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/download/download_prefs.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");

[Marijn Kruisselbrink, 2015/12/18 23:40:10]

It seems really weird to "asynchronously" return (by returning true in the next
line), but really call SendResponse before returning from RunAsync. But it seems
like the rest of the code does the same, so I assume it works correctly.
(If this would be using UIThreadExtensionFunction instead of the deprecated
AsyncExtensionFunction it definitely would not be okay though, but I guess that
doesn't matter)

[Guido Urdaneta, 2015/12/19 00:32:42]

I'll leave it as it is for consistency with the rest of the code.
However, how should it be done if I were using the more modern
UIThreadExtensionFunction?

[Marijn Kruisselbrink, 2015/12/19 00:43:37]

In UIThreadExtensionFunction::Run you'd return RespondNow(Error(...)) to
synchronously fail, or return RespondLater() and after returning from Run call
Respond() with the actual response.

+    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