talk/app/webrtc/mediastreamprovider.h

/*
 * 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,
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 */

#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<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_