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| 1 /* | |
| 2 * libjingle | |
| 3 * Copyright 2012 Google Inc. | |
| 4 * | |
| 5 * Redistribution and use in source and binary forms, with or without | |
| 6 * modification, are permitted provided that the following conditions are met: | |
| 7 * | |
| 8 * 1. Redistributions of source code must retain the above copyright notice, | |
| 9 * this list of conditions and the following disclaimer. | |
| 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | |
| 11 * this list of conditions and the following disclaimer in the documentation | |
| 12 * and/or other materials provided with the distribution. | |
| 13 * 3. The name of the author may not be used to endorse or promote products | |
| 14 * derived from this software without specific prior written permission. | |
| 15 * | |
| 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED | |
| 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF | |
| 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO | |
| 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, | |
| 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, | |
| 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | |
| 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | |
| 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | |
| 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | |
| 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | |
| 26 */ | |
| 27 | |
| 28 #ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ | |
| 29 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ | |
| 30 | |
| 31 #include "webrtc/base/basictypes.h" | |
| 32 #include "webrtc/base/scoped_ptr.h" | |
| 33 #include "webrtc/media/base/videosinkinterface.h" | |
| 34 | |
| 35 namespace cricket { | |
| 36 | |
| 37 class AudioRenderer; | |
| 38 class VideoCapturer; | |
| 39 class VideoFrame; | |
| 40 class VideoRenderer; | |
| 41 struct AudioOptions; | |
| 42 struct VideoOptions; | |
| 43 | |
| 44 } // namespace cricket | |
| 45 | |
| 46 namespace webrtc { | |
| 47 | |
| 48 class AudioSinkInterface; | |
| 49 | |
| 50 // TODO(deadbeef): Change the key from an ssrc to a "sender_id" or | |
| 51 // "receiver_id" string, which will be the MSID in the short term and MID in | |
| 52 // the long term. | |
| 53 | |
| 54 // TODO(deadbeef): These interfaces are effectively just a way for the | |
| 55 // RtpSenders/Receivers to get to the BaseChannels. These interfaces should be | |
| 56 // refactored away eventually, as the classes converge. | |
| 57 | |
| 58 // This interface is called by AudioRtpSender/Receivers to change the settings | |
| 59 // of an audio track connected to certain PeerConnection. | |
| 60 class AudioProviderInterface { | |
| 61 public: | |
| 62 // Enable/disable the audio playout of a remote audio track with |ssrc|. | |
| 63 virtual void SetAudioPlayout(uint32_t ssrc, bool enable) = 0; | |
| 64 // Enable/disable sending audio on the local audio track with |ssrc|. | |
| 65 // When |enable| is true |options| should be applied to the audio track. | |
| 66 virtual void SetAudioSend(uint32_t ssrc, | |
| 67 bool enable, | |
| 68 const cricket::AudioOptions& options, | |
| 69 cricket::AudioRenderer* renderer) = 0; | |
| 70 | |
| 71 // Sets the audio playout volume of a remote audio track with |ssrc|. | |
| 72 // |volume| is in the range of [0, 10]. | |
| 73 virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0; | |
| 74 | |
| 75 // Allows for setting a direct audio sink for an incoming audio source. | |
| 76 // Only one audio sink is supported per ssrc and ownership of the sink is | |
| 77 // passed to the provider. | |
| 78 virtual void SetRawAudioSink( | |
| 79 uint32_t ssrc, | |
| 80 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0; | |
| 81 | |
| 82 protected: | |
| 83 virtual ~AudioProviderInterface() {} | |
| 84 }; | |
| 85 | |
| 86 // This interface is called by VideoRtpSender/Receivers to change the settings | |
| 87 // of a video track connected to a certain PeerConnection. | |
| 88 class VideoProviderInterface { | |
| 89 public: | |
| 90 virtual bool SetCaptureDevice(uint32_t ssrc, | |
| 91 cricket::VideoCapturer* camera) = 0; | |
| 92 // Enable/disable the video playout of a remote video track with |ssrc|. | |
| 93 virtual void SetVideoPlayout( | |
| 94 uint32_t ssrc, | |
| 95 bool enable, | |
| 96 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; | |
| 97 // Enable sending video on the local video track with |ssrc|. | |
| 98 virtual void SetVideoSend(uint32_t ssrc, | |
| 99 bool enable, | |
| 100 const cricket::VideoOptions* options) = 0; | |
| 101 | |
| 102 protected: | |
| 103 virtual ~VideoProviderInterface() {} | |
| 104 }; | |
| 105 | |
| 106 } // namespace webrtc | |
| 107 | |
| 108 #endif // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ | |
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