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}
82 lines
2.6 KiB
C++
82 lines
2.6 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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#include "audio/audio_state.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "rtc_base/atomicops.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "voice_engine/transmit_mixer.h"
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namespace webrtc {
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namespace internal {
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AudioState::AudioState(const AudioState::Config& config)
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: config_(config),
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voe_base_(config.voice_engine),
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audio_transport_proxy_(voe_base_->audio_transport(),
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config_.audio_processing.get(),
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config_.audio_mixer) {
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process_thread_checker_.DetachFromThread();
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RTC_DCHECK(config_.audio_mixer);
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auto* const device = voe_base_->audio_device_module();
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RTC_DCHECK(device);
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// This is needed for the Chrome implementation of RegisterAudioCallback.
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device->RegisterAudioCallback(nullptr);
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device->RegisterAudioCallback(&audio_transport_proxy_);
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}
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AudioState::~AudioState() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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}
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VoiceEngine* AudioState::voice_engine() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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return config_.voice_engine;
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}
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rtc::scoped_refptr<AudioMixer> AudioState::mixer() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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return config_.audio_mixer;
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}
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bool AudioState::typing_noise_detected() const {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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// TODO(solenberg): Remove const_cast once AudioState owns transmit mixer
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// functionality.
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voe::TransmitMixer* transmit_mixer =
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const_cast<AudioState*>(this)->voe_base_->transmit_mixer();
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return transmit_mixer->typing_noise_detected();
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}
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// Reference count; implementation copied from rtc::RefCountedObject.
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void AudioState::AddRef() const {
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rtc::AtomicOps::Increment(&ref_count_);
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}
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// Reference count; implementation copied from rtc::RefCountedObject.
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rtc::RefCountReleaseStatus AudioState::Release() const {
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if (rtc::AtomicOps::Decrement(&ref_count_) == 0) {
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delete this;
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return rtc::RefCountReleaseStatus::kDroppedLastRef;
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}
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return rtc::RefCountReleaseStatus::kOtherRefsRemained;
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}
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} // namespace internal
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rtc::scoped_refptr<AudioState> AudioState::Create(
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const AudioState::Config& config) {
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return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
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}
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} // namespace webrtc
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