AecDump interface

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 37b15c28935f29cb6450f5f5cec3d53fe640f50f..ca12887e870024874df7c07f3028b2afe6954756 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -23,6 +23,7 @@
 #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"
@@ -66,6 +67,7 @@ class AudioProcessingImpl : public AudioProcessing {
   void ApplyConfig(const AudioProcessing::Config& config) override;
   void SetExtraOptions(const webrtc::Config& config) override;
   void UpdateHistogramsOnCallEnd() override;
+  void StartDebugRecording(std::unique_ptr<AecDump> aec_dump) 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;
@@ -299,6 +301,10 @@ class AudioProcessingImpl : public AudioProcessing {
   ApmDebugDumpState debug_dump_;
 #endif
 
+  // AecDump instance used for optionally logging APM config, input
+  // and output to file in the AEC-dump format defined in debug.proto.
+  std::unique_ptr<AecDump> aec_dump_;
+
   // Critical sections.
   rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
   rtc::CriticalSection crit_capture_;