81da488ab6
The audio_device_module field was currently unused. The audio_mixer field is going to be used to pass an AudioMixer to AudioState. In the hopefully-not-very-far future, the toplevel WebRTC API will allow passing a custom AudioMixer, e.g. for spatialized audio (audio in space). If no mixer is passed, a default mixer is created (the one in modules/audio_mixer). The only object which will have a permanent reference to the mixer is AudioState. AudioState is created in WebRTCVoiceEngine with a configuration object, which already contains a VoiceEngine pointer. In this CL, we extend this config object with a mixer pointer. In summary: in an upcoming CL, a mixer will be either created in or passed to WebRTCVoiceEngine. This mixer will be passed to the ctor of AudioState in a config struct. BUG=webrtc:6346 NOTRY=True Review-Url: https://codereview.webrtc.org/2456363002 Cr-Commit-Position: refs/heads/master@{#14973}
51 lines
1.7 KiB
C++
51 lines
1.7 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_API_CALL_AUDIO_STATE_H_
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#define WEBRTC_API_CALL_AUDIO_STATE_H_
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#include "webrtc/api/audio/audio_mixer.h"
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#include "webrtc/base/refcount.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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namespace webrtc {
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class AudioDeviceModule;
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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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};
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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_API_CALL_AUDIO_STATE_H_
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