Fix webrtcAudioPrivate API to handle requests from <webview>s.

BUG=424762

Review URL: https://codereview.chromium.org/880393002

Cr-Commit-Position: refs/heads/master@{#314036}

chrome/browser/extensions/api/webrtc_audio_private/webrtc_audio_private_api.cc

@@ -23,12 +23,14 @@
 namespace extensions {
 
 using content::BrowserThread;
-using content::RenderViewHost;
+using content::RenderProcessHost;
 using media::AudioDeviceNames;
 using media::AudioManager;
 
 namespace wap = api::webrtc_audio_private;
 
+using api::webrtc_audio_private::RequestInfo;
+
 static base::LazyInstance<
     BrowserContextKeyedAPIFactory<WebrtcAudioPrivateEventService> > g_factory =
     LAZY_INSTANCE_INITIALIZER;
@@ -130,27 +132,38 @@ void WebrtcAudioPrivateFunction::OnOutputDeviceNames(
   NOTREACHED();
 }
 
-bool WebrtcAudioPrivateFunction::GetControllerList(int tab_id) {
-  content::WebContents* contents = NULL;
-  if (!ExtensionTabUtil::GetTabById(
-          tab_id, GetProfile(), true, NULL, NULL, &contents, NULL)) {
-    error_ = extensions::ErrorUtils::FormatErrorMessage(
-        extensions::tabs_constants::kTabNotFoundError,
-        base::IntToString(tab_id));
+bool WebrtcAudioPrivateFunction::GetControllerList(const RequestInfo& request) {
+  content::RenderProcessHost* rph = nullptr;
+
+  // If |guest_process_id| is defined, directly use this id to find the
+  // corresponding RenderProcessHost.
+  if (request.guest_process_id.get()) {
+    rph = content::RenderProcessHost::FromID(*request.guest_process_id.get());
+  } else if (request.tab_id.get()) {
+    int tab_id = *request.tab_id.get();
+    content::WebContents* contents = NULL;
+    if (!ExtensionTabUtil::GetTabById(tab_id, GetProfile(), true, NULL, NULL,
+                                      &contents, NULL)) {
+      error_ = extensions::ErrorUtils::FormatErrorMessage(
+          extensions::tabs_constants::kTabNotFoundError,
+          base::IntToString(tab_id));
+      return false;
+    }
+    rph = contents->GetRenderProcessHost();
+  } else {
     return false;
   }
 
-  RenderViewHost* rvh = contents->GetRenderViewHost();
-  if (!rvh)
+  if (!rph)
     return false;
 
-  rvh->GetAudioOutputControllers(base::Bind(
-      &WebrtcAudioPrivateFunction::OnControllerList, this));
+  rph->GetAudioOutputControllers(
+      base::Bind(&WebrtcAudioPrivateFunction::OnControllerList, this));
   return true;
 }
 
 void WebrtcAudioPrivateFunction::OnControllerList(
-    const content::RenderViewHost::AudioOutputControllerList& list) {
+    const content::RenderProcessHost::AudioOutputControllerList& list) {
   NOTREACHED();
 }
 
@@ -254,11 +267,11 @@ bool WebrtcAudioPrivateGetActiveSinkFunction::RunAsync() {
       wap::GetActiveSink::Params::Create(*args_));
   EXTENSION_FUNCTION_VALIDATE(params.get());
 
-  return GetControllerList(params->tab_id);
+  return GetControllerList(params->request);
 }
 
 void WebrtcAudioPrivateGetActiveSinkFunction::OnControllerList(
-    const RenderViewHost::AudioOutputControllerList& controllers) {
+    const RenderProcessHost::AudioOutputControllerList& controllers) {
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
 
   if (controllers.empty()) {
@@ -295,9 +308,8 @@ void WebrtcAudioPrivateGetActiveSinkFunction::OnHMACCalculated(
 }
 
 WebrtcAudioPrivateSetActiveSinkFunction::
-WebrtcAudioPrivateSetActiveSinkFunction()
-    : tab_id_(0),
-      num_remaining_sink_ids_(0) {
+    WebrtcAudioPrivateSetActiveSinkFunction()
+    : num_remaining_sink_ids_(0) {
 }
 
 WebrtcAudioPrivateSetActiveSinkFunction::
@@ -312,22 +324,40 @@ bool WebrtcAudioPrivateSetActiveSinkFunction::RunAsync() {
 
   InitResourceContext();
 
-  tab_id_ = params->tab_id;
+  if (params->request.guest_process_id.get()) {
+    request_info_.guest_process_id.reset(
+        new int(*params->request.guest_process_id.get()));
+  } else if (params->request.tab_id.get()) {
+    request_info_.tab_id.reset(new int(*params->request.tab_id.get()));
+  } else {
+    return false;
+  }
+
   sink_id_ = params->sink_id;
 
-  return GetControllerList(tab_id_);
+  return GetControllerList(request_info_);
 }
 
 void WebrtcAudioPrivateSetActiveSinkFunction::OnControllerList(
-    const RenderViewHost::AudioOutputControllerList& controllers) {
+    const RenderProcessHost::AudioOutputControllerList& controllers) {
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
 
+  std::string requested_process_type;
+  int requested_process_id;
+  if (request_info_.guest_process_id.get()) {
+    requested_process_type = "guestProcessId";
+    requested_process_id = *request_info_.guest_process_id.get();
+  } else {
+    requested_process_type = "tabId";
+    requested_process_id = *request_info_.tab_id.get();
+  }
+
   controllers_ = controllers;
   num_remaining_sink_ids_ = controllers_.size();
   if (num_remaining_sink_ids_ == 0) {
     error_ = extensions::ErrorUtils::FormatErrorMessage(
-        "No active stream for tab with id: *.",
-        base::IntToString(tab_id_));
+        "No active stream for " + requested_process_type + " *",
+        base::IntToString(requested_process_id));
     SendResponse(false);
   } else {
     // We need to get the output device names, and calculate the HMAC
@@ -359,7 +389,7 @@ void WebrtcAudioPrivateSetActiveSinkFunction::OnOutputDeviceNames(
       DVLOG(2) << "Found no matching raw sink ID for HMAC " << sink_id_;
   }
 
-  RenderViewHost::AudioOutputControllerList::const_iterator it =
+  RenderProcessHost::AudioOutputControllerList::const_iterator it =
       controllers_.begin();
   for (; it != controllers_.end(); ++it) {
     (*it)->SwitchOutputDevice(raw_sink_id, base::Bind(

chrome/browser/extensions/api/webrtc_audio_private/webrtc_audio_private_api.h

@@ -9,7 +9,7 @@
 #include "base/system_monitor/system_monitor.h"
 #include "chrome/browser/extensions/chrome_extension_function.h"
 #include "chrome/common/extensions/api/webrtc_audio_private.h"
-#include "content/public/browser/render_view_host.h"
+#include "content/public/browser/render_process_host.h"
 #include "extensions/browser/browser_context_keyed_api_factory.h"
 #include "media/audio/audio_device_name.h"
 #include "url/gurl.h"
@@ -68,11 +68,11 @@ class WebrtcAudioPrivateFunction : public ChromeAsyncExtensionFunction {
   // entire function should fail.
   //
   // Call from any thread. Callback will occur on originating thread.
-  bool GetControllerList(int tab_id);
+  bool GetControllerList(const api::webrtc_audio_private::RequestInfo& request);
 
   // Must override this if you call GetControllerList.
   virtual void OnControllerList(
-      const content::RenderViewHost::AudioOutputControllerList& list);
+      const content::RenderProcessHost::AudioOutputControllerList& list);
 
   // Calculates a single HMAC. Call from any thread. Calls back via
   // OnHMACCalculated on UI thread.
@@ -134,7 +134,7 @@ class WebrtcAudioPrivateGetActiveSinkFunction
 
   bool RunAsync() override;
   void OnControllerList(
-      const content::RenderViewHost::AudioOutputControllerList& controllers)
+      const content::RenderProcessHost::AudioOutputControllerList& controllers)
       override;
   void OnHMACCalculated(const std::string& hmac) override;
 };
@@ -153,18 +153,18 @@ class WebrtcAudioPrivateSetActiveSinkFunction
 
   bool RunAsync() override;
   void OnControllerList(
-      const content::RenderViewHost::AudioOutputControllerList& controllers)
+      const content::RenderProcessHost::AudioOutputControllerList& controllers)
       override;
   void OnOutputDeviceNames(
       scoped_ptr<media::AudioDeviceNames> device_names) override;
   void SwitchDone();
   void DoneOnUIThread();
 
-  int tab_id_;
+  api::webrtc_audio_private::RequestInfo request_info_;
   std::string sink_id_;
 
   // Filled in by OnControllerList.
-  content::RenderViewHost::AudioOutputControllerList controllers_;
+  content::RenderProcessHost::AudioOutputControllerList controllers_;
 
   // Number of sink IDs we are still waiting for. Can become greater
   // than 0 in OnControllerList, decreases on every OnSinkId call.

chrome/browser/extensions/api/webrtc_audio_private/webrtc_audio_private_browsertest.cc

@@ -32,7 +32,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 using base::JSONWriter;
-using content::RenderViewHost;
+using content::RenderProcessHost;
 using content::WebContents;
 using media::AudioDeviceNames;
 using media::AudioManager;
@@ -49,7 +49,7 @@ class AudioWaitingExtensionTest : public ExtensionApiTest {
     // or more AudioOutputController objects for our tab.
     bool audio_playing = false;
     for (size_t remaining_tries = 50; remaining_tries > 0; --remaining_tries) {
-      tab->GetRenderViewHost()->GetAudioOutputControllers(
+      tab->GetRenderProcessHost()->GetAudioOutputControllers(
           base::Bind(OnAudioControllers, &audio_playing));
       base::MessageLoop::current()->RunUntilIdle();
       if (audio_playing)
@@ -65,7 +65,7 @@ class AudioWaitingExtensionTest : public ExtensionApiTest {
   // Used by the test above to wait until audio is playing.
   static void OnAudioControllers(
       bool* audio_playing,
-      const RenderViewHost::AudioOutputControllerList& list) {
+      const RenderProcessHost::AudioOutputControllerList& list) {
     if (!list.empty())
       *audio_playing = true;
   }
@@ -84,9 +84,15 @@ class WebrtcAudioPrivateTest : public AudioWaitingExtensionTest {
   }
 
  protected:
+  void AppendTabIdToRequestInfo(base::ListValue* params, int tab_id) {
+    base::DictionaryValue* request_info = new base::DictionaryValue();
+    request_info->SetInteger("tabId", tab_id);
+    params->Append(request_info);
+  }
+
   std::string InvokeGetActiveSink(int tab_id) {
     base::ListValue parameters;
-    parameters.AppendInteger(tab_id);
+    AppendTabIdToRequestInfo(&parameters, tab_id);
     std::string parameter_string;
     JSONWriter::Write(&parameters, &parameter_string);
 
@@ -221,7 +227,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcAudioPrivateTest, GetActiveSinkNoMediaStream) {
   WebContents* tab = browser()->tab_strip_model()->GetActiveWebContents();
   int tab_id = ExtensionTabUtil::GetTabId(tab);
   base::ListValue parameters;
-  parameters.AppendInteger(tab_id);
+  AppendTabIdToRequestInfo(&parameters, tab_id);
   std::string parameter_string;
   JSONWriter::Write(&parameters, &parameter_string);
 
@@ -244,7 +250,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcAudioPrivateTest, SetActiveSinkNoMediaStream) {
   WebContents* tab = browser()->tab_strip_model()->GetActiveWebContents();
   int tab_id = ExtensionTabUtil::GetTabId(tab);
   base::ListValue parameters;
-  parameters.AppendInteger(tab_id);
+  AppendTabIdToRequestInfo(&parameters, tab_id);
   parameters.AppendString("no such id");
   std::string parameter_string;
   JSONWriter::Write(&parameters, &parameter_string);
@@ -255,7 +261,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcAudioPrivateTest, SetActiveSinkNoMediaStream) {
   std::string error(RunFunctionAndReturnError(function.get(),
                                               parameter_string,
                                               browser()));
-  EXPECT_EQ(base::StringPrintf("No active stream for tab with id: %d.", tab_id),
+  EXPECT_EQ(base::StringPrintf("No active stream for tabId %d", tab_id),
             error);
 }
 
@@ -290,7 +296,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcAudioPrivateTest, GetAndSetWithMediaStream) {
     dict->GetString("sinkId", &target_device);
 
     base::ListValue parameters;
-    parameters.AppendInteger(tab_id);
+    AppendTabIdToRequestInfo(&parameters, tab_id);
     parameters.AppendString(target_device);
     std::string parameter_string;
     JSONWriter::Write(&parameters, &parameter_string);

chrome/browser/resources/hangout_services/thunk.js

@@ -107,12 +107,12 @@ chrome.runtime.onMessageExternal.addListener(
           return true;
         } else if (method == 'getActiveSink') {
           chrome.webrtcAudioPrivate.getActiveSink(
-              sender.tab.id, doSendResponse);
+              requestInfo, doSendResponse);
           return true;
         } else if (method == 'setActiveSink') {
           var sinkId = message['sinkId'];
           chrome.webrtcAudioPrivate.setActiveSink(
-              sender.tab.id, sinkId, doSendResponse);
+              requestInfo, sinkId, doSendResponse);
           return true;
         } else if (method == 'getAssociatedSink') {
           var sourceId = message['sourceId'];

chrome/common/extensions/api/webrtc_audio_private.idl

@@ -4,7 +4,7 @@
 
 // The <code>chrome.webrtcAudioPrivate</code> API allows enumeration
 // of audio output (sink) devices as well as getting and setting the
-// active device for a given tab.
+// active device for a given requesting process.
 //
 // Note that device IDs as used in this API are opaque (i.e. they are
 // not the hardware identifier of the device) and while they are
@@ -41,16 +41,26 @@ namespace webrtcAudioPrivate {
   callback SinkIdCallback = void(DOMString sinkId);
   callback CompletionCallback = void();
 
+  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;
+  };
+
   interface Functions {
     // Retrieves a list of available audio sink devices.
     static void getSinks(GetSinksCallback callback);
 
-    // Retrieves the currently active audio sink for the given tab.
-    static void getActiveSink(long tabId,
+    // Retrieves the currently active audio sink for the given requesting
+    // process.
+    static void getActiveSink(RequestInfo request,
                               SinkIdCallback callback);
 
-    // Sets the active audio sink device for the specified tab.
-    static void setActiveSink(long tabId,
+    // Sets the active audio sink device for the specified requesting process.
+    static void setActiveSink(RequestInfo request,
                               DOMString sinkId,
                               optional CompletionCallback callback);
 
@@ -60,7 +70,7 @@ namespace webrtcAudioPrivate {
     // sink.
     //
     // The associated sink ID can be used as a sink ID for
-    // setActiveSink. It is valid irrespective of which tab you are
+    // setActiveSink. It is valid irrespective of which process you are
     // setting the active sink for.
     static void getAssociatedSink(DOMString securityOrigin,
                                   DOMString sourceIdInOrigin,

content/browser/renderer_host/media/audio_renderer_host.cc

@@ -146,14 +146,11 @@ AudioRendererHost::~AudioRendererHost() {
 }
 
 void AudioRendererHost::GetOutputControllers(
-    int render_view_id,
-    const RenderViewHost::GetAudioOutputControllersCallback& callback) const {
+    const RenderProcessHost::GetAudioOutputControllersCallback&
+        callback) const {
   BrowserThread::PostTaskAndReplyWithResult(
-      BrowserThread::IO,
-      FROM_HERE,
-      base::Bind(&AudioRendererHost::DoGetOutputControllers, this,
-                 render_view_id),
-      callback);
+      BrowserThread::IO, FROM_HERE,
+      base::Bind(&AudioRendererHost::DoGetOutputControllers, this), callback);
 }
 
 void AudioRendererHost::OnChannelClosing() {
@@ -281,17 +278,15 @@ void AudioRendererHost::DoNotifyStreamStateChanged(int stream_id,
   UpdateNumPlayingStreams(entry, is_playing);
 }
 
-RenderViewHost::AudioOutputControllerList
-AudioRendererHost::DoGetOutputControllers(int render_view_id) const {
+RenderProcessHost::AudioOutputControllerList
+AudioRendererHost::DoGetOutputControllers() const {
   DCHECK_CURRENTLY_ON(BrowserThread::IO);
 
-  RenderViewHost::AudioOutputControllerList controllers;
+  RenderProcessHost::AudioOutputControllerList controllers;
   for (AudioEntryMap::const_iterator it = audio_entries_.begin();
        it != audio_entries_.end();
        ++it) {
-    AudioEntry* entry = it->second;
-    if (entry->render_view_id() == render_view_id)
-      controllers.push_back(entry->controller());
+    controllers.push_back(it->second->controller());
   }
 
   return controllers;

content/browser/renderer_host/media/audio_renderer_host.h

@@ -48,7 +48,7 @@
 #include "content/common/content_export.h"
 #include "content/public/browser/browser_message_filter.h"
 #include "content/public/browser/browser_thread.h"
-#include "content/public/browser/render_view_host.h"
+#include "content/public/browser/render_process_host.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_logging.h"
 #include "media/audio/audio_output_controller.h"
@@ -77,8 +77,8 @@ class CONTENT_EXPORT AudioRendererHost : public BrowserMessageFilter {
 
   // Calls |callback| with the list of AudioOutputControllers for this object.
   void GetOutputControllers(
-      int render_view_id,
-      const RenderViewHost::GetAudioOutputControllersCallback& callback) const;
+      const RenderProcessHost::GetAudioOutputControllersCallback&
+          callback) const;
 
   // BrowserMessageFilter implementation.
   void OnChannelClosing() override;
@@ -144,8 +144,7 @@ class CONTENT_EXPORT AudioRendererHost : public BrowserMessageFilter {
   // Send playing/paused status to the renderer.
   void DoNotifyStreamStateChanged(int stream_id, bool is_playing);
 
-  RenderViewHost::AudioOutputControllerList DoGetOutputControllers(
-      int render_view_id) const;
+  RenderProcessHost::AudioOutputControllerList DoGetOutputControllers() const;
 
   // Send an error message to the renderer.
   void SendErrorMessage(int stream_id);

content/browser/renderer_host/render_process_host_impl.cc

@@ -2394,4 +2394,9 @@ void RenderProcessHostImpl::DecrementWorkerRefCount() {
     Cleanup();
 }
 
+void RenderProcessHostImpl::GetAudioOutputControllers(
+    const GetAudioOutputControllersCallback& callback) const {
+  audio_renderer_host()->GetOutputControllers(callback);
+}
+
 }  // namespace content

content/browser/renderer_host/render_process_host_impl.h

@@ -271,6 +271,9 @@ class CONTENT_EXPORT RenderProcessHostImpl
   // immediately after receiving response headers.
   void ResumeResponseDeferredAtStart(const GlobalRequestID& request_id);
 
+  void GetAudioOutputControllers(
+      const GetAudioOutputControllersCallback& callback) const override;
+
  protected:
   // A proxy for our IPC::Channel that lives on the IO thread (see
   // browser_process.h)

content/browser/renderer_host/render_view_host_impl.cc

@@ -1360,13 +1360,6 @@ void RenderViewHostImpl::OnWebkitPreferencesChanged() {
   updating_web_preferences_ = false;
 }
 
-void RenderViewHostImpl::GetAudioOutputControllers(
-    const GetAudioOutputControllersCallback& callback) const {
-  scoped_refptr<AudioRendererHost> audio_host =
-      static_cast<RenderProcessHostImpl*>(GetProcess())->audio_renderer_host();
-  audio_host->GetOutputControllers(GetRoutingID(), callback);
-}
-
 void RenderViewHostImpl::ClearFocusedElement() {
   is_focused_element_editable_ = false;
   Send(new ViewMsg_ClearFocusedElement(GetRoutingID()));

content/browser/renderer_host/render_view_host_impl.h

@@ -168,8 +168,6 @@ class CONTENT_EXPORT RenderViewHostImpl
   WebPreferences GetWebkitPreferences() override;
   void UpdateWebkitPreferences(const WebPreferences& prefs) override;
   void OnWebkitPreferencesChanged() override;
-  void GetAudioOutputControllers(
-      const GetAudioOutputControllersCallback& callback) const override;
   void SelectWordAroundCaret() override;
 
 #if defined(OS_ANDROID)

content/public/browser/render_process_host.h

@@ -5,6 +5,8 @@
 #ifndef CONTENT_PUBLIC_BROWSER_RENDER_PROCESS_HOST_H_
 #define CONTENT_PUBLIC_BROWSER_RENDER_PROCESS_HOST_H_
 
+#include <list>
+
 #include "base/basictypes.h"
 #include "base/id_map.h"
 #include "base/process/kill.h"
@@ -26,6 +28,10 @@ namespace gpu {
 union ValueState;
 }
 
+namespace media {
+class AudioOutputController;
+}
+
 namespace content {
 class BrowserContext;
 class BrowserMessageFilter;
@@ -255,6 +261,16 @@ class CONTENT_EXPORT RenderProcessHost : public IPC::Sender,
   virtual void SendUpdateValueState(
       unsigned int target, const gpu::ValueState& state) = 0;
 
+  // Retrieves the list of AudioOutputController objects associated
+  // with this object and passes it to the callback you specify, on
+  // the same thread on which you called the method.
+  typedef std::list<scoped_refptr<media::AudioOutputController>>
+      AudioOutputControllerList;
+  typedef base::Callback<void(const AudioOutputControllerList&)>
+      GetAudioOutputControllersCallback;
+  virtual void GetAudioOutputControllers(
+      const GetAudioOutputControllersCallback& callback) const = 0;
+
   // Static management functions -----------------------------------------------
 
   // Flag to run the renderer in process.  This is primarily

content/public/browser/render_view_host.h

@@ -5,8 +5,6 @@
 #ifndef CONTENT_PUBLIC_BROWSER_RENDER_VIEW_HOST_H_
 #define CONTENT_PUBLIC_BROWSER_RENDER_VIEW_HOST_H_
 
-#include <list>
-
 #include "base/callback_forward.h"
 #include "content/common/content_export.h"
 #include "content/public/browser/render_widget_host.h"
@@ -31,10 +29,6 @@ namespace gfx {
 class Point;
 }
 
-namespace media {
-class AudioOutputController;
-}
-
 namespace content {
 
 class ChildProcessSecurityPolicy;
@@ -199,16 +193,6 @@ class CONTENT_EXPORT RenderViewHost : virtual public RenderWidgetHost {
   // Passes a list of Webkit preferences to the renderer.
   virtual void UpdateWebkitPreferences(const WebPreferences& prefs) = 0;
 
-  // Retrieves the list of AudioOutputController objects associated
-  // with this object and passes it to the callback you specify, on
-  // the same thread on which you called the method.
-  typedef std::list<scoped_refptr<media::AudioOutputController> >
-      AudioOutputControllerList;
-  typedef base::Callback<void(const AudioOutputControllerList&)>
-      GetAudioOutputControllersCallback;
-  virtual void GetAudioOutputControllers(
-      const GetAudioOutputControllersCallback& callback) const = 0;
-
   // Notify the render view host to select the word around the caret.
   virtual void SelectWordAroundCaret() = 0;
 

content/public/test/mock_render_process_host.h

@@ -112,6 +112,9 @@ class MockRenderProcessHost : public RenderProcessHost {
     process_handle = new_handle.Pass();
   }
 
+  void GetAudioOutputControllers(
+      const GetAudioOutputControllersCallback& callback) const override {}
+
  private:
   // Stores IPC messages that would have been sent to the renderer.
   IPC::TestSink sink_;