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/*
* 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_ref_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,
const rtc::scoped_refptr<webrtc::AudioSinkInterface>& 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_
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