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Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h')
| -rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h | 136 |
1 files changed, 136 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h new file mode 100644 index 00000000000..af4f4d6c88c --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h @@ -0,0 +1,136 @@ +/* + * 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_NETEQ_DSP_HELPER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ + +#include <string.h> // Access to size_t. + +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.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_NETEQ_DSP_HELPER_H_ |