Test submission of complete AEC-dump refactoring.

webrtc/modules/audio_processing/audio_processing_impl.h

 /*
  *  Copyright (c) 2012 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 
 #include <list>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/function_view.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/ignore_wundef.h"
 #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;
 class AudioConverter;
 
 class NonlinearBeamformer;
 
 class AudioProcessingImpl : public AudioProcessing {
  public:
   // Methods forcing APM to run in a single-threaded manner.
   // Acquires both the render and capture locks.
   explicit AudioProcessingImpl(const webrtc::Config& config);
   // AudioProcessingImpl takes ownership of beamformer.
   AudioProcessingImpl(const webrtc::Config& config,
                       NonlinearBeamformer* beamformer);
   ~AudioProcessingImpl() override;
   int Initialize() override;
   int Initialize(int capture_input_sample_rate_hz,
                  int capture_output_sample_rate_hz,
                  int render_sample_rate_hz,
                  ChannelLayout capture_input_layout,
                  ChannelLayout capture_output_layout,
                  ChannelLayout render_input_layout) override;
   int Initialize(const ProcessingConfig& processing_config) override;
   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.
   int ProcessStream(AudioFrame* frame) override;
   int ProcessStream(const float* const* src,
                     size_t samples_per_channel,
                     int input_sample_rate_hz,
                     ChannelLayout input_layout,
                     int output_sample_rate_hz,
                     ChannelLayout output_layout,
@@ skipping 183 matching lines @@
 
   // Render-side exclusive methods possibly running APM in a multi-threaded
   // manner that are called with the render lock already acquired.
   // TODO(ekm): Remove once all clients updated to new interface.
   int AnalyzeReverseStreamLocked(const float* const* src,
                                  const StreamConfig& input_config,
                                  const StreamConfig& output_config)
       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_);
+  int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(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_;
-
-  // 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_);
   rtc::CriticalSection crit_capture_;
 
   // Struct containing the Config specifying the behavior of APM.
   AudioProcessing::Config config_;
 
   // Class containing information about what submodules are active.
   ApmSubmoduleStates submodule_states_;
@@ skipping 117 matching lines @@
   std::unique_ptr<
       SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
       agc_render_signal_queue_;
   std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
       red_render_signal_queue_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_