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Unified Diff: chrome/browser/media/webrtc_logging_handler_host.cc

Issue 1650133002: Start and stop RTC event logs from private extension API. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 11 months ago
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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);
+}

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