Allow audio debug recordings through a private API extension.

chrome/browser/extensions/api/webrtc_logging_private/webrtc_logging_private_api.cc

 // 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.
 
 #include "chrome/browser/extensions/api/webrtc_logging_private/webrtc_logging_private_api.h"
 
+#include "base/command_line.h"
 #include "base/hash.h"
 #include "base/logging.h"
 #include "base/strings/string_number_conversions.h"
 #include "base/supports_user_data.h"
 #include "chrome/browser/extensions/api/tabs/tabs_constants.h"
 #include "chrome/browser/extensions/extension_tab_util.h"
 #include "chrome/browser/profiles/profile.h"
+#include "chrome/common/chrome_switches.h"
 #include "content/public/browser/browser_thread.h"
 #include "content/public/browser/render_process_host.h"
 #include "content/public/browser/site_instance.h"
 #include "content/public/browser/web_contents.h"
 #include "extensions/browser/guest_view/web_view/web_view_guest.h"
 #include "extensions/browser/process_manager.h"
 #include "extensions/common/error_utils.h"
 
 namespace extensions {
 
 using api::webrtc_logging_private::MetaDataEntry;
 using api::webrtc_logging_private::RequestInfo;
 using content::BrowserThread;
 
 namespace Discard = api::webrtc_logging_private::Discard;
 namespace SetMetaData = api::webrtc_logging_private::SetMetaData;
 namespace SetUploadOnRenderClose =
     api::webrtc_logging_private::SetUploadOnRenderClose;
 namespace Start = api::webrtc_logging_private::Start;
 namespace StartRtpDump = api::webrtc_logging_private::StartRtpDump;
 namespace Stop = api::webrtc_logging_private::Stop;
 namespace StopRtpDump = api::webrtc_logging_private::StopRtpDump;
 namespace Store = api::webrtc_logging_private::Store;
 namespace Upload = api::webrtc_logging_private::Upload;
 namespace UploadStored = api::webrtc_logging_private::UploadStored;
+namespace StartAudioDebugRecordings =
+    api::webrtc_logging_private::StartAudioDebugRecordings;
+namespace StopAudioDebugRecordings =
+    api::webrtc_logging_private::StopAudioDebugRecordings;
 
 namespace {
 std::string HashIdWithOrigin(const std::string& security_origin,
                              const std::string& log_id) {
   return base::UintToString(base::Hash(security_origin + log_id));
 }
 }  // namespace
 
 content::RenderProcessHost* WebrtcLoggingPrivateFunction::RphFromRequest(
     const RequestInfo& request, const std::string& security_origin) {
@@ skipping 68 matching lines @@
   if (success) {
     api::webrtc_logging_private::UploadResult result;
     result.report_id = report_id;
     SetResult(result.ToValue().release());
   } else {
     SetError(error_message);
   }
   SendResponse(success);
 }
 
+void WebrtcLoggingPrivateFunctionWithAudioDebugRecordingsCallback::
+    FireErrorCallback(const std::string& error_message) {
+  DCHECK_CURRENTLY_ON(content::BrowserThread::UI);
+  SetError(error_message);
+  SendResponse(false);
+}
+
+void WebrtcLoggingPrivateFunctionWithAudioDebugRecordingsCallback::FireCallback(
+    const std::string& prefix_path,
+    bool did_stop,
+    bool did_manual_stop) {
+  DCHECK_CURRENTLY_ON(content::BrowserThread::UI);
+  api::webrtc_logging_private::AudioDebugRecordingsInfo result;
+  result.prefix_path = prefix_path;
+  result.did_stop = did_stop;
+  result.did_manual_stop = did_manual_stop;
+  SetResult(result.ToValue().release());
+  SendResponse(true);
+}
+
 bool WebrtcLoggingPrivateSetMetaDataFunction::RunAsync() {
   scoped_ptr<SetMetaData::Params> params(SetMetaData::Params::Create(*args_));
   EXTENSION_FUNCTION_VALIDATE(params.get());
 
   WebRtcLoggingHandlerHost::GenericDoneCallback callback;
   scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host =
       PrepareTask(params->request, params->security_origin, &callback);
   if (!webrtc_logging_handler_host.get())
     return false;
 
@@ skipping 204 matching lines @@
 
   BrowserThread::PostTask(BrowserThread::IO,
                           FROM_HERE,
                           base::Bind(&WebRtcLoggingHandlerHost::StopRtpDump,
                                      webrtc_logging_handler_host,
                                      type,
                                      callback));
   return true;
 }
 
+bool WebrtcLoggingPrivateStartAudioDebugRecordingsFunction::RunAsync() {
+  if (!base::CommandLine::ForCurrentProcess()->HasSwitch(
+          switches::kEnableAudioDebugRecordingsFromExtension)) {
+    return false;
+  }
+
+  scoped_ptr<StartAudioDebugRecordings::Params> params(
+      StartAudioDebugRecordings::Params::Create(*args_));
+  EXTENSION_FUNCTION_VALIDATE(params.get());
+
+  if (params->seconds < 0) {
+    FireErrorCallback("seconds must be greater than or equal to 0");
+    return true;
+  }
+
+  content::RenderProcessHost* host =
+      RphFromRequest(params->request, params->security_origin);
+  if (!host)
+    return false;
+
+  scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host(
+      base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host));
+
+  webrtc_logging_handler_host->StartAudioDebugRecordings(
+      host, base::TimeDelta::FromSeconds(params->seconds),
+      base::Bind(
+          &WebrtcLoggingPrivateStartAudioDebugRecordingsFunction::FireCallback,
+          this),
+      base::Bind(&WebrtcLoggingPrivateStartAudioDebugRecordingsFunction::
+                     FireErrorCallback,
+                 this));
+  return true;
+}
+
+bool WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::RunAsync() {
+  if (!base::CommandLine::ForCurrentProcess()->HasSwitch(
+          switches::kEnableAudioDebugRecordingsFromExtension)) {
+    return false;
+  }
+
+  scoped_ptr<StopAudioDebugRecordings::Params> params(
+      StopAudioDebugRecordings::Params::Create(*args_));
+  EXTENSION_FUNCTION_VALIDATE(params.get());
+
+  content::RenderProcessHost* host =
+      RphFromRequest(params->request, params->security_origin);
+  if (!host)
+    return false;
+
+  scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host(
+      base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host));
+
+  webrtc_logging_handler_host->StopAudioDebugRecordings(
+      host,
+      base::Bind(
+          &WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::FireCallback,
+          this),
+      base::Bind(&WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::
+                     FireErrorCallback,
+                 this));
+  return true;
+}
+
 }  // namespace extensions