Cast Extensions API: Major namespace and object renaming

There is no functional change in this patch. This change only does
renaming.

Extensions API namespaces:
webrtc.castSendTransport -> cast.streaming.rtpStream
webrtc.castUdpTransport -> cast.streaming.udpTransport

Class renaming:
WebRtcNativeHandler -> CastStreamingNativeHandler
CastSendTransport -> CastRtpStream

Tests and related files are also renamed to get rid of the webrtc
label.

BUG=301920

Review URL: https://codereview.chromium.org/90083002

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@238403 0039d316-1c4b-4281-b951-d872f2087c98

25 files changed, 247 insertions, 263 deletions

diff --git a/chrome/browser/extensions/webrtc_cast_apitest.cc b/chrome/browser/extensions/cast_streaming_apitest.cc
index 3fee7f8..89d2d18c 100644
--- a/chrome/browser/extensions/webrtc_cast_apitest.cc
+++ b/chrome/browser/extensions/cast_streaming_apitest.cc
@@ -9,7 +9,7 @@
 
 namespace extensions {
 
-class WebrtcCastApiTest : public ExtensionApiTest {
+class CastStreamingApiTest : public ExtensionApiTest {
   virtual void SetUpCommandLine(CommandLine* command_line) OVERRIDE {
     ExtensionApiTest::SetUpCommandLine(command_line);
     command_line->AppendSwitchASCII(
@@ -34,8 +34,8 @@ class WebrtcCastApiTest : public ExtensionApiTest {
 };
 
 // Test running the test extension for Cast Mirroring API.
-IN_PROC_BROWSER_TEST_F(WebrtcCastApiTest, Basics) {
-  ASSERT_TRUE(RunExtensionSubtest("webrtc_cast", "basics.html"));
+IN_PROC_BROWSER_TEST_F(CastStreamingApiTest, Basics) {
+  ASSERT_TRUE(RunExtensionSubtest("cast_streaming", "basics.html"));
 }
 
 }  // namespace extensions
diff --git a/chrome/chrome_renderer.gypi b/chrome/chrome_renderer.gypi
index ce94a42..10b78fb 100644
--- a/chrome/chrome_renderer.gypi
+++ b/chrome/chrome_renderer.gypi
@@ -62,6 +62,8 @@
         'renderer/extensions/binding_generating_native_handler.h',
         'renderer/extensions/blob_native_handler.cc',
         'renderer/extensions/blob_native_handler.h',
+        'renderer/extensions/cast_streaming_native_handler.cc',
+        'renderer/extensions/cast_streaming_native_handler.h',
         'renderer/extensions/chrome_v8_context.cc',
         'renderer/extensions/chrome_v8_context.h',
         'renderer/extensions/chrome_v8_context_set.cc',
@@ -147,15 +149,13 @@
         'renderer/extensions/user_script_slave.h',
         'renderer/extensions/v8_schema_registry.cc',
         'renderer/extensions/v8_schema_registry.h',
-        'renderer/extensions/webrtc_native_handler.cc',
-        'renderer/extensions/webrtc_native_handler.h',
         'renderer/extensions/webstore_bindings.cc',
         'renderer/extensions/webstore_bindings.h',
         'renderer/isolated_world_ids.h',
         'renderer/loadtimes_extension_bindings.cc',
         'renderer/loadtimes_extension_bindings.h',
-        'renderer/media/cast_send_transport.cc',
-        'renderer/media/cast_send_transport.h',
+        'renderer/media/cast_rtp_stream.cc',
+        'renderer/media/cast_rtp_stream.h',
         'renderer/media/cast_session.cc',
         'renderer/media/cast_session.h',
         'renderer/media/cast_session_delegate.cc',
@@ -376,8 +376,8 @@
         }],
         ['enable_webrtc==0', {
           'sources!': [
-            'renderer/extensions/webrtc_native_handler.cc',
-            'renderer/extensions/webrtc_native_handler.h',
+            'renderer/extensions/cast_streaming_native_handler.cc',
+            'renderer/extensions/cast_streaming_native_handler.h',
             'renderer/media/chrome_webrtc_log_message_delegate.cc',
             'renderer/media/chrome_webrtc_log_message_delegate.h',
             'renderer/media/webrtc_logging_message_filter.cc',
diff --git a/chrome/chrome_tests.gypi b/chrome/chrome_tests.gypi
index 0b1774f..bb4a961 100644
--- a/chrome/chrome_tests.gypi
+++ b/chrome/chrome_tests.gypi
@@ -1179,6 +1179,7 @@
         'browser/extensions/browsertest_util.cc',
         'browser/extensions/browsertest_util.h',
         'browser/extensions/browsertest_util_browsertest.cc',
+        'browser/extensions/cast_streaming_apitest.cc',
         'browser/extensions/chrome_app_api_browsertest.cc',
         'browser/extensions/content_script_apitest.cc',
         'browser/extensions/content_security_policy_apitest.cc',
@@ -1242,7 +1243,6 @@
         'browser/extensions/test_extension_dir.cc',
         'browser/extensions/test_extension_dir.h',
         'browser/extensions/web_contents_browsertest.cc',
-        'browser/extensions/webrtc_cast_apitest.cc',
         'browser/extensions/webstore_inline_installer_browsertest.cc',
         'browser/extensions/webstore_installer_test.cc',
         'browser/extensions/webstore_installer_test.h',
diff --git a/chrome/common/extensions/api/_api_features.json b/chrome/common/extensions/api/_api_features.json
index e9a0d57..14c4263 100644
--- a/chrome/common/extensions/api/_api_features.json
+++ b/chrome/common/extensions/api/_api_features.json
@@ -128,10 +128,18 @@
     "dependencies": ["permission:cast"],
     "contexts": ["blessed_extension"]
   },
+  "cast.streaming.rtpStream": {
+    "dependencies": ["permission:cast.streaming"],
+    "contexts": ["blessed_extension"]
+  },
   "cast.streaming.session": {
     "dependencies": ["permission:cast.streaming"],
     "contexts": ["blessed_extension"]
   },
+  "cast.streaming.udpTransport": {
+    "dependencies": ["permission:cast.streaming"],
+    "contexts": ["blessed_extension"]
+  },
   "chromeosInfoPrivate": {
     "platforms": ["chromeos"],
     "dependencies": ["permission:chromeosInfoPrivate"],
@@ -717,14 +725,6 @@
     "dependencies": ["permission:webstorePrivate"],
     "contexts": ["blessed_extension"]
   },
-  "webrtc.castSendTransport": {
-    "dependencies": ["permission:webrtc"],
-    "contexts": ["blessed_extension"]
-  },
-  "webrtc.castUdpTransport": {
-    "dependencies": ["permission:webrtc"],
-    "contexts": ["blessed_extension"]
-  },
   "webview": {
     "internal": true,
     "dependencies": ["permission:webview"],
diff --git a/chrome/common/extensions/api/_permission_features.json b/chrome/common/extensions/api/_permission_features.json
index 175c9bb..0e5a49d 100644
--- a/chrome/common/extensions/api/_permission_features.json
+++ b/chrome/common/extensions/api/_permission_features.json
@@ -843,10 +843,6 @@
     "channel": "stable",
     "extension_types": ["extension", "legacy_packaged_app"]
   },
-  "webrtc": {
-    "channel": "dev",
-    "extension_types": ["extension"]
-  },
   "webrtcAudioPrivate": {
     "channel": "stable",
     "extension_types": ["extension"],
diff --git a/chrome/common/extensions/api/api.gyp b/chrome/common/extensions/api/api.gyp
index c5a56ec..a7c9eb4 100644
--- a/chrome/common/extensions/api/api.gyp
+++ b/chrome/common/extensions/api/api.gyp
@@ -166,9 +166,9 @@
           }],
           ['enable_webrtc==1', {
             'schema_files': [
+              'cast_streaming_rtp_stream.idl',
               'cast_streaming_session.idl',
-              'webrtc_cast_send_transport.idl',
-              'webrtc_cast_udp_transport.idl',
+              'cast_streaming_udp_transport.idl',
               'webrtc_logging_private.idl',
             ],
           }],
diff --git a/chrome/common/extensions/api/cast_streaming_rtp_stream.idl b/chrome/common/extensions/api/cast_streaming_rtp_stream.idl
new file mode 100644
index 0000000..55994d3
--- /dev/null
+++ b/chrome/common/extensions/api/cast_streaming_rtp_stream.idl
@@ -0,0 +1,96 @@
+// 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.
+
+// The <code>chrome.cast.streaming.rtpStream</code> API allows configuration
+// of encoding parameters and RTP parameters used in a Cast streaming
+// session.
+namespace cast.streaming.rtpStream {
+  // Params for audio and video codec.
+  dictionary CodecSpecificParams {
+    DOMString key;
+    DOMString value;
+  };
+
+  // RTP payload param.
+  dictionary RtpPayloadParams {
+    long payloadType;
+
+    DOMString codecName;
+
+    // Synchronization source identifier.
+    long? ssrc;
+
+    long? clockRate;
+
+    long? minBitrate;
+
+    long? maxBitrate;
+
+    // The number of channels.
+    long? channels;
+
+    // Video width in pixels.
+    long? width;
+
+    // Video height in pixels.
+    long? height;
+
+    // A list of codec specific params.
+    CodecSpecificParams[] codecSpecificParams;
+  };
+
+  // Cast RTP capabilities.
+  dictionary RtpCaps {
+    // RTP payload params.
+    RtpPayloadParams[] payloads;
+
+    DOMString[] rtcpFeatures;
+  };
+
+  // Cast RTP parameters.
+  dictionary RtpParams {
+    // RTP payload params.
+    RtpPayloadParams[] payloads;
+
+    DOMString[] rtcpFeatures;
+  };
+
+  // Callback from the <code>create</code> method.
+  // |id| : The ID for the RTP stream.
+  callback CreateCallback = void (long streamId);
+
+  interface Functions {
+    // Destroys a Cast RTP stream.
+    // |streamId| : The RTP stream ID.
+    [nocompile] static void destroy(long streamId);
+
+    // Returns capabilities of the RTP stream.
+    // |streamId| : The RTP stream ID.
+    [nocompile] static RtpCaps getCaps(long streamId);
+
+    // Activates the RTP stream by providing the parameters.
+    // |streamId| : The RTP stream ID.
+    // |params| : Parameters set for this stream.
+    [nocompile] static void start(long streamId, RtpParams params);
+
+    // Stops activity on the specified stream.
+    // |streamId| : The RTP stream ID.
+    [nocompile] static void stop(long streamId);
+  };
+
+  interface Events {
+    // Event fired when a Cast RTP stream has started.
+    // |streamId| : The ID of the RTP stream.
+    static void onStarted(long streamId);
+
+    // Event fired when a Cast RTP stream has stopped.
+    // |streamId| : The ID of the RTP stream.
+    static void onStopped(long streamId);
+
+    // Event fired when a Cast RTP stream has error.
+    // |streamId| : The ID of the RTP stream.
+    // |errorString| : The error info.
+    static void onError(long streamId, DOMString errorString);
+  };
+};
diff --git a/chrome/common/extensions/api/cast_streaming_session.idl b/chrome/common/extensions/api/cast_streaming_session.idl
index 5250c61..47408f0 100644
--- a/chrome/common/extensions/api/cast_streaming_session.idl
+++ b/chrome/common/extensions/api/cast_streaming_session.idl
@@ -7,15 +7,15 @@
 // by RTP streams and a network transport.
 //
 // Calling this API will generate corresponding resources for use with
-// chrome.webrtc.castSendTransport and chrome.webrtc.castUdpTransport
+// chrome.cast.streaming.rtpStream and chrome.cast.streaming.udpTransport
 // APIs.
 namespace cast.streaming.session {
   // Callback from the <code>create</code> method.
-  // |audioTransportId| : The audio transport ID.
-  // |videoTransportId| : The video transport ID.
+  // |audioStreamId| : The audio RTP stream ID.
+  // |videoStreamId| : The video RTP stream ID.
   // |udpTransportId| : The UDP transport ID.
-  callback CreateCallback = void (long audioTransportId,
-                                  long videoTransportId,
+  callback CreateCallback = void (long audioStreamId,
+                                  long videoStreamId,
                                   long udpTransportId);
 
   interface Functions {
diff --git a/chrome/common/extensions/api/webrtc_cast_udp_transport.idl b/chrome/common/extensions/api/cast_streaming_udp_transport.idl
index 26ad6e5..8c14060 100644
--- a/chrome/common/extensions/api/webrtc_cast_udp_transport.idl
+++ b/chrome/common/extensions/api/cast_streaming_udp_transport.idl
@@ -6,7 +6,7 @@
 // transport for Cast RTP streams. This API is not useful when standalone
 // since it does not have send and receive methods.
 // It is used to configure the UDP transport used in Cast session.
-namespace webrtc.castUdpTransport {
+namespace cast.streaming.udpTransport {
   // The UDP socket address and port.
   dictionary UdpParams {
     DOMString address;
diff --git a/chrome/common/extensions/api/webrtc_cast_send_transport.idl b/chrome/common/extensions/api/webrtc_cast_send_transport.idl
deleted file mode 100644
index e2f4fa3..0000000
--- a/chrome/common/extensions/api/webrtc_cast_send_transport.idl
+++ /dev/null
@@ -1,107 +0,0 @@
-// 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.
-
-// The <code>chrome.webrtc.castSendTransport</code> API takes a track as
-// a source of media, and sends that media on the inner transport according to
-// the given RtpParams.
-namespace webrtc.castSendTransport {
-  // Params for audio and video codec.
-  dictionary CodecSpecificParams {
-    DOMString key;
-    DOMString value;
-  };
-
-  // RTP payload param.
-  dictionary RtpPayloadParams {
-    long payloadType;
-
-    DOMString codecName;
-
-    // Synchronization source identifier.
-    long? ssrc;
-
-    long? clockRate;
-
-    long? minBitrate;
-
-    long? maxBitrate;
-
-    // The number of channels.
-    long? channels;
-
-    // Video width in pixels.
-    long? width;
-
-    // Video height in pixels.
-    long? height;
-
-    // A list of codec specific params.
-    CodecSpecificParams[] codecSpecificParams;
-  };
-
-  // Cast transport capabilities
-  dictionary RtpCaps {
-    // RTP payload params.
-    RtpPayloadParams[] payloads;
-
-    DOMString[] rtcpFeatures;
-  };
-
-  // Cast transport params.
-  dictionary RtpParams {
-    // RTP payload params.
-    RtpPayloadParams[] payloads;
-
-    DOMString[] rtcpFeatures;
-  };
-
-  // Callback from the <code>create</code> method.
-  // |id| : The transport id.
-  callback CreateCallback = void (long transportId);
-
-  interface Functions {
-    // Destroys a cast send transport.
-    // |transportId| : The transport ID.
-    [nocompile] static void destroy(long transportId);
-
-    // Returns capabilities of the transport.
-    // |transportId| : The transport ID.
-    [nocompile] static RtpCaps getCaps(long transportId);
-
-    // Starts to use the transport by providing remote params info.
-    // |transportId| : The transport ID.
-    // |params| : Parameters set for this transport.
-    [nocompile] static void start(long transportId, RtpParams params);
-
-    // Stops using the transport.
-    // |transportId| : The transport ID.
-    [nocompile] static void stop(long transportId);
-  };
-
-  interface Events {
-    // Event fired when a cast send transport has started.
-    // |transportId| : The ID of the transport.
-    static void onStarted(long transportId);
-
-    // Event fired when a cast send transport has connected.
-    // After this event, the transport is ready to send the track.
-    // |transportId| : The ID of the transport.
-    static void onConnected(long transportId);
-
-    // Event fired when a cast send transport has stopped.
-    // |transportId| : The ID of the transport.
-    static void onStopped(long transportId);
-
-    // Event fired when a cast send transport has timeout.
-    // This happens when network has been congested for a while, or one side
-    // left.
-    // |transportId| : The ID of the transport.
-    static void onTimeout(long transportId);
-
-    // Event fired when a cast send transport has error.
-    // |transportId| : The ID of the transport.
-    // |errorString| : The error info.
-    static void onError(long transportId, DOMString errorString);
-  };
-};
diff --git a/chrome/common/extensions/permissions/chrome_api_permissions.cc b/chrome/common/extensions/permissions/chrome_api_permissions.cc
index a962096..f82ed5e 100644
--- a/chrome/common/extensions/permissions/chrome_api_permissions.cc
+++ b/chrome/common/extensions/permissions/chrome_api_permissions.cc
@@ -355,7 +355,6 @@ std::vector<APIPermissionInfo*> ChromeAPIPermissions::GetAllPermissions()
     { APIPermission::kPointerLock, "pointerLock" },
     { APIPermission::kFullscreen, "fullscreen" },
     { APIPermission::kAudio, "audio" },
-    { APIPermission::kWebRtc, "webrtc" },
     { APIPermission::kCastStreaming, "cast.streaming" },
     { APIPermission::kOverrideEscFullscreen, "overrideEscFullscreen" },
 
diff --git a/chrome/common/extensions/permissions/permission_set_unittest.cc b/chrome/common/extensions/permissions/permission_set_unittest.cc
index 065310a..b3d4371 100644
--- a/chrome/common/extensions/permissions/permission_set_unittest.cc
+++ b/chrome/common/extensions/permissions/permission_set_unittest.cc
@@ -664,6 +664,7 @@ TEST(PermissionsTest, PermissionMessages) {
   skip.insert(APIPermission::kAppWindow);
   skip.insert(APIPermission::kAudio);
   skip.insert(APIPermission::kBrowsingData);
+  skip.insert(APIPermission::kCastStreaming);
   skip.insert(APIPermission::kContextMenus);
   skip.insert(APIPermission::kDiagnostics);
   skip.insert(APIPermission::kDns);
@@ -687,7 +688,6 @@ TEST(PermissionsTest, PermissionMessages) {
   skip.insert(APIPermission::kSystemStorage);
   skip.insert(APIPermission::kTts);
   skip.insert(APIPermission::kUnlimitedStorage);
-  skip.insert(APIPermission::kWebRtc);
   skip.insert(APIPermission::kWebView);
   skip.insert(APIPermission::kOverrideEscFullscreen);
 
diff --git a/chrome/renderer/extensions/webrtc_native_handler.cc b/chrome/renderer/extensions/cast_streaming_native_handler.cc
index ac3bf18..00316c7 100644
--- a/chrome/renderer/extensions/webrtc_native_handler.cc
+++ b/chrome/renderer/extensions/cast_streaming_native_handler.cc
@@ -2,16 +2,16 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "chrome/renderer/extensions/webrtc_native_handler.h"
+#include "chrome/renderer/extensions/cast_streaming_native_handler.h"
 
 #include <functional>
 
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
-#include "chrome/common/extensions/api/webrtc_cast_send_transport.h"
-#include "chrome/common/extensions/api/webrtc_cast_udp_transport.h"
+#include "chrome/common/extensions/api/cast_streaming_rtp_stream.h"
+#include "chrome/common/extensions/api/cast_streaming_udp_transport.h"
 #include "chrome/renderer/extensions/chrome_v8_context.h"
-#include "chrome/renderer/media/cast_send_transport.h"
+#include "chrome/renderer/media/cast_rtp_stream.h"
 #include "chrome/renderer/media/cast_session.h"
 #include "chrome/renderer/media/cast_udp_transport.h"
 #include "content/public/renderer/v8_value_converter.h"
@@ -22,16 +22,16 @@
 using content::V8ValueConverter;
 
 // Extension types.
-using extensions::api::webrtc_cast_send_transport::CodecSpecificParams;
-using extensions::api::webrtc_cast_send_transport::RtpCaps;
-using extensions::api::webrtc_cast_send_transport::RtpParams;
-using extensions::api::webrtc_cast_send_transport::RtpPayloadParams;
-using extensions::api::webrtc_cast_udp_transport::UdpParams;
+using extensions::api::cast_streaming_rtp_stream::CodecSpecificParams;
+using extensions::api::cast_streaming_rtp_stream::RtpCaps;
+using extensions::api::cast_streaming_rtp_stream::RtpParams;
+using extensions::api::cast_streaming_rtp_stream::RtpPayloadParams;
+using extensions::api::cast_streaming_udp_transport::UdpParams;
 
 namespace extensions {
 
 namespace {
-const char kSendTransportNotFound[] = "The send transport cannot be found";
+const char kRtpStreamNotFound[] = "The RTP stream cannot be found";
 const char kUdpTransportNotFound[] = "The UDP transport cannot be found";
 const char kInvalidUdpParams[] = "Invalid UDP params";
 const char kInvalidRtpParams[] = "Invalid value for RTP params";
@@ -122,37 +122,37 @@ void ToCastRtpParams(const RtpParams& ext_params, CastRtpParams* cast_params) {
 
 }  // namespace
 
-WebRtcNativeHandler::WebRtcNativeHandler(ChromeV8Context* context)
+CastStreamingNativeHandler::CastStreamingNativeHandler(ChromeV8Context* context)
     : ObjectBackedNativeHandler(context),
       last_transport_id_(0),
       weak_factory_(this) {
   RouteFunction("CreateSession",
-      base::Bind(&WebRtcNativeHandler::CreateCastSession,
+      base::Bind(&CastStreamingNativeHandler::CreateCastSession,
                  base::Unretained(this)));
-  RouteFunction("DestroyCastSendTransport",
-      base::Bind(&WebRtcNativeHandler::DestroyCastSendTransport,
+  RouteFunction("DestroyCastRtpStream",
+      base::Bind(&CastStreamingNativeHandler::DestroyCastRtpStream,
                  base::Unretained(this)));
-  RouteFunction("GetCapsCastSendTransport",
-      base::Bind(&WebRtcNativeHandler::GetCapsCastSendTransport,
+  RouteFunction("GetCapsCastRtpStream",
+      base::Bind(&CastStreamingNativeHandler::GetCapsCastRtpStream,
                  base::Unretained(this)));
-  RouteFunction("StartCastSendTransport",
-      base::Bind(&WebRtcNativeHandler::StartCastSendTransport,
+  RouteFunction("StartCastRtpStream",
+      base::Bind(&CastStreamingNativeHandler::StartCastRtpStream,
                  base::Unretained(this)));
-  RouteFunction("StopCastSendTransport",
-      base::Bind(&WebRtcNativeHandler::StopCastSendTransport,
+  RouteFunction("StopCastRtpStream",
+      base::Bind(&CastStreamingNativeHandler::StopCastRtpStream,
                  base::Unretained(this)));
   RouteFunction("DestroyCastUdpTransport",
-      base::Bind(&WebRtcNativeHandler::DestroyCastUdpTransport,
+      base::Bind(&CastStreamingNativeHandler::DestroyCastUdpTransport,
                  base::Unretained(this)));
   RouteFunction("StartCastUdpTransport",
-      base::Bind(&WebRtcNativeHandler::StartCastUdpTransport,
+      base::Bind(&CastStreamingNativeHandler::StartCastUdpTransport,
                  base::Unretained(this)));
 }
 
-WebRtcNativeHandler::~WebRtcNativeHandler() {
+CastStreamingNativeHandler::~CastStreamingNativeHandler() {
 }
 
-void WebRtcNativeHandler::CreateCastSession(
+void CastStreamingNativeHandler::CreateCastSession(
     const v8::FunctionCallbackInfo<v8::Value>& args) {
   CHECK_EQ(3, args.Length());
   CHECK(args[0]->IsObject());
@@ -169,10 +169,10 @@ void WebRtcNativeHandler::CreateCastSession(
     return;
 
   scoped_refptr<CastSession> session(new CastSession());
-  scoped_ptr<CastSendTransport> stream1(
-      new CastSendTransport(track1.component(), session));
-  scoped_ptr<CastSendTransport> stream2(
-      new CastSendTransport(track2.component(), session));
+  scoped_ptr<CastRtpStream> stream1(
+      new CastRtpStream(track1.component(), session));
+  scoped_ptr<CastRtpStream> stream2(
+      new CastRtpStream(track2.component(), session));
   scoped_ptr<CastUdpTransport> udp_transport(
       new CastUdpTransport(session));
 
@@ -181,26 +181,26 @@ void WebRtcNativeHandler::CreateCastSession(
   base::MessageLoop::current()->PostTask(
       FROM_HERE,
       base::Bind(
-          &WebRtcNativeHandler::CallCreateCallback,
+          &CastStreamingNativeHandler::CallCreateCallback,
           weak_factory_.GetWeakPtr(),
           base::Passed(&stream1),
           base::Passed(&stream2),
           base::Passed(&udp_transport)));
 }
 
-void WebRtcNativeHandler::CallCreateCallback(
-    scoped_ptr<CastSendTransport> stream1,
-    scoped_ptr<CastSendTransport> stream2,
+void CastStreamingNativeHandler::CallCreateCallback(
+    scoped_ptr<CastRtpStream> stream1,
+    scoped_ptr<CastRtpStream> stream2,
     scoped_ptr<CastUdpTransport> udp_transport) {
   v8::HandleScope handle_scope(context()->isolate());
   v8::Context::Scope context_scope(context()->v8_context());
 
   const int stream1_id = last_transport_id_++;
-  send_transport_map_[stream1_id] =
-      linked_ptr<CastSendTransport>(stream1.release());
+  rtp_stream_map_[stream1_id] =
+      linked_ptr<CastRtpStream>(stream1.release());
   const int stream2_id = last_transport_id_++;
-  send_transport_map_[stream2_id] =
-      linked_ptr<CastSendTransport>(stream2.release());
+  rtp_stream_map_[stream2_id] =
+      linked_ptr<CastRtpStream>(stream2.release());
   const int udp_id = last_transport_id_++;
   udp_transport_map_[udp_id] =
       linked_ptr<CastUdpTransport>(udp_transport.release());
@@ -214,24 +214,24 @@ void WebRtcNativeHandler::CallCreateCallback(
   create_callback_.reset();
 }
 
-void WebRtcNativeHandler::DestroyCastSendTransport(
+void CastStreamingNativeHandler::DestroyCastRtpStream(
     const v8::FunctionCallbackInfo<v8::Value>& args) {
   CHECK_EQ(1, args.Length());
   CHECK(args[0]->IsInt32());
 
   const int transport_id = args[0]->ToInt32()->Value();
-  if (!GetSendTransportOrThrow(transport_id))
+  if (!GetRtpStreamOrThrow(transport_id))
     return;
-  send_transport_map_.erase(transport_id);
+  rtp_stream_map_.erase(transport_id);
 }
 
-void WebRtcNativeHandler::GetCapsCastSendTransport(
+void CastStreamingNativeHandler::GetCapsCastRtpStream(
     const v8::FunctionCallbackInfo<v8::Value>& args) {
   CHECK_EQ(1, args.Length());
   CHECK(args[0]->IsInt32());
 
   const int transport_id = args[0]->ToInt32()->Value();
-  CastSendTransport* transport = GetSendTransportOrThrow(transport_id);
+  CastRtpStream* transport = GetRtpStreamOrThrow(transport_id);
   if (!transport)
     return;
 
@@ -245,14 +245,14 @@ void WebRtcNativeHandler::GetCapsCastSendTransport(
       caps_value.get(), context()->v8_context()));
 }
 
-void WebRtcNativeHandler::StartCastSendTransport(
+void CastStreamingNativeHandler::StartCastRtpStream(
     const v8::FunctionCallbackInfo<v8::Value>& args) {
   CHECK_EQ(2, args.Length());
   CHECK(args[0]->IsInt32());
   CHECK(args[1]->IsObject());
 
   const int transport_id = args[0]->ToInt32()->Value();
-  CastSendTransport* transport = GetSendTransportOrThrow(transport_id);
+  CastRtpStream* transport = GetRtpStreamOrThrow(transport_id);
   if (!transport)
     return;
 
@@ -276,19 +276,19 @@ void WebRtcNativeHandler::StartCastSendTransport(
   transport->Start(cast_params);
 }
 
-void WebRtcNativeHandler::StopCastSendTransport(
+void CastStreamingNativeHandler::StopCastRtpStream(
     const v8::FunctionCallbackInfo<v8::Value>& args) {
   CHECK_EQ(1, args.Length());
   CHECK(args[0]->IsInt32());
 
   const int transport_id = args[0]->ToInt32()->Value();
-  CastSendTransport* transport = GetSendTransportOrThrow(transport_id);
+  CastRtpStream* transport = GetRtpStreamOrThrow(transport_id);
   if (!transport)
     return;
   transport->Stop();
 }
 
-void WebRtcNativeHandler::DestroyCastUdpTransport(
+void CastStreamingNativeHandler::DestroyCastUdpTransport(
     const v8::FunctionCallbackInfo<v8::Value>& args) {
   CHECK_EQ(1, args.Length());
   CHECK(args[0]->IsInt32());
@@ -299,7 +299,7 @@ void WebRtcNativeHandler::DestroyCastUdpTransport(
   udp_transport_map_.erase(transport_id);
 }
 
-void WebRtcNativeHandler::StartCastUdpTransport(
+void CastStreamingNativeHandler::StartCastUdpTransport(
     const v8::FunctionCallbackInfo<v8::Value>& args) {
   CHECK_EQ(2, args.Length());
   CHECK(args[0]->IsInt32());
@@ -327,19 +327,19 @@ void WebRtcNativeHandler::StartCastUdpTransport(
   transport->Start(net::HostPortPair(udp_params->address, udp_params->port));
 }
 
-CastSendTransport* WebRtcNativeHandler::GetSendTransportOrThrow(
+CastRtpStream* CastStreamingNativeHandler::GetRtpStreamOrThrow(
     int transport_id) const {
-  SendTransportMap::const_iterator iter = send_transport_map_.find(
+  RtpStreamMap::const_iterator iter = rtp_stream_map_.find(
       transport_id);
-  if (iter != send_transport_map_.end())
+  if (iter != rtp_stream_map_.end())
     return iter->second.get();
   v8::Isolate* isolate = context()->v8_context()->GetIsolate();
-  isolate->ThrowException(v8::Exception::RangeError(
-      v8::String::NewFromUtf8(isolate, kSendTransportNotFound)));
+  isolate->ThrowException(v8::Exception::RangeError(v8::String::NewFromUtf8(
+      isolate, kRtpStreamNotFound)));
   return NULL;
 }
 
-CastUdpTransport* WebRtcNativeHandler::GetUdpTransportOrThrow(
+CastUdpTransport* CastStreamingNativeHandler::GetUdpTransportOrThrow(
     int transport_id) const {
   UdpTransportMap::const_iterator iter = udp_transport_map_.find(
       transport_id);
diff --git a/chrome/renderer/extensions/webrtc_native_handler.h b/chrome/renderer/extensions/cast_streaming_native_handler.h
index 99d20cc..4c2360c 100644
--- a/chrome/renderer/extensions/webrtc_native_handler.h
+++ b/chrome/renderer/extensions/cast_streaming_native_handler.h
@@ -2,8 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef CHROME_RENDERER_EXTENSIONS_WEBRTC_NATIVE_HANDLER_H_
-#define CHROME_RENDERER_EXTENSIONS_WEBRTC_NATIVE_HANDLER_H_
+#ifndef CHROME_RENDERER_EXTENSIONS_CAST_STREAMING_NATIVE_HANDLER_H_
+#define CHROME_RENDERER_EXTENSIONS_CAST_STREAMING_NATIVE_HANDLER_H_
 
 #include <map>
 
@@ -13,7 +13,7 @@
 #include "chrome/renderer/extensions/scoped_persistent.h"
 #include "v8/include/v8.h"
 
-class CastSendTransport;
+class CastRtpStream;
 class CastUdpTransport;
 
 namespace extensions {
@@ -21,24 +21,24 @@ namespace extensions {
 class ChromeV8Context;
 
 // Native code that handle chrome.webrtc custom bindings.
-class WebRtcNativeHandler : public ObjectBackedNativeHandler {
+class CastStreamingNativeHandler : public ObjectBackedNativeHandler {
  public:
-  explicit WebRtcNativeHandler(ChromeV8Context* context);
-  virtual ~WebRtcNativeHandler();
+  explicit CastStreamingNativeHandler(ChromeV8Context* context);
+  virtual ~CastStreamingNativeHandler();
 
  private:
   void CreateCastSession(
       const v8::FunctionCallbackInfo<v8::Value>& args);
 
-  void DestroyCastSendTransport(
+  void DestroyCastRtpStream(
       const v8::FunctionCallbackInfo<v8::Value>& args);
-  void CreateParamsCastSendTransport(
+  void CreateParamsCastRtpStream(
       const v8::FunctionCallbackInfo<v8::Value>& args);
-  void GetCapsCastSendTransport(
+  void GetCapsCastRtpStream(
       const v8::FunctionCallbackInfo<v8::Value>& args);
-  void StartCastSendTransport(
+  void StartCastRtpStream(
       const v8::FunctionCallbackInfo<v8::Value>& args);
-  void StopCastSendTransport(
+  void StopCastRtpStream(
       const v8::FunctionCallbackInfo<v8::Value>& args);
 
   void DestroyCastUdpTransport(
@@ -50,30 +50,30 @@ class WebRtcNativeHandler : public ObjectBackedNativeHandler {
 
   // Helper method to call the v8 callback function after a session is
   // created.
-  void CallCreateCallback(scoped_ptr<CastSendTransport> stream1,
-                          scoped_ptr<CastSendTransport> stream2,
+  void CallCreateCallback(scoped_ptr<CastRtpStream> stream1,
+                          scoped_ptr<CastRtpStream> stream2,
                           scoped_ptr<CastUdpTransport> udp_transport);
 
-  // Gets the Send or UDP transport indexed by |transport_id|.
+  // Gets the RTP stream or UDP transport indexed by an ID.
   // If not found, returns NULL and throws a V8 exception.
-  CastSendTransport* GetSendTransportOrThrow(int transport_id) const;
+  CastRtpStream* GetRtpStreamOrThrow(int stream_id) const;
   CastUdpTransport* GetUdpTransportOrThrow(int transport_id) const;
 
   int last_transport_id_;
 
-  typedef std::map<int, linked_ptr<CastSendTransport> > SendTransportMap;
-  SendTransportMap send_transport_map_;
+  typedef std::map<int, linked_ptr<CastRtpStream> > RtpStreamMap;
+  RtpStreamMap rtp_stream_map_;
 
   typedef std::map<int, linked_ptr<CastUdpTransport> > UdpTransportMap;
   UdpTransportMap udp_transport_map_;
 
-  base::WeakPtrFactory<WebRtcNativeHandler> weak_factory_;
+  base::WeakPtrFactory<CastStreamingNativeHandler> weak_factory_;
 
   extensions::ScopedPersistent<v8::Function> create_callback_;
 
-  DISALLOW_COPY_AND_ASSIGN(WebRtcNativeHandler);
+  DISALLOW_COPY_AND_ASSIGN(CastStreamingNativeHandler);
 };
 
 }  // namespace extensions
 
-#endif  // CHROME_RENDERER_EXTENSIONS_WEBRTC_NATIVE_HANDLER_H_
+#endif  // CHROME_RENDERER_EXTENSIONS_CAST_STREAMING_NATIVE_HANDLER_H_
diff --git a/chrome/renderer/extensions/dispatcher.cc b/chrome/renderer/extensions/dispatcher.cc
index 8aefa04..a3414cf 100644
--- a/chrome/renderer/extensions/dispatcher.cc
+++ b/chrome/renderer/extensions/dispatcher.cc
@@ -99,7 +99,7 @@
 #include "v8/include/v8.h"
 
 #if defined(ENABLE_WEBRTC)
-#include "chrome/renderer/extensions/webrtc_native_handler.h"
+#include "chrome/renderer/extensions/cast_streaming_native_handler.h"
 #endif
 
 using blink::WebDataSource;
@@ -944,8 +944,8 @@ void Dispatcher::RegisterNativeHandlers(ModuleSystem* module_system,
   module_system->RegisterNativeHandler("webstore",
       scoped_ptr<NativeHandler>(new WebstoreBindings(this, context)));
 #if defined(ENABLE_WEBRTC)
-  module_system->RegisterNativeHandler("webrtc_natives",
-      scoped_ptr<NativeHandler>(new WebRtcNativeHandler(context)));
+  module_system->RegisterNativeHandler("cast_streaming_natives",
+      scoped_ptr<NativeHandler>(new CastStreamingNativeHandler(context)));
 #endif
 }
 
@@ -1024,12 +1024,13 @@ void Dispatcher::PopulateSourceMap() {
   source_map_.RegisterSource("webRequestInternal",
                              IDR_WEB_REQUEST_INTERNAL_CUSTOM_BINDINGS_JS);
 #if defined(ENABLE_WEBRTC)
+  source_map_.RegisterSource("cast.streaming.rtpStream",
+                             IDR_CAST_STREAMING_RTP_STREAM_CUSTOM_BINDINGS_JS);
   source_map_.RegisterSource("cast.streaming.session",
                              IDR_CAST_STREAMING_SESSION_CUSTOM_BINDINGS_JS);
-  source_map_.RegisterSource("webrtc.castSendTransport",
-                             IDR_WEBRTC_CAST_SEND_TRANSPORT_CUSTOM_BINDINGS_JS);
-  source_map_.RegisterSource("webrtc.castUdpTransport",
-                             IDR_WEBRTC_CAST_UDP_TRANSPORT_CUSTOM_BINDINGS_JS);
+  source_map_.RegisterSource(
+      "cast.streaming.udpTransport",
+      IDR_CAST_STREAMING_UDP_TRANSPORT_CUSTOM_BINDINGS_JS);
 #endif
   source_map_.RegisterSource("webstore", IDR_WEBSTORE_CUSTOM_BINDINGS_JS);
   source_map_.RegisterSource("windowControls", IDR_WINDOW_CONTROLS_JS);
diff --git a/chrome/renderer/media/cast_send_transport.cc b/chrome/renderer/media/cast_rtp_stream.cc
index f48d49b..2590232 100644
--- a/chrome/renderer/media/cast_send_transport.cc
+++ b/chrome/renderer/media/cast_rtp_stream.cc
@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "chrome/renderer/media/cast_send_transport.h"
+#include "chrome/renderer/media/cast_rtp_stream.h"
 
 #include "base/logging.h"
 #include "chrome/renderer/media/cast_session.h"
@@ -118,28 +118,28 @@ CastRtpCaps::CastRtpCaps() {
 CastRtpCaps::~CastRtpCaps() {
 }
 
-CastSendTransport::CastSendTransport(
+CastRtpStream::CastRtpStream(
     const blink::WebMediaStreamTrack& track,
     const scoped_refptr<CastSession>& session)
     : track_(track),
       cast_session_(session) {
 }
 
-CastSendTransport::~CastSendTransport() {
+CastRtpStream::~CastRtpStream() {
 }
 
-CastRtpCaps CastSendTransport::GetCaps() {
+CastRtpCaps CastRtpStream::GetCaps() {
   if (IsAudio())
     return DefaultAudioCaps();
   else
     return DefaultVideoCaps();
 }
 
-CastRtpParams CastSendTransport::GetParams() {
+CastRtpParams CastRtpStream::GetParams() {
   return params_;
 }
 
-void CastSendTransport::Start(const CastRtpParams& params) {
+void CastRtpStream::Start(const CastRtpParams& params) {
   if (IsAudio()) {
     AudioSenderConfig config;
     if (!ToAudioSenderConfig(params, &config)) {
@@ -155,10 +155,10 @@ void CastSendTransport::Start(const CastRtpParams& params) {
   }
 }
 
-void CastSendTransport::Stop() {
+void CastRtpStream::Stop() {
   NOTIMPLEMENTED();
 }
 
-bool CastSendTransport::IsAudio() const {
+bool CastRtpStream::IsAudio() const {
   return track_.source().type() == blink::WebMediaStreamSource::TypeAudio;
 }
diff --git a/chrome/renderer/media/cast_send_transport.h b/chrome/renderer/media/cast_rtp_stream.h
index b30f564..a219234 100644
--- a/chrome/renderer/media/cast_send_transport.h
+++ b/chrome/renderer/media/cast_rtp_stream.h
@@ -2,8 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_
-#define CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_
+#ifndef CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
+#define CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
 
 #include <string>
 #include <vector>
@@ -79,11 +79,11 @@ typedef CastRtpCaps CastRtpParams;
 // Note that this object does not actually output packets. It allows
 // configuration of encoding and RTP parameters and control such a logical
 // stream.
-class CastSendTransport {
+class CastRtpStream {
  public:
-  CastSendTransport(const blink::WebMediaStreamTrack& track,
+  CastRtpStream(const blink::WebMediaStreamTrack& track,
                     const scoped_refptr<CastSession>& session);
-  ~CastSendTransport();
+  ~CastRtpStream();
 
   // Return capabilities currently supported by this transport.
   CastRtpCaps GetCaps();
@@ -107,7 +107,7 @@ class CastSendTransport {
   const scoped_refptr<CastSession> cast_session_;
   CastRtpParams params_;
 
-  DISALLOW_COPY_AND_ASSIGN(CastSendTransport);
+  DISALLOW_COPY_AND_ASSIGN(CastRtpStream);
 };
 
-#endif  // CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_
+#endif  // CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
diff --git a/chrome/renderer/resources/extensions/webrtc_cast_send_transport_custom_bindings.js b/chrome/renderer/resources/extensions/cast_streaming_rtp_stream_custom_bindings.js
index b23e21a..8feb8dd 100644
--- a/chrome/renderer/resources/extensions/webrtc_cast_send_transport_custom_bindings.js
+++ b/chrome/renderer/resources/extensions/cast_streaming_rtp_stream_custom_bindings.js
@@ -2,29 +2,29 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-// Custom binding for the webrtc custom transport API.
+// Custom binding for the Cast Streaming RtpStream API.
 
-var binding = require('binding').Binding.create('webrtc.castSendTransport');
-var webrtc = requireNative('webrtc_natives');
+var binding = require('binding').Binding.create('cast.streaming.rtpStream');
+var natives = requireNative('cast_streaming_natives');
 
 binding.registerCustomHook(function(bindingsAPI, extensionId) {
   var apiFunctions = bindingsAPI.apiFunctions;
 
   apiFunctions.setHandleRequest('destroy',
       function(transportId) {
-        webrtc.DestroyCastSendTransport(transportId);
+        natives.DestroyCastRtpStream(transportId);
   });
   apiFunctions.setHandleRequest('getCaps',
       function(transportId) {
-        return webrtc.GetCapsCastSendTransport(transportId);
+        return natives.GetCapsCastRtpStream(transportId);
   });
   apiFunctions.setHandleRequest('start',
       function(transportId, params) {
-        webrtc.StartCastSendTransport(transportId, params);
+        natives.StartCastRtpStream(transportId, params);
   });
   apiFunctions.setHandleRequest('stop',
       function(transportId) {
-        webrtc.StopCastSendTransport(transportId);
+        natives.StopCastRtpStream(transportId);
   });
 });
 
diff --git a/chrome/renderer/resources/extensions/cast_streaming_session_custom_bindings.js b/chrome/renderer/resources/extensions/cast_streaming_session_custom_bindings.js
index 8455042..c83639c 100644
--- a/chrome/renderer/resources/extensions/cast_streaming_session_custom_bindings.js
+++ b/chrome/renderer/resources/extensions/cast_streaming_session_custom_bindings.js
@@ -2,10 +2,10 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-// Custom binding for the Cast streaming API.
+// Custom binding for the Cast Streaming Session API.
 
 var binding = require('binding').Binding.create('cast.streaming.session');
-var natives = requireNative('webrtc_natives');
+var natives = requireNative('cast_streaming_natives');
 
 binding.registerCustomHook(function(bindingsAPI, extensionId) {
   var apiFunctions = bindingsAPI.apiFunctions;
diff --git a/chrome/renderer/resources/extensions/webrtc_cast_udp_transport_custom_bindings.js b/chrome/renderer/resources/extensions/cast_streaming_udp_transport_custom_bindings.js
index faf4053..a29b321 100644
--- a/chrome/renderer/resources/extensions/webrtc_cast_udp_transport_custom_bindings.js
+++ b/chrome/renderer/resources/extensions/cast_streaming_udp_transport_custom_bindings.js
@@ -2,20 +2,20 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-// Custom binding for the webrtc custom transport API.
+// Custom binding for the Cast Streaming UdpTransport API.
 
-var binding = require('binding').Binding.create('webrtc.castUdpTransport');
-var webrtc = requireNative('webrtc_natives');
+var binding = require('binding').Binding.create('cast.streaming.udpTransport');
+var natives = requireNative('cast_streaming_natives');
 
 binding.registerCustomHook(function(bindingsAPI, extensionId) {
   var apiFunctions = bindingsAPI.apiFunctions;
 
   apiFunctions.setHandleRequest('destroy', function(transportId) {
-    webrtc.DestroyCastUdpTransport(transportId);
+    natives.DestroyCastUdpTransport(transportId);
   });
   apiFunctions.setHandleRequest('start',
     function(transportId, remoteParams) {
-      webrtc.StartCastUdpTransport(transportId, remoteParams);
+      natives.StartCastUdpTransport(transportId, remoteParams);
   });
 });
 
diff --git a/chrome/renderer/resources/renderer_resources.grd b/chrome/renderer/resources/renderer_resources.grd
index 9e37455..d6d6584 100644
--- a/chrome/renderer/resources/renderer_resources.grd
+++ b/chrome/renderer/resources/renderer_resources.grd
@@ -51,7 +51,9 @@ without changes to the corresponding grd file. fb9 -->
         <include name="IDR_APP_RUNTIME_CUSTOM_BINDINGS_JS" file="extensions\app_runtime_custom_bindings.js" type="BINDATA" />
         <include name="IDR_APP_WINDOW_CUSTOM_BINDINGS_JS" file="extensions\app_window_custom_bindings.js" type="BINDATA" />
         <include name="IDR_BROWSER_ACTION_CUSTOM_BINDINGS_JS" file="extensions\browser_action_custom_bindings.js" type="BINDATA" />
+        <include name="IDR_CAST_STREAMING_RTP_STREAM_CUSTOM_BINDINGS_JS" file="extensions\cast_streaming_rtp_stream_custom_bindings.js" type="BINDATA" />
         <include name="IDR_CAST_STREAMING_SESSION_CUSTOM_BINDINGS_JS" file="extensions\cast_streaming_session_custom_bindings.js" type="BINDATA" />
+        <include name="IDR_CAST_STREAMING_UDP_TRANSPORT_CUSTOM_BINDINGS_JS" file="extensions\cast_streaming_udp_transport_custom_bindings.js" type="BINDATA" />
         <include name="IDR_CHROME_DIRECT_SETTING_JS"
         file="extensions\chrome_direct_setting.js" type="BINDATA" />
         <include name="IDR_CHROME_SETTING_JS" file="extensions\chrome_setting.js" type="BINDATA" />
@@ -94,8 +96,6 @@ without changes to the corresponding grd file. fb9 -->
         <include name="IDR_WINDOW_CONTROLS_TEMPLATE_HTML" file="extensions\window_controls_template.html" type="BINDATA" />
         <include name="IDR_WEB_REQUEST_CUSTOM_BINDINGS_JS" file="extensions\web_request_custom_bindings.js" type="BINDATA" />
         <include name="IDR_WEB_REQUEST_INTERNAL_CUSTOM_BINDINGS_JS" file="extensions\web_request_internal_custom_bindings.js" type="BINDATA" />
-        <include name="IDR_WEBRTC_CAST_SEND_TRANSPORT_CUSTOM_BINDINGS_JS" file="extensions\webrtc_cast_send_transport_custom_bindings.js" type="BINDATA" />
-        <include name="IDR_WEBRTC_CAST_UDP_TRANSPORT_CUSTOM_BINDINGS_JS" file="extensions\webrtc_cast_udp_transport_custom_bindings.js" type="BINDATA" />
         <include name="IDR_WEBSTORE_CUSTOM_BINDINGS_JS" file="extensions\webstore_custom_bindings.js" type="BINDATA" />
         <include name="IDR_WEB_VIEW_DENY_JS" file="extensions\web_view_deny.js" type="BINDATA" />
         <include name="IDR_WEB_VIEW_EXPERIMENTAL_JS" file="extensions\web_view_experimental.js" type="BINDATA" />
diff --git a/chrome/test/data/extensions/api_test/webrtc_cast/basics.html b/chrome/test/data/extensions/api_test/cast_streaming/basics.html
index 86106a4..86106a4 100644
--- a/chrome/test/data/extensions/api_test/webrtc_cast/basics.html
+++ b/chrome/test/data/extensions/api_test/cast_streaming/basics.html
diff --git a/chrome/test/data/extensions/api_test/webrtc_cast/basics.js b/chrome/test/data/extensions/api_test/cast_streaming/basics.js
index ade2470..36c1b64 100644
--- a/chrome/test/data/extensions/api_test/webrtc_cast/basics.js
+++ b/chrome/test/data/extensions/api_test/cast_streaming/basics.js
@@ -2,27 +2,27 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-var sendTransport = chrome.webrtc.castSendTransport;
+var rtpStream = chrome.cast.streaming.rtpStream;
 var tabCapture = chrome.tabCapture;
-var udpTransport = chrome.webrtc.castUdpTransport;
+var udpTransport = chrome.cast.streaming.udpTransport;
 var createSession = chrome.cast.streaming.session.create;
 
 chrome.test.runTests([
-  function sendTransportStart() {
+  function rtpStreamStart() {
     tabCapture.capture({audio: true, video: true}, function(stream) {
       console.log("Got MediaStream.");
       chrome.test.assertTrue(!!stream);
       createSession(stream.getAudioTracks()[0],
                     stream.getVideoTracks()[0],
             function(stream, audioId, videoId, udpId) {
-        var audioParams = sendTransport.getCaps(audioId);
-        var videoParams = sendTransport.getCaps(videoId);
-        sendTransport.start(audioId, audioParams);
-        sendTransport.start(videoId, videoParams);
-        sendTransport.stop(audioId);
-        sendTransport.stop(videoId);
-        sendTransport.destroy(audioId);
-        sendTransport.destroy(videoId);
+        var audioParams = rtpStream.getCaps(audioId);
+        var videoParams = rtpStream.getCaps(videoId);
+        rtpStream.start(audioId, audioParams);
+        rtpStream.start(videoId, videoParams);
+        rtpStream.stop(audioId);
+        rtpStream.stop(videoId);
+        rtpStream.destroy(audioId);
+        rtpStream.destroy(videoId);
         udpTransport.destroy(udpId);
         stream.stop();
         chrome.test.assertEq(audioParams.payloads[0].codecName, "OPUS");
diff --git a/chrome/test/data/extensions/api_test/webrtc_cast/manifest.json b/chrome/test/data/extensions/api_test/cast_streaming/manifest.json
index fb001c4..a42cb9c 100644
--- a/chrome/test/data/extensions/api_test/webrtc_cast/manifest.json
+++ b/chrome/test/data/extensions/api_test/cast_streaming/manifest.json
@@ -4,5 +4,5 @@
   "version": "0.1",
   "manifest_version": 2,
   "description": "Tests Cast Mirroring Extensions API.",
-  "permissions": ["webrtc", "tabCapture", "cast", "cast.streaming"]
+  "permissions": ["tabCapture", "cast", "cast.streaming"]
 }
diff --git a/extensions/common/permissions/api_permission.h b/extensions/common/permissions/api_permission.h
index 347f05e..4aacd02 100644
--- a/extensions/common/permissions/api_permission.h
+++ b/extensions/common/permissions/api_permission.h
@@ -159,7 +159,6 @@ class APIPermission {
     kWebRequest,
     kWebRequestBlocking,
     kWebRequestInternal,
-    kWebRtc,
     kWebrtcAudioPrivate,
     kWebrtcLoggingPrivate,
     kWebstorePrivate,