mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00

In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
58 lines
2 KiB
C++
58 lines
2 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
|
|
#define MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
|
|
|
|
#include <memory>
|
|
|
|
#include "modules/audio_processing/include/audio_processing.h"
|
|
#include "rtc_base/constructormagic.h"
|
|
#include "rtc_base/criticalsection.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class AudioBuffer;
|
|
|
|
class VoiceDetectionImpl : public VoiceDetection {
|
|
public:
|
|
explicit VoiceDetectionImpl(rtc::CriticalSection* crit);
|
|
~VoiceDetectionImpl() override;
|
|
|
|
// TODO(peah): Fold into ctor, once public API is removed.
|
|
void Initialize(int sample_rate_hz);
|
|
void ProcessCaptureAudio(AudioBuffer* audio);
|
|
|
|
// VoiceDetection implementation.
|
|
int Enable(bool enable) override;
|
|
bool is_enabled() const override;
|
|
int set_stream_has_voice(bool has_voice) override;
|
|
bool stream_has_voice() const override;
|
|
int set_likelihood(Likelihood likelihood) override;
|
|
Likelihood likelihood() const override;
|
|
int set_frame_size_ms(int size) override;
|
|
int frame_size_ms() const override;
|
|
|
|
private:
|
|
class Vad;
|
|
rtc::CriticalSection* const crit_;
|
|
bool enabled_ RTC_GUARDED_BY(crit_) = false;
|
|
bool stream_has_voice_ RTC_GUARDED_BY(crit_) = false;
|
|
bool using_external_vad_ RTC_GUARDED_BY(crit_) = false;
|
|
Likelihood likelihood_ RTC_GUARDED_BY(crit_) = kLowLikelihood;
|
|
int frame_size_ms_ RTC_GUARDED_BY(crit_) = 10;
|
|
size_t frame_size_samples_ RTC_GUARDED_BY(crit_) = 0;
|
|
int sample_rate_hz_ RTC_GUARDED_BY(crit_) = 0;
|
|
std::unique_ptr<Vad> vad_ RTC_GUARDED_BY(crit_);
|
|
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(VoiceDetectionImpl);
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
|