Allow audio debug recordings through a private API extension.

chrome/common/extensions/api/webrtc_logging_private.idl

Index: chrome/common/extensions/api/webrtc_logging_private.idl
diff --git a/chrome/common/extensions/api/webrtc_logging_private.idl b/chrome/common/extensions/api/webrtc_logging_private.idl
index 5c72c9597ba15e33f52a02f2644256eb8bcc9772..90f7561bb0a79289f55c41b317d20902a0127724 100644
--- a/chrome/common/extensions/api/webrtc_logging_private.idl
+++ b/chrome/common/extensions/api/webrtc_logging_private.idl
@@ -28,7 +28,27 @@ namespace webrtcLoggingPrivate {
     long? guestProcessId;
   };
 
+  // This contains information about the result of an AEC dump operation.
+  // didStop=false and isManualStop=false means startAecDump() was called

[Henrik Grunell, 2015/12/18 12:34:21]

Could these combinations be an enum(-ish variable) instead?

[Guido Urdaneta, 2015/12/18 18:38:58]

I tried it, but the problem is that the handler would need to know about this
enum, which did not look right.
However, I simplified the meanings of the boolean fields.

+  //     without a time limit.
+  // didStop=false and isManualStop=true means stopAecDump() was called, but no
+  //     AEC dump was in progress.
+  // didStop=true and isManualStop=false means an AEC Dump was stopped 
+  //     automatically due to hitting the time limit.
+  // didStop=true and isManualStop=true means an AEC Dump was stopped by a call
+  //     to stopAecDump().
+  dictionary AecDumpInfo {

[Henrik Grunell, 2015/12/18 12:34:21]

Rename "AecDump" to "AudioDebugRecordings" everywhere. AEC dump is only one part
that's being enabled.

[Guido Urdaneta, 2015/12/18 18:38:58]

Done.

+    // Indicates if an AEC dump was actually stopped.
+    boolean didStop;
+    // Indicates if a manual stop was attempted through a stopAecDump() call.
+    boolean isManualStop;
+    // Absolute path prefix for the AEC dump.
+    DOMString prefixPath;
+  };
+
+
   callback GenericDoneCallback = void ();
+  callback AecDumpCallback = void (AecDumpInfo info);
   callback UploadDoneCallback = void (UploadResult result);
 
   interface Functions {
@@ -111,5 +131,19 @@ namespace webrtcLoggingPrivate {
                             boolean incoming,
                             boolean outgoing,
                             GenericDoneCallback callback);
+
+    // Starts an AEC Dump for all renderer processes with open peer connections.

[Henrik Grunell, 2015/12/18 12:34:21]

The more general description is that there's one file per APM, which happens to
be one per render process right now. This will likely change in the future. See
the description in webrtc internals. (Maybe point to that here to avoid
duplication.)

[Guido Urdaneta, 2015/12/18 18:38:58]

Given that we are now recording for a single process, I decided to simplify the
documentation instead.

+    // One file per renderer process will be created, with the renderer
+    // process ID appended to a predefined file path.
+    // |seconds| indicates how many seconds of audio to record. If |seconds| is
+    // zero, recording will continue until stopAecDump() is explicitly called.
+    // If |seconds| is negative, startAecDump() will fail.
+    // If |seconds| is zero |callback| is invoked immediately after the AEC
+    // dump starts. Otherwise, |callback| is invoked when the AEC dump is 
+    // stopped.
+    static void startAecDump(long seconds, AecDumpCallback callback);
+
+    // Stops all AEC Dumps for all renderer processes.
+    static void stopAecDump(AecDumpCallback callback);
   };
 };