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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
55 lines
1.8 KiB
C++
55 lines
1.8 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_CALL_AUDIO_STATE_H_
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#define WEBRTC_CALL_AUDIO_STATE_H_
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#include "webrtc/api/audio/audio_mixer.h"
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#include "webrtc/rtc_base/refcount.h"
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#include "webrtc/rtc_base/scoped_ref_ptr.h"
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namespace webrtc {
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class AudioProcessing;
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class VoiceEngine;
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// WORK IN PROGRESS
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// This class is under development and is not yet intended for for use outside
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// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
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// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
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// AudioState holds the state which must be shared between multiple instances of
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// webrtc::Call for audio processing purposes.
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class AudioState : public rtc::RefCountInterface {
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public:
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struct Config {
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// VoiceEngine used for audio streams and audio/video synchronization.
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// AudioState will tickle the VoE refcount to keep it alive for as long as
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// the AudioState itself.
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VoiceEngine* voice_engine = nullptr;
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// The audio mixer connected to active receive streams. One per
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// AudioState.
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rtc::scoped_refptr<AudioMixer> audio_mixer;
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// The audio processing module.
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rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
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};
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virtual AudioProcessing* audio_processing() = 0;
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// TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
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static rtc::scoped_refptr<AudioState> Create(
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const AudioState::Config& config);
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virtual ~AudioState() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_AUDIO_STATE_H_
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