Chromium Code Reviews| Index: chrome/browser/media/webrtc_logging_handler_host.h |
| diff --git a/chrome/browser/media/webrtc_logging_handler_host.h b/chrome/browser/media/webrtc_logging_handler_host.h |
| index 446d93c546fffd209f4422829c3fdf345f896c45..30d09cf0d05d55089eab94821257b206ac033482 100644 |
| --- a/chrome/browser/media/webrtc_logging_handler_host.h |
| +++ b/chrome/browser/media/webrtc_logging_handler_host.h |
| @@ -82,9 +82,9 @@ class WebRtcLoggingHandlerHost : public content::BrowserMessageFilter { |
| typedef base::Callback<void(bool, const std::string&, const std::string&)> |
| UploadDoneCallback; |
| typedef base::Callback<void(const std::string&)> |
| - AudioDebugRecordingsErrorCallback; |
| + TimeLimitedRecordingErrorCallback; |
| typedef base::Callback<void(const std::string&, bool, bool)> |
| - AudioDebugRecordingsCallback; |
| + TimeLimitedRecordingCallback; |
| WebRtcLoggingHandlerHost(Profile* profile, WebRtcLogUploader* log_uploader); |
| @@ -164,16 +164,38 @@ class WebRtcLoggingHandlerHost : public content::BrowserMessageFilter { |
| void StartAudioDebugRecordings( |
| content::RenderProcessHost* host, |
| base::TimeDelta delay, |
| - const AudioDebugRecordingsCallback& callback, |
| - const AudioDebugRecordingsErrorCallback& error_callback); |
| + const TimeLimitedRecordingCallback& callback, |
| + const TimeLimitedRecordingErrorCallback& error_callback); |
| // Stops an audio debug recording. |callback| is invoked once recording |
| // stops. If no recording was in progress, |error_callback| is invoked instead |
| // of |callback|. |
| void StopAudioDebugRecordings( |
| content::RenderProcessHost* host, |
| - const AudioDebugRecordingsCallback& callback, |
| - const AudioDebugRecordingsErrorCallback& error_callback); |
| + const TimeLimitedRecordingCallback& callback, |
| + const TimeLimitedRecordingErrorCallback& error_callback); |
| + |
| + // Starts an RTC event log. The call writes the most recent events to a |
|
Henrik Grunell
2016/02/01 13:58:13
This class was originally only for the WebRTC nati
terelius1
2016/02/16 20:09:12
Done.
|
| + // file and then starts logging events for the given |delay|. |
| + // If |delay| is zero, the logging will continue until StopRtcEventLogging() |
| + // is explicitly invoked. |
| + // |callback| is invoked once recording stops. If |delay| is zero |
| + // |callback| is invoked once recording starts. |
| + // If a recording was already in progress, |error_callback| is invoked instead |
| + // of |callback|. |
| + void StartRtcEventLogging( |
| + content::RenderProcessHost* host, |
| + base::TimeDelta delay, |
| + const TimeLimitedRecordingCallback& callback, |
| + const TimeLimitedRecordingErrorCallback& error_callback); |
| + |
| + // Stops an RTC event log. |callback| is invoked once recording |
| + // stops. If no recording was in progress, |error_callback| is invoked instead |
| + // of |callback|. |
| + void StopRtcEventLogging( |
| + content::RenderProcessHost* host, |
| + const TimeLimitedRecordingCallback& callback, |
| + const TimeLimitedRecordingErrorCallback& error_callback); |
| private: |
| // States used for protecting from function calls made at non-allowed points |
| @@ -265,8 +287,8 @@ class WebRtcLoggingHandlerHost : public content::BrowserMessageFilter { |
| void DoStartAudioDebugRecordings( |
| content::RenderProcessHost* host, |
| base::TimeDelta delay, |
| - const AudioDebugRecordingsCallback& callback, |
| - const AudioDebugRecordingsErrorCallback& error_callback, |
| + const TimeLimitedRecordingCallback& callback, |
| + const TimeLimitedRecordingErrorCallback& error_callback, |
| const base::FilePath& log_directory); |
| // Helper for stopping audio debug recordings. |
| @@ -274,8 +296,25 @@ class WebRtcLoggingHandlerHost : public content::BrowserMessageFilter { |
| content::RenderProcessHost* host, |
| bool is_manual_stop, |
| uint64_t audio_debug_recordings_id, |
| - const AudioDebugRecordingsCallback& callback, |
| - const AudioDebugRecordingsErrorCallback& error_callback, |
| + const TimeLimitedRecordingCallback& callback, |
| + const TimeLimitedRecordingErrorCallback& error_callback, |
| + const base::FilePath& log_directory); |
| + |
| + // Helper for starting RTC event logs. |
| + void DoStartRtcEventLogging( |
| + content::RenderProcessHost* host, |
| + base::TimeDelta delay, |
| + const TimeLimitedRecordingCallback& callback, |
| + const TimeLimitedRecordingErrorCallback& error_callback, |
| + const base::FilePath& log_directory); |
| + |
| + // Helper for stopping RTC event logs. |
| + void DoStopRtcEventLogging( |
| + content::RenderProcessHost* host, |
| + bool is_manual_stop, |
| + uint64_t audio_debug_recordings_id, |
| + const TimeLimitedRecordingCallback& callback, |
| + const TimeLimitedRecordingErrorCallback& error_callback, |
| const base::FilePath& log_directory); |
| scoped_ptr<WebRtcLogBuffer> log_buffer_; |
| @@ -325,6 +364,12 @@ class WebRtcLoggingHandlerHost : public content::BrowserMessageFilter { |
| // This counter allows saving each debug recording in separate files. |
| uint64_t current_audio_debug_recordings_id_; |
| + // Must be accessed on the UI thread. |
| + bool is_rtc_event_logging_in_progress_; |
| + |
| + // This counter allows saving each log in a separate file. |
| + uint64_t current_rtc_event_log_id_; |
| + |
| DISALLOW_COPY_AND_ASSIGN(WebRtcLoggingHandlerHost); |
| }; |