Chromium Code Reviews| 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..0e3d1ad6a80439c0cc0f7a07933d702d1c9fe0d4 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_; |
|
the sun
2015/11/24 10:10:10
nit: nr_ -> num_
Andrew MacDonald
2015/11/24 17:28:59
+1
ivoc
2015/12/01 15:17:16
Done.
|
| #endif |
| // Format of processing streams at input/output call sites. |