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Moving src/webrtc into src/.

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In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
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58
modules/audio_processing/voice_detection_impl.h Normal file
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/*
* 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 WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#include <memory>
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/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 // WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_