Index: chrome/browser/media/webrtc_logging_handler_host.cc |
diff --git a/chrome/browser/media/webrtc_logging_handler_host.cc b/chrome/browser/media/webrtc_logging_handler_host.cc |
index e0c49cc30877974447a3a42173ca2805b7e3520c..0cee5e8827240f5bdc9dce6dc3aae0080bd41546 100644 |
--- a/chrome/browser/media/webrtc_logging_handler_host.cc |
+++ b/chrome/browser/media/webrtc_logging_handler_host.cc |
@@ -116,6 +116,14 @@ base::FilePath GetAudioDebugRecordingsPrefixPath( |
base::Int64ToString(audio_debug_recordings_id)); |
} |
+// Returns a path name to be used as prefix for RTC event log files. |
+base::FilePath GetRtcEventLogPrefixPath(const base::FilePath& directory, |
+ uint64_t rtc_event_log_id) { |
+ static const char kRtcEventLogFilePrefix[] = "RtcEventLog."; |
+ return directory.AppendASCII(kRtcEventLogFilePrefix + |
+ base::Int64ToString(rtc_event_log_id)); |
+} |
+ |
} // namespace |
WebRtcLogBuffer::WebRtcLogBuffer() |
@@ -160,7 +168,9 @@ WebRtcLoggingHandlerHost::WebRtcLoggingHandlerHost( |
upload_log_on_render_close_(false), |
log_uploader_(log_uploader), |
is_audio_debug_recordings_in_progress_(false), |
- current_audio_debug_recordings_id_(0) { |
+ current_audio_debug_recordings_id_(0), |
+ is_rtc_event_logging_in_progress_(false), |
+ current_rtc_event_log_id_(0) { |
DCHECK(profile_); |
DCHECK(log_uploader_); |
} |
@@ -446,8 +456,8 @@ void WebRtcLoggingHandlerHost::DumpRtpPacketOnIOThread( |
void WebRtcLoggingHandlerHost::StartAudioDebugRecordings( |
content::RenderProcessHost* host, |
base::TimeDelta delay, |
- const AudioDebugRecordingsCallback& callback, |
- const AudioDebugRecordingsErrorCallback& error_callback) { |
+ const TimeLimitedRecordingCallback& callback, |
+ const TimeLimitedRecordingErrorCallback& error_callback) { |
DCHECK_CURRENTLY_ON(BrowserThread::UI); |
BrowserThread::PostTaskAndReplyWithResult( |
@@ -460,8 +470,8 @@ void WebRtcLoggingHandlerHost::StartAudioDebugRecordings( |
void WebRtcLoggingHandlerHost::StopAudioDebugRecordings( |
content::RenderProcessHost* host, |
- const AudioDebugRecordingsCallback& callback, |
- const AudioDebugRecordingsErrorCallback& error_callback) { |
+ const TimeLimitedRecordingCallback& callback, |
+ const TimeLimitedRecordingErrorCallback& error_callback) { |
DCHECK_CURRENTLY_ON(BrowserThread::UI); |
BrowserThread::PostTaskAndReplyWithResult( |
BrowserThread::FILE, FROM_HERE, |
@@ -472,6 +482,35 @@ void WebRtcLoggingHandlerHost::StopAudioDebugRecordings( |
current_audio_debug_recordings_id_, callback, error_callback)); |
} |
+void WebRtcLoggingHandlerHost::StartRtcEventLogging( |
+ content::RenderProcessHost* host, |
+ base::TimeDelta delay, |
+ const TimeLimitedRecordingCallback& callback, |
+ const TimeLimitedRecordingErrorCallback& error_callback) { |
+ DCHECK_CURRENTLY_ON(BrowserThread::UI); |
+ |
+ BrowserThread::PostTaskAndReplyWithResult( |
+ BrowserThread::FILE, FROM_HERE, |
+ base::Bind(&WebRtcLoggingHandlerHost::GetLogDirectoryAndEnsureExists, |
+ this), |
+ base::Bind(&WebRtcLoggingHandlerHost::DoStartRtcEventLogging, this, host, |
+ delay, callback, error_callback)); |
+} |
+ |
+void WebRtcLoggingHandlerHost::StopRtcEventLogging( |
+ content::RenderProcessHost* host, |
+ const TimeLimitedRecordingCallback& callback, |
+ const TimeLimitedRecordingErrorCallback& error_callback) { |
+ DCHECK_CURRENTLY_ON(BrowserThread::UI); |
+ BrowserThread::PostTaskAndReplyWithResult( |
+ BrowserThread::FILE, FROM_HERE, |
+ base::Bind(&WebRtcLoggingHandlerHost::GetLogDirectoryAndEnsureExists, |
+ this), |
+ base::Bind(&WebRtcLoggingHandlerHost::DoStopRtcEventLogging, this, host, |
+ true /* manual stop */, current_audio_debug_recordings_id_, |
+ callback, error_callback)); |
+} |
+ |
void WebRtcLoggingHandlerHost::OnChannelClosing() { |
DCHECK_CURRENTLY_ON(BrowserThread::IO); |
if (logging_state_ == STARTED || logging_state_ == STOPPED) { |
@@ -812,8 +851,8 @@ void WebRtcLoggingHandlerHost::FireGenericDoneCallback( |
void WebRtcLoggingHandlerHost::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) { |
DCHECK_CURRENTLY_ON(BrowserThread::UI); |
@@ -846,8 +885,8 @@ void WebRtcLoggingHandlerHost::DoStopAudioDebugRecordings( |
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) { |
DCHECK_CURRENTLY_ON(BrowserThread::UI); |
DCHECK_LE(audio_debug_recordings_id, current_audio_debug_recordings_id_); |
@@ -874,3 +913,68 @@ void WebRtcLoggingHandlerHost::DoStopAudioDebugRecordings( |
is_audio_debug_recordings_in_progress_ = false; |
callback.Run(prefix_path.AsUTF8Unsafe(), true /* stopped */, is_manual_stop); |
} |
+ |
+void WebRtcLoggingHandlerHost::DoStartRtcEventLogging( |
+ content::RenderProcessHost* host, |
+ base::TimeDelta delay, |
+ const TimeLimitedRecordingCallback& callback, |
+ const TimeLimitedRecordingErrorCallback& error_callback, |
+ const base::FilePath& log_directory) { |
+ DCHECK_CURRENTLY_ON(BrowserThread::UI); |
+ |
+ if (is_rtc_event_logging_in_progress_) { |
+ error_callback.Run("RTC event logging already in progress"); |
+ return; |
+ } |
+ |
+ is_rtc_event_logging_in_progress_ = true; |
+ base::FilePath prefix_path = |
+ GetRtcEventLogPrefixPath(log_directory, ++current_rtc_event_log_id_); |
+ host->EnableEventLogRecordings(prefix_path); |
+ |
+ if (delay.is_zero()) { |
+ callback.Run(prefix_path.AsUTF8Unsafe(), false /* not stopped */, |
+ false /* not manually stopped */); |
+ return; |
+ } |
+ |
+ BrowserThread::PostDelayedTask( |
+ BrowserThread::UI, FROM_HERE, |
+ base::Bind(&WebRtcLoggingHandlerHost::DoStopRtcEventLogging, this, host, |
+ false /* no manual stop */, current_rtc_event_log_id_, |
+ callback, error_callback, prefix_path), |
+ delay); |
+} |
+ |
+void WebRtcLoggingHandlerHost::DoStopRtcEventLogging( |
+ content::RenderProcessHost* host, |
+ bool is_manual_stop, |
+ uint64_t rtc_event_log_id, |
+ const TimeLimitedRecordingCallback& callback, |
+ const TimeLimitedRecordingErrorCallback& error_callback, |
+ const base::FilePath& log_directory) { |
+ DCHECK_CURRENTLY_ON(BrowserThread::UI); |
+ DCHECK_LE(rtc_event_log_id, current_rtc_event_log_id_); |
+ |
+ base::FilePath prefix_path = |
+ GetRtcEventLogPrefixPath(log_directory, rtc_event_log_id); |
+ // Prevent an old posted DoStopRtcEventLogging() call to stop a newer dump. |
+ // This could happen in a sequence like: |
+ // Start(10); //Start dump 1. Post Stop() to run after 10 seconds. |
+ // Stop(); // Manually stop dump 1 before 10 seconds; |
+ // Start(20); // Start dump 2. Posted Stop() for 1 should not stop dump 2. |
+ if (rtc_event_log_id < current_rtc_event_log_id_) { |
+ callback.Run(prefix_path.AsUTF8Unsafe(), false /* not stopped */, |
+ is_manual_stop); |
+ return; |
+ } |
+ |
+ if (!is_rtc_event_logging_in_progress_) { |
+ error_callback.Run("No RTC event logging in progress"); |
+ return; |
+ } |
+ |
+ host->DisableEventLogRecordings(); |
+ is_rtc_event_logging_in_progress_ = false; |
+ callback.Run(prefix_path.AsUTF8Unsafe(), true /* stopped */, is_manual_stop); |
+} |