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 ab924b75343e5d9c086e6a689ca7726b6e830c3d..e04e708b8f98039fb8fede72e3b06ebfe46170be 100644 |
--- a/webrtc/modules/audio_processing/include/audio_processing.h |
+++ b/webrtc/modules/audio_processing/include/audio_processing.h |
@@ -25,6 +25,10 @@ |
#include "webrtc/modules/audio_processing/include/config.h" |
#include "webrtc/typedefs.h" |
+namespace rtc { |
+class TaskQueue; |
+} // namespace rtc |
+ |
namespace webrtc { |
struct AecCore; |
@@ -456,19 +460,25 @@ class AudioProcessing { |
// <= 0, no limit will be used. |
static const size_t kMaxFilenameSize = 1024; |
virtual int StartDebugRecording(const char filename[kMaxFilenameSize], |
- int64_t max_log_size_bytes) = 0; |
+ int64_t max_log_size_bytes, |
+ rtc::TaskQueue* worker_queue) = 0; |
// Same as above but uses an existing file handle. Takes ownership |
// of |handle| and closes it at StopDebugRecording(). |
- virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0; |
+ virtual int StartDebugRecording(FILE* handle, |
+ int64_t max_log_size_bytes, |
+ rtc::TaskQueue* worker_queue) = 0; |
// TODO(ivoc): Remove this function after Chrome stops using it. |
- virtual int StartDebugRecording(FILE* handle) = 0; |
+ virtual int StartDebugRecording(FILE* handle, |
+ rtc::TaskQueue* worker_queue) = 0; |
// Same as above but uses an existing PlatformFile handle. Takes ownership |
// of |handle| and closes it at StopDebugRecording(). |
// TODO(xians): Make this interface pure virtual. |
- virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) = 0; |
+ virtual int StartDebugRecordingForPlatformFile( |
+ rtc::PlatformFile handle, |
+ rtc::TaskQueue* worker_queue) = 0; |
// Stops recording debugging information, and closes the file. Recording |
// cannot be resumed in the same file (without overwriting it). |