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Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.h')
| -rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.h | 136 |
1 files changed, 0 insertions, 136 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.h b/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.h deleted file mode 100644 index 60cd995d840..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.h +++ /dev/null @@ -1,136 +0,0 @@ -/* - * 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_CODING_NETEQ4_DSP_HELPER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DSP_HELPER_H_ - -#include <string.h> // Access to size_t. - -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -// This class contains various signal processing functions, all implemented as -// static methods. -class DspHelper { - public: - // Filter coefficients used when downsampling from the indicated sample rates - // (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12. - static const int16_t kDownsample8kHzTbl[3]; - static const int16_t kDownsample16kHzTbl[5]; - static const int16_t kDownsample32kHzTbl[7]; - static const int16_t kDownsample48kHzTbl[7]; - - // Constants used to mute and unmute over 5 samples. The coefficients are - // in Q15. - static const int kMuteFactorStart8kHz = 27307; - static const int kMuteFactorIncrement8kHz = -5461; - static const int kUnmuteFactorStart8kHz = 5461; - static const int kUnmuteFactorIncrement8kHz = 5461; - static const int kMuteFactorStart16kHz = 29789; - static const int kMuteFactorIncrement16kHz = -2979; - static const int kUnmuteFactorStart16kHz = 2979; - static const int kUnmuteFactorIncrement16kHz = 2979; - static const int kMuteFactorStart32kHz = 31208; - static const int kMuteFactorIncrement32kHz = -1560; - static const int kUnmuteFactorStart32kHz = 1560; - static const int kUnmuteFactorIncrement32kHz = 1560; - static const int kMuteFactorStart48kHz = 31711; - static const int kMuteFactorIncrement48kHz = -1057; - static const int kUnmuteFactorStart48kHz = 1057; - static const int kUnmuteFactorIncrement48kHz = 1057; - - // Multiplies the signal with a gradually changing factor. - // The first sample is multiplied with |factor| (in Q14). For each sample, - // |factor| is increased (additive) by the |increment| (in Q20), which can - // be negative. Returns the scale factor after the last increment. - static int RampSignal(const int16_t* input, - size_t length, - int factor, - int increment, - int16_t* output); - - // Same as above, but with the samples of |signal| being modified in-place. - static int RampSignal(int16_t* signal, - size_t length, - int factor, - int increment); - - // Same as above, but processes |length| samples from |signal|, starting at - // |start_index|. - static int RampSignal(AudioMultiVector* signal, - size_t start_index, - size_t length, - int factor, - int increment); - - // Peak detection with parabolic fit. Looks for |num_peaks| maxima in |data|, - // having length |data_length| and sample rate multiplier |fs_mult|. The peak - // locations and values are written to the arrays |peak_index| and - // |peak_value|, respectively. Both arrays must hold at least |num_peaks| - // elements. - static void PeakDetection(int16_t* data, int data_length, - int num_peaks, int fs_mult, - int* peak_index, int16_t* peak_value); - - // Estimates the height and location of a maximum. The three values in the - // array |signal_points| are used as basis for a parabolic fit, which is then - // used to find the maximum in an interpolated signal. The |signal_points| are - // assumed to be from a 4 kHz signal, while the maximum, written to - // |peak_index| and |peak_value| is given in the full sample rate, as - // indicated by the sample rate multiplier |fs_mult|. - static void ParabolicFit(int16_t* signal_points, int fs_mult, - int* peak_index, int16_t* peak_value); - - // Calculates the sum-abs-diff for |signal| when compared to a displaced - // version of itself. Returns the displacement lag that results in the minimum - // distortion. The resulting distortion is written to |distortion_value|. - // The values of |min_lag| and |max_lag| are boundaries for the search. - static int MinDistortion(const int16_t* signal, int min_lag, - int max_lag, int length, int32_t* distortion_value); - - // Mixes |length| samples from |input1| and |input2| together and writes the - // result to |output|. The gain for |input1| starts at |mix_factor| (Q14) and - // is decreased by |factor_decrement| (Q14) for each sample. The gain for - // |input2| is the complement 16384 - mix_factor. - static void CrossFade(const int16_t* input1, const int16_t* input2, - size_t length, int16_t* mix_factor, - int16_t factor_decrement, int16_t* output); - - // Scales |input| with an increasing gain. Applies |factor| (Q14) to the first - // sample and increases the gain by |increment| (Q20) for each sample. The - // result is written to |output|. |length| samples are processed. - static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor, - int16_t increment, int16_t* output); - - // Starts at unity gain and gradually fades out |signal|. For each sample, - // the gain is reduced by |mute_slope| (Q14). |length| samples are processed. - static void MuteSignal(int16_t* signal, int16_t mute_slope, size_t length); - - // Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input - // has |input_length| samples, and the method will write |output_length| - // samples to |output|. Compensates for the phase delay of the downsampling - // filters if |compensate_delay| is true. Returns -1 if the input is too short - // to produce |output_length| samples, otherwise 0. - static int DownsampleTo4kHz(const int16_t* input, size_t input_length, - int output_length, int input_rate_hz, - bool compensate_delay, int16_t* output); - - private: - // Table of constants used in method DspHelper::ParabolicFit(). - static const int16_t kParabolaCoefficients[17][3]; - - DISALLOW_COPY_AND_ASSIGN(DspHelper); -}; - -} // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DSP_HELPER_H_ |