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 b720bbf1bcd984546abe09bbec84cf1ae7b873ad..3506ac4dc0c5b071f7e7b0e6de0ff8c3d3807ff8 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.h |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h |
@@ -57,9 +57,8 @@ |
int Initialize(const ProcessingConfig& processing_config) override; |
void SetExtraOptions(const Config& config) override; |
void UpdateHistogramsOnCallEnd() 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; |
+ int StartDebugRecording(const char filename[kMaxFilenameSize]) override; |
+ int StartDebugRecording(FILE* handle) override; |
int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override; |
int StopDebugRecording() override; |
@@ -144,9 +143,6 @@ |
struct ApmDebugDumpState { |
ApmDebugDumpState() : debug_file(FileWrapper::Create()) {} |
- // Number of bytes that can still be written to the log before the maximum |
- // size is reached. A value of <= 0 indicates that no limit is used. |
- int64_t num_bytes_left_for_log_ = -1; |
rtc::scoped_ptr<FileWrapper> debug_file; |
ApmDebugDumpThreadState render; |
ApmDebugDumpThreadState capture; |
@@ -225,7 +221,6 @@ |
// TODO(andrew): make this more graceful. Ideally we would split this stuff |
// out into a separate class with an "enabled" and "disabled" implementation. |
static int WriteMessageToDebugFile(FileWrapper* debug_file, |
- int64_t* filesize_limit_bytes, |
rtc::CriticalSection* crit_debug, |
ApmDebugDumpThreadState* debug_state); |
int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |