| Index: chrome/common/extensions/api/cast_streaming_receiver_session.idl
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| 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
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| +++ b/chrome/common/extensions/api/cast_streaming_receiver_session.idl
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| @@ -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 {
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| + DOMString address;
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| + long port;
|
| + };
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| +
|
| + // RTP receiver parameters.
|
| + dictionary RtpReceiverParams {
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| + // 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;
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| +
|
| + DOMString codecName;
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| +
|
| + // Synchronization source identifier for incoming data.
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| + long senderSsrc;
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| +
|
| + // The SSRC used to send RTCP reports back to the sender.
|
| + long receiverSsrc;
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| +
|
| + // RTP time units per second, defaults to 48000 for audio
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| + // and 90000 for video.
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| + long? rtpTimebase;
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| +
|
| + // 32 bytes hex-encoded AES key.
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| + DOMString? aesKey;
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| +
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| + // 32 bytes hex-encoded AES IV (Initialization vector) mask.
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| + DOMString? aesIvMask;
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| + };
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| +
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| + callback ErrorCallback = void (DOMString error);
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| +
|
| + interface Functions {
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| + // 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.
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| + //
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| + // |audioParams| : Audio stream parameters.
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| + // |videoParams| : Video stream parameters.
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| + // |localEndpoint| : Local IP and port to bind to.
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| + // |height| : Video height.
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| + // |width| : Video width.
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| + // |maxFrameRate| : Max video frame rate.
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| + // |mediaStreamURL| : URL of MediaStream to add the audio and video to.
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| + // |transport_options| : Optional transport settings.
|
| + [nocompile] static void createAndBind(
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| + RtpReceiverParams audioParams,
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| + RtpReceiverParams videoParams,
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| + IPEndPoint localEndpoint,
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| + long maxWidth,
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| + long maxHeight,
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| + double maxFrameRate,
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| + DOMString mediaStreamURL,
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| + ErrorCallback error_callback,
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| + optional object transport_options);
|
| + };
|
| +};
|
|
|