665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
104 lines
3.2 KiB
C++
104 lines
3.2 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 AUDIO_AUDIO_STATE_H_
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#define AUDIO_AUDIO_STATE_H_
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#include <map>
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#include <memory>
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#include <unordered_set>
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#include "audio/audio_transport_impl.h"
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#include "audio/null_audio_poller.h"
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#include "call/audio_state.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/refcount.h"
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#include "rtc_base/thread_checker.h"
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namespace webrtc {
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class AudioSendStream;
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class AudioReceiveStream;
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namespace internal {
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class AudioState final : public webrtc::AudioState {
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public:
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explicit AudioState(const AudioState::Config& config);
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~AudioState() override;
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AudioProcessing* audio_processing() override {
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RTC_DCHECK(config_.audio_processing);
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return config_.audio_processing.get();
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}
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AudioTransport* audio_transport() override { return &audio_transport_; }
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void SetPlayout(bool enabled) override;
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void SetRecording(bool enabled) override;
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Stats GetAudioInputStats() const override;
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void SetStereoChannelSwapping(bool enable) override;
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AudioDeviceModule* audio_device_module() {
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RTC_DCHECK(config_.audio_device_module);
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return config_.audio_device_module.get();
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}
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bool typing_noise_detected() const;
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void AddReceivingStream(webrtc::AudioReceiveStream* stream);
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void RemoveReceivingStream(webrtc::AudioReceiveStream* stream);
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void AddSendingStream(webrtc::AudioSendStream* stream,
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int sample_rate_hz,
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size_t num_channels);
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void RemoveSendingStream(webrtc::AudioSendStream* stream);
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private:
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// rtc::RefCountInterface implementation.
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void AddRef() const override;
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rtc::RefCountReleaseStatus Release() const override;
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void UpdateAudioTransportWithSendingStreams();
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rtc::ThreadChecker thread_checker_;
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rtc::ThreadChecker process_thread_checker_;
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const webrtc::AudioState::Config config_;
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bool recording_enabled_ = true;
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bool playout_enabled_ = true;
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// Reference count; implementation copied from rtc::RefCountedObject.
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// TODO(nisse): Use RefCountedObject or RefCountedBase instead.
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mutable volatile int ref_count_ = 0;
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// Transports mixed audio from the mixer to the audio device and
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// recorded audio to the sending streams.
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AudioTransportImpl audio_transport_;
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// Null audio poller is used to continue polling the audio streams if audio
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// playout is disabled so that audio processing still happens and the audio
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// stats are still updated.
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std::unique_ptr<NullAudioPoller> null_audio_poller_;
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std::unordered_set<webrtc::AudioReceiveStream*> receiving_streams_;
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struct StreamProperties {
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int sample_rate_hz = 0;
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size_t num_channels = 0;
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};
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std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
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};
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} // namespace internal
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} // namespace webrtc
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#endif // AUDIO_AUDIO_STATE_H_
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