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 5fcc4d4672965b6ea3c3fe0cf57860259c1551cd..c96f23a2288472477cc90bd263c8cb039c0ba9ab 100644 |
--- a/webrtc/modules/audio_processing/include/audio_processing.h |
+++ b/webrtc/modules/audio_processing/include/audio_processing.h |
@@ -408,13 +408,22 @@ class AudioProcessing { |
// 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. |
- // An already existing file will be overwritten without warning. |
+ // An already existing file will be overwritten without warning. A maximum |
+ // file size (in bytes) for the log can be specified. The logging is stopped |
+ // once the limit has been reached. If max_log_size_bytes is set to a value |
+ // <= 0, no limit will be used. |
static const size_t kMaxFilenameSize = 1024; |
- virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0; |
+ virtual int StartDebugRecording(const char filename[kMaxFilenameSize], |
+ int64_t max_log_size_bytes) = 0; |
// Same as above but uses an existing file handle. Takes ownership |
// of |handle| and closes it at StopDebugRecording(). |
- virtual int StartDebugRecording(FILE* handle) = 0; |
+ virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0; |
+ |
+ // TODO(ivoc): Remove this function after Chromium switches to the one above. |
+ virtual int StartDebugRecording(FILE* handle) { |
+ return StartDebugRecording(handle, -1); |
+ } |
// Same as above but uses an existing PlatformFile handle. Takes ownership |
// of |handle| and closes it at StopDebugRecording(). |