d94659dc27
This is the first public upload of the new NetEq, version 4. It has been through extensive internal review during the course of the project. TEST=trybots Review URL: https://webrtc-codereview.appspot.com/1073005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
113 lines
4.1 KiB
C++
113 lines
4.1 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_CODING_NETEQ4_TIME_STRETCH_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TIME_STRETCH_H_
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#include <assert.h>
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#include <cstring> // memset, size_t
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#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h"
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#include "webrtc/system_wrappers/interface/constructor_magic.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Forward declarations.
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class BackgroundNoise;
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// This is the base class for Accelerate and PreemptiveExpand. This class
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// cannot be instantiated, but must be used through either of the derived
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// classes.
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class TimeStretch {
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public:
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enum ReturnCodes {
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kSuccess = 0,
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kSuccessLowEnergy = 1,
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kNoStretch = 2,
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kError = -1
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};
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TimeStretch(int sample_rate_hz, size_t num_channels,
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const BackgroundNoise& background_noise)
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: sample_rate_hz_(sample_rate_hz),
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fs_mult_(sample_rate_hz / 8000),
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num_channels_(num_channels),
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master_channel_(0), // First channel is master.
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background_noise_(background_noise),
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max_input_value_(0) {
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assert(sample_rate_hz_ == 8000 ||
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sample_rate_hz_ == 16000 ||
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sample_rate_hz_ == 32000 ||
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sample_rate_hz_ == 48000);
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assert(num_channels_ > 0);
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assert(static_cast<int>(master_channel_) < num_channels_);
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memset(auto_correlation_, 0, sizeof(auto_correlation_));
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}
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virtual ~TimeStretch() {}
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// This method performs the processing common to both Accelerate and
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// PreemptiveExpand.
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ReturnCodes Process(const int16_t* input,
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size_t input_len,
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AudioMultiVector<int16_t>* output,
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int16_t* length_change_samples);
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protected:
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// Sets the parameters |best_correlation| and |peak_index| to suitable
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// values when the signal contains no active speech. This method must be
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// implemented by the sub-classes.
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virtual void SetParametersForPassiveSpeech(int input_length,
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int16_t* best_correlation,
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int* peak_index) const = 0;
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// Checks the criteria for performing the time-stretching operation and,
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// if possible, performs the time-stretching. This method must be implemented
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// by the sub-classes.
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virtual ReturnCodes CheckCriteriaAndStretch(
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const int16_t* input, int input_length, size_t peak_index,
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int16_t best_correlation, bool active_speech,
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AudioMultiVector<int16_t>* output) const = 0;
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static const int kCorrelationLen = 50;
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static const int kLogCorrelationLen = 6; // >= log2(kCorrelationLen).
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static const int kMinLag = 10;
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static const int kMaxLag = 60;
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static const int kDownsampledLen = kCorrelationLen + kMaxLag;
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static const int kCorrelationThreshold = 14746; // 0.9 in Q14.
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const int sample_rate_hz_;
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const int fs_mult_; // Sample rate multiplier = sample_rate_hz_ / 8000.
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const int num_channels_;
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const size_t master_channel_;
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const BackgroundNoise& background_noise_;
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int16_t max_input_value_;
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int16_t downsampled_input_[kDownsampledLen];
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// Adding 1 to the size of |auto_correlation_| because of how it is used
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// by the peak-detection algorithm.
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int16_t auto_correlation_[kCorrelationLen + 1];
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private:
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// Calculates the auto-correlation of |downsampled_input_| and writes the
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// result to |auto_correlation_|.
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void AutoCorrelation();
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// Performs a simple voice-activity detection based on the input parameters.
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bool SpeechDetection(int32_t vec1_energy, int32_t vec2_energy,
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int peak_index, int scaling) const;
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DISALLOW_COPY_AND_ASSIGN(TimeStretch);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TIME_STRETCH_H_
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