chrome/common/extensions/api/cast_streaming_receiver_session.idl

   1 // Copyright 2015 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 // The <code>chrome.cast.streaming.receiverSession</code> API creates a Cast
   6 // receiver session and adds the resulting audio and video tracks to a
   7 // MediaStream.
   8 namespace cast.streaming.receiverSession {
   9   // The UDP socket address and port.
  10   dictionary IPEndPoint {
  11     DOMString address;
  12     long port;
  13   };
  14
  15   // RTP receiver parameters.
  16   dictionary RtpReceiverParams {
  17     // Maximum latency in milliseconds. This parameter controls the logic
  18     // of flow control. Implementation can adjust latency adaptively and
  19     // tries to keep it under this threshold. A larger value allows smoother
  20     // playback at the cost of higher latency.
  21     long maxLatency;
  22
  23     DOMString codecName;
  24
  25     // Synchronization source identifier for incoming data.
  26     long senderSsrc;
  27
  28     // The SSRC used to send RTCP reports back to the sender.
  29     long receiverSsrc;
  30
  31     // RTP time units per second, defaults to 48000 for audio
  32     // and 90000 for video.
  33     long? rtpTimebase;
  34
  35     // 32 bytes hex-encoded AES key.
  36     DOMString? aesKey;
  37
  38     // 32 bytes hex-encoded AES IV (Initialization vector) mask.
  39     DOMString? aesIvMask;
  40   };
  41
  42   callback ErrorCallback = void (DOMString error);
  43
  44   interface Functions {
  45     // Creates a Cast receiver session which receives data from a UDP
  46     // socket. The receiver will decode the incoming data into an audio
  47     // and a video track which will be added to the provided media stream.
  48     // The |audioParams| and |videoParams| are generally provided by the
  49     // sender through some other messaging channel.
  50     //
  51     // |audioParams| : Audio stream parameters.
  52     // |videoParams| : Video stream parameters.
  53     // |localEndpoint| : Local IP and port to bind to.
  54     // |height| : Video height.
  55     // |width| : Video width.
  56     // |maxFrameRate| : Max video frame rate.
  57     // |mediaStreamURL| : URL of MediaStream to add the audio and video to.
  58     // |transport_options| : Optional transport settings.
  59     [nocompile] static void createAndBind(
  60         RtpReceiverParams audioParams,
  61         RtpReceiverParams videoParams,
  62         IPEndPoint localEndpoint,
  63         long maxWidth,
  64         long maxHeight,
  65         double maxFrameRate,
  66         DOMString mediaStreamURL,
  67         ErrorCallback error_callback,
  68         optional object transport_options);
  69   };
  70 };