Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : #ifndef WEBRTC_CALL_AUDIO_STATE_H_
      11             : #define WEBRTC_CALL_AUDIO_STATE_H_
      12             : 
      13             : #include "webrtc/api/audio/audio_mixer.h"
      14             : #include "webrtc/base/refcount.h"
      15             : #include "webrtc/base/scoped_ref_ptr.h"
      16             : 
      17             : namespace webrtc {
      18             : 
      19             : class VoiceEngine;
      20             : 
      21             : // WORK IN PROGRESS
      22             : // This class is under development and is not yet intended for for use outside
      23             : // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
      24             : // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
      25             : 
      26             : // AudioState holds the state which must be shared between multiple instances of
      27             : // webrtc::Call for audio processing purposes.
      28           0 : class AudioState : public rtc::RefCountInterface {
      29             :  public:
      30           0 :   struct Config {
      31             :     // VoiceEngine used for audio streams and audio/video synchronization.
      32             :     // AudioState will tickle the VoE refcount to keep it alive for as long as
      33             :     // the AudioState itself.
      34             :     VoiceEngine* voice_engine = nullptr;
      35             : 
      36             :     // The audio mixer connected to active receive streams. One per
      37             :     // AudioState.
      38             :     rtc::scoped_refptr<AudioMixer> audio_mixer;
      39             :   };
      40             : 
      41             :   // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
      42             :   static rtc::scoped_refptr<AudioState> Create(
      43             :       const AudioState::Config& config);
      44             : 
      45           0 :   virtual ~AudioState() {}
      46             : };
      47             : }  // namespace webrtc
      48             : 
      49             : #endif  // WEBRTC_CALL_AUDIO_STATE_H_