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| 1 // Copyright 2013 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 #ifndef CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_ | |
| 6 #define CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_ | |
| 7 | |
| 8 #include <string> | |
| 9 #include <vector> | |
| 10 | |
| 11 #include "base/basictypes.h" | |
| 12 #include "base/memory/ref_counted.h" | |
| 13 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | |
| 14 | |
| 15 class CastSession; | |
| 16 | |
| 17 // A key value pair structure for codec specific parameters. | |
| 18 struct CastCodecSpecificParams { | |
| 19 std::string key; | |
| 20 std::string value; | |
| 21 | |
| 22 CastCodecSpecificParams(); | |
| 23 ~CastCodecSpecificParams(); | |
| 24 }; | |
| 25 | |
| 26 // Defines the basic properties of a payload supported by cast transport. | |
| 27 struct CastRtpPayloadParams { | |
| 28 // RTP specific field that identifies the content type. | |
| 29 int payload_type; | |
| 30 | |
| 31 // RTP specific field to identify a stream. | |
| 32 int ssrc; | |
| 33 | |
| 34 // Update frequency of payload sample. | |
| 35 int clock_rate; | |
| 36 | |
| 37 // Maximum bitrate. | |
| 38 int max_bitrate; | |
| 39 | |
| 40 // Minimum bitrate. | |
| 41 int min_bitrate; | |
| 42 | |
| 43 // Number of audio channels. | |
| 44 int channels; | |
| 45 | |
| 46 // Width and height of the video content. | |
| 47 int width; | |
| 48 int height; | |
| 49 | |
| 50 // Name of the codec used. | |
| 51 std::string codec_name; | |
| 52 | |
| 53 // List of codec specific parameters. | |
| 54 std::vector<CastCodecSpecificParams> codec_specific_params; | |
| 55 | |
| 56 CastRtpPayloadParams(); | |
| 57 ~CastRtpPayloadParams(); | |
| 58 }; | |
| 59 | |
| 60 // Defines the capabilities of the transport. | |
| 61 struct CastRtpCaps { | |
| 62 // Defines a list of supported payloads. | |
| 63 std::vector<CastRtpPayloadParams> payloads; | |
| 64 | |
| 65 // Names of supported RTCP features. | |
| 66 std::vector<std::string> rtcp_features; | |
| 67 | |
| 68 // Names of supported FEC (Forward Error Correction) mechanisms. | |
| 69 std::vector<std::string> fec_mechanisms; | |
| 70 | |
| 71 CastRtpCaps(); | |
| 72 ~CastRtpCaps(); | |
| 73 }; | |
| 74 | |
| 75 typedef CastRtpCaps CastRtpParams; | |
| 76 | |
| 77 // This object represents a RTP stream that encodes and optionally | |
| 78 // encrypt audio or video data from a WebMediaStreamTrack. | |
| 79 // Note that this object does not actually output packets. It allows | |
| 80 // configuration of encoding and RTP parameters and control such a logical | |
| 81 // stream. | |
| 82 class CastSendTransport { | |
| 83 public: | |
| 84 CastSendTransport(const blink::WebMediaStreamTrack& track, | |
| 85 const scoped_refptr<CastSession>& session); | |
| 86 ~CastSendTransport(); | |
| 87 | |
| 88 // Return capabilities currently supported by this transport. | |
| 89 CastRtpCaps GetCaps(); | |
| 90 | |
| 91 // Return parameters set to this transport. | |
| 92 CastRtpParams GetParams(); | |
| 93 | |
| 94 // Begin encoding of media stream and then submit the encoded streams | |
| 95 // to underlying transport. | |
| 96 void Start(const CastRtpParams& params); | |
| 97 | |
| 98 // Stop encoding. | |
| 99 void Stop(); | |
| 100 | |
| 101 private: | |
| 102 // Return true if this track is an audio track. Return false if this | |
| 103 // track is a video track. | |
| 104 bool IsAudio() const; | |
| 105 | |
| 106 blink::WebMediaStreamTrack track_; | |
| 107 const scoped_refptr<CastSession> cast_session_; | |
| 108 CastRtpParams params_; | |
| 109 | |
| 110 DISALLOW_COPY_AND_ASSIGN(CastSendTransport); | |
| 111 }; | |
| 112 | |
| 113 #endif // CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_ | |
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Chromium Code Reviews
Issue 90083002:
Cast Extensions API: Major namespace and object renaming (Closed)
Base URL: svn://svn.chromium.org/chrome/trunk/src