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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}
76 lines
2.6 KiB
C++
76 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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#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 "modules/audio_device/include/audio_device.h"
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#include "modules/audio_processing/include/audio_processing.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 AudioTransport;
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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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Config();
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~Config();
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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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// TODO(solenberg): Temporary: audio device module.
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rtc::scoped_refptr<webrtc::AudioDeviceModule> audio_device_module;
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};
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struct Stats {
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// Audio peak level (max(abs())), linearly on the interval [0,32767].
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int32_t audio_level = -1;
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// See:
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// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
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double total_energy = 0.0f;
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double total_duration = 0.0f;
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};
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virtual AudioProcessing* audio_processing() = 0;
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virtual AudioTransport* audio_transport() = 0;
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// Enable/disable playout of the audio channels. Enabled by default.
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// This will stop playout of the underlying audio device but start a task
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// which will poll for audio data every 10ms to ensure that audio processing
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// happens and the audio stats are updated.
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virtual void SetPlayout(bool enabled) = 0;
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// Enable/disable recording of the audio channels. Enabled by default.
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// This will stop recording of the underlying audio device and no audio
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// packets will be encoded or transmitted.
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virtual void SetRecording(bool enabled) = 0;
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virtual Stats GetAudioInputStats() const = 0;
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virtual void SetStereoChannelSwapping(bool enable) = 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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~AudioState() override {}
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};
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} // namespace webrtc
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#endif // CALL_AUDIO_STATE_H_
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