| 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 72faa26c422cae91039bab89a65660abf6416388..c4d5f084037d89a9523d94308aff4448649321fd 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;
|
|
|
| @@ -143,9 +142,6 @@
|
|
|
| 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;
|
| @@ -220,7 +216,6 @@
|
| // 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_);
|
|
|