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diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/CMakeLists.txt b/lib/libsamplerate/libsamplerate-0.2.2/docs/CMakeLists.txt deleted file mode 100755 index e1e8c697..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/CMakeLists.txt +++ /dev/null @@ -1,19 +0,0 @@ -install( - FILES - api.md - api_callback.md - api_full.md - api_misc.md - api_simple.md - bugs.md - download.md - faq.md - history.md - index.md - license.md - lists.md - quality.md - win32.md - SRC.png - DESTINATION - ${CMAKE_INSTALL_DOCDIR}) diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/SRC.css b/lib/libsamplerate/libsamplerate-0.2.2/docs/SRC.css deleted file mode 100755 index f5c10c86..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/SRC.css +++ /dev/null @@ -1,56 +0,0 @@ -a:link { - color: #FB1465; -} - -a:active, a:visited { - color: #FB1465; -} - -body { - background: black; - color: white; - font-family: arial, helvetica, sans-serif; -} - -h1, h2, h3, h4, h5, h6 { - color: #FB1465; -} - -code, kbd, samp, var { - font-family: courier, monospace; - font-size: 1em; -} - -main { - padding-left: 3%; - padding-right: 3%; -} - -pre { - font-family: courier, monospace; - font-size: 1em; -} - -.image-logo { - display: block; - margin-left: auto; - margin-right: auto; - margin-bottom: 1em; -} - -.indent_block { - margin-left: 10%; - margin-right: 10%; -} - -.nf { - font-weight: bold; -} - -.container { - display: flex; -} - -.content { - flex: 1; -} diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/SRC.png b/lib/libsamplerate/libsamplerate-0.2.2/docs/SRC.png Binary files differdeleted file mode 100755 index 85f327b0..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/SRC.png +++ /dev/null diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/_config.yml b/lib/libsamplerate/libsamplerate-0.2.2/docs/_config.yml deleted file mode 100755 index 79eae420..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/_config.yml +++ /dev/null @@ -1,8 +0,0 @@ ---- -author: Erik de Castro Lopo, erikd@mega-nerd.com -description: An audio Sample Rate Conversion library -keywords: libsamplerate, sound, resample, audio, dsp, Linux -version: 0.2.1 - -markdown: kramdown ---- diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/_layouts/default.html b/lib/libsamplerate/libsamplerate-0.2.2/docs/_layouts/default.html deleted file mode 100755 index 33d41f53..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/_layouts/default.html +++ /dev/null @@ -1,49 +0,0 @@ -<!DOCTYPE html"> -<html lang="en"> - -<head> - <meta charset="utf-8"> - <meta name="viewport" content="width=device-width, initial-scale=1.0"> - <meta name="description" content="{{ site.description | page.description }}"> - <meta name="keywords" content="{{ site.keywords | page.keywords }}"> - <title>{{ page.title }}</title> - <link rel="stylesheet" href="SRC.css" type="text/css" media="all"> -</HEAD> - -<body> - <footer> - <img src="SRC.png" class="image-logo" height=100 width=760 alt="SRC.png"> - </footer> - <div class="container"> - <div class="navbar"> - <nav> - <a href="index.html">Home</a><br> - <a href="license.html">License</a><br> - <a href="history.html">History</a><br> - <a href="download.html">Download</a><br> - <a href="quality.html">Quality</a><br> - <a href="api.html">API</a><br> - <a href="bugs.html">Bug Reporting</a><br> - <a href="win32.html">On Win32</a><br> - <a href="faq.html">FAQ</a><br> - <a href="lists.html">Mailing Lists</a><br> - <a href="https://github.com/libsndfile/libsamplerate/blob/master/ChangeLog">ChangeLog</a><br> - </nav> - <div class="block"> - <p>Author:<br> Erik de Castro Lopo</p> - </div> - </div> - <div class="content"> - <main> - <article> - - {{ content }} - - </article> - </main> - </div> - </div> - -</body> - -</html>
\ No newline at end of file diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/api.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/api.md deleted file mode 100755 index f673e0b7..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/api.md +++ /dev/null @@ -1,46 +0,0 @@ ---- -layout: default ---- - -# Applications Programming Interface - -The publically callable functions of libsamplerate are all listed in the -**\<samplerate.h\>** header file. In order to use any of the functionality of -libsamplerate, you need to add - -```c -#include <samplerate.h> -``` - -to the top of any function that call any of the following functions. You will -also need to link you binary with the libsamplerate library. - -The API allows three methods for accessing the capabilities of the library: - -- A [simple interface](api_simple.md) which can sample rate convert a single - block of samples (one or more channels) in one go. The simple API is less - capable than the full API. -- A [more fully featured interface](api_full.md) which allows time varying - sample rate conversion on streaming data (again one or more channels). -- A [callback interface](api_callback.md) which has the same functionality as - the interface above but allows the details of input and output to be - separated. The output is generated by call a read function and the library - calls a user supplied callback function to obtain its input. This interface is - particularly well suited to applications where the output sample rate is - varied with time. - -**NB :** All three access methods are able to process multi channel interleaved -data. - -The parts of the API which are common to all three interfaces are: - -- The [error reporting](api_misc.md#error-reporting) mechanisim. -- The available [converter](api_misc.md#converters) types. -- The [SRC_DATA](api_misc.md#src_data) struct. - -All three versions of the API are restricted to operating on buffers of ISO C -Standard **float** data. However, there are two [auxiliary functions](api_misc.md#auxiliary-functions) -for converting arrays of float data to and from short data. - -**Note:** The **tests/** and **examples/** directories of the source code -distribution contain numerous example programs showing the use of the library. diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/api_callback.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/api_callback.md deleted file mode 100755 index 45ca868e..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/api_callback.md +++ /dev/null @@ -1,88 +0,0 @@ ---- -layout: default ---- - -# Callback API - -The callback API consists of the following functions : - -```c -typedef long (*src_callback_t) (void *cb_data, float **data) ; - -SRC_STATE* src_callback_new (src_callback_t func, - int converter_type, int channels, - int *error, void* cb_data) ; - -SRC_STATE* src_delete (SRC_STATE *state) ; - -long src_callback_read (SRC_STATE *state, double src_ratio, - long frames, float *data) ; - -int src_reset (SRC_STATE *state) ; -int src_set_ratio (SRC_STATE *state, double new_ratio) ; -``` - -Like the [simple API](api_simple.md) and the [full API](api_full.md), the -callback based API is able to operate on interleaved multi channel data. - -An example of the use of the callback based API can be found in the -**varispeed-play** program in the **examples/** directory of the source code -tarball. - -## Initialisation - -```c -SRC_STATE* src_callback_new (src_callback_t func, - int converter_type, int channels, - int *error, void* cb_data) ; -``` - -The **src_callback_new** function returns an anonymous pointer to a sample rate -converter callback object, src_state. This is the same type of object as that -returned by [src\_new](api_full.md#initialisation), but with different internal -state. Although these are the same object type, they cannot be used -interchangeably. If an error occurs the function returns a NULL pointer and -fills in the error value pointed to by the **error** pointer supplied by the -caller. - -The caller then passes the SRC_STATE object to the **src_callback_read** -function to pull data out of the converter. When the caller is finished with the -converter they should pass it to the clean up function [src_delete](api_full.md#cleanup). - -The **func** parameter is a user supplied function which must match the -**src_callback_t** type signature while **cb_data** is a pointer to data which -be passed as the first parameter to the user supplied callback function. This -function is called by the converter whenever it needs input data as a result of -being calls to **src_callback_read**. - -If the converter was initialised to work with more than one channel, the -callback function must work with mutiple channels of interleaved data. The -callback function should return the number of frames it supplying to the -converter. For multi channel files, this return value should be the number of -floats divided by the number of channels. - -The converter must be one of the supplied converter types documented [here](api_misc.md#converters). - -The caller then passes the SRC_STATE pointer to the **src_callback_read** -function to pull data out of the converter. - -## Callback Read - -```c -long src_callback_read (SRC_STATE *state, double src_ratio, - long frames, float *data) ; -``` - -The **src_callback_read** function is passed the [**SRC_STATE**](api_misc.md#src_data) -pointer returned by **src_callback_new**, the coversion ratio -(output_sample_rate / input_sample_rate), the maximum number of output frames to -generate and a pointer to a buffer in which to place the output data. For multi -channel files, the data int the output buffer is stored in interleaved format. - -The **src_callback_read** function returns the number of frames generated or -zero if an error occurs or it runs out of input (ie the user supplied callback -function returns zero and there is no more data buffered internally). If an -error has occurred, the function [src_error](api_misc.md#error-reporting) -will return non-zero. - -See also : [**src_set_ratio**](api_full.md#set-ratio) diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/api_full.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/api_full.md deleted file mode 100755 index 63eacdc9..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/api_full.md +++ /dev/null @@ -1,146 +0,0 @@ ---- -layout: default ---- - -# Full API - -The full API consists of the following functions : - -```c -SRC_STATE* src_new (int converter_type, int channels, int *error) ; -SRC_STATE* src_delete (SRC_STATE *state) ; - -int src_process (SRC_STATE *state, SRC_DATA *data) ; -int src_reset (SRC_STATE *state) ; -int src_set_ratio (SRC_STATE *state, double new_ratio) ; -``` - -## Initialisation - -```c -SRC_STATE* src_new (int converter_type, int channels, int *error) ; -``` - -The **src\_new** function returns an anonymous pointer to a sample rate -converter object, src\_state. If an error occurs the function returns a -NULL pointer and fills in the error value pointed to by the **error** -pointer supplied by the caller. The converter must be one of the -supplied converter types documented [here](api_misc.md#converters). - -## Cleanup - -```c -SRC_STATE* src_delete (SRC_STATE *state) ; -``` - -The **src\_delete** function frees up all memory allocated for the given -sample rate converter object and returns a NULL pointer. The caller is -responsible for freeing any memory passed to the sample rate converter -via the pointer to the **SRC\_DATA** struct. - -## Process - -```c -int src_process (SRC_STATE *state, SRC_DATA *data) ; -``` - -The **src\_process** function processes the data provided by the caller -in an **SRC\_DATA** struct using the sample rate converter object -specified by the **SRC\_STATE** pointer. When operating on streaming -data, this function can be called over and over again, with each new -call providing new input data and returning new output data. - -The **SRC\_DATA** struct passed as the second parameter to the -**src\_process** function has the following fields: - -```c -typedef struct -{ const float *data_in; - float *data_out; - - long input_frames, output_frames ; - long input_frames_used, output_frames_gen ; - - int end_of_input ; - - double src_ratio ; -} SRC_DATA ; -``` - -The fields of this struct which must be filled in by the caller are: - -data_in -: A pointer to the input data samples. - -input_frames -: The number of frames of data pointed to by data_in. - -data_out -: A pointer to the output data samples. - -output_frames -: Maximum number of frames pointer to by data_out. - -src_ratio -: Equal to output_sample_rate / input_sample_rate. - -end_of_input -: Equal to 0 if more input data is available and 1 otherwise. - -Note that the data\_in and data\_out arrays may not overlap. If they do, -the library will return an error code. - -When the **src\_process** function returns **output\_frames\_gen** will -be set to the number of output frames generated and -**input\_frames\_used** will be set to the number of input frames -consumed to generate the provided number of output frames. - -The **src\_process** function returns non-zero if an error occurs. The -non-zero error return value can be decoded into a text string using the -function documented [here](api_misc.md#error-reporting). - -## Reset - -```c -int src_reset (SRC_STATE *state) ; -``` - -The **src\_reset** function resets the internal state of the sample rate -converter object to the same state it had immediately after its creation -using **src\_new**. This should be called whenever a sample rate -converter is to be used on two separate, unrelated pieces of audio. - -## Clone - -```c -SRC_STATE* src_clone (SRC_STATE *state, int *error) ; -``` - -The **src_clone** function creates a copy of the internal state of the sample -rate converter object. The output of the next call to **src\_process** will be -identical for both the original and cloned state (given the same **SRC_DATA** -input). This could be used to later resume sample rate conversion at a specific -location in a stream with the same state, which may be useful in real-time -applications. - -If an error occurs the function returns a NULL pointer and fills in the -error value pointed to by the **error** pointer supplied by the caller. - -## Set Ratio - -```c -int src_set_ratio (SRC_STATE *state, double new_ratio) ; -``` - -When using the **src_process** or **src_callback_process** APIs and updating the -**src_ratio** field of the **SRC_STATE** struct, the library will try to -smoothly transition between the conversion ratio of the last call and the -conversion ratio of the current call. - -If the user wants to bypass this smooth transition and achieve a step response in -the conversion ratio, the **src_set_ratio** function can be used to set the -starting conversion ratio of the next call to **src_process** or -**src_callback_process**. - -This function returns non-zero on error and the error return value can be -decoded into a text string using the function documented [here](api_misc.md#error-reporting). diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/api_misc.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/api_misc.md deleted file mode 100755 index 9e9a53ba..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/api_misc.md +++ /dev/null @@ -1,145 +0,0 @@ ---- -layout: default ---- - -# Miscellaneous API Documentation - -## Error Reporting - -Most of the API functions either return an integer error (ie **src_simple** and -**src_process**) or return an integer error value via an int pointer parameter -(**src_new**). These integer error values can be converted into human-readable -text strings by calling the function: - -```c -const char* src_strerror (int error) ; -``` - -which will return an error string for valid error numbers, the string -\"No Error\" for an error value of zero or a NULL pointer if no error message -has been defined for that error value. - -## Converters - -Secret Rabbit Code has a number of different converters which can be selected -using the **converter_type** parameter when calling **src_simple** or -**src_new**. Currently, the five converters available are: - -```c -enum -{ - SRC_SINC_BEST_QUALITY = 0, - SRC_SINC_MEDIUM_QUALITY = 1, - SRC_SINC_FASTEST = 2, - SRC_ZERO_ORDER_HOLD = 3, - SRC_LINEAR = 4 -} ; -``` - -As new converters are added, they will be given a number corresponding to the next -integer. - -The details of these converters are as follows: - -- **SRC_SINC_BEST_QUALITY** - This is a bandlimited interpolator derived from - the mathematical **sinc** function and this is the highest quality sinc based - converter, providing a worst case Signal-to-Noise Ratio (SNR) of 97 decibels - (dB) at a bandwidth of 97%. All three SRC_SINC_* converters are based on the - techniques of [Julius O. Smith](http://ccrma.stanford.edu/~jos/resample/) - although this code was developed independently. -- **SRC_SINC_MEDIUM_QUALITY** - This is another bandlimited interpolator much - like the previous one. It has an SNR of 97dB and a bandwidth of 90%. The speed - of the conversion is much faster than the previous one. -- **SRC_SINC_FASTEST** - This is the fastest bandlimited interpolator and has an - SNR of 97dB and a bandwidth of 80%. -- **SRC_ZERO_ORDER_HOLD** - A Zero Order Hold converter (interpolated value is - equal to the last value). The quality is poor but the conversion speed is - blindlingly fast. Be aware that this interpolator is not bandlimited, and the - user is responsible for adding anti-aliasing filtering. -- **SRC_LINEAR** - A linear converter. Again the quality is poor, but the - conversion speed is blindingly fast. This interpolator is also not bandlimited, - and the user is responsible for adding anti-aliasing filtering. - -There are two functions that give either a (text string) name or description for -each converter: - -```c -const char *src_get_name (int converter_type) ; -const char *src_get_description (int converter_type) ; -``` - -The name will typically be a short string for use in a dialog box, while the -description string is longer. - -Both of these functions return a NULL pointer if there is no converter for the -given **converter_type** value. Since the converters have consecutive -**converter_type** values, the caller is easily able to figure out the number of -converters at run time. This enables a binary dynamically linked against an old -version of the library to know about converters from later versions of the -library as they become available. - -## SRC_DATA - -Both the simple and the full featured versions of the API use the **SRC_DATA** -struct to pass audio and control data into the sample rate converter. This -struct is defined as: - -```c -typedef struct -{ const float *data_in; - float *data_out ; - - long input_frames, output_frames ; - long input_frames_used, output_frames_gen ; - - int end_of_input ; - - double src_ratio ; -} SRC_DATA ; -``` - -The **data_in** pointer is used to pass audio data into the converter while the -**data_out** pointer supplies the converter with an array to hold the -converter's output. For a converter which has been configured for multichannel -operation, these pointers need to point to a single array of interleaved data. - -The **input_frames** and **output_frames** fields supply the converter with the -lengths of the arrays (in frames) pointed to by the **data_in** and **data_out** -pointers respectively. For monophonic data, these values would indicate the -length of the arrays while for multi channel data these values would be equal to -the length of the array divided by the number of channels. - -The **end_of_input** field is only used when the sample rate converter is used -by calling the **src_process** function. In this case it should be set to zero -if more buffers are to be passed to the converter and 1 if the current buffer is -the last. - -Finally, the **src_ratio** field specifies the conversion ratio defined as the -output sample rate divided by the input sample rate. For a connected set of -buffers, this value can vary on each call to **src_process** resulting in a -time varying sample rate conversion process. For time varying sample rate -conversions, the ratio will be linearly interpolated between the **src_ratio** -value of the previous call to **src_process** and the value for the current -call. - -The **input_frames_used** and **output_frames_gen** fields are set by the -converter to inform the caller of the number of frames consumed from the -**data_in** array and the number of frames generated in the **data_out** array -respectively. These values are for the current call to **src_process** only. - -## Auxiliary Functions - -There are four auxiliary functions for converting arrays of float data to and -from short or int data. These functions are defined as: - -```c -void src_short_to_float_array (const short *in, float *out, int len) ; -void src_float_to_short_array (const float *in, short *out, int len) ; -void src_int_to_float_array (const int *in, float *out, int len) ; -void src_float_to_int_array (const float *in, int *out, int len) ; -``` - -The float data is assumed to be in the range [-1.0, 1.0] and it is automatically -scaled on the conversion to and from float. On the float to short/int conversion -path, any data values which would overflow the range of short/int data are -clipped. diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/api_simple.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/api_simple.md deleted file mode 100755 index 9947f3b6..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/api_simple.md +++ /dev/null @@ -1,71 +0,0 @@ ---- -layout: default ---- - -# Simple API - -**Important Note:** The simple API is not designed to work on small chunks of a -larger piece of audio. If you attempt to use it this way you are doing it wrong -and will not get the results you want. For processing audio data in chunks you -**must** use the [full api](api_full.md) or the [callback based api](api_callback.md). - -The simple API consists of a single function: - -```c -int src_simple (SRC_DATA *data, int converter_type, int channels) ; -``` - -The use of this function rather than the more fully featured API requires the -caller to know the total length of the input data before hand and that all input -and output data can be held in the system's memory at once. It also assumes that -there is a single constant ratio between input and output sample rates. - -Dealing with the easy stuff first, the **converter_type** parameter should be -one of the values defined in **samplerate.h** and documented [here](api_misc.md#converters) -while the **channels** parameter specifies the number of interleaved channels -that the sample rate converter is being asked to process (number of input -channels and output channels is always equal). There is no hard upper limit on -the number of channels; it is limited purely by the amount of memory available. - -The first parameter to **src_simple** is a pointer to an **SRC_DATA** struct -(more info [here](api_misc.md#src_data)) defined as follows: - -```c -typedef struct -{ const float *data_in; - float *data_out ; - - long input_frames, output_frames ; - long input_frames_used, output_frames_gen ; - - int end_of_input ; - - double src_ratio ; -} SRC_DATA ; -``` - -The fields of this struct which must be filled in by the caller are: - -data_in -: A pointer to the input data samples. - -input_frames -: The number of frames of data pointed to by data_in. - -data_out -: A pointer to the output data samples. - -output_frames -: Maximum number of frames pointer to by data_out. - -src_ratio -: Equal to output_sample_rate / input_sample_rate. - -When the **src_simple** function returns **output_frames_gen** will be set to -the number of output frames generated and **input_frames_used** will be set to -the number of input frames used to generate the provided number of output -frames. - -The **src_simple** function returns a non-zero value when an error occurs. See -[here](api_misc.md#error-reporting) for how to convert the error value into a -text string. diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/bugs.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/bugs.md deleted file mode 100755 index ef24c459..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/bugs.md +++ /dev/null @@ -1,37 +0,0 @@ ---- -layout: default ---- - -# Bug Reporting - -If you are a user and have a problem using libsamplerate with another piece of -software, you should contact the author of that other software and get them to -explore their use of this library and possibly submit a bug report. If you are a -coder and think you have found a bug in libsamplerate then read on. - -Secret Rabbit Code is an extremely complex piece of code but I do think that it -is relatively bug free. In addition, the source code distribution includes a -comprehensive test suite for regression testing. This means it is extremely -unlikely that new bugs creep in when modifications are made to the code. - -SRC is also not the most simple library to use which means that I do get a -number of bug reports which turn out to be bugs in the user's program rather -than bugs in SRC. Up until now, I have investigated each bug report as -thoroughly as possible. Unfortunately, this chews up a lot of my time which -could otherwise be spent improving SRC, working on other Free Software or -spending time with my family. - -I have therefore decided, that I cannot investigate any bug report unless the -person reporting the problem can supply me with a short self contained test -program or a modification to one of the existing test programs in the tests/ -directory of the source code distribution. The test program should meet the -following criteria: - -- Written in C or C++. -- Does not use any libraries or header files other than the ones which are - standard for the relevant languages. (Of course libsamplerate can be used :-)). -- It is the minimal program which can adequately display the problem. -- It clearly displays the criteria for pass or fail. - -Supplying a good test program will maximize the speed with which your bug report -gets dealt with. diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/download.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/download.md deleted file mode 100755 index 7f0d32d0..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/download.md +++ /dev/null @@ -1,15 +0,0 @@ ---- -layout: default ---- - -# Download - -The latest version of Secret Rabbit Code is {{ site.version }}. - -Download it on [GitHub releases page](https://github.com/libsndfile/libsamplerate/releases/latest). - -Compiling libsamplerate is relatively easy. The INSTALL file in the top level -directory gives instructions on compiling and installing libsamplerate on -Unix-like systems (including MacOSX). For Win32 there are instructions in the -docs/ directory of the tarball. These instructions are mirrored -[here](win32.md). diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/faq.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/faq.md deleted file mode 100755 index da3d87a9..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/faq.md +++ /dev/null @@ -1,193 +0,0 @@ ---- -layout: default ---- - -# Frequently Asked Questions - -1. [Q1 : Is it normal for the output of libsamplerate to be louder than its input?](#Q001) -2. [Q2 : On Unix/Linux/MacOSX, what is the best way of detecting the presence and location of libsamplerate and its header file using autoconf?](#Q002) -3. [Q3 : If I upsample and downsample to the original rate, for example 44.1-\>96-\>44.1, do I get an identical signal as the one before the up/down resampling?](#Q003) -4. [Q4 : If I ran src\_simple (libsamplerate) on small chunks (160 frames) would that sound bad?](#Q004) -5. [Q5 : I\'m using libsamplerate but the high quality settings sound worse than the SRC\_LINEAR converter. Why?](#Q005) -6. [Q6 : I\'m use the SRC\_SINC\_\* converters and up-sampling by a ratio of 2. I reset the converter and put in 1000 samples and I expect to get 2000 samples out, but I\'m getting less than that. Why?](#Q006) -7. [Q7 : I have input and output sample rates that are integer values, but the API wants me to divide one by the other and put the result in a floating point number. Won\'t this case problems for long running conversions?](#Q007) - -## Q1 : Is it normal for the output of libsamplerate to be louder than its input? {#Q001} - -The output of libsamplerate will be roughly the same volume as the input. -However, even if the input is strictly in the range (-1.0, 1.0), it is still -possible for the output to contain peak values outside this range. - -Consider four consecutive samples of [0.5 0.999 0.999 0.5]. If we are up -sampling by a factor of two we need to insert samples between each of the -existing samples. Its pretty obvious then, that the sample between the two 0.999 -values should and will be bigger than 0.999. - -This means that anyone using libsamplerate should normalize its output before -doing things like saving the audio to a 16 bit WAV file. - -## Q2 : On Unix/Linux/MacOSX, what is the best way of detecting the presence and location of libsamplerate and its header file using autoconf? {#Q002} - -libsamplerate uses the pkg-config (man pkg-config) method of registering itself -with the host system. The best way of detecting its presence is using something -like this in configure.ac (or configure.in): - - PKG_CHECK_MODULES(SAMPLERATE, samplerate >= 0.1.3, - ac_cv_samplerate=1, ac_cv_samplerate=0) - - AC_DEFINE_UNQUOTED([HAVE_SAMPLERATE],${ac_cv_samplerate}, - [Set to 1 if you have libsamplerate.]) - - AC_SUBST(SAMPLERATE_CFLAGS) - AC_SUBST(SAMPLERATE_LIBS) - -This will automatically set the **SAMPLERATE_CFLAGS** and **SAMPLERATE_LIBS** -variables which can be used in Makefile.am or Makefile.in like this: - - SAMPLERATE_CFLAGS = @SAMPLERATE_CFLAGS@ - SAMPLERATE_LIBS = @SAMPLERATE_LIBS@ - -If you install libsamplerate from source, you will probably need to set the -**PKG_CONFIG_PATH** environment variable's suggested at the end of the -libsamplerate configure process. For instance on my system I get this: - - -=-=-=-=-=-=-=-=-=-= Configuration Complete =-=-=-=-=-=-=-=-=-=-=- - - Configuration summary : - - Version : ..................... 0.1.3 - Enable debugging : ............ no - - Tools : - - Compiler is GCC : ............. yes - GCC major version : ........... 3 - - Extra tools required for testing and examples : - - Have FFTW : ................... yes - Have libsndfile : ............. yes - Have libefence : .............. no - - Installation directories : - - Library directory : ........... /usr/local/lib - Program directory : ........... /usr/local/bin - Pkgconfig directory : ......... /usr/local/lib/pkgconfig - -## Q3 : If I upsample and downsample to the original rate, for example 44.1->96->44.1, do I get an identical signal as the one before the up/down resampling? {#Q003} - -The short answer is that for the general case, no, you don't. The long answer is -that for some signals, with some converters, you will get very, very close. - -In order to resample correctly (ie using the **SRC_SINC_*** converters), -filtering needs to be applied, regardless of whether its upsampling or -downsampling. This filter needs to attenuate all frequencies above 0.5 times the -minimum of the source and destination sample rate (call this fshmin). Since the -filter needed to achieve full attenuation at this point, it has to start rolling -off a some frequency below this point. It is this rolloff of the very highest -frequencies which causes some of the loss. - -The other factor is that the filter itself can introduce transient artifacts -which causes the output to be different to the input. - -## Q4 : If I ran src_simple on small chunks (say 160 frames) would that sound bad? {#Q004} - -Well if you are after odd sound effects, it might sound OK. If you are after -high quality sample rate conversion you will be disappointed. - -The src_simple() was designed to provide a simple to use interface for people -who wanted to do sample rate conversion on say, a whole file all at once. - -## Q5 : I'm using libsamplerate but the high quality settings sound worse than the SRC_LINEAR converter. Why? {#Q005} - -There are two possible problems. Firstly, if you are using the src_simple() -function on successive blocks of a stream of samples, you will get bad results. -The src_simple() function is designed for use on a whole sound file, all at -once, not on contiguous segments of the same sound file. To fix the problem, you -need to move to the src_process() API or the callback based API. - -If you are already using the src_process() API or the callback based API and the -high quality settings sound worse than SRC_LINEAR, then you have other problems. -Read on for more debugging hints. - -All of the higher quality converters need to keep state while doing conversions -on segments of a large chunk of audio. This state information is kept inside the -private data pointed to by the SRC_STATE pointer returned by the src_new() -function. This means, that when you want to start doing sample rate conversion -on a stream of data, you should call src_new() to get a new SRC_STATE pointer -(or alternatively, call src_reset() on an existing SRC_STATE pointer). You -should then pass this SRC_STATE pointer to the src_process() function with each -new block of audio data. When you have completed the conversion, you can then -call src_delete() on the SRC_STATE pointer. - -If you are doing all of the above correctly, you need to examine your usage of -the values passed to src\_process() in the [SRC_DATA](api_misc.md#src_data) -struct. Specifically: - -- Check that input_frames and output_frames fields are being set in terms of - frames (number of sample values times channels) instead of just the number of - samples. -- Check that you are using the return values input_frames_used and - output_frames_gen to update your source and destination pointers correctly. -- Check that you are updating the data_in and data_out pointers correctly for - each successive call. - -While doing the above, it is probably useful to compare what you are doing to -what is done in the example programs in the examples/ directory of the source -code tarball. - -If you have done all of the above and are still having problems then its -probably time to email the author with the smallest chunk of code that -adequately demonstrates your problem. This chunk should not need to be any more -than 100 lines of code. - -## Q6 : I'm use the SRC_SINC_* converters and up-sampling by a ratio of 2. I reset the converter and put in 1000 samples and I expect to get 2000 samples out, but I'm getting less than that. Why? {#Q006} - -The short answer is that there is a transport delay inside the converter itself. -Long answer follows. - -By way of example, the first time you call src_process() you might only get 1900 -samples out. However, after that first call all subsequent calls will probably -get you about 2000 samples out for every 1000 samples you put in. - -The main problems people have with this transport delay is that they need to -read out an exact number of samples and the transport delay scews this up. The -best way to overcome this problem is to always supply more samples on the input -than is actually needed to create the required number of output samples. With -reference to the example above, if you always supply 1500 samples at the input, -you will always get 2000 samples at the output. You will always need to keep -track of the number of input frames used on each call to src_process() and deal -with these values appropriately. - -## Q7 : I have input and output sample rates that are integer values, but the API wants me to divide one by the other and put the result in a floating point number. Won't this case problems for long running conversions? {#Q007} - -The short answer is no, the precision of the ratio is many orders of magnitude -more than is really needed. - -For the long answer, lets do come calculations. Firstly, the `src_ratio` field -is double precision floating point number which has [53 bits of precision](http://en.wikipedia.org/wiki/Double_precision). - -That means that the maximum error in your ratio converted to a double is one bit -in 2^53 which means the double float value would be wrong by one sample -after 9007199254740992 samples have passed or wrong by more than half a sample -wrong after half that many (4503599627370496 samples) have passed. - -Now if for example our output sample rate is 96kHz then - - 4503599627370496 samples at 96kHz is 46912496118 seconds - 46912496118 seconds is 781874935 minutes - 781874935 minutes is 13031248 hours - 13031248 hours is 542968 days - 542968 days is 1486 years - -So, after 1486 years, the input will be wrong by more than half of one sampling -period. - -All this assumes that the crystal oscillators uses to sample the audio stream is -perfect. This is not the case. According to [this web site](http://www.ieee-uffc.org/freqcontrol/quartz/vig/vigcomp.htm), -the accuracy of standard crystal oscillators (XO, TCXO, OCXO) is at best 1 in -100 million. The `src_ratio` is therefore 45035996 times more accurate than the -crystal clock source used to sample the original audio signal and any potential -problem with the `src_ratio` being a floating point number will be completely -swamped by sampling inaccuracies. diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/foo_dsp_src9.png b/lib/libsamplerate/libsamplerate-0.2.2/docs/foo_dsp_src9.png Binary files differdeleted file mode 100755 index 285c694c..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/foo_dsp_src9.png +++ /dev/null diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/history.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/history.md deleted file mode 100755 index 46686b9b..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/history.md +++ /dev/null @@ -1,38 +0,0 @@ ---- -layout: default ---- - -# History - -- Version 0.0.0 (Oct 06 2002) First alpha release (friends and family only). -- Version 0.0.2 (Oct 10 2002) Second alpha release (fools rush in ....). -- ... -- Version 0.0.7 (Nov 17 2002) The last mono only version. -- Version 0.0.9 (Nov 21 2002) Full multi channel support. First pass at - documentation. Still not ready for release. -- Version 0.0.11 (Nov 26 2002) Incorporated some suggestions from Conrad Parker. -- Version 0.0.12 (Nov 28 2002) First public release. -- Version 0.0.13 (Dec 03 2002) Fixes for MacOSX and Solaris. -- Version 0.0.14 (Jan 13 2003) Now compiles on Win32. Major code speedup. Minor - bug fixes. -- Version 0.0.15 (May 02 2003) Minor bug fixes. -- Version 0.1.0 (Mar 14 2004) Add callback API, functions for short to float and - float to short conversion. Minor bug fixes. -- Version 0.1.1 (Jul 17 2004) Callback API bug fix. Bugfix for aggressive - gcc-3.4 optimisations. -- Version 0.1.2 (Sep 12 2004) Callback API reset bug fix. -- Version 0.1.3 (Mar 23 2008) Huge quality improvements to two best SINC based - converters. -- Version 0.1.4 (Jul 02 2008) Fix segfault when using extremely low conversion - ratios. -- Version 0.1.5 (Jan 11 2009) Optimisation resulting in dramatic throughput - improvements ([See here.](http://www.mega-nerd.com/erikd/Blog/CodeHacking/SecretRabbitCode/rel_0_1_5.html)). -- Version 0.1.6 (Jan 27 2009) Minor bug fix in test suite (account for rounding - error on x86_64). -- Version 0.1.7 (Feb 14 2009) Fix a segfault bug. Fix compilation under MSVC. -- Version 0.1.8 (Aug 15 2011) Minor bug fixes and updates. -- Version 0.1.9 (Sep 19 2016) Fix for a segfault. Relicense under BSD license. -- Version 0.2.0 (Jan 21 2021) Cleaned up build system. -- Version 0.2.1 (Jan 23 2021) Fix libtool ABI versioning. -- Version 0.2.2 (Sep 05 2021) Fix ABI version incompatibility between Autotools - and CMake build on Apple platforms. Minor bug fixes and updates. diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/index.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/index.md deleted file mode 100755 index 4546d7bd..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/index.md +++ /dev/null @@ -1,160 +0,0 @@ ---- -layout: default -title: Libsamplerate home page ---- - -{: .indent_block} -*"Choosing a sample rate coverter wasn't easy. We ran numerous tests with Secret -Rabbit Code and other sample rate converters, then compared them all. In the -end, SRC outperformed the others, including some extremely well known and -popular software. We had one issue with SRC, but after emailing Erik, he got -back to us immediately with an answer. Choosing SRC was a no brainer."* -Ryan Smith, International Marketing Manager, -[Emersys Corp.](http://emersys.co.kr/), South Korea. -Product : [Maven3D](http://maven3d.com) - -**S**ecret **R**abbit **C**ode (aka libsamplerate) is a **S**ample **R**ate -**C**onverter for audio. One example of where such a thing would be useful is -converting audio from the CD sample rate of 44.1kHz to the 48kHz sample rate -used by DAT players. - -**SRC** is capable of arbitrary and time varying conversions, from downsampling -by a factor of 256 to upsampling by the same factor. Arbitrary in this case -means that the ratio of input and output sample rates can be an irrational -number. The conversion ratio can also vary with time for speeding up and slowing -down effects. - -**SRC** provides a small set of converters to allow quality to be traded off -against computation cost. The current best converter provides a signal-to-noise -ratio of 145dB with -3dB passband extending from DC to 96% of the theoretical -best bandwidth for a given pair of input and output sample rates. - -Since the library has few dependencies beyond that provided by the standard C -library, it should compile and work on just about any operating system. It is -known to work on Linux, MacOSX, [Win32](win32.md) and Solaris. With some -relatively minor hacking it should also be relatively easy to port it to -embedded systems and digital signal processors. - -In addition, the library comes with a comprehensive test suite which can -validate the performance of the library on new platforms. - -## Download - -Check latest version on -[GitHub Releases page](https://github.com/libsndfile/libsamplerate/releases). - -Binaries and source packages are signed by current release manager David Seifert aka -@SoapGentoo. You can verify signatures with his public GPG key (`0xA47620E801E47E95`): - -``` ------BEGIN PGP PUBLIC KEY BLOCK----- -Version: GnuPG v2 - -mQINBFppABgBEAC42ZiNvV7BTIgR6TQy0YnF54fx3mVRP1u8Mq00UZa7reAsNKh7 -1H60j0W4s6+4pVVIKGfpVGxLwUdJe+KVCYw1Cd3YW6uMf5zZrC/ZWqnJiH/n6S6o -1l4INII2o6YbGBnzIWBPRo7PlOL+mvgKTLpBSJPnhD8XDGN5wRiV8rL2+6Dptg0F -nJt7oxECGF3OD3gk6HMel0o82CVkIqMtNaX1L/bhcdF7K0Rp2MXPZMmpn1izW5sI -asN1G9+w+Zwj7kMJzq1Aw3ac+rsX4SEYdvXjS2QhDHQUIr6LXri3D2WbcEqIZj2R -JVoVwblsrG11dYXFDBbgrq4NhgTBsxHYDlkr/qF2W+kbPC/nhSqTVZeCYvTBZbOQ -+RqyN/I0izukglnWmV1jGijFA8snyP8efx732hw/24zRYmtXOtnEITUpw8WOeZCq -6uiHaQ+eopnY2ojBg9BI7WZm0AFn58xxT9soMsyFOUFgXTqaWFZWlJ3fhZE8/0v8 -JEu/kPGE5aJReT3b34B+Bojkj74XR+h2u7iJJBHMTE8RwGoUOZHer/XsL9xlcdks -I+7TCjiq++ShaSSt2XsJmw2BhREohrjW/2KkwmvT3b44RMpKPB4WTH+++aqJQNeM -IqmswOMoZvzEZezInj7WVY/r0WEei1Y6wt1tBrJ/cFf1oQBM1UmphxcrfQARAQAB -tB9EYXZpZCBTZWlmZXJ0IDxzb2FwQGdlbnRvby5vcmc+iQJUBBMBCgA+BQsJCAcD -BRUKCQgLBRYCAwEAAh4BAheAAhsBFiEEMdlcq22A0mIkShdQpHYg6AHkfpUFAl/V 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--- a/lib/libsamplerate/libsamplerate-0.2.2/docs/lists.md +++ /dev/null @@ -1,17 +0,0 @@ ---- -layout: default ---- - -# Mailing Lists - -There are currently two mailings lists for Secret Rabbit Code: - -- **src-announce\@mega-nerd.com** - [Subscribe](mailto:src-announce-request@mega-nerd.com?subject=subscribe) - - A read only announcement list. -- **src\@mega-nerd.com** - [Subscribe](mailto:src-request@mega-nerd.com?subject=subscribe) - - A general list which will also carry all the email from the announce - list. Posting to this list is restricted to subscribers. diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/paypal.png b/lib/libsamplerate/libsamplerate-0.2.2/docs/paypal.png Binary files differdeleted file mode 100755 index 2706d25b..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/paypal.png +++ /dev/null diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/quality.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/quality.md deleted file mode 100755 index 5c69578e..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/quality.md +++ /dev/null @@ -1,84 +0,0 @@ ---- -layout: default ---- - -# SRC Quality - -**This document not yet complete.** - -When measuring the performance of a Sample Rate Converter, there are three -factors to consider: - -- **Signal-to-Noise Ratio** - a measure of how much noise the sample rate - conversion process adds to the signal. This is measured in decibels (dB) and - the higher this value the better. For most sample rate converters, the SNR - will vary depending on the input signal and the ratio between input and output - sample rates. The only valid comparison of SNR is between the worst case for - each converter. -- **Bandwidth** - most sample rate converters attenuate high frequencies as part - of their operation. Bandwidth can be measured by finding the frequency where - the attenuation is 3dB and expressing that as a percentage of the full - bandwidth at that sampling rate. -- **Speed** - the faster the better **:-)**. - -There are a number of sample rate converters available for downloading but I -will limit the comparison of Secret Rabbit Code to the following: - -- [sndfile-resample](http://libsndfile.github.io/libsamplerate/download.html) - which is a program (which uses libsamplerate) from the **examples/** directory - of the Secret Rabbit Code source code distribution. -- [Resample](https://ccrma.stanford.edu/~jos/resample/) by Julius O Smiths which - seems to have been the first high quality converter available as source code. -- [ResampAudio](http://www.tsp.ece.mcgill.ca/MMSP/Documents/Software/AFsp/ResampAudio.html) - which is part of [Audio File Programs and Routines](http://www.tsp.ece.mcgill.ca/MMSP/Documents/Software/AFsp/AFsp.html) - by Peter Kabal. -- [SoX](http://home.sprynet.com/~cbagwell/sox.html) which is maintained by Chris - Bagwell. SoX is also able to perform some low quality sample rate conversions - but these will not be investigated. -- [Shibatch](http://shibatch.sourceforge.net/) which seems to be a frequency - domain sample rate converter. Unfortunately, this converter does not handle - arbitrary conversion ratios and hence could not be properly compared to the - other converters. -- [sr-convert](http://sr-convert.sourceforge.net/) is another converter which - does not handle arbitrary conversion ratios. - -It should be noted that the first three converters above are based on the -algorithm by [Julius O. Smith](http://www-ccrma.stanford.edu/~jos/resample/) -which emulates the conversion of the digital signal to an analogue one and then -samples the analogue signal at the new sample rate. - -## Methodology - -Measuring the SNR of a converter is relatively straightforward. Generate an -input signal consisting of a windowed sine wave, sample rate convert it and -measure the signal-to-noise ratio of the output signal. A typical length for the -original file is 30000 samples. - -The bandwidth of a sample rate converter is a little more difficult to measure. -Currently this is done by generating two short files containing a windowed sine -wave. The frequencies of the sine waves are 0.35 and 0.495 of the sample rate. -These files are then upsampled by a factor of 2 using the converter under test. -If the attenuation of the lower frequency is less than 3dB and higher frequency -is more than 3dB, it is then possible to iteratively increase the lower -frequency and decrease the upper frequency keeping the -3dB point bracketed. -When the distance between the upper and lower frequency is sufficiently small, -it is possible to obtain a very accurate estimate of the -3dB frequency. - -The speed of a sample rate converter is easy to measure; simply perform a -conversion on a large file or a number of smaller files and time the conversion -process. - -The above measurement techniques are built into a test program which is -delivered with the Secret Rabbit Code source code distribution. This program is -able to test the first four of the above converters. - -## SoX - -SoX provides three methods of resampling; a linear interpolator, a polyphase -resampler and the Julius O. Smith simulated analogue filter method. - -## Shibatch - -Shibach - -**More Coming Soon.** diff --git a/lib/libsamplerate/libsamplerate-0.2.2/docs/win32.md b/lib/libsamplerate/libsamplerate-0.2.2/docs/win32.md deleted file mode 100755 index 6f93347d..00000000 --- a/lib/libsamplerate/libsamplerate-0.2.2/docs/win32.md +++ /dev/null @@ -1,15 +0,0 @@ ---- -layout: default ---- - -# Compiling on Win32 - -You can use [CMake](https://cmake.org/) to generate Visual Studio project. The -configuration process is described [here](https://cmake.org/runningcmake/). - -The libsamplerate library itself does not require any dependencies, but if you -want to build examples and tests, you will need the [libsndfile](https://github.com/libsndfile/libsndfile) -and [FFTW](http://www.fftw.org/) libraries. The easiest way to install them is -to use a package manager, such as [Vcpkg](https://github.com/microsoft/vcpkg). Their README.md contains detailed -installation instructions for supported platforms. Libsamplerate requires the -`libsndfile` and `fftw3` packages. |
