diff --git a/reaper-tmp/REAPER/CMakeLists.txt b/reaper-tmp/REAPER/CMakeLists.txt deleted file mode 100644 index 58d95b6e17b5f45223f8c2654913c4f05b4e004d..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/CMakeLists.txt +++ /dev/null @@ -1,39 +0,0 @@ -cmake_minimum_required(VERSION 2.8) -project(Reaper) - -# Use C11 for unique_ptr -include(CheckCXXCompilerFlag) -CHECK_CXX_COMPILER_FLAG("-std=c++11" COMPILER_SUPPORTS_CXX11) -CHECK_CXX_COMPILER_FLAG("-std=c++0x" COMPILER_SUPPORTS_CXX0X) -if(COMPILER_SUPPORTS_CXX11) - set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -std=c++11") -elseif(COMPILER_SUPPORTS_CXX0X) - set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -std=c++0x") -else() - message(STATUS "The compiler ${CMAKE_CXX_COMPILER} has no C++11 support. Please use a different C++ compiler.") -endif() - -include_directories(".") - -# Core functionality. -add_library(core STATIC core/file_resource.h - core/file_resource.cc core/float_matrix.h - core/float_matrix-inl.h core/float_matrix.cc core/track.h - core/track.cc) - -# Wave functionality. -add_library(wave STATIC wave/codec_riff.h wave/codec_riff.cc - wave/codec_api.h wave/codec_api-inl.h wave/wave.h wave/wave.cc - wave/wave_io.h wave/wave_io-inl.h wave/wave_io.cc) - -# Epoch tracker. -add_library(epoch_tracker STATIC epoch_tracker/fft.h epoch_tracker/fft.cc - epoch_tracker/fd_filter.h epoch_tracker/fd_filter.cc - epoch_tracker/lpc_analyzer.cc epoch_tracker/lpc_analyzer.h - epoch_tracker/epoch_tracker.cc epoch_tracker/epoch_tracker.h) - -# Binary to extract F0. -add_executable(reaper epoch_tracker_main.cc) -target_link_libraries(wave core) -target_link_libraries(epoch_tracker wave) -target_link_libraries(reaper epoch_tracker) diff --git a/reaper-tmp/REAPER/CONTRIBUTING.md b/reaper-tmp/REAPER/CONTRIBUTING.md deleted file mode 100644 index d6508261ad2c3fde4ddcf166eeb88af62a31abc5..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/CONTRIBUTING.md +++ /dev/null @@ -1,13 +0,0 @@ -Anyone contributing code must sign a Contributor License Agreement. - -Individual contributors can complete the Individual Contributor -License Agreement online: - -https://developers.google.com/open-source/cla/individual - -If you are contributing on behalf of a corporation, fill out the -Corporate Contributor License Agreement and send it in as described on -this page: - -https://developers.google.com/open-source/cla/corporate - diff --git a/reaper-tmp/REAPER/LICENSE b/reaper-tmp/REAPER/LICENSE deleted file mode 100644 index d645695673349e3947e8e5ae42332d0ac3164cd7..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/LICENSE +++ /dev/null @@ -1,202 +0,0 @@ - - Apache License - Version 2.0, January 2004 - http://www.apache.org/licenses/ - - TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION - - 1. 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The _reaper_ program uses the -EpochTracker class to simultaneously estimate the location of -voiced-speech "epochs" or glottal closure instants (GCI), voicing -state (voiced or unvoiced) and fundamental frequency (F0 or "pitch"). -We define the local (instantaneous) F0 as the inverse of the time -between successive GCI. - -This code was developed by David Talkin at Google. This is not an -official Google product (experimental or otherwise), it is just -code that happens to be owned by Google. - -## Downloading and Building _reaper_ -``` -cd convenient_place_for_repository -git clone https://github.com/google/REAPER.git -cd REAPER -mkdir build # In the REAPER top-level directory -cd build -cmake .. -make -``` - -_reaper_ will now be in `convenient_place_for_repository/REAPER/build/reaper` - -You may want to add that path to your PATH environment variable or -move _reaper_ to your favorite bin repository. - -Example: - -To compute F0 (pitch) and pitchmark (GCI) tracks and write them out as ASCII files: - -`reaper -i /tmp/bla.wav -f /tmp/bla.f0 -p /tmp/bla.pm -a` - - -## Input Signals: - -As written, the input stage expects 16-bit, signed integer samples. -Any reasonable sample rate may be used, but rates below 16 kHz will -introduce increasingly coarse quantization of the results, and higher -rates will incur quadratic increase in computational requirements -without gaining much in output accuracy. - -While REAPER is fairly robust to recording quality, it is designed for -use with studio-quality speech signals, such as those recorded for -concatenation text-to-speech systems. Phase distortion, such as that -introduced by some close-talking microphones or by well-intended -recording-studio filtering, including rumble removal, should be -avoided, for best results. A rumble filter is provided within REAPER -as the recommended (default) high-pass pre-filtering option, and is -implemented as a symmetric FIR filter that introduces no phase -distortion. - -The help text _(-h)_ provided by the _reaper_ program describes -various output options, including debug output of some of the feature -signals. Of special interest is the residual waveform which may be -used to check for the expected waveshape. (The residual has a -_.resid_ filename extension.) During non-nasalized, open vocal tract -vocalizations (such as /a/), each period should show a somewhat noisy -version of the derivative of the idealized glottal flow. If the computed -residual deviates radically from this ideal, the Hilbert transform -option _(-t)_ might improve matters. - -## The REAPER Algorithm: - -The process can be broken down into the following phases: -* Signal Conditioning -* Feature Extraction -* Lattice Generation -* Dynamic Programming -* Backtrace and Output Generation - - -## Signal Conditioning - -DC bias and low-frequency noise are removed by high-pass filtering, -and the signal is converted to floating point. If the input is known -to have phase distortion that is impacting tracker performance, a -Hilbert transform, optionally done at this point, may improve -performance. - - -## Feature Extraction - -The following feature signals are derived from the conditioned input: -* Linear Prediction residual: - This is computed using the autocorrelation method and continuous - interpolation of the filter coefficients. It is checked for the - expected polarity (negative impulses), and inverted, if necessary. -* Amplitude-normalized prediction residual: - The normalization factor is based on the running, local RMS. -* Pseudo-probability of voicing: - This is based on a local measure of low-frequency energy normalized - by the peak energy in the utterance. -* Pseudo-probability of voicing onset: - Based on a forward delta of lowpassed energy. -* Pseudo-probability of voicing offset: - Based on a backward delta of lowpassed energy. -* Graded GCI candidates: - Each negative peak in the normalized residual is compared with the - local RMS. Peaks exceeding a threshold are selected as GCI candidates, - and then graded by a weighted combination of peak amplitude, skewness, - and sharpness. Each of the resulting candidates is associated with the - other feature values that occur closest in time to the candidate. -* Normalized cross-correlation functions (NCCF) for each GCI candidate: - The correlations are computed on a weighted combination of the speech - signal and its LP residual. The correlation reference window for - each GCI candidate impulse is centered on the inpulse, and - correlations are computed for all lags in the expected pitch period range. - - -## Lattice Generation - -Each GCI candidate (pulse) is set into a lattice structure that links -preceding and following pulses that occur within minimum and maximum -pitch period limits that are being considered for the utterance. -These links establish all of the period hypotheses that will be -considered for the pulse. Each hypothesis is scored on "local" -evidence derived from the NCCF and peak quality measures. Each pulse -is also assigned an unvoiced hypothesis, which is also given a score -based on the available local evidence. The lattice is checked, and -modified, if necessary to ensure that each pulse has at least one -voiced and one unvoiced hypothesis preceding and following it, to -maintain continuity for the dynamic programming to follow. -(Note that the "scores" are used as costs during dynamic programming, -so that low scores encourage selection of hypotheses.) - - -## Dynamic Programming - -``` -For each pulse in the utterance: - For each period hypotheses following the pulse: - For each period hypothesis preceding the pulse: - Score the transition cost of connecting the periods. Choose the - minimum overall cost (cumulative+local+transition) preceding - period hypothesis, and save its cost and a backpointer to it. - The costs of making a voicing state change are modulated by the - probability of voicing onset and offset. The cost of - voiced-to-voiced transition is based on the delta F0 that - occurs, and the cost of staying in the unvoiced state is a - constant system parameter. -``` - -## Backtrace and Output Generation - -Starting at the last peak in the utterance, the lowest cost period -candidate ending on that peak is found. This is the starting point -for backtracking. The backpointers to the best preceding period -candidates are then followed backwards through the utterance. As each -"best candidate" is found, the time location of the terminal peak is -recorded, along with the F0 corresponding to the period, or 0.0 if the -candidate is unvoiced. Instead of simply taking the inverse of the -period between GCI estimates as F0, the system refers back to the NCCF -for that GCI, and takes the location of the NCCF maximum closest to -the GCI-based period as the actual period. The output array of F0 and -estimated GCI location is then time-reversed for final output. - diff --git a/reaper-tmp/REAPER/build/CMakeCache.txt b/reaper-tmp/REAPER/build/CMakeCache.txt deleted file mode 100644 index 4b08d21c70005861166ef21f55677a2e490c95cd..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/build/CMakeCache.txt +++ /dev/null @@ -1,389 +0,0 @@ -# This is the CMakeCache file. -# For build in directory: /content/NeuCoSVC-Colab/REAPER/build -# It was generated by CMake: /usr/local/lib/python3.10/dist-packages/cmake/data/bin/cmake -# You can edit this file to change values found and used by cmake. -# If you do not want to change any of the values, simply exit the editor. -# If you do want to change a value, simply edit, save, and exit the editor. -# The syntax for the file is as follows: -# KEY:TYPE=VALUE -# KEY is the name of a variable in the cache. -# TYPE is a hint to GUIs for the type of VALUE, DO NOT EDIT TYPE!. -# VALUE is the current value for the KEY. - 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- -/* Construct a string literal encoding the version number components. */ -#elif defined(COMPILER_VERSION_MAJOR) -char const info_version[] = { - 'I', 'N', 'F', 'O', ':', - 'c','o','m','p','i','l','e','r','_','v','e','r','s','i','o','n','[', - COMPILER_VERSION_MAJOR, -# ifdef COMPILER_VERSION_MINOR - '.', COMPILER_VERSION_MINOR, -# ifdef COMPILER_VERSION_PATCH - '.', COMPILER_VERSION_PATCH, -# ifdef COMPILER_VERSION_TWEAK - '.', COMPILER_VERSION_TWEAK, -# endif -# endif -# endif - ']','\0'}; -#endif - -/* Construct a string literal encoding the internal version number. */ -#ifdef COMPILER_VERSION_INTERNAL -char const info_version_internal[] = { - 'I', 'N', 'F', 'O', ':', - 'c','o','m','p','i','l','e','r','_','v','e','r','s','i','o','n','_', - 'i','n','t','e','r','n','a','l','[', - COMPILER_VERSION_INTERNAL,']','\0'}; -#elif defined(COMPILER_VERSION_INTERNAL_STR) -char const* info_version_internal = "INFO" ":" "compiler_version_internal[" COMPILER_VERSION_INTERNAL_STR "]"; -#endif - -/* Construct a string literal encoding the version number components. */ -#ifdef SIMULATE_VERSION_MAJOR -char const info_simulate_version[] = { - 'I', 'N', 'F', 'O', ':', - 's','i','m','u','l','a','t','e','_','v','e','r','s','i','o','n','[', - SIMULATE_VERSION_MAJOR, -# ifdef SIMULATE_VERSION_MINOR - '.', SIMULATE_VERSION_MINOR, -# ifdef SIMULATE_VERSION_PATCH - '.', SIMULATE_VERSION_PATCH, -# ifdef SIMULATE_VERSION_TWEAK - '.', SIMULATE_VERSION_TWEAK, -# endif -# endif -# endif - ']','\0'}; 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All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#include "core/file_resource.h" - -FileResource::FileResource(const std::string &filename, const std::string &mode) - : filename_(filename), mode_(mode) { -} - -FileResource::~FileResource() { - fclose(fp_); -} - -bool FileResource::Get() { - fp_ = fopen(filename_.c_str(), mode_.c_str()); - return fp_ != NULL; -} diff --git a/reaper-tmp/REAPER/core/file_resource.h b/reaper-tmp/REAPER/core/file_resource.h deleted file mode 100644 index 271e9ce660700eb64de8281ccc14d95f16fa5f3c..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/core/file_resource.h +++ /dev/null @@ -1,44 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#ifndef _FILE_RESOURCE_H_ -#define _FILE_RESOURCE_H_ - -#include -#include - -class FileResource { - public: - // Mode follows the standard (eg. "wb", "r"). - FileResource(const std::string &filename, const std::string &mode); - ~FileResource(); - - // Opens a file with the specified mode. Returns false if it fails. - bool Get(); - - FILE *fp() { return fp_; } - - bool eof() const { - return feof(fp_) != 0; - } - - private: - std::string filename_; - std::string mode_; - FILE *fp_; -}; - -#endif // _FILE_RESOURCE_H_ diff --git a/reaper-tmp/REAPER/core/float_matrix-inl.h b/reaper-tmp/REAPER/core/float_matrix-inl.h deleted file mode 100644 index 86cb13fce94a2fdaf1c556f21f31bac846278dd9..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/core/float_matrix-inl.h +++ /dev/null @@ -1,72 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#ifndef _FLOAT_MATRIX_INL_H_ -#define _FLOAT_MATRIX_INL_H_ - -// IWYU pragma: private, include "float_matrix.h" - - - -inline uint32_t FloatMatrix::get_x_size(void) const { - return x_size_; -} - -inline uint32_t FloatMatrix::get_y_size(void) const { - return y_size_; -} - -// y1x1 y1x2 y1x3 y1x4 y2x1 y2x2 y2x3 y2x4 -inline float FloatMatrix::Get(uint32_t x, uint32_t y) const { - return data_[y * x_size_ + x]; -} - -inline float &FloatMatrix::Get(uint32_t x, uint32_t y) { -#ifdef _DEBUG - if (y >= y_size_ || x >= x_size_) { - fprintf(stderr, "Matrix boundary overrun error"); - } -#endif - return data_[y * x_size_ + x]; -} - -inline void FloatMatrix::Set(uint32_t x, uint32_t y, float val) { -#ifdef _DEBUG - if (y >= y_size_ || x >= x_size_) { - fprintf(stderr, "Matrix boundary overrun error"); - } -#endif - data_[y *x_size_ + x] = val; -} - -inline uint32_t FloatMatrix::size() const { - return x_size_ * y_size_; -} - -inline const float *FloatMatrix::data(void) const { - return data_; -} - -inline void FloatMatrix::StringWrite(std::string *out) const { - if (size()) { - char *source = reinterpret_cast(data_); - out->append(source, size() * sizeof(*data_)); - } -} - - - -#endif // SPEECH_PATTS_ENGINE_COMMON_CORE_FLOAT_MATRIX_INL_H_ diff --git a/reaper-tmp/REAPER/core/float_matrix.cc b/reaper-tmp/REAPER/core/float_matrix.cc deleted file mode 100644 index ab163ee6b43ccaa117d07100fb604797f08825e5..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/core/float_matrix.cc +++ /dev/null @@ -1,78 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#include "core/float_matrix.h" - -#include - - - -FloatMatrix::FloatMatrix() { - data_ = NULL; - x_size_ = 0; - y_size_ = 0; -} - -FloatMatrix::FloatMatrix(const FloatMatrix &m) { - clear(); - resize(m.x_size_, m.y_size_); - for (uint32_t i = 0; i < x_size_ * y_size_; i++) { - data_[i] = m.data_[i]; - } -} - -FloatMatrix::~FloatMatrix() { - clear(); -} - -void FloatMatrix::operator=(const FloatMatrix &m) { - clear(); - resize(m.x_size_, m.y_size_); - for (uint32_t i = 0; i < x_size_ * y_size_; i++) { - data_[i] = m.data_[i]; - } -} - -void FloatMatrix::clear() { - if (data_ != NULL) { - delete [] data_; - data_ = NULL; - } - x_size_ = 0; - y_size_ = 0; -} - -void FloatMatrix::resize(uint32_t x_size, uint32_t y_size) { - if (data_ == NULL) { - data_ = new float [x_size * y_size]; - memset(data_, 0, sizeof(float)*x_size * y_size); - } else if (x_size == x_size_ && y_size == y_size_) { - return; - } else { - float *newdata = new float [x_size * y_size]; - memset(newdata, 0, sizeof(float) * x_size * y_size); - int min_x = x_size < x_size_ ? x_size : x_size_; - int min_y = y_size < y_size_ ? y_size : y_size_; - for (int y = 0; y < min_y; y++) { - memcpy(newdata + y * x_size, data_ + y * x_size_, min_x * sizeof(float)); - } - delete [] data_; - data_ = newdata; - } - - x_size_ = x_size; - y_size_ = y_size; -} diff --git a/reaper-tmp/REAPER/core/float_matrix.h b/reaper-tmp/REAPER/core/float_matrix.h deleted file mode 100644 index 8b253757ae2afe057f375e9691d413a00eec89ff..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/core/float_matrix.h +++ /dev/null @@ -1,57 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ -// A memory mappable float matrix with restricted use, composite audio and -// track classes being its current uses. Its further use should be avoided -// as its designed to meet the specific needs of these classes. - -#ifndef _FLOAT_MATRIX_H_ -#define _FLOAT_MATRIX_H_ - -#include -#include - -class FloatMatrix { - public: - FloatMatrix(); - FloatMatrix(const FloatMatrix &); - ~FloatMatrix(); - - void operator = (const FloatMatrix &); - - void resize(uint32_t x_size, uint32_t y_size); - void clear(); - - float Get(uint32_t x, uint32_t y) const; - float &Get(uint32_t x, uint32_t y); - void Set(uint32_t x, uint32_t y, float val); - - uint32_t get_x_size() const; - uint32_t get_y_size() const; - - uint32_t size() const; - const float *data(void) const; - - void StringWrite(std::string *out) const; - - private: - float *data_; - uint32_t x_size_; - uint32_t y_size_; -}; - -#include "float_matrix-inl.h" - -#endif // _FLOAT_MATRIX_H_ diff --git a/reaper-tmp/REAPER/core/track.cc b/reaper-tmp/REAPER/core/track.cc deleted file mode 100644 index 1ab0a995e4e37e042015d597124b1dded4738eb7..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/core/track.cc +++ /dev/null @@ -1,581 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ -#include "core/track.h" - -#include -#include -#include - -#define __STDC_FORMAT_MACROS -#include - -#if !defined(INVALID_LOG) -#define INVALID_LOG -1.0E+10F -#endif // INVALID_LOG - -namespace { - -int ToInt(const std::string &s) { - char *p = 0; - int32_t r = strtol(s.c_str(), &p, 10); - return static_cast(r); -} - -std::string ToString(float n, int p) { - char buf[1000]; - buf[0] = 0; - snprintf(buf, sizeof(buf), "%.*f", p, n); - return buf; -} - -std::string ToString(uint32_t n) { - char buf[1000]; - buf[0] = 0; - snprintf(buf, sizeof(buf), "%" PRIu32, n); - return buf; -} - -double ToDouble(const std::string &s) { - char *p = 0; - return strtod(s.c_str(), &p); -} - -float ToFloat(const std::string &s) { - return static_cast(ToDouble(s)); -} - -// -// Default size of character I/O buffers: -// -const uint32_t kMaxCharBufSize = 8192; // 8k - -std::string GetToken(FileResource *fr) { - char buff[kMaxCharBufSize]; - uint32_t i = 0; - bool foundData = false; - do { - buff[i] = fgetc(fr->fp()); - if ((iscntrl((unsigned char)buff[i]) || isspace((unsigned char)buff[i])) && - !foundData) { - continue; - } - foundData = true; - if (iscntrl((unsigned char)buff[i]) || isspace((unsigned char)buff[i])) { - break; - } - i++; - } while (i < kMaxCharBufSize && !fr->eof()); - buff[i] = 0; - return std::string(buff); -} - -// Find the first whitespace/newline delimited token in the character -// array c. Return the stripped token in s. return the position of -// the first character following the token in tok_end. If the end of -// c is found during the search, return true, else return false. If no -// token is found, return 0 in tok_end. -bool GetTokenFromChars(const char *c, std::string *s, int32_t *tok_end) { - int32_t i = 0; - s->clear(); - *tok_end = 0; - // skip initial whitespace - while (c[i] && ((c[i] == ' ') || (c[i] == '\t') || (c[i] == '\n'))) { - i++; - } - if (!c[i]) { - return true; - } - int32_t start = i; - while (c[i] && (c[i] != ' ') && (c[i] != '\t') && (c[i] != '\n')) { - i++; - } - *tok_end = i; - - if (c[i-1] == '\r') { - --i; - } - s->assign(c + start, i - start); - return c[i] == 0; -} - -} // namespace - - -Track::Track() { - num_frames_ = 0; - num_channels_ = 0; - shift_ = 0.0F; - voicing_enabled_ = false; -} - -void Track::FillFrom(const Track &t) { - num_frames_ = t.num_frames_; - num_channels_ = t.num_channels_; - shift_ = t.shift_; - data_ = t.data_; - val_ = t.val_; - voicing_enabled_ = t.voicing_enabled_; -} - - -Track::~Track() { - Clear(); -} - -bool Track::Save(FileResource *fr) const { - fprintf(fr->fp(), "EST_File Track\n"); - fprintf(fr->fp(), "DataType binary2\n"); - fprintf(fr->fp(), "NumFrames %d\n", num_frames()); - fprintf(fr->fp(), "NumChannels %d\n", num_channels()); - fprintf(fr->fp(), "FrameShift %f\n", shift()); - if (voicing_enabled_) { - fprintf(fr->fp(), "VoicingEnabled true\n"); - } else { - fprintf(fr->fp(), "VoicingEnabled false\n"); - } - fprintf(fr->fp(), "EST_Header_End\n"); - - // write time information - for (int i = 0; i < num_frames(); ++i) { - float f = t(i); - if (fwrite(&f, sizeof(f), 1, fr->fp()) != 1) { - fprintf(stderr, "Failed to write to track file"); - return false; - } - } - - // write voicing information - if (voicing_enabled_) { - for (int i = 0; i < num_frames(); ++i) { - char vu = v(i) ? 1 : 0; - if (fwrite(&vu, sizeof(vu), 1, fr->fp()) != 1) { - fprintf(stderr, "Failed to write to track file"); - return false; - } - } - } - - // write coefficient information - for (int i = 0; i < num_frames(); ++i) { - for (int j = 0; j < num_channels(); ++j) { - float f = a(i, j); - if (fwrite(&f, sizeof(f), 1, fr->fp()) != 1) { - fprintf(stderr, "Failed to write to track file"); - return false; - } - } - } - return true; -} - -int Track::Index(float x) const { - if (num_frames() > 1) { // if single frame, return that index (0) - int bst, bmid, bend; - bst = 1; - bend = num_frames(); - if (x < t(bst)) { - bmid = bst; - } - if (x >= t(bend - 1)) { - bmid = bend - 1; - } else { - while (1) { - bmid = bst + (bend - bst) / 2; - if (bst == bmid) { - break; - } else if (x < t(bmid)) { - if (x >= t(bmid - 1)) { - break; - } - bend = bmid; - } else { - bst = bmid; - } - } - } - if (fabs(x - t(bmid)) < fabs(x - t(bmid - 1))) { - return bmid; - } else { - return bmid - 1; - } - } - return num_frames() - 1; -} - -int Track::IndexBelow(float x) const { - for (int i = 1; i < num_frames(); ++i) { - if (x <= t(i)) { - return i - 1; - } - } - return num_frames() - 1; -} - -int Track::IndexAbove(float x) const { - for (int i = 0; i < num_frames(); ++i) { - if (x <= t(i)) { - return i; - } - } - return num_frames() - 1; -} - -std::string Track::HeaderToString() const { - std::string ret; - ret = "EST_File Track\n"; - ret += "DataType ascii\n"; - ret += "NumFrames " + ::ToString(num_frames()) + "\n"; - ret += "NumChannels " + ::ToString(num_channels()) + "\n"; - ret += "FrameShift " + ::ToString(shift(), 5) + "\n"; - ret += "VoicingEnabled "; - if (voicing_enabled_) { - ret += "true\n"; - } else { - ret += "false\n"; - } - ret += "EST_Header_End\n"; - return ret; -} - -std::string Track::ToString(uint32_t precision) const { - std::string ret; - ret = HeaderToString(); - char buf[1024]; - for (int i = 0; i < num_frames(); ++i) { - snprintf(buf, 1024, "%f %d", t(i), v(i) ? 1 : 0); - ret += buf; - for (int j = 0; j < num_channels(); ++j) { - ret += " "; - ret += ::ToString(a(i, j), precision); - } - ret += std::string("\n"); - } - return ret; -} - -std::string Track::ToString() const { - const uint32_t kPrintOutPrecision = 6; - return ToString(kPrintOutPrecision); -} - -float Track::shift() const { - return shift_; -} - -void Track::FillTime(float frame_shift, int start) { - shift_ = frame_shift; - for (int i = 0; i < num_frames_; ++i) { - data_.Set(0, i, frame_shift * (i + start)); - } -} - -void Track::FillTime(const Track &t) { - for (int i = 0; i < num_frames_; ++i) { - data_.Set(0, i, t.t(i)); - } -} - -void Track::SetTimes(float *times, int length) { - if (length != num_frames_) { - fprintf(stderr, "Track::SetTimes: input `times` has different number " - "of frames (%d != %d)", length, num_frames_); - } - for (int i = 0; i < num_frames_; ++i) { - data_.Set(0, i, times[i]); - } -} - -bool Track::SetVoicing(const std::vector &vuv) { - if (vuv.size() != num_frames_) { - fprintf(stderr, "Track::SetVoicing: input has different number " - "of frames (%zu != %d)", vuv.size(), num_frames_); - return false; - } - val_ = vuv; - return true; -} - -void Track::FrameOut(std::vector *fv, int n) const { - fv->resize(num_channels_); - for (int i = 0; i < num_channels_; i++) { - (*fv)[i] = a(n, i); - } -} - -void Track::FrameOut(std::vector *fv, int n) const { - fv->resize(num_channels_); - for (int i = 0; i < num_channels_; i++) { - (*fv)[i] = a(n, i); - } -} - -void Track::FrameIn(const float &f, int n) { - a(n) = f; -} - -void Track::FrameIn(const std::vector &fv, int n) { - for (int i = 0; i < num_channels_; i++) { - a(n, i) = fv[i]; - } -} - -void Track::FrameIn(const std::vector &fv, int n) { - for (int i = 0; i < num_channels_; i++) { - a(n, i) = fv[i]; - } -} - -Track *Track::GetSubTrack(float start, - float end, - int ch_offset, - int ch_size) const { - int start_frame = -1; - int end_frame = -1; - int n_sub_frames; - int i, j; - - for (i = 0; i < this->num_frames(); ++i) { - if ((start_frame == -1) && (this->t(i) > start)) { - start_frame = i - 1; - } - if ((end_frame == -1) && (this->t(i) > end)) { - end_frame = i; - } - } - if (start_frame == -1) { - start_frame = this->num_frames() - 1; - } - if (end_frame == -1) { - end_frame = this->num_frames() - 1; - } - n_sub_frames = end_frame - start_frame; - if (n_sub_frames < 1) { - // make sure we have at least one frame - n_sub_frames = 1; - } - if (ch_size == -1 && ch_offset == -1) { - ch_offset = 0; - ch_size = this->num_channels(); - } else { - if (ch_size > this->num_channels()) { - fprintf(stderr, "Incorrect number of channels for sub track"); - return NULL; - } - } - Track *sub_track = new Track; - sub_track->resize(n_sub_frames, ch_size); - for (i = 0; i < n_sub_frames; ++i) { - sub_track->t(i) = this->t(i + start_frame); - for (j = ch_offset; j < ch_size; ++j) { - sub_track->a(i, j) = this->a(i + start_frame, j); - } - } - return sub_track; -} - - -Track *Track::GetSubTrack(int start_frame_index, - int end_frame_index, - int start_channel_index, - int end_channel_index) const { - if (start_frame_index > end_frame_index || - start_frame_index < 0 || start_frame_index >= this->num_frames() || - end_frame_index < 0 || end_frame_index >= this->num_frames()) { - fprintf(stderr, "Incorrect frame indices for sub-track 0<%d<=%d<%d\n", - start_frame_index, - end_frame_index, - this->num_frames()); - return NULL; - } - - if (start_channel_index > end_channel_index || - start_channel_index < 0 || start_channel_index >= this->num_channels() || - end_channel_index < 0 || end_channel_index >= this->num_channels()) { - fprintf(stderr, "Incorrect channel indices for sub-track 0<%d<=%d<%d\n", - start_channel_index, - end_channel_index, - this->num_channels()); - return NULL; - } - - Track *sub_track = new Track; - - int number_frames = end_frame_index - start_frame_index + 1; - int number_channels = end_channel_index - start_channel_index + 1; - sub_track->resize(number_frames, number_channels); - - for (int f = 0; f < number_frames; f++) { - for (int p = 0; p < number_channels; p++) { - sub_track->a(f, p) = this->a(start_frame_index + f, - start_channel_index + p); - } - sub_track->t(f) = this->t(start_frame_index + f); - sub_track->set_v(f, this->v(start_frame_index + f)); - } - - sub_track->set_shift(this->shift()); - return sub_track; -} - -void Track::resize(int n, int c) { - if (n != num_frames_) { - val_.resize(n, 1); - } - - data_.resize(c + 1, n); - num_frames_ = n; - num_channels_ = c; -} - -void Track::Clear() { - num_frames_ = 0; - num_channels_ = 0; - shift_ = 0.0F; - data_.clear(); - val_.clear(); -} - -void Track::CopyFrom(const char *v, int32_t num_frames, int32_t num_channels) { - resize(num_frames, num_channels); - const float* source = reinterpret_cast(v); - for (int32_t i = 0; i < num_frames; ++i) { - for (int32_t j = 0; j < num_channels; ++j) { - a(i, j) = *source++; - } - } -} - -// Sets current track to be combination of two tracks -bool Track::SetCombinedTrack(const Track &track_a, const Track &track_b) { - if (track_a.num_frames() != track_b.num_frames()) { - fprintf(stderr, "Mismatching number of frames: %d vs. %d", - track_a.num_frames(), - track_b.num_frames()); - return false; - } - - const int n_frames = track_a.num_frames(); - const int n_channels_a = track_a.num_channels(); - const int n_channels_b = track_b.num_channels(); - resize(n_frames, n_channels_a + n_channels_b); - for (int i = 0; i < n_frames; i++) { - for (int j = 0; j < n_channels_a; j++) { - a(i, j) = track_a.a(i, j); - } - for (int j = 0; j < n_channels_b; j++) { - a(i, n_channels_a + j) = track_b.a(i, j); - } - set_v(i, track_a.v(i) && track_b.v(i)); - } - return true; -} - -bool Track::SetCombinedTrack(const Track &track_a, - const Track &track_b, - const Track &track_c) { - Track track_ab; - if (!track_ab.SetCombinedTrack(track_a, track_b)) { - return false; - } - return SetCombinedTrack(track_ab, track_c); -} - -// Pads a track with frames: -bool Track::Pad(int num_pads) { - if (num_pads <= 0) { - return false; - } - const int num_original_frames = num_frames(); - const int num_original_channels = num_channels(); - - Track new_track; - new_track.resize(num_original_frames + num_pads, num_original_channels); - for (int i = 0; i < num_original_frames; ++i) { - for (int j = 0; j < num_original_channels; ++j) { - new_track.a(i, j) = a(i, j); - } - new_track.set_v(i, v(i)); - } - for (int i = 0; i < num_pads; ++i) { - for (int j = 0; j < num_original_channels; ++j) { - new_track.a(num_original_frames + i, j) = a(num_original_frames - 1, j); - } - new_track.set_v(i, v(num_original_frames - 1)); - } - - const int num_new_frames = new_track.num_frames(); - Clear(); - resize(num_new_frames, num_original_channels); - for (int i = 0; i < num_new_frames; ++i) { - for (int j = 0; j < num_original_channels; ++j) { - a(i, j) = new_track.a(i, j); - } - set_v(i, new_track.v(i)); - } - return true; -} - -// Forces the two tracks to have the same size, padding if necessary. -void Track::MakeSameSize(Track *track) { - int max_frames = num_frames(); - if (track->num_frames() > max_frames) { - max_frames = track->num_frames(); - } - Pad(max_frames - num_frames()); - track->Pad(max_frames - track->num_frames()); -} - -// Forces the three tracks to have same size, padding if necessary. -void Track::MakeSameSize(Track *track_a, Track *track_b) { - int max_frames = num_frames(); - if (track_a->num_frames() > max_frames) { - max_frames = track_a->num_frames(); - } - if (track_b->num_frames() > max_frames) { - max_frames = track_b->num_frames(); - } - Pad(max_frames - num_frames()); - track_a->Pad(max_frames - track_a->num_frames()); - track_b->Pad(max_frames - track_b->num_frames()); -} - -//------------------------------------------------------------------------------ -// Utils. -//------------------------------------------------------------------------------ - -void ConvertToLogarithmic(Track *t) { - for (int32_t i = 0; i < t->num_frames(); ++i) { - for (int32_t j = 0; j < t->num_channels(); ++j) { - if (t->a(i, j) <= 0.0F) { - t->a(i, j) = INVALID_LOG; - } else { - t->a(i, j) = log(t->a(i, j)); - } - } - } -} - -float *TrackToFloatPointer(const Track &track, int *num_samples) { - *num_samples = track.num_channels() * track.num_frames(); - float *array = new float[*num_samples]; - uint32_t k = 0; - for (int i = 0; i < track.num_frames(); ++i) { - for (int j = 0; j < track.num_channels(); ++j) { - array[k++] = track.a(i, j); - } - } - return array; -} diff --git a/reaper-tmp/REAPER/core/track.h b/reaper-tmp/REAPER/core/track.h deleted file mode 100644 index c66dfdac257a4ffbd883aa0cc7a8b4764dfcae8c..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/core/track.h +++ /dev/null @@ -1,320 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#ifndef _TRACK_H_ -#define _TRACK_H_ - -#include -#include -#include - -#include "core/file_resource.h" -#include "core/float_matrix.h" - -class Track { - public: - Track(); - ~Track(); - - void FillFrom(const Track &track); - - // Returns the number of frames contained within this track instance. - int num_frames() const; - - // Returns the number of channels contained within this track instance. - int num_channels() const; - - // Returns whether the voicing is enabled on this track instance. - bool voicing_enabled() const; - - // Sets the voicing enable flag on this track instance. - void set_voicing_enabled(bool b); - - // Gets values for frame "f" and channel "c". - float &a(int f, int c = 0); - - // Gets values for frame "f" and channel "c". - float a(int f, int c = 0) const; - - // Gets voiced flags for frame "f". - void set_v(int f, bool value); - - // Gets voiced flags for frame "f". - bool v(int f) const; - - // Gets time instant for frame "f". - float &t(int f); - - // Gets time instant for frame "f". - float t(int f) const; - - // Sets the interval between frames in seconds. - void set_shift(float shift); - - // Returns the interval between frames in seconds. - float shift() const; - - // Modifies this track instances frame times with that given, from the - // given frame index. - void FillTime(float frame_shift, int start = 1); - - // Modifies this track instances frame times with the value corresponding - // to its frame index. - void FillTime(const Track &t); - - // Modifies the times of this track instance with the given values. - // NOTE: the length must be equal to the number of frames in the track. - void SetTimes(float *times, int length); - - // Modifies the voicing flag of this track instance with the given values. -bool SetVoicing(const std::vector &vuv); - - // Sets the track from the given time and value vectors. - // All frames are set to voiced. - template - void Set(const std::vector ×, const std::vector &values); - - // Sets the track from the given time and value vectors. - // All frames are set to voiced. - template - void Set(float time_shift, const std::vector &values); - - // Populates the given vector with the values for all channels for - // the given frame index. - void FrameOut(std::vector *fv, int n) const; - - // Populates the given vector with the values for all channels for - // the given frame index. - void FrameOut(std::vector *fv, int n) const; - - // Updates the values for all channels for the given frame index with - // the values given. - void FrameIn(const float &f, int n); - - // Updates the values for all channels for the given frame index with - // the values given. - void FrameIn(const std::vector &fv, int n); - - // Updates the values for all channels for the given frame index with - // the values given. - void FrameIn(const std::vector &fv, int n); - - // Returns the index of the frame at the given time index. - int Index(float x) const; - - // Returns the index of the first frame below that at the given time index, x. - int IndexBelow(float x) const; - - // Returns the index of the first frame above that at the given time index, x. - int IndexAbove(float x) const; - - // Resizes the data and voicing flags to the given value, n, and the - // number of channels to that given, c. - void resize(int n, int c = 1); - - // Resets all the data associated with this track instance. - void Clear(); - - // Returns a portion of this track instance as defined by the input - // start and end times along with the channels required. - // Note: Ownership of the returned track is required to prevent - // memory leaks downstream. - Track *GetSubTrack(float start, - float end, - int ch_offset = -1, - int ch_size = -1) const; - - // Returns a portion of this track instance as defined by the input - // start and end frame indices along with the channels required. - // Note: Ownership of the returned track is required to prevent - // memory leaks downstream. - Track *GetSubTrack(int start_frame_index, - int end_frame_index, - int start_channel_index, - int end_channel_index) const; - - // Returns a std::string representation of the header of this track: - // The number of frames in this track - // The number of channels in this track - // The interval between frames - // Whether voicing is enabled or not - std::string HeaderToString() const; - - // Returns a std::string representation of this track instance containing: - // The number of frames in this track - // The number of channels in this track - // The interval between frames - // Whether voicing is enabled or not - // values - // If precision is specified, float numbers are printed out up to - // the specified number of decimal places. - std::string ToString(uint32_t precision) const; - std::string ToString() const; - - // Saves this track instance to the given FileResource. - bool Save(FileResource *fr) const; - - // Saves this track instance to the given FileResource. - bool Save(const std::string &filename, bool ascii) const; - - const FloatMatrix &data() const; - - // Resizes the track to the number of frames and channels given, - // copying the given data into the track. - void CopyFrom(const char *v, int32_t num_frames, int32_t num_channels); - - // Sets this track to be a combination of two input tracks - bool SetCombinedTrack(const Track &track_a, - const Track &track_b); - - // Sets this track to be combination of three input tracks - bool SetCombinedTrack(const Track &track_a, - const Track &track_b, - const Track &track_c); - - // Pads a track by repeating the last frame num_pads times. - bool Pad(int num_pads); - - // Makes the current track and ref track equal lengths, use the longest length - // and zero pad the shorter track if the are different lengths. - void MakeSameSize(Track *track); - - // Makes the current track and ref track equal lengths, use the longest length - // and zero pad the shorter tracks if any are shorter than the longest. - void MakeSameSize(Track *track_a, - Track *track_b); - - private: - // Holds the number of frames of this track instance. - int num_frames_; - - // Holds the number of channels of this track instance. - int num_channels_; - - // Holds the frames values and times of this track instance. - FloatMatrix data_; - - // Holds the voicing flags of this track instance. - std::vector val_; - - // Holds whether this track instance has voicing enabled. - bool voicing_enabled_; - - // The frame interval in seconds. - float shift_; -}; - -// Applies logarithm to all data values. Invalid values are set to INVALID_LOG. -void ConvertToLogarithmic(Track *t); - -// Returns a new vector with the track data as a single sequence of -// values. The caller takes ownership fo the pointer. -float *TrackToFloatPointer(const Track &track, int *num_samples); - -inline int Track::num_frames() const { - return num_frames_; -} - -inline int Track::num_channels() const { - return num_channels_; -} - -inline float &Track::a(int f, int c) { - return data_.Get(c + 1, f); -} - -inline float Track::a(int f, int c) const { - return data_.Get(c + 1, f); -} - -inline void Track::set_v(int f, bool value) { - // If a writable reference is used, force the track into - // voicing enabled mode. - voicing_enabled_ = true; - val_[f] = value; -} - -inline bool Track::v(int f) const { - return val_[f]; -} - -inline float &Track::t(int f) { - return data_.Get(0, f); -} - -inline float Track::t(int f) const { - return data_.Get(0, f); -} - -inline void Track::set_shift(float shift) { - shift_ = shift; -} - -inline const FloatMatrix &Track::data() const { - return data_; -} - -inline bool Track::voicing_enabled() const { - return voicing_enabled_; -} - -inline void Track::set_voicing_enabled(bool b) { - voicing_enabled_ = b; -} - -inline bool Track::Save(const std::string &filename, bool ascii) const { - FileResource fr(filename, "wb"); - if (!fr.Get()) { - fprintf(stderr, "Failed to write '%s'", filename.c_str()); - return false; - } - if (!ascii) { - return Save(&fr); - } - const std::string data = ToString(); - if (fprintf(fr.fp(), "%s", data.c_str()) != data.size()) { - return false; - } - return true; -} - -template -void Track::Set(const std::vector ×, - const std::vector &values) { - if (times.size() != values.size()) { - fprintf(stderr, "Length of time and value vectors should equal (%d != %d)", - times.size(), values.size()); - return; - } - resize(times.size()); - for (int n = 0; n < times.size(); ++n) { - t(n) = times[n]; - a(n) = values[n]; - set_v(n, true); - } -} - -template -void Track::Set(float time_shift, const std::vector &values) { - resize(values.size()); - FillTime(time_shift, 0); - for (int n = 0; n < values.size(); ++n) { - a(n) = values[n]; - set_v(n, true); - } -} - -#endif // _TRACK_H_ diff --git a/reaper-tmp/REAPER/epoch_tracker/epoch_tracker.cc b/reaper-tmp/REAPER/epoch_tracker/epoch_tracker.cc deleted file mode 100644 index 0b640f536adbbe2ab8abce7133632617290e5fc1..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/epoch_tracker/epoch_tracker.cc +++ /dev/null @@ -1,1255 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ -// Author: dtalkin@google.com (David Talkin) - -// Implementation of the EpochTracker class. This does all of the -// processing necessary to estimate the F0, voicing state and epochs -// (glottal-closure instants) in human speech signals. See -// epoch_tracker.h for details. - -#include "epoch_tracker/epoch_tracker.h" - -#include -#include - -#include "epoch_tracker/fd_filter.h" -#include "epoch_tracker/lpc_analyzer.h" -#include "epoch_tracker/fft.h" - -const int kMinSampleRate = 6000; - -EpochTracker::EpochTracker(void) : sample_rate_(-1.0) { - SetParameters(); -} - -EpochTracker::~EpochTracker(void) { - CleanUp(); -} - -static inline int32_t RoundUp(float val) { - return static_cast(val + 0.5); -} - -void EpochTracker::CleanUp(void) { - for (size_t i = 0; i < resid_peaks_.size(); ++i) { - for (size_t j = 0; j < resid_peaks_[i].future.size(); ++j) { - delete resid_peaks_[i].future[j]; - } - } - resid_peaks_.clear(); - output_.clear(); - best_corr_.clear(); -} - -void EpochTracker::SetParameters(void) { - // Externally-settable control parameters: - // Period for the returned F0 signal. - external_frame_interval_ = kExternalFrameInterval; - do_highpass_ = kDoHighpass; // Enables highpassing of input signal. - // Enables Hilbert transformation of the input data. - do_hilbert_transform_ = kDoHilbertTransform; - max_f0_search_ = kMaxF0Search; // Maximum F0 to search for. - min_f0_search_ = kMinF0Search; // Minimum F0 to search for. - // Pulse spacing to use in unvoiced regions of the returned epoch signal. - unvoiced_pulse_interval_ = kUnvoicedPulseInterval; - debug_name_ = kDebugName; // base path for all debugging signals. - - // Internal feature-computation parameters: - // For internal feature computations - internal_frame_interval_ = kInternalFrameInterval; - // for the high-pass filter - corner_frequency_ = 80.0; - filter_duration_ = 0.05; - // for the LPC inverse filter. - frame_duration_ = 0.02; // window size (sec) - lpc_frame_interval_ = 0.01; // (sec) - preemphasis_ = 0.98; // preemphasis for LPC analysis - noise_floor_ = 70.0; // SNR in dB simulated during LPC analysis. - // for computing LPC residual peak quality. - peak_delay_ = 0.0004; // for measuring prominence - skew_delay_ = 0.00015; // for measuring shape - peak_val_wt_ = 0.1; - peak_prominence_wt_ = 0.3; - peak_skew_wt_ = 0.1; - peak_quality_floor_ = 0.01; - // for computing voice-transition pseudo-probabilities - time_span_ = 0.020; // the interval (sec) centered on the - // measurement point, used to compute parameter - // deltas. - level_change_den_ = 30.0; // max. dB level change expected over - // time_span_ for bandpassed RMS for - // computing pseudo-probability of - // voicing. - min_rms_db_ = 20.0; // level floor in dB - // window size for computing amplitude-normalizing RMS - ref_dur_ = 0.02; - // low and high frequency limits for bandpassed RMS used in voicing indicator - min_freq_for_rms_ = 100.0; - max_freq_for_rms_ = 1000.0; - // duration of integrator for bandpassed RMS - rms_window_dur_ = 0.025; - // window duration, in seconds, for NCCF computations - correlation_dur_ = 0.0075; - // ignore any NCCF peaks less than this - correlation_thresh_ = 0.2; - - // Parametrs used by the dynamic-programming tracker: - // reward for inserting another period - reward_ = -1.5; - // weight given to deviation of inter-pulse interval from the - // closest NCCF peak lag - period_deviation_wt_ = 1.0; - // weight given to the quality of the residual peak - peak_quality_wt_ = 1.3; - // cost of the unvoiced hypothesis - unvoiced_cost_ = kUnvoicedCost; - // cost of high NCCF values in hypothetical unvoiced regions - nccf_uv_peak_wt_ = 0.9; - // weight given to period length - period_wt_ = 0.0002; - // weight given to the pseudo-probability of voicing feature - level_wt_ = 0.8; - // weight given to period-length differences between adjacent periods. - freq_trans_wt_ = 1.8; - // cost of switching between voicing states; modulated by voicing - // onset/offset probs. - voice_transition_factor_ = 1.4; - - // Parameters used to generate final outputs: - // pad time in seconds to add to the last measured period during - // output of periodically-resampled data - endpoint_padding_ = 0.01; -} - -bool EpochTracker::Init(const int16_t* input, int32_t n_input, float sample_rate, - float min_f0_search, float max_f0_search, - bool do_highpass, bool do_hilbert_transform) { - if (input && (sample_rate > 6000.0) && (n_input > (sample_rate * 0.05)) && - (min_f0_search < max_f0_search) && (min_f0_search > 0.0)) { - CleanUp(); - min_f0_search_ = min_f0_search; - max_f0_search_ = max_f0_search; - sample_rate_ = sample_rate; - int16_t* input_p = const_cast(input); - if (do_highpass) { - input_p = HighpassFilter(input_p, n_input, sample_rate, - corner_frequency_, filter_duration_); - } - signal_.resize(n_input); - if (do_hilbert_transform) { - HilbertTransform(input_p, n_input, &(signal_.front())); - } else { - for (int32_t i = 0; i < n_input; ++i) { - signal_[i] = input_p[i]; - } - } - if (input_p != input) { - delete [] input_p; - } - return true; - } - return false; -} - -void EpochTracker::HilbertTransform(int16_t* input, int32_t n_input, - float* output) { - FFT ft = FFT(FFT::fft_pow2_from_window_size(n_input)); - int32_t n_fft = ft.get_fftSize(); - float* re = new float[n_fft]; - float* im = new float[n_fft]; - for (int i = 0; i < n_input; ++i) { - re[i] = input[i]; - im[i] = 0.0; - } - for (int i = n_input; i < n_fft; ++i) { - re[i] = 0.0; - im[i] = 0.0; - } - ft.fft(re, im); - for (int i = 1; i < n_fft/2; ++i) { - float tmp = im[i]; - im[i] = -re[i]; - re[i] = tmp; - } - re[0] = im[0] = 0.0; - for (int i = n_fft/2 + 1; i < n_fft; ++i) { - float tmp = im[i]; - im[i] = re[i]; - re[i] = -tmp; - } - ft.ifft(re, im); - for (int i = 0; i < n_input; ++i) { - output[i] = re[i] / n_fft; - } - delete [] re; - delete [] im; -} - - -int16_t* EpochTracker::HighpassFilter(int16_t* input, int32_t n_input, - float sample_rate, float corner_freq, - float fir_duration) { - FdFilter filter(sample_rate, corner_freq, true, fir_duration, false); - int16_t* filtered_data = new int16_t[n_input]; - int32_t max_buffer_size = filter.GetMaxInputSize(); - int32_t to_process = n_input; - bool start = true; - bool end = false; - int32_t input_index = 0; - int32_t output_index = 0; - while (to_process > 0) { - int32_t to_send = to_process; - if (to_send > max_buffer_size) { - to_send = max_buffer_size; - } else { - end = true; - } - int32_t samples_returned = filter.FilterArray(input + input_index, to_send, - start, end, - filtered_data + output_index, - n_input - output_index); - input_index += to_send; - to_process -= to_send; - output_index += samples_returned; - start = false; - } - return filtered_data; -} - - -static float LpcDcGain(float* lpc, int32_t order) { - float sum = 0.0; - for (int32_t i = 0; i <= order; ++i) { - sum += lpc[i]; - } - if (sum > 0.0) { - return sum; - } else { - return 1.0; - } -} - - -static void MakeDeltas(float* now, float* next, int32_t size, int32_t n_steps, - float* deltas) { - for (int32_t i = 0; i < size; ++i) { - deltas[i] = (next[i] - now[i]) / n_steps; - } -} - - -bool EpochTracker::GetLpcResidual(const std::vector& input, float sample_rate, - std::vector* output) { - int32_t n_input = input.size(); - if (!((n_input > 0) && (sample_rate > 0.0) && output)) { - return false; - } - output->resize(n_input); - int32_t frame_step = RoundUp(sample_rate * lpc_frame_interval_); - int32_t frame_size = RoundUp(sample_rate * frame_duration_); - int32_t n_frames = 1 + ((n_input - frame_size) / frame_step); - int32_t n_analyzed = ((n_frames - 1) * frame_step) + frame_size; - // Must have one more than frame size to do a complete frame. - if (n_analyzed <= n_input) { - n_frames--; - if (n_frames <= 0) { - return false; - } - } - LpcAnalyzer lp; - int32_t order = lp.GetLpcOrder(sample_rate); - float* lpc = new float[order + 1]; - float* old_lpc = new float[order + 1]; - float* delta_lpc = new float[order + 1]; - float norm_error = 0.0; - float preemp_rms = 0.0; - -#define RELEASE_MEMORY() { \ - delete [] lpc; \ - delete [] old_lpc; \ - delete [] delta_lpc; \ - } - - if (!lp.ComputeLpc(order, noise_floor_, frame_size, &(input.front()), - old_lpc, NULL, NULL, &norm_error, &preemp_rms, - preemphasis_)) { - RELEASE_MEMORY(); - return false; - } - for (int32_t i = 0; i <= order; ++i) { - delta_lpc[i] = 0.0; - (*output)[i] = 0.0; - } - float old_gain = LpcDcGain(old_lpc, order); - float new_gain = 1.0; - int32_t n_to_filter = (frame_size / 2) - order; // How many samples - // to process before - // computing the next - // LPC frame. - int32_t input_p = 0; // Where to get the next frame for LPC analysis - int32_t output_p = order; // where to store output samples. - int32_t proc_p = 0; // Where to pick up samples for input to the filter - - // Main processing loop: - // Compute a new frame of LPC - // Compute the DC gain for the new LPC - // Compute delta DC gain. - // Compute the LPC deltas. - // For each point in the frame: - // Use old_lpc to produce an output point. - // Update the old LPCs and the DC gain - // As soon as the center of the current frame is reached, compute - // the LPC for the next frame. - for ( ; n_frames > 0; --n_frames, input_p += frame_step, - n_to_filter = frame_step) { - if (!lp.ComputeLpc(order, noise_floor_, frame_size, - (&(input.front())) + input_p, lpc, NULL, NULL, - &norm_error, &preemp_rms, preemphasis_)) { - RELEASE_MEMORY(); - return false; - } - new_gain = LpcDcGain(lpc, order); - float delta_gain = (new_gain - old_gain) / n_to_filter; - MakeDeltas(old_lpc, lpc, order+1, n_to_filter, delta_lpc); - for (int32_t sample = 0; sample < n_to_filter; ++sample, ++proc_p, - ++output_p) { - float sum = 0.0; - int32_t mem = proc_p; - for (int32_t k = order; k > 0; --k, ++mem) { - sum += (old_lpc[k] * input[mem]); - old_lpc[k] += delta_lpc[k]; - } - sum += input[mem]; // lpc[0] is always 1.0 - (*output)[output_p] = sum / old_gain; - old_gain += delta_gain; - } - } - RELEASE_MEMORY(); - return true; -} - -// Note that GetResidualPulses assumes the LPC residual is in the -// "correct" polarity, with the GCI pulses of interest being negative -// pulses with a gradual fall and an abrupt rise. -void EpochTracker::GetResidualPulses(void) { - int32_t peak_ind = RoundUp(peak_delay_ * sample_rate_); - int32_t skew_ind = RoundUp(skew_delay_ * sample_rate_); - float min_peak = -1.0; // minimum value that will be considered as a peak - int32_t limit = norm_residual_.size() - peak_ind; - resid_peaks_.resize(0); - peaks_debug_.resize(residual_.size()); - for (size_t i = 0; i < peaks_debug_.size(); ++i) { - peaks_debug_[i] = 0.0; - } - for (int32_t i = peak_ind; i < limit; ++i) { - float val = norm_residual_[i]; - if (val > min_peak) { - continue; - } - if ((norm_residual_[i-1] > val) && (val <= norm_residual_[i+1])) { - float vm_peak = norm_residual_[i - peak_ind]; - float vp_peak = norm_residual_[i + peak_ind]; - if ((vm_peak < val) || (vp_peak < val)) { - continue; - } - float vm_skew = norm_residual_[i - skew_ind]; - float vp_skew = norm_residual_[i + skew_ind]; - float sharp = (0.5 * (vp_peak + vm_peak)) - val; - float skew = -(vm_skew - vp_skew); - ResidPeak p; - p.resid_index = i; - float time = static_cast(i) / sample_rate_; - p.frame_index = RoundUp(time / internal_frame_interval_); - if (p.frame_index >= n_feature_frames_) { - p.frame_index = n_feature_frames_ - 1; - } - p.peak_quality = (-val * peak_val_wt_) + (skew * peak_skew_wt_) + - (sharp * peak_prominence_wt_); - if (p.peak_quality < peak_quality_floor_) { - p.peak_quality = peak_quality_floor_; - } - resid_peaks_.push_back(p); - peaks_debug_[i] = p.peak_quality; - } - } -} - - -void EpochTracker::GetVoiceTransitionFeatures(void) { - int32_t frame_offset = RoundUp(0.5 * time_span_ / internal_frame_interval_); - if (frame_offset <= 0) { - frame_offset = 1; - } - voice_onset_prob_.resize(n_feature_frames_); - voice_offset_prob_.resize(n_feature_frames_); - int32_t limit = n_feature_frames_ - frame_offset; - for (int32_t frame = frame_offset; frame < limit; ++frame) { - float delta_rms = (bandpassed_rms_[frame + frame_offset] - - bandpassed_rms_[frame - frame_offset]) / level_change_den_; - if (delta_rms > 1.0) { - delta_rms = 1.0; - } else { - if (delta_rms < -1.0) { - delta_rms = -1.0; - } - } - float prob_onset = delta_rms; - float prob_offset = -prob_onset; - if (prob_onset > 1.0) { - prob_onset = 1.0; - } else { - if (prob_onset < 0.0) { - prob_onset = 0.0; - } - } - if (prob_offset > 1.0) { - prob_offset = 1.0; - } else { - if (prob_offset < 0.0) { - prob_offset = 0.0; - } - } - voice_onset_prob_[frame] = prob_onset; - voice_offset_prob_[frame] = prob_offset; - } - // Just set the onset and offset probs to zero in the end zones. - for (int32_t frame = 0; frame < frame_offset; ++frame) { - int32_t bframe = n_feature_frames_ - 1 - frame; - voice_onset_prob_[frame] = voice_offset_prob_[frame] = 0.0; - voice_onset_prob_[bframe] = voice_offset_prob_[bframe] = 0.0; - } -} - - -void EpochTracker::GetRmsVoicingModulator(void) { - float min_val = bandpassed_rms_[0]; - float max_val = min_val; - - prob_voiced_.resize(bandpassed_rms_.size()); - // Find the max and min over the whole RMS array. Scale and offset - // the RMS values to all fall in th range of 0.0 to 1.0. - for (size_t i = 1; i < bandpassed_rms_.size(); ++i) { - float val = bandpassed_rms_[i]; - if (val < min_val) { - min_val = val; - } else { - if (val > max_val) { - max_val = val; - } - } - } - if (min_val < min_rms_db_) { - min_val = min_rms_db_; - } - float range = max_val - min_val; - if (range < 1.0) { - range = 1.0; - } - for (size_t i = 0; i < bandpassed_rms_.size(); ++i) { - prob_voiced_[i] = (bandpassed_rms_[i] - min_val) / range; - if (prob_voiced_[i] < 0.0) { - prob_voiced_[i] = 0.0; - } - } -} - - -int32_t EpochTracker::FindNccfPeaks(const std::vector& input, float thresh, - std::vector* output) { - int32_t limit = input.size() - 1; - uint32_t n_peaks = 0; - float max_val = 0.0; - int16_t max_index = 1; - int16_t max_out_index = 0; - output->resize(0); - for (int16_t i = 1; i < limit; ++i) { - float val = input[i]; - if ((val > thresh) && (val > input[i-1]) && (val >= input[i+1])) { - if (val > max_val) { - max_val = val; - max_out_index = n_peaks; - max_index = i; - } - n_peaks++; - output->push_back(i); - } - } - // Be sure the highest peak is the first one in the array. - if ((n_peaks > 1) && (max_out_index > 0)) { - int16_t hold = (*output)[0]; - (*output)[0] = (*output)[max_out_index]; - (*output)[max_out_index] = hold; - } else { - if (n_peaks <= 0) { - n_peaks = 1; - output->push_back(max_index); - } - } - return n_peaks; -} - - -void EpochTracker::CrossCorrelation(const std::vector& data, int32_t start, - int32_t first_lag, int32_t n_lags, - int32_t size, std::vector* corr) { - const float* input = (&(data.front())) + start; - corr->resize(n_lags); - float energy = 0.0; // Zero-lag energy part of the normalizer. - for (int32_t i = 0; i < size; ++i) { - energy += input[i] * input[i]; - } - if (energy == 0.0) { // Bail out if no energy is found. - for (int32_t i = 0; i < n_lags; ++i) { - (*corr)[i] = 0.0; - } - return; - } - int32_t limit = first_lag + size; - double lag_energy = 0.0; // Energy at the period hypothesis lag. - for (int32_t i = first_lag; i < limit; ++i) { - lag_energy += input[i] * input[i]; - } - int32_t last_lag = first_lag + n_lags; - int32_t oind = 0; // Index for storing output values. - for (int32_t lag = first_lag; lag < last_lag; ++lag, ++oind) { - float sum = 0.0; - int32_t lag_ind = lag; - for (int32_t i = 0; i < size; ++i, ++lag_ind) { - sum += input[i] * input[lag_ind]; - } - if (lag_energy <= 0.0) - lag_energy = 1.0; - (*corr)[oind] = sum / sqrt(energy * lag_energy); - lag_energy -= input[lag] * input[lag]; // Discard old sample. - lag_energy += input[lag_ind] * input[lag_ind]; // Pick up the new sample. - } - return; -} - - -void EpochTracker::GetPulseCorrelations(float window_dur, float peak_thresh) { - first_nccf_lag_ = RoundUp(sample_rate_ / max_f0_search_); - int32_t max_lag = RoundUp(sample_rate_ / min_f0_search_); - n_nccf_lags_ = max_lag - first_nccf_lag_; - int32_t window_size = RoundUp(window_dur * sample_rate_); - int32_t half_wind = window_size / 2; - int32_t frame_size = window_size + max_lag; - - std::vector mixture; - mixture.resize(residual_.size()); - const float kMinCorrelationStep = 0.001; // Pulse separation - // before computing new - // correlation values. - const float kResidFract = 0.7; // Fraction of the residual to use. - const float kPcmFract = 1.0 - kResidFract; // Fraction of the input to use. - for (size_t i = 0; i < residual_.size(); ++i) { - mixture[i] = (kResidFract * residual_[i]) + (kPcmFract * signal_[i]); - } - - int32_t min_step = RoundUp(sample_rate_ * kMinCorrelationStep); - int32_t old_start = - (2.0 * min_step); - for (size_t peak = 0; peak < resid_peaks_.size(); ++peak) { - int32_t start = resid_peaks_[peak].resid_index - half_wind; - if (start < 0) { - start = 0; - } - size_t end = start + frame_size; - if ((end >= mixture.size()) || ((start - old_start) < min_step)) { - resid_peaks_[peak].nccf = resid_peaks_[peak - 1].nccf; - resid_peaks_[peak].nccf_periods = resid_peaks_[peak - 1].nccf_periods; - } else { - CrossCorrelation(mixture, start, first_nccf_lag_, n_nccf_lags_, - window_size, &(resid_peaks_[peak].nccf)); - FindNccfPeaks(resid_peaks_[peak].nccf, peak_thresh, - &(resid_peaks_[peak].nccf_periods)); - // Turn the peak indices from FindNccfPeaks into NCCF period hyps. - for (size_t i = 0; i < resid_peaks_[peak].nccf_periods.size(); ++i) { - resid_peaks_[peak].nccf_periods[i] += first_nccf_lag_; - } - old_start = start; - } - } -} - - -void EpochTracker::Window(const std::vector& input, int32_t offset, size_t size, - float* output) { - if (size != window_.size()) { - window_.resize(size); - float arg = 2.0 * M_PI / size; - for (size_t i = 0; i < size; ++i) { - window_[i] = 0.5 - (0.5 * cos((i + 0.5) * arg)); - } - } - const float* data = (&(input.front())) + offset; - for (size_t i = 0; i < size; ++i) { - output[i] = data[i] * window_[i]; - } -} - - -bool EpochTracker::GetBandpassedRmsSignal(const std::vector& input, - float sample_rate, - float low_limit, float high_limit, - float frame_interval, - float frame_dur, - std::vector* output_rms) { - size_t frame_step = RoundUp(sample_rate * frame_interval); - size_t frame_size = RoundUp(sample_rate * frame_dur); - size_t n_frames = 1 + ((input.size() - frame_size) / frame_step); - if (n_frames < 2) { - fprintf(stderr, "input too small (%d) in GetBandpassedRmsSignal\n", - static_cast(input.size())); - output_rms->resize(0); - return false; - } - output_rms->resize(n_frames); - FFT ft(FFT::fft_pow2_from_window_size(frame_size)); - int32_t fft_size = ft.get_fftSize(); - int32_t first_bin = RoundUp(fft_size * low_limit / sample_rate); - int32_t last_bin = RoundUp(fft_size * high_limit / sample_rate); - float* re = new float[fft_size]; - float* im = new float[fft_size]; - size_t first_frame = frame_size / (2 * frame_step); - if ((first_frame * 2 * frame_step) < frame_size) { - first_frame++; - } - for (size_t frame = first_frame; frame < n_frames; ++frame) { - Window(input, (frame - first_frame) * frame_step, frame_size, re); - for (size_t i = 0; i < frame_size; ++i) { - im[i] = 0.0; - } - for (int32_t i = frame_size; i < fft_size; ++i) { - re[i] = im[i] = 0.0; - } - ft.fft(re, im); - float rms = 20.0 * - log10(1.0 + ft.get_band_rms(re, im, first_bin, last_bin)); - (*output_rms)[frame] = rms; - if (frame == first_frame) { - for (size_t bframe = 0; bframe < first_frame; ++bframe) { - (*output_rms)[bframe] = rms; - } - } - } - delete [] re; - delete [] im; - return true; -} - - -void EpochTracker::GetSymmetryStats(const std::vector& data, float* pos_rms, - float* neg_rms, float* mean) { - int32_t n_input = data.size(); - double p_sum = 0.0; - double n_sum = 0.0; - double sum = 0.0; - int32_t n_p = 0; - int32_t n_n = 0; - for (int32_t i = 0; i < n_input; ++i) { - sum += data[i]; - } - *mean = sum / n_input; - for (int32_t i = 0; i < n_input; ++i) { - double val = data[i] - *mean; - if (val > 0.0) { - p_sum += (val * val); - n_p++; - } else { - if (val < 0.0) { - n_sum += (val * val); - n_n++; - } - } - } - *pos_rms = sqrt(p_sum / n_p); - *neg_rms = sqrt(n_sum / n_n); -} - - -void EpochTracker::NormalizeAmplitude(const std::vector& input, - float sample_rate, - std::vector* output) { - int32_t n_input = input.size(); - int32_t ref_size = RoundUp(sample_rate * ref_dur_); - std::vector wind; - - output->resize(n_input); - // Just calling Window here to create a Hann window in window_. - Window(input, 0, ref_size, &(output->front())); - int32_t ref_by_2 = ref_size / 2; - int32_t frame_step = RoundUp(sample_rate * internal_frame_interval_); - int32_t limit = n_input - ref_size; - int32_t frame_limit = ref_by_2; - int32_t data_p = 0; - int32_t frame_p = 0; - double old_inv_rms = 0.0; - while (frame_p < limit) { - double ref_energy = 1.0; // to prevent divz - for (int32_t i = 0; i < ref_size; ++i) { - double val = window_[i] * input[i + frame_p]; - ref_energy += (val * val); - } - double inv_rms = sqrt(static_cast(ref_size) / ref_energy); - double delta_inv_rms = 0.0; - if (frame_p > 0) { - delta_inv_rms = (inv_rms - old_inv_rms) / frame_step; - } else { - old_inv_rms = inv_rms; - } - for (int i = 0; i < frame_limit; ++i, ++data_p) { - (*output)[data_p] = input[data_p] * old_inv_rms; - old_inv_rms += delta_inv_rms; - } - frame_limit = frame_step; - frame_p += frame_step; - } - for ( ; data_p < n_input; ++data_p) { - (*output)[data_p] = input[data_p] * old_inv_rms; - } -} - -bool EpochTracker::ComputePolarity(int *polarity) { - if (sample_rate_ <= 0.0) { - fprintf(stderr, "EpochTracker not initialized in ComputeFeatures\n"); - return false; - } - if (!GetBandpassedRmsSignal(signal_, sample_rate_, min_freq_for_rms_, - max_freq_for_rms_, internal_frame_interval_, - rms_window_dur_, &bandpassed_rms_)) { - fprintf(stderr, "Failure in GetBandpassedRmsSignal\n"); - return false; - } - if (!GetLpcResidual(signal_, sample_rate_, &residual_)) { - fprintf(stderr, "Failure in GetLpcResidual\n"); - return false; - } - float mean = 0.0; - GetSymmetryStats(residual_, &positive_rms_, &negative_rms_, &mean); - *polarity = -1; - if (positive_rms_ > negative_rms_) { - *polarity = 1; - } - return true; -} - -bool EpochTracker::ComputeFeatures(void) { - if (sample_rate_ <= 0.0) { - fprintf(stderr, "EpochTracker not initialized in ComputeFeatures\n"); - return false; - } - if (!GetBandpassedRmsSignal(signal_, sample_rate_, min_freq_for_rms_, - max_freq_for_rms_, internal_frame_interval_, - rms_window_dur_, &bandpassed_rms_)) { - fprintf(stderr, "Failure in GetBandpassedRmsSignal\n"); - return false; - } - if (!GetLpcResidual(signal_, sample_rate_, &residual_)) { - fprintf(stderr, "Failure in GetLpcResidual\n"); - return false; - } - n_feature_frames_ = bandpassed_rms_.size(); - float mean = 0.0; - GetSymmetryStats(residual_, &positive_rms_, &negative_rms_, &mean); - fprintf(stdout, "Residual symmetry: P:%f N:%f MEAN:%f\n", - positive_rms_, negative_rms_, mean); - if (positive_rms_ > negative_rms_) { - fprintf(stdout, "Inverting signal\n"); - for (size_t i = 0; i < residual_.size(); ++i) { - residual_[i] = -residual_[i]; - signal_[i] = -signal_[i]; - } - } - NormalizeAmplitude(residual_, sample_rate_, &norm_residual_); - GetResidualPulses(); - GetPulseCorrelations(correlation_dur_, correlation_thresh_); - GetVoiceTransitionFeatures(); - GetRmsVoicingModulator(); - return true; -} - - -bool EpochTracker::TrackEpochs(void) { - CreatePeriodLattice(); - DoDynamicProgramming(); - return BacktrackAndSaveOutput(); -} - - -void EpochTracker::CreatePeriodLattice(void) { - int32_t low_period = RoundUp(sample_rate_ / max_f0_search_); - int32_t high_period = RoundUp(sample_rate_ / min_f0_search_); - int32_t total_cands = 0; - - // For each pulse in the normalized residual... - for (size_t peak = 0; peak < resid_peaks_.size(); ++peak) { - size_t frame_index = resid_peaks_[peak].frame_index; - size_t resid_index = resid_peaks_[peak].resid_index; - int32_t min_period = resid_index + low_period; - int32_t max_period = resid_index + high_period; - float lowest_cost = 1.0e30; - float time = resid_index / sample_rate_; - int32_t best_nccf_period = resid_peaks_[peak].nccf_periods[0]; - float best_cc_val = - resid_peaks_[peak].nccf[best_nccf_period - first_nccf_lag_]; - best_corr_.push_back(time); - best_corr_.push_back(best_cc_val); - EpochCand* uv_cand = new EpochCand; // pre-allocate an unvoiced candidate. - uv_cand->voiced = false; - uv_cand->start_peak = peak; - uv_cand->cost_sum = 0.0; - uv_cand->local_cost = 0.0; - uv_cand->best_prev_cand = -1; - int32_t next_cands_created = 0; - // For each of the next residual pulses in the search range... - for (size_t npeak = peak + 1; npeak < resid_peaks_.size(); ++npeak) { - int32_t iperiod = resid_peaks_[npeak].resid_index - resid_index; - if (resid_peaks_[npeak].resid_index >= min_period) { - float fperiod = iperiod; - // Find the NCCF period that most closely matches. - int32_t cc_peak = 0; - float min_period_diff = fabs(log(fperiod / best_nccf_period)); - for (size_t cc_peak_ind = 1; - cc_peak_ind < resid_peaks_[peak].nccf_periods.size(); - ++cc_peak_ind) { - int32_t nccf_period = resid_peaks_[peak].nccf_periods[cc_peak_ind]; - float test_diff = fabs(log(fperiod / nccf_period)); - if (test_diff < min_period_diff) { - min_period_diff = test_diff; - cc_peak = cc_peak_ind; - } - } - // Generate a forward-period candidate. Grade the candidate - // on closeness to a NCCF period hyp, value of the NCCF, - // values of the candidate endpoint peaks. - EpochCand* v_cand = new EpochCand; - v_cand->voiced = true; - v_cand->period = iperiod; - int32_t cc_index = iperiod - first_nccf_lag_; - float cc_value = 0.0; - // If this period is in the normal search range, retrieve the - // actual NCCF value for that lag. - if ((cc_index >= 0) && (cc_index < n_nccf_lags_)) { - cc_value = resid_peaks_[peak].nccf[cc_index]; - } else { // punt and use the "closest" nccf peak - int32_t peak_cc_index = resid_peaks_[peak].nccf_periods[cc_peak] - - first_nccf_lag_; - cc_value = resid_peaks_[peak].nccf[peak_cc_index]; - } - float per_dev_cost = period_deviation_wt_ * min_period_diff; - float level_cost = level_wt_ * (1.0 - prob_voiced_[frame_index]); - float period_cost = fperiod * period_wt_; - float peak_qual_cost = peak_quality_wt_ / - (resid_peaks_[npeak].peak_quality + resid_peaks_[peak].peak_quality); - float local_cost = (1.0 - cc_value) + per_dev_cost + peak_qual_cost + - level_cost + period_cost + reward_; - v_cand->local_cost = local_cost; - if (local_cost < lowest_cost) { - lowest_cost = local_cost; - // Evaluate this best voiced period as an unvoiced - // hypothesis. (There are always plenty of poor - // voiced candidates!) - uv_cand->period = iperiod; - level_cost = level_wt_ * prob_voiced_[frame_index]; - uv_cand->local_cost = (nccf_uv_peak_wt_ * cc_value) + - level_cost + unvoiced_cost_ + reward_; - uv_cand->end_peak = npeak; - uv_cand->closest_nccf_period = - resid_peaks_[peak].nccf_periods[cc_peak]; - } - v_cand->start_peak = peak; - v_cand->end_peak = npeak; - v_cand->closest_nccf_period = resid_peaks_[peak].nccf_periods[cc_peak]; - v_cand->cost_sum = 0.0; - v_cand->best_prev_cand = -1; - resid_peaks_[peak].future.push_back(v_cand); - resid_peaks_[npeak].past.push_back(v_cand); - total_cands++; - next_cands_created++; - if (resid_peaks_[npeak].resid_index >= max_period) { - break; // Exit the search only after at least one peak has - // been found, even if it is necessary to go beyond - // the nominal maximum period. - } - } // end if this period is >= minimum search period. - } // end for each next pulse in the global period-search range. - // Install the unvoiced candidate for this pulse. - if (next_cands_created) { // Register the unvoiced hyp iff there - // was at least one voiced hyp. - resid_peaks_[peak].future.push_back(uv_cand); - resid_peaks_[uv_cand->end_peak].past.push_back(uv_cand); - total_cands++; - } else { - delete uv_cand; - } - // Now all residual-pulse period hyps that start at the current - // pulse have been generated. - - // If this pulse is one of the first few in the residual that had - // no possible preceeding periods, mark it as an origin. - if (resid_peaks_[peak].past.size() == 0) { // Is this pulse an origin? - for (size_t pp = 0; pp < resid_peaks_[peak].future.size(); ++pp) { - resid_peaks_[peak].future[pp]->cost_sum = - resid_peaks_[peak].future[pp]->local_cost; - resid_peaks_[peak].future[pp]->best_prev_cand = -1; - } - } else { // There are previous period hyps to consider... - // Check if at least one UV hyp is included in the period hyps - // that end on this peak. If there are none, generate one by - // cloning the best voiced hyp in the collection, but score it - // as unvoiced. - int32_t uv_hyps_found = 0; - float lowest_cost = resid_peaks_[peak].past[0]->local_cost; - size_t lowest_index = 0; - for (size_t pcand = 0; pcand < resid_peaks_[peak].past.size(); ++pcand) { - if (!resid_peaks_[peak].past[pcand]->voiced) { - uv_hyps_found++; - } else { - if (resid_peaks_[peak].past[pcand]->local_cost < lowest_cost) { - lowest_index = pcand; - lowest_cost = resid_peaks_[peak].past[pcand]->local_cost; - } - } - } - if (!uv_hyps_found) { // clone an UV hyp from the best V hyp found. - size_t start_peak = resid_peaks_[peak].past[lowest_index]->start_peak; - EpochCand* uv_cand = new EpochCand; - uv_cand->voiced = false; - uv_cand->start_peak = start_peak; - uv_cand->end_peak = peak; - uv_cand->period = resid_peaks_[peak].past[lowest_index]->period; - uv_cand->closest_nccf_period = - resid_peaks_[peak].past[lowest_index]->closest_nccf_period; - uv_cand->cost_sum = 0.0; - uv_cand->local_cost = 0.0; - uv_cand->best_prev_cand = -1; - float llevel_cost = level_wt_ * - prob_voiced_[resid_peaks_[start_peak].frame_index]; - int32_t lcc_index = uv_cand->period - first_nccf_lag_; - float lcc_value = 0.0; - // If this period is in the normal search range, retrieve the - // actual NCCF value for that lag. - if ((lcc_index >= 0) && (lcc_index < n_nccf_lags_)) { - lcc_value = resid_peaks_[start_peak].nccf[lcc_index]; - } else { - int32_t peak_cc_index = uv_cand->closest_nccf_period - first_nccf_lag_; - lcc_value = resid_peaks_[start_peak].nccf[peak_cc_index]; - } - uv_cand->local_cost = (nccf_uv_peak_wt_ * lcc_value) + llevel_cost + - unvoiced_cost_ + reward_; - resid_peaks_[start_peak].future.push_back(uv_cand); - resid_peaks_[peak].past.push_back(uv_cand); - total_cands++; - } - } - } // end of the first pass at all pulses in the residual. - // All forward period hypotheses that start on all residual pulses - // in the signal have now been generated, and both voiced and - // unvoiced continuity throughout the lattice of hyps have been - // assured. -} - - -void EpochTracker::DoDynamicProgramming(void) { - // Perform the dynamic programming iterations over all pulses in - // the residual. - // For each pulse in the residual.... - for (size_t peak = 0; peak < resid_peaks_.size(); ++peak) { - if (resid_peaks_[peak].past.size() == 0) { // Is this peak an origin? - continue; - } - // For each forward period hypothesis starting at this pulse... - for (size_t fhyp = 0; fhyp < resid_peaks_[peak].future.size(); ++fhyp) { - float min_cost = 1.0e30; // huge - size_t min_index = 0; - float forward_period = resid_peaks_[peak].future[fhyp]->period; - // For each of the previous period hyps ending on this pulse... - for (size_t phyp = 0; phyp < resid_peaks_[peak].past.size(); ++phyp) { - float sum_cost = 0.0; - // There are 4 voicing hyps to consider: V->V V->UV UV->V UV->UV - if (resid_peaks_[peak].future[fhyp]->voiced && - resid_peaks_[peak].past[phyp]->voiced) { // v->v - float f_trans_cost = freq_trans_wt_ * - fabs(log(forward_period / resid_peaks_[peak].past[phyp]->period)); - sum_cost = f_trans_cost + resid_peaks_[peak].past[phyp]->cost_sum; - } else { - if (resid_peaks_[peak].future[fhyp]->voiced && - !resid_peaks_[peak].past[phyp]->voiced) { // uv->v - float v_transition_cost = voice_transition_factor_ * - (1.0 - voice_onset_prob_[resid_peaks_[peak].frame_index]); - sum_cost = resid_peaks_[peak].past[phyp]->cost_sum + - v_transition_cost; - } else { - if ((!resid_peaks_[peak].future[fhyp]->voiced) && - resid_peaks_[peak].past[phyp]->voiced) { // v->uv - float v_transition_cost = voice_transition_factor_ * - (1.0 - voice_offset_prob_[resid_peaks_[peak].frame_index]); - sum_cost = resid_peaks_[peak].past[phyp]->cost_sum + - v_transition_cost; - } else { // UV->UV - sum_cost = resid_peaks_[peak].past[phyp]->cost_sum; - } - } - } - if (sum_cost < min_cost) { - min_cost = sum_cost; - min_index = phyp; - } - } // end for each previous period hyp - resid_peaks_[peak].future[fhyp]->cost_sum = - resid_peaks_[peak].future[fhyp]->local_cost + min_cost; - resid_peaks_[peak].future[fhyp]->best_prev_cand = min_index; - } // end for each foreward period hyp - } // end for each pulse in the residual signal. - // Here ends the dynamic programming. -} - - -bool EpochTracker::BacktrackAndSaveOutput(void) { - if (resid_peaks_.size() == 0) { - fprintf(stderr, "Can't backtrack with no residual peaks\n"); - return false; - } - // Now find the best period hypothesis at the end of the signal, - // and backtrack from there. - float min_cost = 1.0e30; - int32_t min_index = 0; - // First, find a terminal peak which is the end of more than one - // period candidate. - size_t end = 0; - for (size_t peak = resid_peaks_.size() - 1; peak > 0; --peak) { - if ((resid_peaks_[peak].past.size() > 1)) { - for (size_t ind = 0; ind < resid_peaks_[peak].past.size(); ++ind) { - if (resid_peaks_[peak].past[ind]->cost_sum < min_cost) { - min_cost = resid_peaks_[peak].past[ind]->cost_sum; - min_index = ind; - } - } - end = peak; - break; - } - } - if (end == 0) { - fprintf(stderr, "No terminal peak found in DynamicProgramming\n"); - return false; - } - output_.clear(); - // Backtrack through the best pointers to retrieve the optimum - // period and voicing candidates. Save the GCI and voicing - // estimates. - while (1) { - int32_t start_peak = resid_peaks_[end].past[min_index]->start_peak; - TrackerResults tr; - tr.resid_index = resid_peaks_[start_peak].resid_index; - if (resid_peaks_[end].past[min_index]->voiced) { - float nccf_period = - resid_peaks_[end].past[min_index]->closest_nccf_period; - // TODO(dtalkin) If the closest NCCF period is more than epsilon - // different from the inter-pulse interval, use the inter-pulse - // interval instead. - tr.f0 = sample_rate_ / nccf_period; - tr.voiced = true; - } else { - tr.f0 = 0.0; - tr.voiced = false; - } - int32_t cc_index = resid_peaks_[end].past[min_index]->period - - first_nccf_lag_; - // If this period is in the normal search range, retrieve the - // actual NCCF value for that lag. - if ((cc_index >= 0) && (cc_index < n_nccf_lags_)) { - tr.nccf_value = resid_peaks_[start_peak].nccf[cc_index]; - } else { - int32_t peak_cc_index = - resid_peaks_[end].past[min_index]->closest_nccf_period - - first_nccf_lag_; - tr.nccf_value = resid_peaks_[start_peak].nccf[peak_cc_index]; - } - output_.push_back(tr); - size_t new_end = resid_peaks_[end].past[min_index]->start_peak; - min_index = resid_peaks_[end].past[min_index]->best_prev_cand; - if (min_index < 0) { // Has an origin pulse been reached? - break; - } - end = new_end; - } - // NOTE: The output_ array is in reverse time order! - return true; -} - - -void EpochTracker::GetFilledEpochs(float unvoiced_pm_interval, - std::vector* times, - std::vector* voicing) { - times->clear(); - voicing->clear(); - float final_time = norm_residual_.size() / sample_rate_; - int32_t limit = output_.size() - 1; - int32_t i = limit; - // Produce the output in normal time order. - while (i >= 0) { - int32_t i_old = i; - float time = output_[i].resid_index / sample_rate_; - // Note that the pulse locations of both the beginning and end - // of any voiced period are of interest. - if (output_[i].voiced || ((i < limit) && (output_[i+1].voiced))) { - times->push_back(time); - voicing->push_back(1); - i--; - } - if (i == limit) { - time = 0.0; - } - if ((i > 0) && (!output_[i].voiced) && (time < final_time)) { - for ( ; i > 0; --i) { - if (output_[i].voiced) { - break; - } - } - float next_time = final_time; - int32_t fill_ind = 1; - if (i > 0) { - next_time = (output_[i].resid_index / sample_rate_) - - (1.0 / max_f0_search_); - } - float now = time + (fill_ind * unvoiced_pm_interval); - while (now < next_time) { - times->push_back(now); - voicing->push_back(0); - fill_ind++; - now = time + (fill_ind * unvoiced_pm_interval); - } - } - if (i == i_old) { - i--; - } - } -} - - -bool EpochTracker::ResampleAndReturnResults(float resample_interval, - std::vector* f0, - std::vector* correlations) { - if ((sample_rate_ <= 0.0) || (output_.size() == 0)) { - fprintf(stderr, - "Un-initialized EpochTracker or no output_ in ResampleAndReturnF0\n"); - return false; - } - if (resample_interval <= 0.0) { - fprintf(stderr, "resample_interval <= 0.0 in ResampleAndReturnF0\n"); - return false; - } - float last_time = (output_[0].resid_index / sample_rate_) + endpoint_padding_; - int32_t n_frames = RoundUp(last_time / resample_interval); - f0->resize(0); - correlations->resize(0); - f0->insert(f0->begin(), n_frames, 0.0); - correlations->insert(correlations->begin(), n_frames, 0.0); - int32_t limit = output_.size() - 1; - int32_t prev_frame = 0; - float prev_f0 = output_[limit].f0; - float prev_corr = output_[limit].nccf_value; - for (int32_t i = limit; i >= 0; --i) { - int32_t frame = RoundUp(output_[i].resid_index / - (sample_rate_ * resample_interval)); - (*f0)[frame] = output_[i].f0; - (*correlations)[frame] = output_[i].nccf_value; - if ((frame - prev_frame) > 1) { - for (int32_t fr = prev_frame + 1; fr < frame; ++fr) { - (*f0)[fr] = prev_f0; - (*correlations)[fr] = prev_corr; - } - } - prev_frame = frame; - prev_corr = output_[i].nccf_value; - prev_f0 = output_[i].f0; - } - for (int32_t frame = prev_frame; frame < n_frames; ++frame) { - (*f0)[frame] = prev_f0; - (*correlations)[frame] = prev_corr; - } - return true; -} - - -bool EpochTracker::WriteDebugData(const std::vector& data, - const std::string& extension) { - if (debug_name_.empty()) { - return true; - } - std::string filename = debug_name_ + "." + extension; - if (data.size() == 0) { - fprintf(stdout, "Data size==0 for %s in WriteDebugData\n", - filename.c_str()); - return false; - } - FILE* out = fopen(filename.c_str(), "w"); - if (!out) { - fprintf(stderr, "Can't open %s for debug output\n", filename.c_str()); - return false; - } - size_t written = fwrite(&(data.front()), sizeof(data.front()), - data.size(), out); - fclose(out); - if (written != data.size()) { - fprintf(stderr, "Problems writing debug data (%d %d)\n", - static_cast(written), static_cast(data.size())); - return false; - } - return true; -} - -bool EpochTracker::WriteDiagnostics(const std::string& file_base) { - if (!file_base.empty()) { - set_debug_name(file_base); - } - WriteDebugData(signal_, "pcm"); - WriteDebugData(residual_, "resid"); - WriteDebugData(norm_residual_, "nresid"); - WriteDebugData(bandpassed_rms_, "bprms"); - WriteDebugData(voice_onset_prob_, "onsetp"); - WriteDebugData(voice_offset_prob_, "offsetp"); - WriteDebugData(peaks_debug_, "pvals"); - WriteDebugData(prob_voiced_, "pvoiced"); - // best_corr_ is only available after CreatePeriodLattice. - WriteDebugData(best_corr_, "bestcorr"); - // NOTE: if WriteDiagnostics is called before the - // DynamicProgramming, there will be nothing in output_. - if ((!debug_name_.empty()) && (output_.size() > 2)) { - std::string pm_name = debug_name_ + ".pmlab"; - FILE* pmfile = fopen(pm_name.c_str(), "w"); - fprintf(pmfile, "#\n"); - std::vector f0; - int32_t limit = output_.size() - 1; - // Produce debug output in normal time order. - for (int32_t i = limit; i >= 0; --i) { - float time = output_[i].resid_index / sample_rate_; - // Note that the pulse locations of both the beginning and end - // of any voiced period are of interest. - if (output_[i].voiced || ((i < limit) && (output_[i+1].voiced))) { - fprintf(pmfile, "%f blue \n", time); - } else { - fprintf(pmfile, "%f red \n", time); - } - f0.push_back(time); - f0.push_back(output_[i].f0); - f0.push_back(output_[i].nccf_value); - } - fclose(pmfile); - WriteDebugData(f0, "f0ap"); - } - return true; -} diff --git a/reaper-tmp/REAPER/epoch_tracker/epoch_tracker.h b/reaper-tmp/REAPER/epoch_tracker/epoch_tracker.h deleted file mode 100644 index 22d7a20111f1339ac94e20402676e2f13e77856c..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/epoch_tracker/epoch_tracker.h +++ /dev/null @@ -1,478 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ -// Author: dtalkin@google.com (David Talkin) - -// EpochTracker estimates the location of glottal closure instants -// (GCI), also known as "epochs" from digitized acoustic speech -// signals. It simultaneously estimates the local fundamental -// frequency (F0) and voicing state of the speech on a per-epoch -// basis. Various output methods are available for retrieving the -// results. -// -// The processing stages are: -// * Optionally highpass the signal at 80 Hz to remove rumble, etc. -// * Compute the LPC residual, obtaining an approximation of the -// differentiated glottal flow. -// * Normalize the amplitude of the residual by a local RMS measure. -// * Pick the prominent peaks in the glottal flow, and grade them by -// peakiness, skew and relative amplitude. -// * Compute correlates of voicing to serve as pseudo-probabilities -// of voicing, voicing onset and voicing offset. -// * For every peak selected from the residual, compute a normalized -// cross-correlation function (NCCF) of the LPC residual with a -// relatively short reference window centered on the peak. -// * For each peak in the residual, hypothesize all following peaks -// within a specified F0 seaqrch range, that might be the end of a -// period starting on that peak. -// * Grade each of these hypothesized periods on local measures of -// "voicedness" using the NCCF and the pseudo-probability of voicing -// feature. -// * Generate an unvoiced hypothesis for each period and grade it -// for "voicelessness". -// * Do a dynamic programming iteration to grade the goodness of -// continuity between all hypotheses that start on a peak and -// those that end on the same peak. For voiced-voiced -// connections add a cost for F0 transitions. For -// unvoiced-voiced and voiced-unvoiced transitions add a cost -// that is modulated by the voicing onset or offset inverse -// pseudo-probability. Unvoiced-unvoiced transitions incur no cost. -// * Backtrack through the lowest-cost path developed during the -// dynamic-programming stage to determine the best peak collection -// in the residual. At each voiced peak, find the peak in the NCCF -// (computed above) that corresponds to the duration closest to the -// inter-peak interval, and use that as the inverse F0 for the -// peak. -// -// A typical calling sequence might look like: -/* ============================================================== - EpochTracker et; - et.Init(); // Prepare the instance for, possibly, multiple calls. - Track* f0; // for returning the F0 track - Track* pm; // for returning the epoch track - if (!et.ComputeEpochs(my_input_waveform, &pm, &f0)) { - exit(-2); // problems in the epoch computations - } - DoSomethingWithTracks(f0, pm); - delete f0; - delete pm; - ============================================================== */ -// -// NOTE: Any client of this code inherits the Google command-line flags -// defined in epoch_tracker.cc. These flags are processed in the Init() -// method, and override both default and params-sourced settings. -// -// As currently written, this is a batch process. Very little has -// been done to conserve either memory or CPU. The aim was simply to -// make the best possible tracker. As will be seen in the -// implementation, there are many parameters that can be adjusted to -// influence the processing. It is very unlikely that the best -// parameter setting is currently expressed in the code! However, the -// performance, as written, appears to be quite good on a variety of voices. - -#ifndef _EPOCH_TRACKER_H_ -#define _EPOCH_TRACKER_H_ - -#include -#include -#include -#include - -static const float kExternalFrameInterval = 0.005; -static const float kInternalFrameInterval = 0.002; -static const float kMinF0Search = 40.0; -static const float kMaxF0Search = 500.0; -static const float kUnvoicedPulseInterval = 0.01; -static const float kUnvoicedCost = 0.9; -static const bool kDoHighpass = true; -static const bool kDoHilbertTransform = false; -static const char kDebugName[] = ""; - - -class EpochTracker { - public: - EpochTracker(void); - - virtual ~EpochTracker(void); - - // Set the default operating parameters of the tracker. - void SetParameters(void); - - // NOTE: The following methods are exposed primarily for algorithm - // development purposes, where EpochTracker is used in a developer's test - // harness. These need not/should not be called directly in normal use. - - // Prepare the instance for use. Some sanity check is made on the - // parameters, and the instance is reset so it can be reused - // multiple times by simply calling Init() for each new input - // signal. frame_interval determines the framing for some of the - // internal feature computations, and for the periodic resampling of - // F0 that will occur during final tracking result output. min_ - // and max_f0_search are the bounding values, in Hz, for the F0 - // search. - // NOTE: This Init method is DEPRECATED, and is only retained to - // support legacy code. IT MAY GO AWAY SOON. This is NOT to be - // used with ComputeEpochs(). - bool Init(const int16_t* input, int32_t n_input, float sample_rate, - float min_f0_search, float max_f0_search, - bool do_highpass, bool do_hilbert_transform); - - // Set the name for various intermediate features and other signals - // that may be written to files used during debug and development of - // the algorithm. If this is set to the empty std::string, no debug - // signals will be output. - void set_debug_name(const std::string& debug_name) { - debug_name_ = debug_name; - } - - std::string debug_name(void) { return debug_name_; } - - // Compute the Hilbert transform of the signal in input, and place - // the floating-point results in output. output must be at least - // n_input samples long. TODO(dtalkin): Make these vector inputs - // and outputs. - void HilbertTransform(int16_t* input, int32_t n_input, float* output); - - // Apply a highpass filter to the signal in input. The filter - // corner frequency is corner_freq, and the duration, in seconds, of - // the Hann-truncated symmetric FIR is in fir_duration. The - // transition bandwidth is the inverse of fir_duration. The return - // value is a pointer to the filtered result, which is the same - // length as the input (n_input). It is up to the caller to free - // this returned memory. TODO(dtalkin): Make this vector I/O and - // supply the output as floats. - int16_t* HighpassFilter(int16_t* input, int32_t n_input, - float sample_rate, float corner_freq, - float fir_duration); - - // Compute the LPC residual of the speech signal in input. - // sample_rate is the rate of both the input and the residual to be - // placed in output. The order of the LPC analysis is automatically - // set to be appropriate for the sample rate, and the output is - // integrated so it approximates the derivative of the glottal flow. - bool GetLpcResidual(const std::vector& input, float sample_rate, - std::vector* output); - - // Compute the normalized cross-correlation function (NCCF) of the - // signal in data, starting at sample start. size is the number of - // samples to include in the inner product. Compute n_lags - // contiguous correlations starting at a delay of first_lag samples. - // Return the resulting n_lags correlation values in corr. Note - // that the NCCF is bounded by +-1.0. - void CrossCorrelation(const std::vector& data, int32_t start, - int32_t first_lag, int32_t n_lags, - int32_t size, std::vector* corr); - - // Compute the band-limited RMS of the signal in input, which is - // sampled at sample_rate. low_limit and high_limit are the - // frequency bounds, in Hz, within which th RMS is measured. - // frame_interval is the period of the RMS signal returned in - // output_rms. frame_dur is the duration, in seconds, of the - // Hanning window used for each measurement. - bool GetBandpassedRmsSignal(const std::vector& input, float sample_rate, - float low_limit, float high_limit, float frame_interval, - float frame_dur, std::vector* output_rms); - - // Compute the RMS of positive and negative signal values separately. - // The signal is made to be zero mean before this computation. Any - // imbalance in these measures is an indication of asymmetrical peak - // distribution, which is charactristic of the LPC residual of voiced speech. - void GetSymmetryStats(const std::vector& data, float* pos_rms, - float* neg_rms, float* mean); - - // Normalize the input signal based on a local measure of its RMS. - void NormalizeAmplitude(const std::vector& input, float sample_rate, - std::vector* output); - - // Apply a Hann weighting to the signal in input starting at - // sample index offset. The window will contain size samples, and - // the windowed signal is placed in output. - void Window(const std::vector& input, int32_t offset, size_t size, - float* output); - - // Computes signal polarity (-1 for negative, +1 for - // positive). Requires data to be initialized via Init(...). Returns - // false if there's an error. - bool ComputePolarity(int *polarity); - - // Compute NCCF, NCCF peak locations and values, bandpass RMS, - // residual, symmetry statistics (and invert residual, if necessary), - // normalized residual, residual peaks and values. Finally, generate - // the pulse working array in preparation for dynamic programming. - bool ComputeFeatures(void); - - // Write all of data to a file, wht name of which is - // debug_name_ _ "." + extension. If debug_name_ is empty, do nothing. - bool WriteDebugData(const std::vector& data, - const std::string& extension); - - // Write a collection of debugging signals to separate files with - // various, internally-defined name extensions. If file_base is not - // empty, use this as the base path for all of the files. If file - // base is empty, use debug_name_ as the base path. If both are - // empty, do nothing. - bool WriteDiagnostics(const std::string& file_base); - - // After Init, ComputeFeatures and CreatePeriodLattice have been - // successfully called, TrackEpochs should be called to do the - // actual tracking of epochs (GCI) and to estimate the corresponding - // F0. This method integrates the information from all of the - // features, including the LPC residual peaks and the NCCF values, - // to find the optimum period assignments and voicing state - // assignments over the entire signal. The results are left in - // internal storage, pending retrieval by other methods. - bool TrackEpochs(void); - - // Create a lattice of glottal period hypotheses in preparation for - // dynamic programming. This fills out most of the data fields in - // resid_peaks_. This must be called after ComputeFeatures. - void CreatePeriodLattice(void); - - // Apply the Viterbi dynamic programming algorithm to find the best - // path through the period hypothesis lattice created by - // CreatePeriodLattice. The backpointers and cumulative scores are - // left in the relevant fields in resid_peaks_. - void DoDynamicProgramming(void); - - // Backtrack through the best pointers in the period hypothesis - // lattice created by CreatePeriodLattice and processed by - // DoDynamicProgramming. The estimated GCI locations - // (epochs) and the corresponding F0 and voicing-states are placed - // in the output_ array pending retrieval using other methods. - bool BacktrackAndSaveOutput(void); - - // Resample the per-period F0 and correlation data that results from - // the tracker to a periodic signal at an interval of - // resample_interval seconds. Samples returned are those nearest in - // time to an epoch. Thus, if the resample_interval is greater than - // the local epoch interval, some epochs, and their period - // information, will be skipped. Conversely, if the - // resample_interval is less than the local epoch interval, - // measurements will be replicated as required. - bool ResampleAndReturnResults(float resample_interval, - std::vector* f0, - std::vector* correlations); - - // Convert the raw backtracking results in output_ into - // normal-time-order epoch markers. In unvoiced regions, fill with - // regularly-spaced pulses separated by unvoiced_pm_interval - // seconds. The epoch/pulse times are returned in times re the - // utterance beginning, and the corresponding voicing states in - // voicing (0=unvoiced; 1=voiced). This can only be called after - // TrackEpochs. - void GetFilledEpochs(float unvoiced_pm_interval, std::vector* times, - std::vector* voicing); - - // Setters. - void set_do_hilbert_transform(bool v) { do_hilbert_transform_ = v; } - void set_do_highpass(bool v) { do_highpass_ = v; } - void set_external_frame_interval(float v) { external_frame_interval_ = v; } - void set_unvoiced_pulse_interval(float v) { unvoiced_pulse_interval_ = v; } - void set_min_f0_search(float v) { min_f0_search_ = v; } - void set_max_f0_search(float v) { max_f0_search_ = v; } - void set_unvoiced_cost(float v) { unvoiced_cost_ = v; } - - private: - // Search the signal in norm_residual_ for prominent negative peaks. - // Grade the peaks on a combination of amplitude, "peakiness" and - // skew. (It is expected that the glottal pulses will - // have a relatively slow fall, and a rapid rise.) Place the - // selected and graded pulses in resid_peaks_. - void GetResidualPulses(void); - - // Create pseudo-probability functions in voice_onset_prob_ and - // voice_offset_prob_ that attempt to indicate the time-varying - // probability that a voice onset or offset is occurring. - // Presently, this is based solely on the derivative of the - // bandpassed RMS signal, bandpassed_rms_. - void GetVoiceTransitionFeatures(void); - - // Generate a pseudo-probability function that attempts to corespond - // to the probability that voicing is occurring. This is presently - // based solely on the bandpassed RMS signal, bandpassed_rms_. - void GetRmsVoicingModulator(void); - - // Free memory, and prepare the instance for a new signal. - void CleanUp(void); - - // Scan the signal in input searching for all local maxima that - // exceed thresh. The indices corresponding to the location of the - // peaks are placed in output. The first entry in output is always - // the location of the largest maximum found. - int32_t FindNccfPeaks(const std::vector& input, float thresh, - std::vector* output); - - // Compute the NCCF with the reference window centered on each of - // the residual pulses identified in GetResidualPulses. window_dur - // is the duration in seconds of the correlation inner product. - // After the NCCF for each residual pulse is computed, it is - // searched for local maxima that exceed peak_thresh. These peak - // locations and the full NCCF are saved in the corresponding - // elements of the resid_peaks_ array of structures. - void GetPulseCorrelations(float window_dur, float peak_thresh); - - - private: - // EpochCand stores all of the period hypotheses that can be - // generated from the peaks found in the LPC residual. It also - // maintains the cumulative path costs and backpointers generated - // during dynamic programming. - struct EpochCand { - int32_t period; // # of samples in this period candidate - float local_cost; // cost of calling this a period (or unvoiced) - float cost_sum; // cumulative cost from DP - int32_t start_peak; // index in resid_peaks_ where this period hyp starts - int32_t end_peak; // where this period ends - int32_t best_prev_cand; // backpointer used after DP - int32_t closest_nccf_period; // per. implied by the closest correlation peak - bool voiced; // hypothesized voicing state for this cand. - }; - - typedef std::vector CandList; - - // The ResidPeak stores data for each residual impulse. The array - // of these in resid_peaks_ serves as input to the dynamic - // programming search for GCI, voicing state and F0. - struct ResidPeak { - int32_t resid_index; // index into the resid_ array of this peak - int32_t frame_index; // index into the feature arrays for this peak - float peak_quality; // "goodness" measure for this peak - std::vector nccf; // the NCCF computed centered on this peak - std::vector nccf_periods; // periods implied by major peaks in nccf - CandList future; // period candidates that start on this peak - CandList past; // period candidates that end on this peak - }; - - struct TrackerResults { - bool voiced; - float f0; - int32_t resid_index; - float nccf_value; - }; - typedef std::vector TrackerOutput; - - protected: - std::vector resid_peaks_; // array of structures used to - // store the peak search lattice - TrackerOutput output_; // Array of time stamped results of the tracker. - // signal_, residual_, norm_residual and peaks_debug_ are all - // sampled at the original signal input sample_rate_. - std::vector signal_; // floating version of input speech signal - std::vector residual_; // LPC residual normalized for constant DC. - std::vector norm_residual_; // LPC residual normalized by its local RMS. - std::vector peaks_debug_; // for debug output of residual peak candidates - // bandpassed_rms_, voice_onset_prob_, voice_offset_prob_ and - // prob_voiced_ are all sampled with a period of internal_frame_interval_. - std::vector bandpassed_rms_; // RMS sampled at internal_frame_interval_ - std::vector voice_onset_prob_; // prob that a voice onset is occurring - std::vector voice_offset_prob_; // prob that a voice offset is occurring - std::vector prob_voiced_; // prob that voicing is occurring - std::vector best_corr_; // An array of best NCCF vals for all resid peaks. - std::vector window_; // Hann weighting array for Window() - float sample_rate_; // original input signal sample rate in Hz - - float positive_rms_; // RMS of all positive, non-zero samples in residual_ - float negative_rms_; // RMS of all negative, non-zero samples in residual_ - int32_t n_feature_frames_; // The number of feature frames available - // for all features computed at - // internal_frame_interval_. - int32_t first_nccf_lag_; // The index of the first correlation of the - // NCCF. This is determined by - // max_f0_search_. - int32_t n_nccf_lags_; // The number of correlations computed at each - // residual peak candidate. This is determined - // by max_f0_search_ and min_f0_search_. - std::string debug_name_; // The base path name for all debug output files. - - // Below are all of the parameters that control the functioning of - // the tracker. These are all set to default known-to-work values in - // SetParameters(). - - // Control parameters available to clients of EpochTracker. - float external_frame_interval_; // Frame interval for final output of F0. - float unvoiced_pulse_interval_; // Pulse interval in unvoiced regions - float min_f0_search_; // minimum F0 to search for (Hz) - float max_f0_search_; // maximum F0 to search for (Hz) - bool do_highpass_; // Highpass input sighal iff true. - bool do_hilbert_transform_; // Hilbert trans. input data iff true. - - // Internal feature-computation Parameters: - float internal_frame_interval_; // interval, in seconds, between frame onsets - // for the high-pass filter - float corner_frequency_; - float filter_duration_; - // for the LPC inverse filter. - float frame_duration_; // window size (sec) - float lpc_frame_interval_; // (sec) - float preemphasis_; // preemphasis for LPC analysis - float noise_floor_; // SNR in dB simulated during LPC analysis. - // for computing LPC residual peak quality. - float peak_delay_; // for measuring prominence - float skew_delay_; // for measuring shape - float peak_val_wt_; - float peak_prominence_wt_; - float peak_skew_wt_; - float peak_quality_floor_; - // for computing voice-transition pseudo-probabilities - float time_span_; // the interval (sec) centered on the - // measurement point, used to - // compute parameter deltas - float level_change_den_; // max. dB level change - // expected over time_span_ for - // bandpassed RMS - // for computing pseudo-probability of voicing - float min_rms_db_; // level floor in dB - // window size for computing amplitude-normalizing RMS - float ref_dur_; - // low and high frequency limits for bandpassed RMS used in voicing indicator - float min_freq_for_rms_; - float max_freq_for_rms_; - // duration of integrator for bandpassed RMS - float rms_window_dur_; - // window duration, in seconds, for NCCF computations - float correlation_dur_; - // ignore any NCCF peaks less than this - float correlation_thresh_; - - // Parametrs used by the dynamic-programming tracker: - // reward for inserting another period - float reward_; - // weight given to deviation of inter-pulse interval from the - // closest NCCF peak lag - float period_deviation_wt_; - // weight given to the quality of the residual peak - float peak_quality_wt_; - // cost of the unvoiced hypothesis - float unvoiced_cost_; - // cost of high NCCF values in hypothetical unvoiced regions - float nccf_uv_peak_wt_; - // weight given to period length - float period_wt_; - // weight given to the pseudo-probability of voicing feature - float level_wt_; - // weight given to period-length differences between adjacent periods. - float freq_trans_wt_; - // cost of switching between voicing states; modulated by voicing - // onset/offset probs. - float voice_transition_factor_; - - // Parameters used to generate final outputs: - // pad time in seconds to add to the last measured period during - // output of periodically-resampled data - float endpoint_padding_; -}; - - -#endif // _EPOCH_TRACKER_H_ diff --git a/reaper-tmp/REAPER/epoch_tracker/fd_filter.cc b/reaper-tmp/REAPER/epoch_tracker/fd_filter.cc deleted file mode 100644 index f37787aeb7dde82f067d30430fceda8e70305e5e..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/epoch_tracker/fd_filter.cc +++ /dev/null @@ -1,767 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -/* FdFilter: In-Line Filter class */ -/* -Author: David Talkin (dtalkin@google.com) -*/ - -/* -This class implements a general-purpose streaming FIR filter that is -applied in the frequency domain for speed/efficiency. Using -FFT-multiply-IFFT rather than a simple time-domain convolution greatly -speeds up the computations. The longer the impulse response of the -filter, the greater the advantage of this FFT approach over -time-domain convolution. - -This class also supports sample-rate conversion. For simple -input/output rate ratios, it is also quite efficient, but becomes -rather inefficient for ratios with denominator terms greater than 5. -It therefore limits the maximum denominator value to kMaxDecimation, -and thus can only approximate rate conversions that require larger -denominator terms. For large denominator ratios, or extreme -decimation, indexed time-domain implementations can be much faster. - -All filters implemented with FdFilter are symmetric FIR, and thus have -linear phase response. The filter is implemented non-causally, so -that there is no signal delay introduced by filtering or downsampling -operations. - -See the individual methods for details of use. See fd_filter_msin.cc -for a test harness and use examples. -*/ - -#include "epoch_tracker/fd_filter.h" - -#include -#include - -#include "epoch_tracker/fft.h" - -// kIoBufferSize can be any reasonable size. -static const int kIoBufferSize = 10000; - -// kMaxDecimation sets the limit on the maximum value of the denominator in the -// ratio that approximates the sample rate ratios when rate conversion -// is done. FdFilter is really not appropriate if the ratio needs to be -// greater than about 7, because time-domain approaches are then likely to be -// faster. However, perverse or adventuresome souls might want to try -// increasing this if they have extra computer cycles to waste. -static const int kMaxDecimation = 12; - -// Constructor when FdFilter is to be used as a high- or low-pass -// filter or as a sample-rate converter. 'input_freq' is the sample -// rate of the input signal in Hz. If input_freq != corner_freq and -// do_rate_conversion is true, FdFilter is confgured to convert the -// sample rate to corner_freq using a filter of length specified by -// filter_dur. -// -// If do_rate_conversion is false, FdFilter is configured to be a -// high- or low-pass filter. In this case, corner_freq is interpreted -// as the corner frequency of the filter, and must be in the range 0 < -// corner_freq < (input_freq/2). If do_highpass is true, the filter -// will be high-pass, else, low-pass. Again, filter_dur determines -// the filter length. in all cases, the filter length is specified in -// seconds, and determines the transition band width. The band width -// is approximately 1.0/filter_dur Hz wide. -/* ******************************************************************** */ -FdFilter::FdFilter(float input_freq, float corner_freq, bool do_highpass, - float filter_dur, bool do_rate_conversion) { - FdFilterInitialize(input_freq, corner_freq, do_highpass, filter_dur, - do_rate_conversion, NULL, NULL, 0); -} - -// Constructor when FdFilter will be used to implement a filter whose -// spectral magnitude values are listed in the plain text file with -// the name specified by 'spectrum_shape'. input_freq is the sample -// rate in Hz of the input signal. The file must have the following -// format: -// -// Line 1: pow2 nMags -// Lines 2 through (nMags+1): ind val -// -// nMags must equal ((1 << pow2)/2)+1, -// pow2 is typically in the range 3 < pow2 < 15. - -// The lines in the file containing (ind, val) pairs specify the -// magnitude response of the filter uniformly sampling the spectrum -// from 0 Hz to (input_freq/2) Hz. the 'ind' column is just a line index -// to make the file easily human readable, the 'val' column contains -// (positive) magnitude scaling values. Note that values > 1.0 cause -// an increase in output signal amplitude re the input signal at the -// corresponding frequency, and have the potential to cause clipping, -// if the input signal is too energetic at those frequencies. -/* ******************************************************************** */ -FdFilter::FdFilter(float input_freq, char *spectrum_shape) { - FdFilterInitialize(input_freq, input_freq, 0, 0.01, 0, spectrum_shape, - NULL, 0); -} - -// Constructor when FdFilter will be used to implement a filter whose -// 'n_magnitudes' spectral magnitude values are in the array 'spectrum_array'. -// n_magnitudes must equal ((1 << pow2)+1) for some pow2 in the -// range 2 < pow2 < 16. The values in spectrum_array are interpreted as -// above for the 'val' column in the 'spectrum_shape' file. -/* ******************************************************************** */ -FdFilter::FdFilter(float input_freq, float *spectrum_array, int n_magnitudes) { - FdFilterInitialize(input_freq, input_freq, 0, 0.01, 0, NULL, - spectrum_array, n_magnitudes); -} - -// This is the private method that configures FdFilter to satisfy the -// requirements of the constructor. -/* ----------------------------------------------------------------------- */ -void FdFilter::FdFilterInitialize(float input_freq, float corner_freq, - bool do_highpass, float filter_dur, - bool do_rate_conversion, - char *spectrum_shape, float *spectrum_array, - int n_magnitudes) { - float beta = 0.0, *b = NULL; - float ratio_t, ratio, freq1 = input_freq; - int pow2, i; - bool do_eq_filtering = false; - bool b_allocated = true; - - insert_ = 1; - decimate_ = 1; - filter_state_ = 1; - array_leftover_ = 0; - array_index_ = 0; - array_samples_used_ = 0; - true_output_rate_ = input_freq; - - output_buffer_ = new int16_t[kIoBufferSize * 2]; - input_buffer_ = new int16_t[kIoBufferSize]; - - if (spectrum_shape || (spectrum_array && (n_magnitudes > 1))) { - do_eq_filtering = true; - } else { - if (do_rate_conversion) { - freq1 = input_freq; - } else { /* it is just a symmetric FIR */ - if (corner_freq >= (freq1 = input_freq) / 2.0) { - fprintf(stderr, - "Unreasonable corner frequency specified to filter() (%f)\n", - corner_freq); - } - } - } - - if (spectrum_shape) { - FILE *spec_stream = fopen(spectrum_shape, "r"); - if (spec_stream) { - int n_spect, ind; - char line[500]; - float fs; - if (fgets(line, 500, spec_stream)) { - int nItems = sscanf(line, "%d %d %f", &pow2, &n_spect, &fs); - if ((nItems == 3) && (fs != input_freq)) { // This should be a - // fatal error! - fprintf(stderr, - "Filter spec (%f) does not match input frequency (%f)\n", - fs, input_freq); - fprintf(stderr, - "The filtering results will probably not be what you want!\n"); - } - b = new float[n_spect]; - n_filter_coeffs_ = n_spect - 1; // n_filter_coeffs_ represents actual - // filter-kernel length, - // instead of half filter - // length when using external - // filter. - for (i = 0; i < n_spect; i++) { - if ((!fgets(line, 500, spec_stream)) || - (sscanf(line, "%d %f", &ind, &(b[i])) != 2)) { - fprintf(stderr, "Parsing error in spect ratio file %s\n", - spectrum_shape); - } - } - } else { - fprintf(stderr, "Bad format in spectrum file %s\n", - spectrum_shape); - } - fclose(spec_stream); - } else { - fprintf(stderr, "Can't open %s as a spectrum file\n", - spectrum_shape); - } - } else { - // Note: n_magnitudes MUST be ((2^k)+1) for k > 1. - if (spectrum_array && (n_magnitudes > 1)) { - n_filter_coeffs_ = n_magnitudes - 1; - int nft = n_filter_coeffs_ * 2; - pow2 = 1; - while ((1 << pow2) < nft) { - pow2++; - } - b = spectrum_array; // Note: b must not be deleted in this case! - b_allocated = false; - } else { /* it is not an eq filter */ - if (do_rate_conversion) { - /* get a ratio of integers close to desired freq. ratio. */ - ratio = corner_freq/freq1; - RationalApproximation(ratio, &insert_, &decimate_, kMaxDecimation); - ratio_t = static_cast(insert_) / decimate_; - - if (fabs(1.0 - ratio_t) < .01) { - fprintf(stderr, - "Input and output frequencies are essentially equal!\n"); - } - true_output_rate_ = ratio_t * freq1; - // if (corner_freq != true_output_rate_) { - // fprintf(stderr, - // "Warning: Output frequency obtained(%f) is not as requested(%f)\n", - // true_output_rate_, corner_freq); - // } - corner_freq = true_output_rate_; - n_filter_coeffs_ = static_cast(freq1 * insert_ * filter_dur) | 1; - if (corner_freq < freq1) - beta = (.5 * corner_freq)/(insert_ * freq1); - else - beta = .5/insert_; - } else { - beta = corner_freq/freq1; - n_filter_coeffs_ = static_cast(freq1 * filter_dur) | 1; - } - - /* Generate the symmetric FIR filter coefficients. */ - b = new float[1 + (n_filter_coeffs_ / 2)]; - MakeLinearFir(beta, &n_filter_coeffs_, b); - - if (insert_ > 1) { // Scale up filter coeffs. to maintain - // precision in output. - float fact = insert_; - for (i = n_filter_coeffs_ / 2; i >= 0; i--) b[i] *= fact; - } - } // end else it is not an eq filter. - } // end else (a spectrum shape was not specified). - - n_filter_coeffs_by2_ = n_filter_coeffs_ / 2; - - if (!do_eq_filtering) { /* Is it a simple high- or low-pass filter? */ - MirrorFilter(b, do_highpass); - int nf2 = n_filter_coeffs_ << 1; - fft_size_ = 128; - pow2 = 7; - while (nf2 > fft_size_) { - fft_size_ *= 2; - pow2++; - } - } else { // It is a filter with the magnitude response specified in b. - fft_size_ = n_filter_coeffs_ * 2; - pow2 = 2; - while ((1 << pow2) < fft_size_) - pow2++; - } - - fft_size_by2_ = fft_size_ / 2; - - x_ = new float[fft_size_]; - y_ = new float[fft_size_]; - xf_ = new float[fft_size_]; - yf_ = new float[fft_size_]; - - leftovers_ = new int16_t[fft_size_]; - max_input_ = kIoBufferSize / insert_; - output_delayed_ = new float[(2 * kIoBufferSize)+n_filter_coeffs_+fft_size_]; - - float ftscale = 1.0 / fft_size_; - fft_ = new FFT(pow2); - if (!do_eq_filtering) { - // position the filter kernel to be symmetric about time=0 - // Note that this assumes an odd number of symmetric filter coefficients. - for (i = 0; i <= n_filter_coeffs_by2_; i++) { - xf_[i] = ftscale * filter_coeffs_[i+n_filter_coeffs_by2_]; - yf_[i] = 0.0; - } - for (; i < n_filter_coeffs_; i++) { - xf_[fft_size_ - n_filter_coeffs_ + i] = ftscale * - filter_coeffs_[i - n_filter_coeffs_by2_ - 1]; - yf_[fft_size_ - n_filter_coeffs_ + i] = 0.0; - } - for (i = n_filter_coeffs_by2_; i < (fft_size_-n_filter_coeffs_by2_); i++) - xf_[i] = yf_[i] = 0.0; - fft_->fft(xf_, yf_); - } else { /* Install the magnitude response symmetrically. */ - for (i = 0; i <= n_filter_coeffs_; i++) { - xf_[i] = ftscale * b[i]; - yf_[i] = 0.0; - } - for (; i < fft_size_; i++) { - xf_[i] = xf_[fft_size_ - i]; - yf_[i] = 0.0; - } - } - /* The filter, regardless of its origin, is now in the frequency domain. */ - - if (b_allocated) { - delete [] b; - } -} - -// Destructor -/* ******************************************************************** */ -FdFilter::~FdFilter() { - delete [] input_buffer_; - delete [] output_buffer_; - delete [] x_; - delete [] y_; - delete [] xf_; - delete [] yf_; - delete [] filter_coeffs_; - delete [] leftovers_; - delete [] output_delayed_; - delete fft_; -} - - -// Given the half filter in fc, store the full symmetric kernel in -// filter_coeffs_. -/* ******************************************************************** */ -void FdFilter::MirrorFilter(float *fc, bool invert) { - float *dp1, *dp2, *dp3, sum, integral; - int i, ncoefb2; - - filter_coeffs_ = new float[n_filter_coeffs_]; - ncoefb2 = 1 + n_filter_coeffs_ / 2; - // Copy the half-filter and its mirror image into the coefficient array. - for (i = ncoefb2 - 1, dp3 = fc+ncoefb2 - 1, dp2 = filter_coeffs_, - dp1 = filter_coeffs_ + n_filter_coeffs_ - 1, integral = 0.0; i-- > 0;) { - if (!invert) { - *dp1-- = *dp2++ = *dp3--; - } else { - integral += (sum = *dp3--); - *dp1-- = *dp2++ = -sum; - } - } - if (!invert) { - *dp1 = *dp3; /* point of symmetry */ - } else { - integral *= 2; - integral += *dp3; - *dp1 = integral - *dp3; - } -} - -// This is a complex vector multiply. If the second vector (r2, i2) is -// known to be real, set 'i2' to NULL for faster computation. The -// result is returned in (r3, i3). -/* ******************************************************************** */ -void FdFilter::ComplexDotProduct(int n, float *r1, float *i1, float *r2, - float *i2, float *r3, float *i3) { - float tr1, ti1; - - /* This full complex multiply is only necessary for non-symmetric kernels */ - if (i2) { // Only supply the i2 vector if you need to do a full - // complex multiply. - while (n--) { - tr1 = (*r1 * *r2) - (*i1 * *i2); - ti1 = (*r1++ * *i2++) + (*r2++ * *i1++); - *i3++ = ti1; - *r3++ = tr1; - } - } else { - /* Can do this iff the filter is symmetric, zero phase. */ - while (n--) { - tr1 = (*r1++ * *r2); - ti1 = (*r2++ * *i1++); - *i3++ = ti1; - *r3++ = tr1; - } - } -} - -/* ******************************************************************** */ -// Process samples from 'input' array to 'output' array. 'nIn' -// contains the number of samples available in 'input'. 'maxOut' is -// the maximum number of samples that the caller allows to be -// transferred to 'output' (usually the size of 'output'). Set -// 'first' TRUE if the first sample of a new signal is in 'input'. -// Set 'last' TRUE of the last sample of the signal is in 'input' (to -// cause flushing of processing pipeline). The simplest setup for use -// of this method requires that the caller use an input buffer no -// larger than the size returned by getMaxInputSize(), and an output -// array twice the size returned by getMaxInputSize(). Then, each -// subsequent call to filterArray will use all 'input' samples, and -// will transfer all available samples to 'output'. Examples of this -// and the other case, of arbitrarily small caller buffers can be seen -// in the test harness at the end of this file. -// This method returns the number of output samples transferred to 'output'. -int FdFilter::FilterArray(int16_t *input, int nIn, bool first, bool last, - int16_t *output, int max_to_output) { - int i, j, nLeft = nIn, nOut = 0, nToGo = max_to_output; - int toRead, toWrite, available; - int16_t *p = input, *q, *r = output; - - if (first) { - filter_state_ = 1; // indicate start of new signal - array_leftover_ = 0; - array_index_ = 0; - } - if (array_leftover_) { // First, move any output remaining from the - // previous call. - int toCopy = array_leftover_; - if (toCopy > nToGo) - toCopy = nToGo; - for (i = array_index_, j = 0; j < toCopy; i++, j++) - *r++ = output_buffer_[i]; - nToGo -= toCopy; - nOut += toCopy; - array_leftover_ -= toCopy; - array_index_ = i; - } - - if (nToGo <= 0) { - array_samples_used_ = 0; // Can't process any input this time; no - // room in output array. - return max_to_output; - } - - /* Process data from array to array. */ - while (nLeft > 0) { - toRead = nLeft; - if (toRead > max_input_) - toRead = max_input_; - if (insert_ > 1) { - for (q = input_buffer_, i = 0; i < toRead; i++) { - *q++ = *p++; - for (j = 1; j < insert_; j++) - *q++ = 0; - } - } else { - for (q = input_buffer_, i = 0; i < toRead; i++) - *q++ = *p++; - } - nLeft -= toRead; - if ((nLeft <= 0) && last) - filter_state_ |= 2; // Indicate that end of signal is (also) in - // this bufferful. - FilterBuffer(toRead * insert_, &available); - filter_state_ = 0; // Clear the initialization bit, if any for the - // next iteration. - - toWrite = available; - if (toWrite > nToGo) - toWrite = nToGo; - - for (i = 0; i < toWrite; i++) - *r++ = output_buffer_[i]; - nOut += toWrite; - available -= toWrite; - if (available > 0) { // Ran out of output space; suspend processing - array_leftover_ = available; // Save the remaining output - // samples for the next call. - array_index_ = i; - array_samples_used_ = nIn - nLeft; // Record the number of input - // samples actually used. - return(nOut); - } - } - array_samples_used_ = nIn; - return(nOut); -} - -// Use after a call to filterArray() to determine the number of input -// samples actually processed. -int FdFilter::GetArraySamplesUsed() { - return(array_samples_used_); -} - -// Use after a call to filterArray() to determine how many output -// samples were NOT transferred to caller's output array due to lack -// of space in caller's array. -int FdFilter::GetArrayOutputLeftover() { - return(array_leftover_); -} - -// Given the input stream 'input_stream' and the output stream 'output_stream' -// process all samples until EOF is reached on 'input_stream'. Returns 1 -// on success, 0 on failure. All processing is done in a single call -// to this method. -/* ******************************************************************** */ -int FdFilter::FilterStream(FILE *input_stream, FILE *output_stream) { - int i, j; - int toread, towrite, nread, rVal = 1, testc; - - toread = max_input_; - filter_state_ = 1; // indicate start of new signal - /* process data from a stream */ - while ((nread = fread(input_buffer_, sizeof(*input_buffer_), toread, - input_stream))) { - testc = getc(input_stream); - if (feof(input_stream)) - filter_state_ |= 2; - else - ungetc(testc, input_stream); - if (insert_ > 1) { - int16_t *p, *q; - for (p = input_buffer_ + nread - 1, - q = input_buffer_ + (nread * insert_) - 1, i = nread; i--;) { - for (j = insert_ - 1; j--;) *q-- = 0; - *q-- = *p--; - } - } - FilterBuffer(nread*insert_, &towrite); - if ((i = fwrite(output_buffer_, sizeof(*output_buffer_), towrite, - output_stream)) < towrite) { - fprintf(stderr, "Problems writing output in FilterStream\n"); - rVal = 0; - } - filter_state_ = 0; - } - return(rVal); -} - -// This is a private method that supports FilterStream() and -// FilterArray(). It assumes that the input samples have been -// transferred to this->input_buffer_. It places the filtered results in -// this->output_buffer_, and returns the number of output samples in -// *n_output. -/* ******************************************************************** */ -void FdFilter::FilterBuffer(int n_input, int *n_output) { - int16_t *p, *r, *p2; - float *dp1, *dp2, *q, half = 0.5; - int i, j, k, npass; - int totaln; - - if (filter_state_ & 1) { /* first call with this signal and filter? */ - first_out_ = 0; - to_skip_ = 0; - left_over_ = 0; - for (i = max_input_+n_filter_coeffs_, q = output_delayed_; i--;) - *q++ = 0.0; - } /* end of once-per-filter-invocation initialization */ - - npass = (n_input + left_over_) / fft_size_; - if (!npass && (filter_state_ & 2)) { // if it's the end... - // Append the input to the leftovers, then pad with zeros. - p = input_buffer_; - for (p = input_buffer_, r = leftovers_ + left_over_, i = n_input; i--;) { - *r++ = *p++; /* append to leftovers_ from prev. call */ - } - int to_pad = fft_size_ - (left_over_ + n_input); - for (int i = 0; i < to_pad; ++i) { - *r++ = 0; - } - npass = 1; - } else { - /* This is the normal non-boundary course of action. */ - for (p = input_buffer_, r = leftovers_ + left_over_, - i = fft_size_ - left_over_; i--;) - *r++ = *p++; /* append to leftovers_ from prev. call */ - } - if (!npass && !(filter_state_ & 2)) { // it's not the end, but don't - // have enough data for a loop. - left_over_ += n_input; - *n_output = 0; - first_out_ |= filter_state_ & 1; // flag that start of sig is still here. - return; - } - filter_state_ |= first_out_; - first_out_ = 0; - - /* >>>>>>>>>>>>>> Here's the main processing loop. <<<<<<<<<<<<< */ - for (/* p set up above */ q = output_delayed_, i = 0; i < npass; - i++, q += fft_size_) { - if (i) { - for (r = p + fft_size_by2_, j = fft_size_by2_, dp1 = x_, dp2 = y_; j--;) { - *dp1++ = *p++; - *dp2++ = *r++; - } - p += fft_size_by2_; - } else { - for (p2 = leftovers_, r = p2 + fft_size_by2_, j = fft_size_by2_, - dp1 = x_, dp2 = y_; j--;) { - *dp1++ = *p2++; - *dp2++ = *r++; - } - } - for (j = fft_size_by2_; j--;) - *dp1++ = *dp2++ = 0.0; - - /* Filtering is done in the frequency domain; transform two real arrays. */ - fft_->fft(x_, y_); - ComplexDotProduct(fft_size_, x_, y_, xf_, NULL, x_, y_); - fft_->ifft(x_, y_); - - /* Overlap and add. */ - for (dp2 = q, j = fft_size_ - n_filter_coeffs_by2_; j < fft_size_; j++) - *dp2++ += x_[j]; - for (j = 0, k = n_filter_coeffs_by2_; j < k; j++) - *dp2++ += x_[j]; - for (j = n_filter_coeffs_by2_, k = fft_size_ - n_filter_coeffs_by2_; - j < k; j++) - *dp2++ = x_[j]; - - for (dp2 = q+fft_size_by2_, j = fft_size_ - n_filter_coeffs_by2_; - j < fft_size_; j++) - *dp2++ += y_[j]; - for (j = 0, k = n_filter_coeffs_by2_; j < k; j++) - *dp2++ += y_[j]; - for (j = n_filter_coeffs_by2_, k = fft_size_ - n_filter_coeffs_by2_; - j < k; j++) - *dp2++ = y_[j]; - } /* end of main processing loop */ - - left_over_ = n_input - (p-input_buffer_); - for (i = left_over_, r = leftovers_; i--;) // Save unused input - // samples for next call. - *r++ = *p++; - /* If signal end is here, must process any unused input. */ - if (left_over_ && (filter_state_ & 2)) { // Must do one more zero-pad pass. - for (p2 = leftovers_ + left_over_, i = fft_size_ - left_over_; i--;) - *p2++ = 0; - for (p2 = leftovers_, r = p2 + fft_size_by2_, j = fft_size_by2_, - dp1 = x_, dp2 = y_; j--;) { - *dp1++ = *p2++; - *dp2++ = *r++; - } - for (j = fft_size_by2_; j--;) - *dp1++ = *dp2++ = 0.0; - /* Filtering is done in the frequency domain; transform two real arrays. */ - fft_->fft(x_, y_); - ComplexDotProduct(fft_size_, x_, y_, xf_, NULL, x_, y_); - fft_->ifft(x_, y_); - - /* Overlap and add. */ - for (dp2 = q, j = fft_size_ - n_filter_coeffs_by2_; j < fft_size_; j++) - *dp2++ += x_[j]; - for (j = 0, k = n_filter_coeffs_by2_; j < k; j++) - *dp2++ += x_[j]; - for (j = n_filter_coeffs_by2_, k = fft_size_ - n_filter_coeffs_by2_; - j < k; j++) - *dp2++ = x_[j]; - - for (dp2 = q+fft_size_by2_, j = fft_size_ - n_filter_coeffs_by2_; - j < fft_size_; j++) - *dp2++ += y_[j]; - for (j = 0, k = n_filter_coeffs_by2_; j < k; j++) - *dp2++ += y_[j]; - for (j = n_filter_coeffs_by2_, k = fft_size_ - n_filter_coeffs_by2_; - j < k; j++) - *dp2++ = y_[j]; - } - /* total good output samples in ob: */ - totaln = (((npass * fft_size_) - to_skip_ - - ((1 & filter_state_)? n_filter_coeffs_by2_ : 0)) + - ((filter_state_ & 2)? left_over_ + n_filter_coeffs_by2_ : 0)); - /* number returned to caller: */ - *n_output = 1 + (totaln - 1) / decimate_; // possible decimation for - // downsampling - /* Round, decimate and output the samples. */ - float f_temp; - q = (filter_state_ & 1)? output_delayed_ + (n_filter_coeffs_by2_) : - output_delayed_ + to_skip_; - for (j = decimate_, i = *n_output, p = output_buffer_; i-- ; q += j) { - if ((f_temp = *q) > 32767.0) - f_temp = 32767.0; - if (f_temp < -32768.0) - f_temp = -32768.0; - *p++ = static_cast((f_temp > 0.0) ? half + f_temp : f_temp - half); - } - - for (dp1 = output_delayed_ + npass*fft_size_, j = n_filter_coeffs_, - dp2 = output_delayed_; j--;) /*save mem for next call */ - *dp2++ = *dp1++; - /* If decimating, number to skip on next call. */ - to_skip_ = (*n_output * decimate_) - totaln; -} - -// Return the largest number of samples that can be processed in a -// single call to FdFilter::filterArray(). This can be used to configure -// the caller's buffer sizes to simplify subsequent processing. The -// simplest use of filterArray() is when the caller sends chunks of -// size getMaxInputSize() (or less) as input, and has an output buffer -// of size >= (2 * getMaxInputSize()). In this case, no checking is -// required to synchronize input and output buffering. This, and the -// less ideal case of arbitrary caller buffer sizes are illustrated in -// the test harness at the end of this file. -/* ******************************************************************** */ -int FdFilter::GetMaxInputSize() { - return(max_input_); -} - -// When sample-rate conversion is attempted, it is possible that the -// output frequency realizable with the FdFilter configuration does not -// exactly match the requested rate. getActualOutputFreq() retrieves -// the rate achieved, and allows the caller to decide whether to -// proceed with the filtering or not, and to correctly set the rate og -// the output stream for processes later in the chain. This method -// may be called immediately after instantiation of the FdFilter, or at any -// later time. -float FdFilter::GetActualOutputFreq() { - return(true_output_rate_); -} - -// A private method that finds the closest ratio to the fraction in -// 'a' (0.0 < a < 1.0). The numerator is returned in 'k', the -// denominator in 'l'. The largest allowed denominator is specified -// by 'qlim'. -/* ---------------------------------------------------------- */ -void FdFilter::RationalApproximation(float a, int *k, int *l, int qlim) { - float aa, af, q, em, qq = 1.0, pp = 1.0, ps, e; - int ai, ip, i; - - aa = fabs(a); - ai = static_cast(aa); - i = ai; - af = aa - i; - q = 0; - em = 1.0; - while (++q <= qlim) { - ps = q * af; - ip = static_cast(ps + 0.5); - e = fabs((ps - static_cast(ip)) / q); - if (e < em) { - em = e; - pp = ip; - qq = q; - } - } - *k = static_cast((ai * qq) + pp); - *k = (a > 0)? *k : -(*k); - *l = static_cast(qq); -} - -// A private method to create the coefficients for a symmetric FIR -// lowpass filter using the window technique with a Hanning window. -// Half of the symmetric kernel is returned in ''coef'. The desired -// number of filter coefficients is in 'nf', but is forced to be odd -// by adding one, if the requesred number is even. 'fc' is the -// normalized corner frequency (0 < fc < 1). -/* ---------------------------------------------------------- */ -void FdFilter::MakeLinearFir(float fc, int *nf, float *coef) { - int i, n; - double twopi, fn, c; - - if (((*nf % 2) != 1)) - *nf = *nf + 1; - n = (*nf + 1) / 2; - - /* Compute part of the ideal impulse response (the sin(x)/x kernel). */ - twopi = M_PI * 2.0; - coef[0] = 2.0 * fc; - c = M_PI; - fn = twopi * fc; - for (i = 1; i < n; i++) - coef[i] = sin(i * fn) / (c * i); - - /* Now apply a Hanning window to the (infinite) impulse response. */ - /* (Could use other windows, like Kaiser, Gaussian...) */ - fn = twopi / *nf; - for (i = 0; i < n; i++) - coef[n - i - 1] *= (.5 - (.5 * cos(fn * (i + 0.5)))); -} diff --git a/reaper-tmp/REAPER/epoch_tracker/fd_filter.h b/reaper-tmp/REAPER/epoch_tracker/fd_filter.h deleted file mode 100644 index 6b73411a7244383ef694e8bc0278e7c634befe48..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/epoch_tracker/fd_filter.h +++ /dev/null @@ -1,131 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ -/* In-Line FIR filter class */ -/* -Author: David Talkin (dtalkin@google.com) -*/ - -#ifndef _FD_FILTER_H_ -#define _FD_FILTER_H_ - -#include -#include - -class FFT; - -class FdFilter { - public: - // Constructor when doing arbitrary FIR filtering using the response - // in file 'spectrum_shape' - FdFilter(float input_freq, char* spectrum_shape); - // Constructor when doing arbitrary FIR filtering using the response - // in array 'spectrum_array' - FdFilter(float input_freq, float* spectrum_rray, int n_magnitudes); - // Constructor when doing highpass or lowpass filtering or when - // doing rate conversion - FdFilter(float input_freq, float corner_freq, bool do_highpass, - float filter_dur, bool do_rate_conversion); - - ~FdFilter(); - - // Read and filter/convert all input from inStream and write to - // outStream until inStream is exhausted. Return 1 for success 0 - // for failure. - int FilterStream(FILE* fIn, FILE* fOut); - - // Process nIn samples from 'input' array into 'output' array. Set - // 'first' true if this is the first call for a new signal. Set - // 'last' true if the end of the signal is in 'input'. The - // number of samples transferred to 'output' is the return value. - // 'maximum_to_output' is the size of caller's output array. - int FilterArray(int16_t* input, int n_input, bool first, bool last, - int16_t* output, int maximum_to_output); - - // Return number of samples left after a call to filterArray() in the - // case where 'maximum_to_output' is smaller than the number of output samples - // produced. - int GetArrayOutputLeftover(); - - // Use after a call to filterArray() to get number of 'input' - // samples actually used (in cases where the caller's output array - // size has limited the number used, or when the caller's input size - // is larger than getMaxInputSize()). - int GetArraySamplesUsed(); - - // Return the largest batch of input samples that can be processed - // with a single call to filterBuffer() - int GetMaxInputSize(); - - // When sample-rate conversion is being done, return the actual - // output frequency, which may differ from that requested. - float GetActualOutputFreq(); - - private: - // The main initialization function. Called by all constructors. - void FdFilterInitialize(float input_freq, float corner_freq, bool do_highpass, - float filter_dur, bool do_rate_conversion, - char* spectrum_shape, float* spectrum_array, - int n_magnitudes); - - // Use the half filter in fc to create the full filter in filter_coeff_ - void MirrorFilter(float* fc, bool invert); - - // Window approach to creating a high- or low-pass FIR kernel - void MakeLinearFir(float fc, int* nf, float* coef); - - // Find the closest integer ratio to the float 'a' - void RationalApproximation(float a, int* k, int* l, int qlim); - - // Complex inner product of length 'n'; result goes to (r3,i3) - void ComplexDotProduct(int n, float* r1, float* i1, float* r2, float* i2, - float* r3, float* i3); - - // Assuming n_input samples are in in_buffer_, process the signal - // into out_buffer_. Returns number transferred to out_buffer_ in - // 'n_output'. - void FilterBuffer(int n_input, int* n_output); - - - int16_t* output_buffer_; // Processing I/O buffers - int16_t* input_buffer_; - float* filter_coeffs_; // becomes the filter coefficient array - int n_filter_coeffs_; // The number of filter coefficients - int n_filter_coeffs_by2_; // Precompute for convenience/speed - int left_over_; // bookkeeping for the OLA filtering operations - int first_out_; // non-zero indicates the first call to the filter kernel - int to_skip_; // saves state for the decimator when downsampling - int array_leftover_; // Bookkeeping for buffered processing. - int array_index_; // internal book keeping - int array_samples_used_; // internal book keeping - int filter_state_; // State of processing WRT input and completion - float true_output_rate_; // For sample rate conversion, this can - // differ from requested freq. - int16_t* leftovers_; // saves residual input samples that are mod fft_size - float* output_delayed_; // saves samples that exceed the caller's output capacity. - float* x_; // FFT processing arrays - float* y_; - float* xf_; - float* yf_; - int fft_size_; // Number of points in the FFTs - int fft_size_by2_; // Precompute for convenience/speed - int insert_; // for sample rate conversion, these are the - // interpolation/decimations - int decimate_; - int max_input_; // maximum allowed input with each call to filterBuffer - FFT* fft_; // The FFT instance used by the filter. -}; - -#endif // _FD_FILTER_H_ diff --git a/reaper-tmp/REAPER/epoch_tracker/fft.cc b/reaper-tmp/REAPER/epoch_tracker/fft.cc deleted file mode 100644 index 3a4b0bd4477571bcdb3ff1409c3f3d4dc92968fd..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/epoch_tracker/fft.cc +++ /dev/null @@ -1,181 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ -// Author: David Talkin (dtalkin@google.com) - -#include "epoch_tracker/fft.h" - -/* Construct a FFT to perform a DFT of size 2^power. */ -FFT::FFT(int power) { - makefttable(power); -} - -FFT::~FFT() { - delete [] fsine; - delete [] fcosine; -} - -/*-----------------------------------------------------------------------*/ -/* z <- (10 * log10(x^2 + y^2)) for n elements */ -bool FFT::flog_mag(float *x, float *y, float *z, int n) { - float *xp, *yp, *zp, t1, t2, ssq; - - if (x && y && z && n) { - for (xp = x + n, yp = y + n, zp = z + n; zp > z;) { - t1 = *--xp; - t2 = *--yp; - ssq = (t1 * t1) + (t2 * t2); - *--zp = (ssq > 0.0)? 10.0 * log10(ssq) : -200.0; - } - return true; - } else { - return false; - } -} - -/*-----------------------------------------------------------------------*/ -float FFT::get_band_rms(float *x, float*y, int first_bin, int last_bin) { - double sum = 0.0; - for (int i = first_bin; i <= last_bin; ++i) { - sum += (x[i] * x[i]) + (y[i] * y[i]); - } - return sqrt(sum / (last_bin - first_bin + 1)); -} - -/*-----------------------------------------------------------------------*/ -int FFT::makefttable(int pow2) { - int lmx, lm; - float *c, *s; - double scl, arg; - - fftSize = 1 << pow2; - fft_ftablesize = lmx = fftSize/2; - fsine = new float[lmx]; - fcosine = new float[lmx]; - scl = (M_PI * 2.0) / fftSize; - for (s = fsine, c = fcosine, lm = 0; lm < lmx; ++lm) { - arg = scl * lm; - *s++ = sin(arg); - *c++ = cos(arg); - } - kbase = (fft_ftablesize * 2) / fftSize; - power2 = pow2; - return(fft_ftablesize); -} - -/*-----------------------------------------------------------------------*/ -/* Compute the discrete Fourier transform of the 2**l complex sequence - * in x (real) and y (imaginary). The DFT is computed in place and the - * Fourier coefficients are returned in x and y. - */ -void FFT::fft(float *x, float *y) { - float c, s, t1, t2; - int j1, j2, li, lix, i; - int lmx, lo, lixnp, lm, j, nv2, k = kbase, im, jm, l = power2; - - for (lmx = fftSize, lo = 0; lo < l; lo++, k *= 2) { - lix = lmx; - lmx /= 2; - lixnp = fftSize - lix; - for (i = 0, lm = 0; lm < lmx; lm++, i += k) { - c = fcosine[i]; - s = fsine[i]; - for (li = lixnp + lm, j1 = lm, j2 = lm + lmx; j1 <= li; - j1 += lix, j2 += lix) { - t1 = x[j1] - x[j2]; - t2 = y[j1] - y[j2]; - x[j1] += x[j2]; - y[j1] += y[j2]; - x[j2] = (c * t1) + (s * t2); - y[j2] = (c * t2) - (s * t1); - } - } - } - - /* Now perform the bit reversal. */ - j = 1; - nv2 = fftSize / 2; - for (i = 1; i < fftSize; i++) { - if (j < i) { - jm = j - 1; - im = i - 1; - t1 = x[jm]; - t2 = y[jm]; - x[jm] = x[im]; - y[jm] = y[im]; - x[im] = t1; - y[im] = t2; - } - k = nv2; - while (j > k) { - j -= k; - k /= 2; - } - j += k; - } -} - -/*-----------------------------------------------------------------------*/ -/* Compute the discrete inverse Fourier transform of the 2**l complex - * sequence in x (real) and y (imaginary). The DFT is computed in - * place and the Fourier coefficients are returned in x and y. Note - * that this DOES NOT scale the result by the inverse FFT size. - */ -void FFT::ifft(float *x, float *y) { - float c, s, t1, t2; - int j1, j2, li, lix, i; - int lmx, lo, lixnp, lm, j, nv2, k = kbase, im, jm, l = power2; - - for (lmx = fftSize, lo = 0; lo < l; lo++, k *= 2) { - lix = lmx; - lmx /= 2; - lixnp = fftSize - lix; - for (i = 0, lm = 0; lm < lmx; lm++, i += k) { - c = fcosine[i]; - s = -fsine[i]; - for (li = lixnp + lm, j1 = lm, j2 = lm + lmx; j1 <= li; - j1 += lix, j2 += lix) { - t1 = x[j1] - x[j2]; - t2 = y[j1] - y[j2]; - x[j1] += x[j2]; - y[j1] += y[j2]; - x[j2] = (c * t1) + (s * t2); - y[j2] = (c * t2) - (s * t1); - } - } - } - - /* Now perform the bit reversal. */ - j = 1; - nv2 = fftSize / 2; - for (i = 1; i < fftSize; i++) { - if (j < i) { - jm = j-1; - im = i-1; - t1 = x[jm]; - t2 = y[jm]; - x[jm] = x[im]; - y[jm] = y[im]; - x[im] = t1; - y[im] = t2; - } - k = nv2; - while (j > k) { - j -= k; - k /= 2; - } - j += k; - } -} diff --git a/reaper-tmp/REAPER/epoch_tracker/fft.h b/reaper-tmp/REAPER/epoch_tracker/fft.h deleted file mode 100644 index 58ce89a44194dca180796ce83d8dc2f6b70383ef..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/epoch_tracker/fft.h +++ /dev/null @@ -1,73 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ -// Author: David Talkin (dtalkin@google.com) - -#ifndef _FFT_H_ -#define _FFT_H_ - -#include -#include -#include - -#ifndef M_PI -#define M_PI 3.1415927 -#endif - -#define TRUE 1 -#define FALSE 0 - -class FFT { - public: - // Constructor: Prepare for radix-2 FFT's of size (1< -#include - -#ifndef M_PI -#define M_PI (3.14159265359) -#endif - -/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/ -/* Generate a Hanning window, if one does not already exist. */ -void LpcAnalyzer::HannWindow(const float* din, float* dout, int n, - float preemp) { - int i; - const float *p; - - // Need to create a new Hanning window? */ - if (window_.size() != static_cast(n)) { - double arg, half = 0.5; - window_.resize(n); - for (i = 0, arg = M_PI * 2.0 / n; i < n; ++i) - window_[i] = (half - half * cos((half + i) * arg)); - } - /* If preemphasis is to be performed, this assumes that there are n+1 valid - samples in the input buffer (din). */ - if (preemp != 0.0) { - for (i = 0, p = din + 1; i < n; ++i) - *dout++ = window_[i] * (*p++ - (preemp * *din++)); - } else { - for (i = 0; i < n; ++i) - *dout++ = window_[i] * *din++; - } -} - -/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/ -/* Place a time-weighting window of length n in energywind_. -*/ -void LpcAnalyzer::GetWindow(int n) { - // Need to create a new Hanning window? - if (energywind_.size() != static_cast(n)) { - double arg = M_PI * 2.0 / n, half = 0.5; - energywind_.resize(n); - for (int i = 0; i < n; ++i) - energywind_[i] = (half - half * cos((half + i) * arg)); - } -} - -/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/ -/* Compute the pp+1 autocorrelation lags of the windowsize samples in s. -* Return the normalized autocorrelation coefficients in r. -* The rms is returned in e. -*/ -void LpcAnalyzer::Autoc(int windowsize, float* s, int p, float* r, float* e) { - int i, j; - float *q, *t, sum, sum0; - - for (i = windowsize, q = s, sum0 = 0.0; i--;) { - sum = *q++; - sum0 += sum*sum; - } - *r = 1.; /* r[0] will always = 1.0 */ - if (sum0 == 0.0) { /* No energy: fake low-energy white noise. */ - *e = 1.; /* Arbitrarily assign 1 to rms. */ - /* Now fake autocorrelation of white noise. */ - for (i = 1; i <= p; i++) { - r[i] = 0.; - } - return; - } - *e = sqrt(sum0 / windowsize); - sum0 = 1.0 / sum0; - for (i = 1; i <= p; i++) { - for (sum = 0.0, j = windowsize - i, q = s, t = s + i; j--;) - sum += (*q++) * (*t++); - *(++r) = sum * sum0; // normalizing by the inverse energy - } -} - -/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/ -/* Using Durbin's recursion, convert the autocorrelation sequence in r -* to reflection coefficients in k and predictor coefficients in a. -* The prediction error energy (gain) is left in *ex. -* Note: Durbin returns the coefficients in normal sign format. -* (i.e. a[0] is assumed to be = +1.) -*/ -void LpcAnalyzer::Durbin(float* r, float* k, float* a, int p, float* ex) { - float bb[BIGSORD]; - int i, j; - float e, s, *b = bb; - - e = *r; - *k = -r[1] / e; - *a = *k; - e *= (1.0 - (*k) * (*k)); - for (i = 1; i < p; i++) { - s = 0; - for (j = 0; j < i; j++) { - s -= a[j] * r[i - j]; - } - k[i] = (s - r[i + 1]) / e; - a[i] = k[i]; - for (j = 0; j <= i; j++) { - b[j] = a[j]; - } - for (j = 0; j < i; j++) { - a[j] += k[i] * b[i - j - 1]; - } - e *= (1.0 - (k[i] * k[i])); - } - *ex = e; -} - -/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/ -/* Compute the autocorrelations of the p LP coefficients in a. -* (a[0] is assumed to be = 1 and not explicitely accessed.) -* The magnitude of a is returned in c. -* 2* the other autocorrelation coefficients are returned in b. -*/ -void LpcAnalyzer::PcToAutocorPc(float* a, float* b, float* c, int p) { - float s, *ap, *a0; - int i, j; - - for (s = 1., ap = a, i = p; i--; ap++) - s += *ap * *ap; - - *c = s; - for (i = 1; i <= p; i++) { - s = a[i - 1]; - for (a0 = a, ap = a + i, j = p - i; j--;) - s += (*a0++ * *ap++); - *b++ = 2.0 * s; - } -} - -/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/ -/* Compute the Itakura LPC distance between the model represented -* by the signal autocorrelation (r) and its residual (gain) and -* the model represented by an LPC autocorrelation (c, b). -* Both models are of order p. -* r is assumed normalized and r[0]=1 is not explicitely accessed. -* Values returned by the function are >= 1. -*/ -float LpcAnalyzer::ItakuraDistance(int p, float* b, float* c, float* r, - float gain) { - float s; - - for (s = *c; p--;) - s += *r++ * *b++; - - return s / gain; -} - -/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/ -/* Compute the time-weighted RMS of a size segment of data. The data -* is weighted by a Hanning window before RMS computation. -*/ -float LpcAnalyzer::WindowedRms(float* data, int size) { - float sum, f; - int i; - - GetWindow(size); - for (i = 0, sum = 0.0; i < size; i++) { - f = energywind_[i] * (*data++); - sum += f * f; - } - return sqrt(sum / size); -} - -/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/ -// Generic autocorrelation LPC analysis of the floating-point -// sequence in data. -// -// int lpc_ord, /* Analysis order */ -// wsize, /* window size in points */ -// type; /* window type (decoded in window() above) */ -// float noise_floor, /* Simulated white noise floor in dB (SNR). */ -// *lpca, /* if non-NULL, return vector for predictors */ -// *ar, /* if non-NULL, return vector for normalized autoc. */ -// *lpck, /* if non-NULL, return vector for PARCOR's */ -// *normerr, /* return scalar for normalized error */ -// *rms, /* return scalar for energy in preemphasized window */ -// preemp; -// float *data; /* input data sequence; assumed to be wsize+1 long */ -int LpcAnalyzer::ComputeLpc(int lpc_ord, float noise_floor, int wsize, - const float* data, float* lpca, float* ar, - float* lpck, float* normerr, float* rms, - float preemp) { - float rho[BIGSORD+1], k[BIGSORD], a[BIGSORD+1], *r, *kp, *ap, en, er; - - if ((wsize <= 0) || (!data) || (lpc_ord > BIGSORD)) - return false; - - float *dwind = new float[wsize]; - - HannWindow(data, dwind, wsize, preemp); - if (!(r = ar)) r = rho; /* Permit optional return of the various */ - if (!(kp = lpck)) kp = k; /* coefficients and intermediate results. */ - if (!(ap = lpca)) ap = a; - Autoc(wsize, dwind, lpc_ord, r, &en); - if (noise_floor > 1.0) { // Add some to the diagonal to simulate white noise. - int i; - float ffact; - ffact = 1.0 / (1.0 + exp((-noise_floor / 20.0) * log(10.0))); - for (i = 1; i <= lpc_ord; i++) - rho[i] = ffact * r[i]; - *rho = *r; - r = rho; - if (ar) { - for (i = 0; i <= lpc_ord; i++) - ar[i] = r[i]; - } - } - Durbin(r, kp, ap + 1, lpc_ord, &er); - float wfact = .612372; // ratio of Hanning RMS to rectangular RMS - ap[0] = 1.0; - if (rms) - *rms = en / wfact; - if (normerr) - *normerr = er; - delete [] dwind; - return true; -} diff --git a/reaper-tmp/REAPER/epoch_tracker/lpc_analyzer.h b/reaper-tmp/REAPER/epoch_tracker/lpc_analyzer.h deleted file mode 100644 index 56378c6435e819b4d5f8cbcfb36ed9f9e55cdd31..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/epoch_tracker/lpc_analyzer.h +++ /dev/null @@ -1,113 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -// LpcAnalyzer -// A collection of methods commonly used in linear-prediction analysis. - -#ifndef _LPC_ANALYZER_H_ -#define _LPC_ANALYZER_H_ - -#include - -// Largest order allowed for any linear predictor analysis -#define BIGSORD 100 - -class LpcAnalyzer { - public: - LpcAnalyzer(void) { } - - ~LpcAnalyzer(void) { } - - - // Apply Hanning weighting to the input data sequence in data_in, - // put the results in data_out. Array lengths are assumed as - // data_in has windowsize+1 points; data_out has windowsize - // elements. - void HannWindow(const float* data_in, float* data_out, int windowsize, - float preemp); - - - // Compute the order+1 autocorrelation lags of the windowsize - // samples in data_in. Return the normalized autocorrelation - // coefficients in autoc. The rms is returned in rms. - void Autoc(int windowsize, float* data_in, int order, float* autoc, - float* rms); - - - // Using Durbin's recursion, convert the autocorrelation sequence in autocor - // to reflection coefficients in refcof and predictor coefficients in lpc. - // The prediction error energy (gain) is left in *gain. - // Note: Durbin returns the coefficients in normal sign format. - // (i.e. lpca[0] is assumed to be = +1.) - void Durbin(float* autocor, float* refcof, float* lpc, int order, - float* gain); - - - // Compute the autocorrelations of the order LP coefficients in lpc. - // (lpc[0] is assumed to be = 1 and not explicitely accessed.) - // The magnitude of lpc is returned in mag. - // 2* the other autocorrelation coefficients are returned in lpc_auto. - void PcToAutocorPc(float* lpc, float* lpc_auto, float* mag, int order); - - - // Compute the Itakura LPC distance between the model represented - // by the signal autocorrelation (autoc) and its residual (gain) and - // the model represented by an LPC autocorrelation (mag, lpc_auto). - // Both models are of order. - // r is assumed normalized and r[0]=1 is not explicitely accessed. - // Values returned by the function are >= 1. - float ItakuraDistance(int order, float* lpc_auto, float* mag, float* autoc, - float gain); - - - // Compute the time-weighted RMS of a size segment of data. The data - // is weighted by a window of type w_type before RMS computation. w_type - // is decoded above in window(). - float WindowedRms(float* data, int size); - - - // Generic autocorrelation LPC analysis of the floating-point - // sequence in data. - // - // int lpc_ord, /* Analysis order - // wsize; /* window size in points - // float noise_floor, /* To simulate a white noise floor (dB SNR). - // *lpca, /* if non-NULL, return vector for predictors - // *ar, /* if non-NULL, return vector for normalized autoc. - // *lpck, /* if non-NULL, return vector for PARCOR's - // *normerr, /* return scalar for normalized error - // *rms, /* return scalar for energy in preemphasized window - // preemp; - // float *data; /* input data sequence; assumed to be wsize+1 long - int ComputeLpc(int lpc_ord, float noise_floor, int wsize, const float* data, - float* lpca, float* ar, float* lpck, float* normerr, float* rms, - float preemp); - - // Use the standard speech analysis formula to determine the - // appropriate LPC order, given a sample rate. - static int GetLpcOrder(float sample_rate) { - return static_cast(2.5 + (sample_rate / 1000.0)); - } - - private: - // Puts a time-weighting window of length n in energywind. - void GetWindow(int n); - - std::vector energywind_; - std::vector window_; -}; - -#endif // _LPC_ANALYZER_H_ diff --git a/reaper-tmp/REAPER/epoch_tracker_main.cc b/reaper-tmp/REAPER/epoch_tracker_main.cc deleted file mode 100644 index 56fcd35932fcf6505942028eb832fa72c3c6157c..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/epoch_tracker_main.cc +++ /dev/null @@ -1,260 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#include -#include -#include -#include -#include - -#include "core/file_resource.h" -#include "core/track.h" -#include "epoch_tracker/epoch_tracker.h" -#include "wave/wave.h" - - -const char* kHelp = "Usage: -i " - "[-f -p \\\n" - "-c " - "-t " - "-s " - "-e " - "-x " - "-m \\\n" - "-u " - "-w " - "-a " - "-d ] " - "\n\n Help:\n" - "-t enables a Hilbert transform that may reduce phase distortion\n" - "-s suppress applying high pass filter at 80Hz " - "(rumble-removal highpass filter)\n" - "-e specifies the output frame interval for F0\n" - "-x maximum f0 to look for\n" - "-m minimum f0 to look for\n" - "-u regular inter-mark interval to use in UV pitchmark regions\n" - "-w set the cost for unvoiced segments\n" - " (def. 0.9, the higher the value the more f0 estimates in noise)\n" - "-a saves F0 and PM output in ascii mode\n" - "-d write diagnostic output to this file pattern\n" -"\nOutputs:\n"\ -"The output files specified with -f and -p are Edinburgh Speech Tool\n" -"(EST) style files. These are output as binary by default, ASCII with\n" -"the -a option. The F0 values in the f0_output file are resampled with\n" -"the frame interval specified by option -e (default .005 s).\n" -"The unvoiced regions of the pitchmark file are filled with marks spaced\n" -"by the interval specified with -u (default .01 s).\n" -"\nIt is strongly recommended that the default high-pass filter be\n" -"applied to all input signals to remove DC and low-frequency\n" -"components. For signals that have been recorded using close-talking\n" -"microphones, or those that have been subjected to various other\n" -"non-linear phase distortions in the studio, or in post-production, it\n" -"is often helpful to apply a Hilbert transform (-t option). The .resid\n" -"file output when -d is specified can be examined to determine if the\n" -"voice pulses look anything like the classical glottal-flow derivative,\n" -"and the Hilbert transform enabled, or not.\n" -"\n" -"In the discussion below, the following notation is used:\n" -"Fs: sample rate\n" -"Fs0: sample rate of input file\n" -"nd: n-dimensional vector signal\n" -"ts: time-stamped vector signal; 1st ele. is the time of the sample in sec.\n" -"float, int16, etc.: atomic type of the data in the file\n" -"\nIf -d is specified, the following raw binary output files are produced,\n" -"with the base path as specified in , and the indicated extensions:\n" -".bestcorr 2d float ts the highest NCC value found at each residual peak\n" -".bprms 1d float Fs=500 RMS of 100-1000 Hz bandpassed input signal\n" -".f0ap 3d float ts F0 and NCC value found for each period\n" -".resid 1d float Fs=Fs0 LPC residual of conditioned input signal\n" -".nresid 1d float Fs=Fs0 LPC residual with local gain normalization\n" -".offsetp 1d float Fs=500 pseudo-probability that voicing is terminating\n" -".onsetp 1d float Fs=500 pseudo-probability that voicing is starting\n" -".pvoiced 1d float Fs=500 pseudo-probability that voicing is occurring\n" -".pcm 1d float Fs=Fs0 conditioned input signal\n" -".pmlab ASCII 'xlabel' format file of epoch marks\n" -".pvals 1d float Fs=Fs0 graded residual peak candidates\n"; - -Track* MakeEpochOutput(EpochTracker &et, float unvoiced_pm_interval) { - std::vector times; - std::vector voicing; - et.GetFilledEpochs(unvoiced_pm_interval, ×, &voicing); - Track* pm_track = new Track; - pm_track->resize(times.size()); - for (int32_t i = 0; i < times.size(); ++i) { - pm_track->t(i) = times[i]; - pm_track->set_v(i, voicing[i]); - } - return pm_track; -} - -Track* MakeF0Output(EpochTracker &et, float resample_interval, Track** cor) { - std::vector f0; - std::vector corr; - if (!et.ResampleAndReturnResults(resample_interval, &f0, &corr)) { - return NULL; - } - - Track* f0_track = new Track; - Track* cor_track = new Track; - f0_track->resize(f0.size()); - cor_track->resize(corr.size()); - for (int32_t i = 0; i < f0.size(); ++i) { - float t = resample_interval * i; - f0_track->t(i) = t; - cor_track->t(i) = t; - f0_track->set_v(i, (f0[i] > 0.0) ? true : false); - cor_track->set_v(i, (f0[i] > 0.0) ? true : false); - f0_track->a(i) = (f0[i] > 0.0) ? f0[i] : -1.0; - cor_track->a(i) = corr[i]; - } - *cor = cor_track; - return f0_track; -} - -bool ComputeEpochsAndF0(EpochTracker &et, float unvoiced_pulse_interval, - float external_frame_interval, - Track** pm, Track** f0, Track** corr) { - if (!et.ComputeFeatures()) { - return false; - } - bool tr_result = et.TrackEpochs(); - et.WriteDiagnostics(""); // Try to save them here, even after tracking failure. - if (!tr_result) { - fprintf(stderr, "Problems in TrackEpochs"); - return false; - } - - // create pm and f0 objects, these need to be freed in calling client. - *pm = MakeEpochOutput(et, unvoiced_pulse_interval); - *f0 = MakeF0Output(et, external_frame_interval, corr); - return true; -} - -int main(int argc, char* argv[]) { - int opt = 0; - std::string filename; - std::string f0_output; - std::string pm_output; - std::string corr_output; - bool do_hilbert_transform = kDoHilbertTransform; - bool do_high_pass = kDoHighpass; - float external_frame_interval = kExternalFrameInterval; - float max_f0 = kMaxF0Search; - float min_f0 = kMinF0Search; - float inter_pulse = kUnvoicedPulseInterval; - float unvoiced_cost = kUnvoicedCost; - bool ascii = false; - std::string debug_output; - if (argc < 3) { - fprintf(stdout, "\n%s\n", kHelp); - return 1; - } - while ((opt = getopt(argc, argv, "i:f:p:c:htse:x:m:u:w:ad:")) != -1) { - switch(opt) { - case 'i': - filename = optarg; - break; - case 'f': - f0_output = optarg; - break; - case 'p': - pm_output = optarg; - break; - case 'c': - corr_output = optarg; - break; - case 't': - do_hilbert_transform = true; - break; - case 's': - do_high_pass = false; - break; - case 'e': - external_frame_interval = atof(optarg); - break; - case 'x': - max_f0 = atof(optarg); - break; - case 'm': - min_f0 = atof(optarg); - break; - case 'u': - inter_pulse = atof(optarg); - break; - case 'w': - unvoiced_cost = atof(optarg); - break; - case 'a': - ascii = true; - break; - case 'd': - debug_output = optarg; - break; - case 'h': - fprintf(stdout, "\n%s\n", kHelp); - return 0; - } - } - - // Load input. - Wave wav; - if (!wav.Load(filename)) { - fprintf(stderr, "Failed to load waveform '%s'\n", filename.c_str()); - return 1; - } - - EpochTracker et; - et.set_unvoiced_cost(unvoiced_cost); - int16_t* wave_datap = const_cast(wav.data()->data()); - int32_t n_samples = wav.num_samples(); - float sample_rate = wav.sample_rate(); - if (!et.Init(wave_datap, n_samples, sample_rate, - min_f0, max_f0, do_high_pass, do_hilbert_transform)) { - return 1; - } - if (!debug_output.empty()) { - et.set_debug_name(debug_output); - } - // Compute f0 and pitchmarks. - Track *f0 = NULL; - Track *pm = NULL; - Track *corr = NULL; - if (!ComputeEpochsAndF0(et, inter_pulse, external_frame_interval, &pm, &f0, &corr)) { - fprintf(stderr, "Failed to compute epochs\n"); - return 1; - } - - // Save outputs. - if (!f0_output.empty() && !f0->Save(f0_output, ascii)) { - delete f0; - fprintf(stderr, "Failed to save f0 to '%s'\n", f0_output.c_str()); - return 1; - } - if (!pm_output.empty() && !pm->Save(pm_output, ascii)) { - delete pm; - fprintf(stderr, "Failed to save pitchmarks to '%s'\n", pm_output.c_str()); - return 1; - } - if (!corr_output.empty() && !corr->Save(corr_output, ascii)) { - delete corr; - fprintf(stderr, "Failed to save correlations to '%s'\n", corr_output.c_str()); - return 1; - } - delete f0; - delete pm; - delete corr; - return 0; -} diff --git a/reaper-tmp/REAPER/wave/codec_api-inl.h b/reaper-tmp/REAPER/wave/codec_api-inl.h deleted file mode 100644 index dc2dd4f626204e9d96e9c1a3f2e09acf3c035efc..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/wave/codec_api-inl.h +++ /dev/null @@ -1,135 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#ifndef _CODEC_API_INL_H_ -#define _CODEC_API_INL_H_ - -#include "core/file_resource.h" - - -template -inline bool CodecApi::SetHeaderInfo( - int sampling_rate, int num_samples_per_channel) { - bool status = set_sampling_rate(sampling_rate); - status &= set_num_samples_per_channel(num_samples_per_channel); - return status; -} - -template -inline bool CodecApi::GetHeaderInfo( - int *sampling_rate, int *num_samples_per_channel) { - *sampling_rate = sampling_rate_; - *num_samples_per_channel = num_samples_per_channel_; - return true; -} - -template -inline bool CodecApi::ReadHeader(FileResource *fr) { - return codec_.ReadHeader(fr, &num_samples_per_channel_, &sampling_rate_); -} - -template -bool CodecApi::ReadAudioData( - int wave_start, int num_samples, - std::vector *samples, FileResource *fr) { - if (samples == NULL) { - fprintf(stderr, "CodecApi::ReadAudioData: empty pointer was given"); - return true; - } - int64_t offset_audio_container = ftell(fr->fp()); - bool status = codec_.ReadAudioData(wave_start, num_samples, samples, fr); - // Reset the file resource pointer back to the beginning of the - // audio container. - if (fseek(fr->fp(), offset_audio_container, SEEK_SET) != 0) { - fprintf(stderr, "CodecApi::ReadAudioData: error seeking the beginning of the " - "audio container"); - return false; - } - return status; -} - -template -bool CodecApi::ReadAudioContainer( - int container_size_in_bytes, - std::vector *samples, - FileResource *fr) { - int64_t offset_audio_container = ftell(fr->fp()); - bool status = codec_.ReadAudioContainer(container_size_in_bytes, samples, fr); - // Reset the FileResource pointer back to the beginning of the audio container - if (fseek(fr->fp(), offset_audio_container, SEEK_SET) != 0) { - fprintf(stderr, "CodecApi::ReadAudioData: error seeking the beginning of the " - "audio container"); - return false; - } - return status; -} - -template -inline int CodecApi::get_num_samples_per_channel() const { - return num_samples_per_channel_; -} - -template -inline int CodecApi::get_sampling_rate() const { - return sampling_rate_; -} - -template -inline -bool CodecApi::set_num_samples_per_channel(int value) { - num_samples_per_channel_ = value; - return true; -} - -template -inline -bool CodecApi::set_sampling_rate(int value) { - sampling_rate_ = value; - return true; -} - -template -bool CodecApi::Load( - FileResource *fr, std::vector *samples) { - if (!codec_.ReadHeader(fr, &num_samples_per_channel_, &sampling_rate_)) { - return false; - } - samples->resize(num_samples_per_channel_); - return codec_.ReadAudioData( - 0, num_samples_per_channel_, PCM16, samples, fr); -} - -template -template -bool CodecApi::Load( - const std::string &filename, std::vector *samples) { - FileResource *fr = FileResourceType::Open(filename); - if (fr != NULL) { - return Load(fr, samples); - } else { - fprintf(stderr, "CodecApi::Load: Failed to open \"%s\"", filename.c_str()); - return false; - } -} - -template -bool CodecApi::Initialize( - WaveCodingType coding_type) { - return codec_.Initialize(coding_type); -} - - -#endif // SPEECH_PATTS_LIBS_IO_CODEC_API_INL_H_ diff --git a/reaper-tmp/REAPER/wave/codec_api.h b/reaper-tmp/REAPER/wave/codec_api.h deleted file mode 100644 index 09e496c3575201ac7b23eefb60eadbbf61ca5145..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/wave/codec_api.h +++ /dev/null @@ -1,98 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#ifndef _CODEC_API_H_ -#define _CODEC_API_H_ - -#include -#include -#include - -#include "wave/codec_riff.h" - - - -class FileResource; - -template -class CodecApi { - public: - CodecApi() { } - ~CodecApi() { } - - // Loads audio samples from a file-resource. It sets the internal header - // parameters. - bool Load(FileResource *fr, std::vector *samples); - - // Loads audio samples from a file. It sets the internal header parameters. - template - bool Load(const std::string &filename, std::vector *samples); - - // Saves audio samples to a file-resource using the internal header - // parameters. - bool Save(const std::vector &samples, FileResource *fr) const; - - // Saves audio samples to a file, using the internal header parameters. - template - bool Save(const std::vector &samples, const std::string &filename) const; - - // Sets generic header information. - bool SetHeaderInfo(int sampling_rate, int num_samples_per_channel); - - // Gets generic header information. - bool GetHeaderInfo(int *sampling_rate, int *num_samples_per_channel); - - // Reads wave header from file-resource. - bool ReadHeader(FileResource *fr); - - // Reads the audio data from the FileResource, under the condition that the - // current position of the FileResource always points to the beginning of the - // audio-container. The latter condition is preserved by this function. When - // a codec with internal-state is used (i.e. iSAC), the internal state is kept - // between sequential reads. The codec is reset prior non-sequential reads. - // By definition, the codecs DO NOT check whether you are trying to read - // beyond the boundaries of the audio-data container, so, be careful. - bool ReadAudioData(int wave_start, int num_samples, - std::vector *samples, FileResource *fr); - - // Reads all audio data contained in the audio container held at the current - // position of the FileResource. The functions assumes and preserves the - // condition that the current position of the FileResource always points to - // the beginning of the audio container. - bool ReadAudioContainer(int container_size_in_bytes, - std::vector *samples, - FileResource *fr); - - int get_num_samples_per_channel() const; - int get_sampling_rate() const; - - bool set_num_samples_per_channel(int value); - bool set_sampling_rate(int value); - - // Initializes the codec for the particular coding type - bool Initialize(WaveCodingType coding_type); - - private: - int num_samples_per_channel_; - int sampling_rate_; - CodecImplementation codec_; - -}; - - -#include "codec_api-inl.h" - -#endif // _CODEC_API_H_ diff --git a/reaper-tmp/REAPER/wave/codec_riff.cc b/reaper-tmp/REAPER/wave/codec_riff.cc deleted file mode 100644 index 91213fc814c1dad1d43361ca132f057fe9928b68..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/wave/codec_riff.cc +++ /dev/null @@ -1,270 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#include "wave/codec_riff.h" - -#include - -#include "core/file_resource.h" - -uint8_t UlawCodec::Int16ToUlaw(int16_t pcm_val) const { - int16_t mask; - int16_t seg; - uint8_t uval; - // u-law inverts all bits - // Get the sign and the magnitude of the value. - if (pcm_val < 0) { - pcm_val = -pcm_val; - mask = 0x7f; - } else { - mask = 0xff; - } - if (pcm_val > kClip) { - pcm_val = kClip; // clip the magnitude - } - pcm_val += (kBias >> 2); - // Convert the scaled magnitude to segment number. - seg = SegmentSearch(pcm_val, seg_uend, 8); - // Combine the sign, segment, quantization bits, - // and complement the code word. - if (seg >= 8) { // out of range, return maximum value. - return (uint8_t)(0x7f ^ mask); - } else { - uval = (uint8_t) (seg << 4) | ((pcm_val >> (seg + 1)) & 0xF); - return uval ^ mask; - } -} - -int16_t UlawCodec::SegmentSearch(int16_t val, const int16_t *table, - int size) const { - for (int i = 0; i < size; ++i) { - if (val <= *table++) { - return i; - } - } - return size; -} - -int16_t UlawCodec::UlawToInt16(uint8_t ulaw) const { - // Complement to obtain normal u-law value. - ulaw = ~ulaw; - // Extract and bias the quantization bits. Then - // shift up by the segment number and subtract out the bias. - int16_t t = ((ulaw & kQuantMask) << 3) + kBias; - t <<= ((unsigned)ulaw & kSegMask) >> kSegShift; - return ((ulaw & kSignBit) ? (kBias - t) : (t - kBias)) >> 2; -} - -// -// Table and constants for mu-law coding and decoding: -// -const int16_t UlawCodec::seg_uend[8] = { 0x3f, 0x7f, 0xff, 0x1ff, 0x3ff, 0x7ff, - 0xfff, 0x1fff}; -const int16_t UlawCodec::kBias = 0x84; // Bias for linear code. -const int16_t UlawCodec::kClip = 8159; // max. linear value -const int16_t UlawCodec::kSignBit = 0x80; // Sign bit for a A-law byte. -const int16_t UlawCodec::kQuantMask = 0xf; // Quantization field mask. -const int16_t UlawCodec::kSegShift = 4; // Left shift for segment number. -const int16_t UlawCodec::kSegMask = 0x70; // Segment field mask. - -bool WavRiffCodec::ReadChunk(FileResource *fr, std::string *chunk) const { - if (chunk == NULL) { - return false; - } - char ch[5]; - if (fread(static_cast(&ch), 1, 4, fr->fp()) != 4) { - return false; - } - ch[4] = '\0'; - (*chunk) = ch; - return true; -} - -bool WavRiffCodec::CheckChunk(FileResource *fr, const char *data) const { - std::string chunk; - if (ReadChunk(fr, &chunk)) { - return chunk == data; - } - return false; -} - -bool WavRiffCodec::ReadHeader(FileResource *fr, - int32_t *num_samples_per_channel, - int32_t *sampling_rate) { - if (!CheckChunk(fr, "RIFF")) { - fprintf(stderr, "Invalid file: Expected \"RIFF\" in header"); - return false; - } - // File length minus first 8 bytes of RIFF description, we don't use it - int32_t len = 0; - if (fread(static_cast(&len), 4, 1, fr->fp()) != 1) { - fprintf(stderr, "Failed to read file length"); - return false; - } - - if (!CheckChunk(fr, "WAVE")) { - fprintf(stderr, "Invalid file: Expected \"WAVE\" in header"); - return false; - } - - std::string riff_type; - if (!ReadChunk(fr, &riff_type)) { - fprintf(stderr, "Invalid file: Missing RIFF type header"); - return false; - } - - if (riff_type != "fmt " && riff_type != "bext") { - fprintf(stderr, - "Invalid file: RIFF type must be fmt or bext"); - return false; - } - - // If there is a broadcast format extension, skip it. - if (riff_type == "bext") { - int32_t bext_length; - fread(static_cast(&bext_length), sizeof(bext_length), 1, fr->fp()); - fseek(fr->fp(), bext_length, SEEK_CUR); - std::string sub_type; - if (!CheckChunk(fr, "fmt ")) { - fprintf(stderr, "Invalid file: fmt subtype not found in bext header"); - return false; - } - } - - // Now skip wave format header only reading the number of channels and sample - // rate: - if (fread(static_cast(&len), sizeof(len), 1, fr->fp()) != 1) { - fprintf(stderr, "failed to skip file length"); - return false; - } - if (len < 16) { // bad format chunk length - fprintf(stderr, "Invalid of the wave format buffer"); - return false; - } - - int16_t n_channels = 0; - fseek(fr->fp(), sizeof(n_channels), SEEK_CUR); - if (fread(static_cast(&n_channels), sizeof(n_channels), 1, fr->fp()) != 1) { - fprintf(stderr, "Failed to read number of channels"); - return false; - } - if (n_channels != 1) { - fprintf(stderr, "Attempt to load multi channel audio"); - return false; - } - int32_t sample_rate; - if (fread(&sample_rate, sizeof(sample_rate), 1, fr->fp()) != 1) { - fprintf(stderr, "Failed to read sample rate"); - return false; - } - fseek(fr->fp(), 8, SEEK_CUR); - - // advance in the stream to skip the wave format block - fseek(fr->fp(), len - 16, SEEK_CUR); - - // now go to the end of "data" section, if found - while (!fr->eof()) { - if (fgetc(fr->fp()) == 'd' && - fgetc(fr->fp()) == 'a' && - fgetc(fr->fp()) == 't' && - fgetc(fr->fp()) == 'a') { - break; - } - } - if (fr->eof()) { - fprintf(stderr, "Unexpected end of file: no data"); - return false; - } - - int32_t num_bytes = 0; - if (fread(&num_bytes, sizeof(num_bytes), 1, fr->fp()) != 1) { - fprintf(stderr, "Failed to read number of bytes"); - return false; - } - - *sampling_rate = sample_rate; - *num_samples_per_channel = num_bytes / 2; - return true; -} - -bool WavRiffCodec::ReadAudioData(int32_t wave_start, - int32_t num_samples, - std::vector *samples, - FileResource *fr) { - samples->resize(num_samples); - bool status = true; - switch (coding_type_) { - case PCM16: { - status &= (fseek(fr->fp(), - wave_start * sizeof((*samples)[0]), SEEK_CUR) == 0); - uint32_t read = fread(&(*samples)[0], sizeof(int16_t), num_samples, fr->fp()); - if (read != num_samples) { - fprintf(stderr, "WaveIO::ReadSamples: only %d out of %d values read", - read, num_samples); - status = false; - } - } - break; - case ULAW8: { - UlawCodec sample_codec; - uint8_t *buffer = new uint8_t[num_samples]; - status &= (fseek(fr->fp(), wave_start * sizeof(*buffer), SEEK_CUR) == 0); - uint32_t read = fread(buffer, sizeof(uint8_t), num_samples, fr->fp()); - if (read != num_samples) { - fprintf(stderr, "WaveIO::ReadSamples: only %d out of %d values read", - read, num_samples); - status = false; - } - for (int i = 0 ; i < num_samples; ++i) { - (*samples)[i] = sample_codec.UlawToInt16(buffer[i]); - } - delete [] buffer; - } - break; - default: - fprintf(stderr, "WaveIO::ReadSamples: Unsupported coding type (%d)", - coding_type_); - status = false; - } - return status; -} - -bool WavRiffCodec::ReadAudioContainer(int container_size_in_bytes, - std::vector *samples, - FileResource *fr) { - int number_samples; - switch (coding_type_) { - case PCM16: - number_samples = container_size_in_bytes / 2; - break; - case PCM8: - case ULAW8: - number_samples = container_size_in_bytes; - break; - default: - return false; - } - return ReadAudioData(0, number_samples, samples, fr); -} - -bool WavRiffCodec::Initialize(WaveCodingType coding_type) { - if ((coding_type != PCM16) && (coding_type != PCM8) && - (coding_type != ULAW8)) { - return false; - } - coding_type_ = coding_type; - return true; -} diff --git a/reaper-tmp/REAPER/wave/codec_riff.h b/reaper-tmp/REAPER/wave/codec_riff.h deleted file mode 100644 index 78e3cc1058f8e2c6765cd43d2e408170412dea16..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/wave/codec_riff.h +++ /dev/null @@ -1,91 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License.*/ - -#ifndef _CODEC_RIFF_H_ -#define _CODEC_RIFF_H_ - -#include -#include -#include - -class FileResource; - -enum WaveCodingType { - PCM16 = 0, - PCM8, - ULAW8, - ISAC_18kbps, - ISAC_20kbps, - ISAC_22kbps, - ISAC_24kbps, - ISAC_26kbps, - ISAC_30kbps -}; - -class WavRiffCodec { - public: - WavRiffCodec() { } - ~WavRiffCodec() { } - - // The header is expected to be in the format: - // - // offset size description value - // 0x00 4 Chunk ID "RIFF" - // 0x04 4 Chunk data size file size - 8 - // 0x08 4 RIFF Type "WAVE" - // 0x10 Wave chunks - // - // A boolean success indicator can be used to check that the header was read. - bool ReadHeader(FileResource *fr, - int32_t *num_samples_per_channel, - int32_t *sample_rate); - - bool ReadAudioData(int32_t wave_start, - int32_t num_samples, - std::vector *samples, - FileResource *fr); - - bool ReadAudioContainer(int container_size_in_bytes, - std::vector *samples, - FileResource *fr); - - bool Initialize(WaveCodingType coding_type); - - private: - bool ReadChunk(FileResource *fr, std::string *chunk) const; - bool CheckChunk(FileResource *fr, const char *data) const; - - WaveCodingType coding_type_; -}; - -class UlawCodec { - public: - uint8_t Int16ToUlaw(int16_t pcm_val) const; - int16_t UlawToInt16(uint8_t ulaw) const; - - private: - int16_t SegmentSearch(int16_t val, const int16_t *table, int size) const; - - // Table and constants for mu-law coding and decoding - static const int16_t seg_uend[8]; - static const int16_t kBias; // Bias for linear code. - static const int16_t kClip; // max. linear value - static const int16_t kSignBit; // Sign bit for a A-law byte. - static const int16_t kQuantMask; // Quantization field mask. - static const int16_t kSegShift; // Left shift for segment number. - static const int16_t kSegMask; // Segment field mask. -}; - -#endif // _CODEC_RIFF_H_ diff --git a/reaper-tmp/REAPER/wave/wave.cc b/reaper-tmp/REAPER/wave/wave.cc deleted file mode 100644 index 74bcb19ba1d5961a80d785da1bb9fc895ad22d5e..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/wave/wave.cc +++ /dev/null @@ -1,158 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#include "wave/wave.h" - -#include -#include -#include - -#include "core/file_resource.h" -#include "wave/wave_io.h" - - - -WaveData::WaveData() : sample_rate_(16000) { -} -WaveData::~WaveData() { -} - -void WaveData::Set(int number_samples, int sampling_rate, const int16_t *data) { - set_sample_rate(sampling_rate); - resize(number_samples); - for (int n = 0; n < number_samples; ++n) { - (*this)[n] = data[n]; - } -} - -int WaveData::sample_rate(void) const { - return sample_rate_; -} - -void WaveData::set_sample_rate(int sample_rate) { - sample_rate_ = sample_rate; -} - -bool WaveData::Equals(const WaveData &wave_data, int threshold) const { - if (wave_data.size() != size()) { - fprintf(stderr, "Different number of samples"); - return false; - } - if (wave_data.sample_rate() != sample_rate()) { - fprintf(stderr, "Different sample rate"); - return false; - } - for (int i = 0; i < size(); ++i) { - if (abs(wave_data[i] - (*this)[i]) > threshold) { - fprintf(stderr, "Sample %d differs", i); - return false; - } - } - return true; -} - -Wave::Wave() { - data_ = new WaveData; - owner_ = true; -} - -Wave::Wave(const WaveData *data) { - // Shallow copy - data_ = const_cast(data); - owner_ = false; -} - -Wave::~Wave() { - Clear(); -} - -void Wave::Clear() { - if (owner_) { - delete data_; - } - data_ = NULL; - owner_ = true; -} - -void Wave::set_data(const WaveData *data) { - if (owner_) { - delete data_; - } - data_ = const_cast(data); - owner_ = false; -} - -void Wave::copy_data(const WaveData &data) { - if (owner_) { - delete data_; - } - data_ = new WaveData(data); - owner_ = true; -} - -void Wave::resize(int n, bool clear) { - if (!owner_) { - // copy the data to a local copy - WaveData *new_wave_data = new WaveData(*data_); - new_wave_data->set_sample_rate(data_->sample_rate()); - data_ = new_wave_data; - } - data_->resize(n); - owner_ = true; -} - -void Wave::ZeroFill() { - if (!owner_) { - // copy the data to a local copy - WaveData *new_wave_data = new WaveData(*data_); - new_wave_data->set_sample_rate(data_->sample_rate()); - data_ = new_wave_data; - owner_ = true; - } - std::fill(data_->begin(), data_->end(), 0); -} - -bool Wave::Load(FileResource *fr) { - WaveIO wave_io; - std::vector *samples = reinterpret_cast *>(data_); - int sampling_rate; - bool status = wave_io.Load(fr, samples, &sampling_rate); - set_sample_rate(sampling_rate); - return status; -} - -const int16_t Wave::kMaxShort = std::numeric_limits::max() - 1; -const int16_t Wave::kMinShort = std::numeric_limits::min() + 1; - -bool Wave::Amplify(float gain) { - return AmplifyBuffer(gain, &(*data_)[0], data_->size()); -} - -bool Wave::AmplifyBuffer(float gain, int16_t *buf, uint32_t size) { - bool clipped = false; - for (uint32_t i = 0; i < size; ++i) { - float sample = static_cast(buf[i]) * gain; - if (sample > kMaxShort) { - sample = kMaxShort; - clipped = true; - } else if (sample < kMinShort) { - sample = kMinShort; - clipped = true; - } - buf[i] = static_cast(sample); - } - return !clipped; -} diff --git a/reaper-tmp/REAPER/wave/wave.h b/reaper-tmp/REAPER/wave/wave.h deleted file mode 100644 index 6fc2c1199e97f713a2c68e9d62a9913254febbde..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/wave/wave.h +++ /dev/null @@ -1,142 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#ifndef _WAVE_H_ -#define _WAVE_H_ - -#include "core/file_resource.h" -#include -#include - - - -class WaveData : public std::vector { - public: - WaveData(); - ~WaveData(); - - void Set(int number_samples, int sampling_rate, const int16_t *data); - int sample_rate(void) const; - void set_sample_rate(int sample_rate); - bool Equals(const WaveData &wave_data, int threshold = 0) const; - - private: - int sample_rate_; -}; - -class Wave { - public: - Wave(); - explicit Wave(const WaveData *data); - ~Wave(); - - // Sets internal WaveData to the new object. It loses ownership. - void set_data(const WaveData *data); - - // Sets values of WaveData to the new data. It takes ownership. - void copy_data(const WaveData &data); - - // Returns the number of samples contained in the wave. - int num_samples() const { - return data_->size(); - } - - // Returns the sample rate of the wave. - int sample_rate() const { - return data_->sample_rate(); - } - - // Sets the sample rate for the wave. - void set_sample_rate(int sample_rate) { - data_->set_sample_rate(sample_rate); - } - - // Returns the value of the data at the given offset within the wave. - int16_t get(int f) const { - return (*data_)[f]; - } - - // Sets the wave data at the given position with the given value. - void set(int f, int16_t v) { - (*data_)[f] = v; - } - - // Returns the wave data. - const WaveData *data() const { - return data_; - } - - // Ensures that this wave 'parents' the wave data, - // so deleting it when this instance is deleted. - void adopt() { - owner_ = true; - } - - const int16_t &operator [] (int f) const { - return (*data_)[f]; - } - - int16_t &operator [] (int f) { - return (*data_)[f]; - } - - // Resizes - void resize(int n, bool clear = false); - - // Zero fills the wave data. - void ZeroFill(); - - // Applies the given gain to the wave data, limited to int16_t limits. - // Returns true if no clipping occurred. - // Returns false if the amplification resulted in clipping. - // The gain factor is a simple multiplicative factor, NOT a dB value. - bool Amplify(float gain_factor); - - // Applies the given gain to the given buffer location and the given - // number of entries in that buffer. - // Returns true if no clipping occurred. - // Returns false if the amplification resulted in clipping. - // The gain factor is a simple multiplicative factor, NOT a dB value. - static bool AmplifyBuffer(float gain_factor, - int16_t *buffer, uint32_t num_entries); - - bool Load(const std::string &filename); - bool Load(FileResource *fr); - - protected: - // Clears the audio data down and sets the owner. - void Clear(); - - private: - static const int16_t kMaxShort; - static const int16_t kMinShort; - // The wave data. - WaveData *data_; - // Holds whether this instance is to sole owner of the wave data, allowing - // for correct object deletion. - bool owner_; -}; - -inline bool Wave::Load(const std::string &filename) { - FileResource fr(filename, "rb"); - if (!fr.Get()) { - fprintf(stderr, "Failed to open \"%s\"", filename.c_str()); - return false; - } - return Load(&fr); -} - -#endif // SPEECH_PATTS_ENGINE_LING_ARCH_WAVE_H_ diff --git a/reaper-tmp/REAPER/wave/wave_io-inl.h b/reaper-tmp/REAPER/wave/wave_io-inl.h deleted file mode 100644 index ddccb3fd4196055807f24739f5990a4b6e0f83c6..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/wave/wave_io-inl.h +++ /dev/null @@ -1,39 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - -http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software - distributed under the License is distributed on an "AS IS" BASIS, - WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#ifndef _WAVE_IO_INL_H_ -#define _WAVE_IO_INL_H_ - -#include -#include - -inline WaveCodingType WaveIO::get_coding_type() const { - return coding_type_; -} - -template -bool WaveIO::Load(const std::string &filename, - std::vector *samples, - int32_t *sample_rate) { - FileResource fr(filename, "rb"); - if (!fr.Get()) { - fprintf(stderr, "Failed to open \"%s\"", filename.c_str()); - return false; - } - return Load(&fr, samples, sample_rate); -} - -#endif // _WAVE_IO_INL_H_ diff --git a/reaper-tmp/REAPER/wave/wave_io.cc b/reaper-tmp/REAPER/wave/wave_io.cc deleted file mode 100644 index d34f24a73bd7facbaa433da22e931634f08722e6..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/wave/wave_io.cc +++ /dev/null @@ -1,111 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#include "wave/wave_io.h" - -#include - -#include "core/file_resource.h" - - - -WaveIO::WaveIO(void) { - Initialize(PCM16); -} - -WaveIO::WaveIO(WaveCodingType coding_type) { - Initialize(coding_type); -} - -bool WaveIO::ReadAudioData(int32_t wave_start, - int32_t num_samples, - std::vector *samples, - FileResource *fr) { - switch (coding_type_) { - case PCM16: - case PCM8: - case ULAW8: - return codec_riff_.ReadAudioData(wave_start, num_samples, samples, fr); - default: - fprintf(stderr, "WaveIO:ReadAudioData: unknown coding type %d", - coding_type_); - } - return false; -} - -bool WaveIO::ReadAudioContainer(int container_size_in_bytes, - std::vector *samples, - FileResource *fr) { - switch (coding_type_) { - case PCM16: - case PCM8: - case ULAW8: - return codec_riff_.ReadAudioContainer(container_size_in_bytes, - samples, fr); - fprintf(stderr, "WaveIO:ReadAudioData: unknown coding type %d", - coding_type_); - default: - fprintf(stderr, "WaveIO: coding type not supported %d", coding_type_); - return false; - } - return false; -} - -bool WaveIO::ReadHeader(FileResource *fr, - int32_t *num_samples_per_channel, - int32_t *sampling_rate) { - switch (coding_type_) { - case PCM16: - case PCM8: - case ULAW8: { - bool status = codec_riff_.ReadHeader(fr); - if (!status) { - return false; - } - *num_samples_per_channel = codec_riff_.get_num_samples_per_channel(); - *sampling_rate = codec_riff_.get_sampling_rate(); - return true; - } - default: - fprintf(stderr, "WaveIO:ReadHeader: unknown coding type %d", - coding_type_); - } - return false; -} - -bool WaveIO::Load(FileResource *fr, - std::vector *samples, - int32_t *sample_rate) { - int32_t num_samples; - if (!ReadHeader(fr, &num_samples, sample_rate)) { - return false; - } - samples->resize(num_samples); - return ReadAudioData(0, num_samples, samples, fr); -} - -bool WaveIO::Initialize(WaveCodingType coding_type) { - coding_type_ = coding_type; - switch (coding_type_) { - case PCM16: - case PCM8: - case ULAW8: - return codec_riff_.Initialize(coding_type); - default: - fprintf(stderr, "WaveIO:WaveIO: unknown coding type %d", coding_type_); - return false; - } -} diff --git a/reaper-tmp/REAPER/wave/wave_io.h b/reaper-tmp/REAPER/wave/wave_io.h deleted file mode 100644 index 04f80da0fb8289322de0f83e74187b8cd73adbe2..0000000000000000000000000000000000000000 --- a/reaper-tmp/REAPER/wave/wave_io.h +++ /dev/null @@ -1,79 +0,0 @@ -/* -Copyright 2015 Google Inc. All rights reserved. - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -#ifndef _WAVE_IO_H_ -#define _WAVE_IO_H_ - -#include "wave/codec_api.h" -#include "wave/codec_riff.h" - -class FileResource; - -// This class provides the main interface for waveform input/output. The class -// abstracts the implementation of the codec from the rest of the system. -class WaveIO { - public: - WaveIO(); - explicit WaveIO(WaveCodingType coding_type); - ~WaveIO() { } - - template - bool Load(const std::string &filename, - std::vector *samples, - int32_t *sample_rate); - - // Loads the header-info and the audio data from a file, starting from the - // current position of the FileResource. - bool Load(FileResource *fr, - std::vector *samples, - int32_t *sample_rate); - - // Reads the header-info from the FileResource, starting from the current - // position. - bool ReadHeader(FileResource *fr, - int32_t *num_samples_per_channel, - int32_t *sample_rate); - - // Reads the audio data from the FileResource, under the condition that the - // current position of the FileResource always points to the beginning of the - // audio-container. The latter condition is preserved by this function. When - // a codec with internal-state is used (i.e. iSAC), the internal state is kept - // between sequential reads. The codec is reset prior non-sequential reads. - bool ReadAudioData(int32_t wave_start, - int32_t num_samples, - std::vector *samples, - FileResource *fr); - - // Reads all audio data contained in the audio container held at the current - // position of the FileResource. - bool ReadAudioContainer(int container_size_in_bytes, - std::vector *samples, - FileResource *fr); - - // Sets WaveIO to the corresponding codec, and also initializes the codec - // itself. - bool Initialize(WaveCodingType coding_type); - - WaveCodingType get_coding_type() const; - - private: - CodecApi codec_riff_; - WaveCodingType coding_type_; -}; - -#include "wave_io-inl.h" - -#endif // _WAVE_IO_H_