dce40cf804
use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
65 lines
2.1 KiB
C++
65 lines
2.1 KiB
C++
/*
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* Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/processing_component.h"
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namespace webrtc {
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class AudioBuffer;
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class CriticalSectionWrapper;
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class VoiceDetectionImpl : public VoiceDetection,
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public ProcessingComponent {
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public:
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VoiceDetectionImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit);
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virtual ~VoiceDetectionImpl();
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int ProcessCaptureAudio(AudioBuffer* audio);
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// VoiceDetection implementation.
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bool is_enabled() const override;
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// ProcessingComponent implementation.
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int Initialize() override;
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private:
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// VoiceDetection implementation.
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int Enable(bool enable) override;
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int set_stream_has_voice(bool has_voice) override;
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bool stream_has_voice() const override;
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int set_likelihood(Likelihood likelihood) override;
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Likelihood likelihood() const override;
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int set_frame_size_ms(int size) override;
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int frame_size_ms() const override;
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// ProcessingComponent implementation.
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void* CreateHandle() const override;
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int InitializeHandle(void* handle) const override;
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int ConfigureHandle(void* handle) const override;
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void DestroyHandle(void* handle) const override;
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int num_handles_required() const override;
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int GetHandleError(void* handle) const override;
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const AudioProcessing* apm_;
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CriticalSectionWrapper* crit_;
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bool stream_has_voice_;
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bool using_external_vad_;
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Likelihood likelihood_;
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int frame_size_ms_;
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size_t frame_size_samples_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
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