Allow audio debug recordings through a private API extension.

chrome/common/extensions/api/webrtc_logging_private.idl

 // 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.
 
 // Use the <code>chrome.webrtcLoggingPrivate</code> API to control diagnostic
 // WebRTC logging.
 namespace webrtcLoggingPrivate {
   dictionary MetaDataEntry {
     // The meta data entry key.
     DOMString key;
@@ skipping 10 matching lines @@
   dictionary RequestInfo {
     // The tab identifier from the chrome.tabs API, if the request is from a
     // tab.
     long? tabId;
 
     // The guest process id for the requester, if the request is from a
     // webview.
     long? guestProcessId;
   };
 
+  // This contains information about the result of audio debug recordings.
+  dictionary AudioDebugRecordingsInfo {
+    // Absolute path prefix for the files with the audio debug recordings.
+    DOMString prefixPath;
+
+    // Indicates if recording was stopped.
+    boolean didStop;
+
+    // Indicates if recording was stopped manually through a
+    // stopAudioDebugRecordings() call.
+    boolean didManualStop;
+  };
+
+
   callback GenericDoneCallback = void ();
+  callback AudioDebugRecordingsCallback =
+      void (AudioDebugRecordingsInfo info);
   callback UploadDoneCallback = void (UploadResult result);
 
   interface Functions {
     // For all functions, |request| determines which render process to apply
     // the operation on. |request| identifies the requesting process.
     // |securityOrigin| is the security origin for the tab identified by |tabId|
     // and is used for verifying that the tab is the correct one and has not
     // been navigated away from.
 
     // Sets additional custom meta data that will be uploaded along with the
@@ skipping 62 matching lines @@
                              GenericDoneCallback callback);
 
     // Stops RTP dumping. After stop has finished, the dumps will be
     // uploaded with the log if upload is called. Otherwise, the dumps will be
     // discarded.
     static void stopRtpDump(RequestInfo request,
                             DOMString securityOrigin,
                             boolean incoming,
                             boolean outgoing,
                             GenericDoneCallback callback);
+
+    // Starts audio debug recordings.
+    // |seconds| indicates how many seconds of audio to record. |callback|
+    // is invoked once recording stops.
+    // If |seconds| is zero, recording will continue until
+    // stopAudioDebugRecordings() is explicitly called. In this case,
+    // |callback| is invoked once recording starts and will report
+    // that recording has not stopped.
+    // If |seconds| is negative, startAudioDebugRecordings() fails.
+    static void startAudioDebugRecordings(RequestInfo request,
+                             DOMString securityOrigin,
+                             long seconds,
+                             AudioDebugRecordingsCallback callback);
+
+    // Stops audio debug recordings.  |callback| is invoked once recording
+    // stops. If there is no recording in progress, stopAudioDebugRecordings()
+    // fails.
+    static void stopAudioDebugRecordings(RequestInfo request,
+                            DOMString securityOrigin,
+                            AudioDebugRecordingsCallback callback);
   };
 };