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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
51 lines
1.4 KiB
C++
51 lines
1.4 KiB
C++
/*
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* Copyright (c) 2014 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 COMMON_AUDIO_VAD_INCLUDE_VAD_H_
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#define COMMON_AUDIO_VAD_INCLUDE_VAD_H_
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#include <memory>
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#include "common_audio/vad/include/webrtc_vad.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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class Vad {
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public:
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enum Aggressiveness {
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kVadNormal = 0,
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kVadLowBitrate = 1,
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kVadAggressive = 2,
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kVadVeryAggressive = 3
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};
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enum Activity { kPassive = 0, kActive = 1, kError = -1 };
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virtual ~Vad() = default;
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// Calculates a VAD decision for the given audio frame. Valid sample rates
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// are 8000, 16000, and 32000 Hz; the number of samples must be such that the
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// frame is 10, 20, or 30 ms long.
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virtual Activity VoiceActivity(const int16_t* audio,
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size_t num_samples,
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int sample_rate_hz) = 0;
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// Resets VAD state.
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virtual void Reset() = 0;
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};
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// Returns a Vad instance that's implemented on top of WebRtcVad.
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std::unique_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness);
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} // namespace webrtc
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#endif // COMMON_AUDIO_VAD_INCLUDE_VAD_H_
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