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 bf29bf36332003bb62bbae0cededa3a500fe1db0..2c201f9299df8063ee8d7029743bdb3b05fb6f4a 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.h |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h |
@@ -102,8 +102,9 @@ class AudioProcessingImpl : public AudioProcessing { |
void set_delay_offset_ms(int offset) override; |
int delay_offset_ms() const override; |
void set_stream_key_pressed(bool key_pressed) override; |
- int StartDebugRecording(const char filename[kMaxFilenameSize]) override; |
- int StartDebugRecording(FILE* handle) 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 StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override; |
int StopDebugRecording() override; |
void UpdateHistogramsOnCallEnd() override; |
@@ -174,6 +175,10 @@ class AudioProcessingImpl : public AudioProcessing { |
// Serialized string of last saved APM configuration. |
std::string last_serialized_config_; |
+ |
+ // 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 nr_bytes_left_for_log; |
hlundin-webrtc
2015/11/11 15:55:05
Trailing underscore on member variable names.
ivoc
2015/11/11 16:44:31
Right, fixed.
|
#endif |
// Format of processing streams at input/output call sites. |