Index: webrtc/modules/audio_processing/include/audio_processing.h |
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h |
index cc6772bf718fac20351c3b55a1b92c5970e0809a..fab75a0a9cd10d0e32e763271517ab2319e81743 100644 |
--- a/webrtc/modules/audio_processing/include/audio_processing.h |
+++ b/webrtc/modules/audio_processing/include/audio_processing.h |
@@ -29,6 +29,7 @@ namespace webrtc { |
struct AecCore; |
+class AecDump; |
class AudioFrame; |
class NonlinearBeamformer; |
@@ -447,6 +448,15 @@ class AudioProcessing { |
virtual void set_delay_offset_ms(int offset) = 0; |
virtual int delay_offset_ms() const = 0; |
+ // Starts recording debugging information through the provided |
+ // webrtc::AecDump. Log file and maximum file size logic is supposed |
+ // to be handled by implementing instance of AecDump. Calling this |
+ // method when logging is active resets the active AecDump with a |
+ // new one. This causes the d-tor of the earlier AecDump to be |
+ // called. The d-tor call may block until all pending logging tasks |
+ // are completed. |
+ virtual void StartDebugRecording(std::unique_ptr<AecDump> aec_dump) = 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. |