fcfc0f36332790765fd530f267690cfc9a..24a56658b03ccf750c0af24bae931b834cf70efd 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -23,22 +23,12 @@
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/include/aec_dump.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-// Files generated at build-time by the protobuf compiler.
-RTC_PUSH_IGNORING_WUNDEF()
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
-#else
-#include "webrtc/modules/audio_processing/debug.pb.h"
-#endif
-RTC_POP_IGNORING_WUNDEF()
-#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
-
 namespace webrtc {
 
 class AgcManagerDirect;
@@ -66,12 +56,8 @@ class AudioProcessingImpl : public AudioProcessing {
   void ApplyConfig(const AudioProcessing::Config& config) override;
   void SetExtraOptions(const webrtc::Config& config) override;
   void UpdateHistogramsOnCallEnd() 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;
+  void StartDebugRecording(std::unique_ptr<AecDump> aec_dump) override;
+  void StopDebugRecording() override;
 
   // Capture-side exclusive methods possibly running APM in a
   // multi-threaded manner. Acquire the capture lock.
@@ -275,29 +261,10 @@ class AudioProcessingImpl : public AudioProcessing {
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
   int ProcessRenderStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
 
-// Debug dump methods that are internal and called without locks.
-// TODO(peah): Make thread safe.
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  // TODO(andrew): make this more graceful. Ideally we would split this stuff
-  // out into a separate class with an "enabled" and "disabled" implementation.
-  static int WriteMessageToDebugFile(FileWrapper* debug_file,
-                                     int64_t* filesize_limit_bytes,
-                                     rtc::CriticalSection* crit_debug,
-                                     ApmDebugDumpThreadState* debug_state);
-  int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
-
-  // Writes Config message. If not |forced|, only writes the current config if
-  // it is different from the last saved one; if |forced|, writes the config
-  // regardless of the last saved.
-  int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
-
-  // Critical section.
-  rtc::CriticalSection crit_debug_;
+  int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
-  // Debug dump state.
-  ApmDebugDumpState debug_dump_;
-#endif
+  // TODO(aleloi) doc.
+  std::unique_ptr<AecDump> aec_dump_;
 
   // Critical sections.
   rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);