Start and stop RTC event logs from private extension API.

chrome/browser/media/webrtc_internal_log_handler_host.h

Index: chrome/browser/media/webrtc_internal_log_handler_host.h
diff --git a/chrome/browser/media/webrtc_internal_log_handler_host.h b/chrome/browser/media/webrtc_internal_log_handler_host.h
new file mode 100644
index 0000000000000000000000000000000000000000..584fe73564fd3b4ac56399aeb1e249a8b40a9bed
--- /dev/null
+++ b/chrome/browser/media/webrtc_internal_log_handler_host.h
@@ -0,0 +1,141 @@
+// Copyright 2013 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#ifndef CHROME_BROWSER_MEDIA_WEBRTC_INTERNAL_LOG_HANDLER_HOST_H_
+#define CHROME_BROWSER_MEDIA_WEBRTC_INTERNAL_LOG_HANDLER_HOST_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "base/macros.h"
+#include "build/build_config.h"
+#include "content/public/browser/browser_message_filter.h"
+#include "content/public/browser/render_process_host.h"
+#include "net/base/network_interfaces.h"
+
+class Profile;
+
+// WebRtcInternalLogHandlerHost provides an interface to the internal logs
+// in WebRTC:
+// - Starts and stops AudioDebugRecordings, aka AEC dumps.
+// - Starts and stops RTC event logs.
+class WebRtcInternalLogHandlerHost

[Henrik Grunell, 2016/02/17 12:33:02]

I'd prefer separate classes for the audio recordings and the rtc event log,
named say AudioDebugRecordingsHandler and RtcEventLogHandler. Is there a lot of
boilerplate code?

[terelius-chromium, 2016/02/22 09:28:55]

Done. Please take a look.

+    : public base::RefCountedThreadSafe<WebRtcInternalLogHandlerHost> {
+ public:
+  typedef base::Callback<void(bool, const std::string&)> GenericDoneCallback;
+  typedef base::Callback<void(const std::string&)>
+      TimeLimitedRecordingErrorCallback;
+  typedef base::Callback<void(const std::string&, bool, bool)>
+      TimeLimitedRecordingCallback;
+
+  // Key used to attach the handler to the RenderProcessHost
+  static const char kWebRtcInternalLogHandlerHostKey[];
+
+  explicit WebRtcInternalLogHandlerHost(Profile* profile);
+
+  // Starts an audio debug recording. The recording lasts the given |delay|,
+  // unless |delay| is zero, in which case recording will continue until
+  // StopAudioDebugRecordings() 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 StartAudioDebugRecordings(
+      content::RenderProcessHost* host,
+      base::TimeDelta delay,
+      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 TimeLimitedRecordingCallback& callback,
+      const TimeLimitedRecordingErrorCallback& error_callback);
+
+  // Starts an RTC event log. The call writes the most recent events to a
+  // 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:
+  friend class content::BrowserThread;
+  friend class base::DeleteHelper<WebRtcInternalLogHandlerHost>;
+  friend class base::RefCountedThreadSafe<WebRtcInternalLogHandlerHost>;
+
+  virtual ~WebRtcInternalLogHandlerHost();
+
+  base::FilePath GetLogDirectoryAndEnsureExists();
+
+  // Helper for starting audio debug recordings.
+  void DoStartAudioDebugRecordings(
+      content::RenderProcessHost* host,
+      base::TimeDelta delay,
+      const TimeLimitedRecordingCallback& callback,
+      const TimeLimitedRecordingErrorCallback& error_callback,
+      const base::FilePath& log_directory);
+
+  // Helper for stopping audio debug recordings.
+  void DoStopAudioDebugRecordings(
+      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);
+
+  // 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);
+
+  // The profile associated with our renderer process.
+  Profile* const profile_;
+
+  // Must be accessed on the UI thread.
+  bool is_audio_debug_recordings_in_progress_;
+
+  // 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(WebRtcInternalLogHandlerHost);
+};
+
+#endif  // CHROME_BROWSER_MEDIA_WEBRTC_INTERNAL_LOG_HANDLER_HOST_H_