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 7fc81f8899df3219537bc51289fad6fbebc105c4..0472edf094cb1ed02f203c9dd1f9cafc4b21d771 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"
@@ -64,6 +65,8 @@ class AudioProcessingImpl : public AudioProcessing {
   void ApplyConfig(const AudioProcessing::Config& config) override;
   void SetExtraOptions(const webrtc::Config& config) override;
   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;
@@ -278,6 +281,34 @@ class AudioProcessingImpl : public AudioProcessing {
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
   int ProcessRenderStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
 
+  // Collects configuration settings from public and private
+  // submodules to be saved as an audioproc::Config message on the
+  // AecDump if it is attached.  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.
+  void WriteAecDumpConfigMessage(bool forced)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  // Notifies attached AecDump of current configuration and capture data.
+  void RecordUnprocessedCaptureStream(const float* const* capture_stream)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  void RecordUnprocessedCaptureStream(const AudioFrame& capture_frame)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  // Notifies attached AecDump of current configuration and
+  // processed capture data and issues a capture stream recording
+  // request.
+  void RecordProcessedCaptureStream(
+      const float* const* processed_capture_stream)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  void RecordProcessedCaptureStream(const AudioFrame& processed_capture_frame)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  // Notifies attached AecDump about current state (delay, drift, etc).
+  void RecordAudioProcessingState() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
 // Debug dump methods that are internal and called without locks.
 // TODO(peah): Make thread safe.
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
@@ -302,6 +333,14 @@ 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_;
+
+  // Hold the last config written with AecDump for avoiding writing
+  // the same config twice.
+  InternalAPMConfig apm_config_for_aec_dump_ GUARDED_BY(crit_capture_);
+
   // Critical sections.
   rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
   rtc::CriticalSection crit_capture_;