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| 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 }; |
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