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Unified Diff: chrome/common/extensions/api/cast_streaming_receiver_session.idl

Issue 938903003: Cast: Javascript bindings for cast receiver (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@cast_receiver_session
Patch Set: comments addressed Created 5 years, 10 months ago
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Index: chrome/common/extensions/api/cast_streaming_receiver_session.idl
diff --git a/chrome/common/extensions/api/cast_streaming_receiver_session.idl b/chrome/common/extensions/api/cast_streaming_receiver_session.idl
new file mode 100644
index 0000000000000000000000000000000000000000..15a7f0a7f493b23fb3a3ad6036e54620d9f8a600
--- /dev/null
+++ b/chrome/common/extensions/api/cast_streaming_receiver_session.idl
@@ -0,0 +1,70 @@
+// Copyright 2015 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.receiverSession</code> API creates a Cast
+// receiver session and adds the resulting audio and video tracks to a
+// MediaStream.
+namespace cast.streaming.receiverSession {
+ // The UDP socket address and port.
+ dictionary IPEndPoint {
+ DOMString address;
+ long port;
+ };
+
+ // RTP receiver parameters.
+ dictionary RtpReceiverParams {
+ // Maximum latency in milliseconds. This parameter controls the logic
+ // of flow control. Implementation can adjust latency adaptively and
+ // tries to keep it under this threshold. A larger value allows smoother
+ // playback at the cost of higher latency.
+ long maxLatency;
+
+ DOMString codecName;
+
+ // Synchronization source identifier for incoming data.
+ long senderSsrc;
+
+ // The SSRC used to send RTCP reports back to the sender.
+ long receiverSsrc;
+
+ // RTP time units per second, defaults to 48000 for audio
+ // and 90000 for video.
+ long? rtpTimebase;
+
+ // 32 bytes hex-encoded AES key.
+ DOMString? aesKey;
+
+ // 32 bytes hex-encoded AES IV (Initialization vector) mask.
+ DOMString? aesIvMask;
+ };
+
+ callback ErrorCallback = void (DOMString error);
+
+ interface Functions {
+ // Creates a Cast receiver session which receives data from a UDP
+ // socket. The receiver will decode the incoming data into an audio
+ // and a video track which will be added to the provided media stream.
+ // The |audioParams| and |videoParams| are generally provided by the
+ // sender through some other messaging channel.
+ //
+ // |audioParams| : Audio stream parameters.
+ // |videoParams| : Video stream parameters.
+ // |localEndpoint| : Local IP and port to bind to.
+ // |height| : Video height.
+ // |width| : Video width.
+ // |maxFrameRate| : Max video frame rate.
+ // |mediaStreamURL| : URL of MediaStream to add the audio and video to.
+ // |transport_options| : Optional transport settings.
+ [nocompile] static void createAndBind(
+ RtpReceiverParams audioParams,
+ RtpReceiverParams videoParams,
+ IPEndPoint localEndpoint,
+ long maxWidth,
+ long maxHeight,
+ double maxFrameRate,
+ DOMString mediaStreamURL,
+ ErrorCallback error_callback,
+ optional object transport_options);
+ };
+};
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