/* * libjingle * Copyright 2012 Google Inc. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions are met: * * 1. Redistributions of source code must retain the above copyright notice, * this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright notice, * this list of conditions and the following disclaimer in the documentation * and/or other materials provided with the distribution. * 3. The name of the author may not be used to endorse or promote products * derived from this software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ #ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ #include "webrtc/base/basictypes.h" #include "webrtc/base/scoped_ptr.h" namespace cricket { class AudioRenderer; class VideoCapturer; class VideoRenderer; struct AudioOptions; struct VideoOptions; } // namespace cricket namespace webrtc { class AudioSinkInterface; // TODO(deadbeef): Change the key from an ssrc to a "sender_id" or // "receiver_id" string, which will be the MSID in the short term and MID in // the long term. // TODO(deadbeef): These interfaces are effectively just a way for the // RtpSenders/Receivers to get to the BaseChannels. These interfaces should be // refactored away eventually, as the classes converge. // This interface is called by AudioRtpSender/Receivers to change the settings // of an audio track connected to certain PeerConnection. class AudioProviderInterface { public: // Enable/disable the audio playout of a remote audio track with |ssrc|. virtual void SetAudioPlayout(uint32_t ssrc, bool enable) = 0; // Enable/disable sending audio on the local audio track with |ssrc|. // When |enable| is true |options| should be applied to the audio track. virtual void SetAudioSend(uint32_t ssrc, bool enable, const cricket::AudioOptions& options, cricket::AudioRenderer* renderer) = 0; // Sets the audio playout volume of a remote audio track with |ssrc|. // |volume| is in the range of [0, 10]. virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0; // Allows for setting a direct audio sink for an incoming audio source. // Only one audio sink is supported per ssrc and ownership of the sink is // passed to the provider. virtual void SetRawAudioSink( uint32_t ssrc, rtc::scoped_ptr sink) = 0; protected: virtual ~AudioProviderInterface() {} }; // This interface is called by VideoRtpSender/Receivers to change the settings // of a video track connected to a certain PeerConnection. class VideoProviderInterface { public: virtual bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) = 0; // Enable/disable the video playout of a remote video track with |ssrc|. virtual void SetVideoPlayout(uint32_t ssrc, bool enable, cricket::VideoRenderer* renderer) = 0; // Enable sending video on the local video track with |ssrc|. virtual void SetVideoSend(uint32_t ssrc, bool enable, const cricket::VideoOptions* options) = 0; protected: virtual ~VideoProviderInterface() {} }; } // namespace webrtc #endif // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_