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-rw-r--r--chrome/common/extensions/api/_api_features.json16
-rw-r--r--chrome/common/extensions/api/_permission_features.json4
-rw-r--r--chrome/common/extensions/api/api.gyp4
-rw-r--r--chrome/common/extensions/api/cast_streaming_rtp_stream.idl96
-rw-r--r--chrome/common/extensions/api/cast_streaming_session.idl10
-rw-r--r--chrome/common/extensions/api/cast_streaming_udp_transport.idl (renamed from chrome/common/extensions/api/webrtc_cast_udp_transport.idl)2
-rw-r--r--chrome/common/extensions/api/webrtc_cast_send_transport.idl107
7 files changed, 112 insertions, 127 deletions
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);
- };
-};