990d6b875e
This reverts commit 90bace095806a635411edd40fb8490a144e59e63. Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it. Original change's description: > Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API > > (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201) > > This SetAudioPlayout method lets applications disable audio playout while > still processing incoming audio data and generating statistics on the > received audio. > > This may be useful if the application wants to set up media flows as > soon as possible, but isn't ready to play audio yet. Currently, native > applications don't have any API point to control this, unless they > completely implement their own AudioDeviceModule. > > The SetAudioRecording works in a similar fashion but for the recorded > audio. One difference is that calling SetAudioRecording(false) does not > keep any audio processing alive. > > TBR=solenberg > > Bug: webrtc:7313 > Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa > Reviewed-on: https://webrtc-review.googlesource.com/16180 > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Henrik Andreassson <henrika@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20499} TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7313 Reviewed-on: https://webrtc-review.googlesource.com/17701 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20512}
56 lines
1.8 KiB
C++
56 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 CALL_AUDIO_STATE_H_
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#define CALL_AUDIO_STATE_H_
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#include "api/audio/audio_mixer.h"
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#include "rtc_base/refcount.h"
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#include "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 // CALL_AUDIO_STATE_H_
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