History

Update site documentation

* Use GitHub's Jekyll static site generator. * Move all documentation to docs/ directory. * Convert HTML to Markdown.

evpobr committed 6 years ago

Commit 260c3f61 parent 01284e738 files · +1057 −2227Browse files

Showing the changes to docs (20 of 38 files). Show all

Unified · Split

20 files
  1. docs/SRC.css +48 −0
  2. docs/SRC.png +0 −0
  3. docs/_config.yml +8 −0
  4. docs/_layouts/default.html +51 −0
  5. docs/api.md +46 −0
  6. docs/api_callback.md +88 −0
  7. docs/api_full.md +146 −0
  8. docs/api_misc.md +143 −0
  9. docs/api_simple.md +71 −0
  10. docs/bugs.md +37 −0
  11. docs/download.md +15 −0
  12. docs/faq.md +193 −0
  13. docs/foo_dsp_src9.png +0 −0
  14. docs/history.md +34 −0
  15. docs/index.md +39 −0
  16. docs/license.md +12 −0
  17. docs/lists.md +17 −0
  18. docs/paypal.png +0 −0
  19. docs/quality.md +84 −0
  20. docs/win32.md +15 −0
docs/SRC.css+48−0
@@ -0,0 +1,48 @@
1+a:link {
2+ color: #FB1465;
3+}
4+
5+a:active, a:visited {
6+ color: #FB1465;
7+}
8+
9+body {
10+ background: black;
11+ color: white;
12+ font-family: arial, helvetica, sans-serif;
13+}
14+
15+h1, h2, h3, h4, h5, h6 {
16+ color: #FB1465;
17+}
18+
19+code, kbd, samp, var {
20+ font-family: courier, monospace;
21+ font-size: 1em;
22+}
23+
24+main {
25+ padding-left: 3%;
26+ padding-right: 3%;
27+}
28+
29+pre {
30+ font-family: courier, monospace;
31+ font-size: 1em;
32+}
33+
34+.image-logo {
35+ display: block;
36+ margin-left: auto;
37+ margin-right: auto;
38+ margin-bottom: 1em;
39+}
40+
41+.indent_block {
42+ margin-left: 10%;
43+ margin-right: 10%;
44+}
45+
46+.nf {
47+ font-weight: bold;
48+}
docs/SRC.png+0−0

Binary file not shown.

docs/_config.yml+8−0
@@ -0,0 +1,8 @@
1+---
2+author: Erik de Castro Lopo, erikd@mega-nerd.com
3+description: An audio Sample Rate Conversion library
4+keywords: libsamplerate, sound, resample, audio, dsp, Linux
5+version: 0.1.8
6+
7+markdown: kramdown
8+---
docs/_layouts/default.html+51−0
@@ -0,0 +1,51 @@
1+<!DOCTYPE html">
2+<html lang="en">
3+
4+<head>
5+ <meta charset="utf-8">
6+ <meta name="viewport" content="width=device-width, initial-scale=1.0">
7+ <meta name="description" content="{{ site.description | page.description }}">
8+ <meta name="keywords" content="{{ site.keywords | page.keywords }}">
9+ <title>{{ page.title }}</title>
10+ <link rel="stylesheet" href="SRC.css" type="text/css" media="all">
11+</HEAD>
12+
13+<body>
14+ <footer>
15+ <img src="SRC.png" class="image-logo" height=100 width=760 alt="SRC.png">
16+ </footer>
17+ <table align="center" width="98%">
18+ <tr>
19+ <td valign="top">
20+ <nav>
21+ <a href="index.html">Home</a><br>
22+ <a href="license.html">License</a><br>
23+ <a href="history.html">History</a><br>
24+ <a href="download.html">Download</a><br>
25+ <a href="quality.html">Quality</a><br>
26+ <a href="api.html">API</a><br>
27+ <a href="bugs.html">Bug Reporting</a><br>
28+ <a href="win32.html">On Win32</a><br>
29+ <a href="faq.html">FAQ</a><br>
30+ <a href="lists.html">Mailing Lists</a><br>
31+ <a href="https://github.com/libsndfile/libsamplerate/blob/master/ChangeLog">ChangeLog</a><br>
32+ <nav>
33+ <div class="block">
34+ <p>Author:<br>&nbsp;Erik&nbsp;de&nbsp;Castro&nbsp;Lopo</p>
35+ </div>
36+ </td>
37+ <td valign="top">
38+ <main>
39+ <article>
40+
41+ {{ content }}
42+
43+ </article>
44+ </main>
45+ </td>
46+ </tr>
47+ </table>
48+
49+</body>
50+
51+</html>
docs/api.md+46−0
@@ -0,0 +1,46 @@
1+---
2+layout: default
3+---
4+
5+# Applications Programming Interface
6+
7+The publically callable functions of libsamplerate are all listed in the
8+**\<samplerate.h\>** header file. In order to use any of the functionality of
9+libsamplerate, you need to add
10+
11+```c
12+#include <samplerate.h>
13+```
14+
15+to the top of any function that call any of the following functions. You will
16+also need to link you binary with the libsamplerate library.
17+
18+The API allows three methods for accessing the capabilies of the library:
19+
20+- A [simple interface](api_simple.md) which can sample rate convert a single
21+ block of samples (one or more channels) in one go. The simple API is less
22+ capable than the full API.
23+- A [more fully featured interface](api_full.md) which allows time varying
24+ sample rate conversion on streaming data (again one or more channels).
25+- A [callback interface](api_callback.md) which has the same functionality as
26+ the interface above but allows the details of input and output to be
27+ separated. The output is generated by call a read function and the library
28+ calls a user supplied callback function to obtain its input. This interface is
29+ particularly well suited to applications where the output sample rate is
30+ varied with time.
31+
32+**NB :** All three access methods are able to process multi channel interleaved
33+data.
34+
35+The parts of the API which are common to all three interfaces are:
36+
37+- The [error reporting](api_misc.md#error-reporting) mechanisim.
38+- The available [converter](api_misc.md#converters) types.
39+- The [SRC_DATA](api_misc.md#src_data) struct.
40+
41+All three versions of the API are restricted to operating on buffers of ISO C
42+Standard **float** data. However, there are two [auxillary functions](api_misc.md#auxillary-functions)
43+for converting arrays of float data to and from short data.
44+
45+**Note:** The **tests/** and **examples/** directories of the source code
46+distribution contain numerous example programs showing the use of the library.
docs/api_callback.md+88−0
@@ -0,0 +1,88 @@
1+---
2+layout: default
3+---
4+
5+# Callback API
6+
7+The callback API consists of the following functions :
8+
9+```c
10+typedef long (*src_callback_t) (void *cb_data, float **data) ;
11+
12+SRC_STATE* src_callback_new (src_callback_t func,
13+ int converter_type, int channels,
14+ int *error, void* cb_data) ;
15+
16+SRC_STATE* src_delete (SRC_STATE *state) ;
17+
18+long src_callback_read (SRC_STATE *state, double src_ratio,
19+ long frames, float *data) ;
20+
21+int src_reset (SRC_STATE *state) ;
22+int src_set_ratio (SRC_STATE *state, double new_ratio) ;
23+```
24+
25+Like the [simple API](api_simple.md) and the [full API](api_full.md), the
26+callback based API is able to operate on interleaved multi channel data.
27+
28+An example of the use of the callback based API can be found in the
29+**varispeed-play** program in the **examples/** directory of the source code
30+tarball.
31+
32+## Initialisation
33+
34+```c
35+SRC_STATE* src_callback_new (src_callback_t func,
36+ int converter_type, int channels,
37+ int *error, void* cb_data) ;
38+```
39+
40+The **src_callback_new** function returns an anonymous pointer to a sample rate
41+converter callback object, src_state. This is the same type of object as that
42+returned by [src\_new](api_full.md#initialisation), but with different internal
43+state. Although these are the same object type, they cannot be used
44+interchangeably. If an error occurs the function returns a NULL pointer and
45+fills in the error value pointed to by the **error** pointer supplied by the
46+caller.
47+
48+The caller then passes the SRC_STATE object to the **src_callback_read**
49+function to pull data out of the converter. When the caller is finished with the
50+converter they should pass it to the clean up function [src_delete](api_full.md#cleanup).
51+
52+The **func** parameter is a user supplied function which must match the
53+**src_callback_t** type signature while **cb_data** is a pointer to data which
54+be passed as the first parameter to the user supplied callback function. This
55+function is called by the converter whenever it needs input data as a result of
56+being calls to **src_callback_read**.
57+
58+If the converter was initialised to work with more than one channel, the
59+callback function must work with mutiple channels of interleaved data. The
60+callback function should return the number of frames it supplying to the
61+converter. For multi channel files, this return value should be the number of
62+floats divided by the number of channels.
63+
64+The converter must be one of the supplied converter types documented [here](api_misc.md#converters).
65+
66+The caller then passes the SRC_STATE pointer to the **src_callback_read**
67+function to pull data out of the converter.
68+
69+## Callback Read
70+
71+```c
72+long src_callback_read (SRC_STATE *state, double src_ratio,
73+ long frames, float *data) ;
74+```
75+
76+The **src_callback_read** function is passed the [**SRC_STATE**](api_misc.md#src_data)
77+pointer returned by **src_callback_new**, the coversion ratio
78+(output_sample_rate / input_sample_rate), the maximum number of output frames to
79+generate and a pointer to a buffer in which to place the output data. For multi
80+channel files, the data int the output buffer is stored in interleaved format.
81+
82+The **src_callback_read** function returns the number of frames generated or
83+zero if an error occurs or it runs out of input (ie the user supplied callback
84+function returns zero and there is no more data buffered internally). If an
85+error has occurred, the function [src_error](api_misc.md#error-reporting)
86+will return non-zero.
87+
88+See also : [**src_set_ratio**](api_full.md#set-ratio)
docs/api_full.md+146−0
@@ -0,0 +1,146 @@
1+---
2+layout: default
3+---
4+
5+# Full API
6+
7+The full API consists of the following functions :
8+
9+```c
10+SRC_STATE* src_new (int converter_type, int channels, int *error) ;
11+SRC_STATE* src_delete (SRC_STATE *state) ;
12+
13+int src_process (SRC_STATE *state, SRC_DATA *data) ;
14+int src_reset (SRC_STATE *state) ;
15+int src_set_ratio (SRC_STATE *state, double new_ratio) ;
16+```
17+
18+## Initialisation
19+
20+```c
21+SRC_STATE* src_new (int converter_type, int channels, int *error) ;
22+```
23+
24+The **src\_new** function returns an anonymous pointer to a sample rate
25+converter object, src\_state. If an error occurs the function returns a
26+NULL pointer and fills in the error value pointed to by the **error**
27+pointer supplied by the caller. The converter must be one of the
28+supplied converter types documented [here](api_misc.md#converters).
29+
30+## Cleanup
31+
32+```c
33+SRC_STATE* src_delete (SRC_STATE *state) ;
34+```
35+
36+The **src\_delete** function frees up all memory allocated for the given
37+sample rate converter object and returns a NULL pointer. The caller is
38+responsible for freeing any memory passed to the sample rate converter
39+via the pointer to the **SRC\_DATA** struct.
40+
41+## Process
42+
43+```c
44+int src_process (SRC_STATE *state, SRC_DATA *data) ;
45+```
46+
47+The **src\_process** function processes the data provided by the caller
48+in an **SRC\_DATA** struct using the sample rate converter object
49+specified by the **SRC\_STATE** pointer. When operating on streaming
50+data, this function can be called over and over again, with each new
51+call providing new input data and returning new output data.
52+
53+The **SRC\_DATA** struct passed as the second parameter to the
54+**src\_process** function has the following fields:
55+
56+```c
57+typedef struct
58+{ const float *data_in;
59+ float *data_out;
60+
61+ long input_frames, output_frames ;
62+ long input_frames_used, output_frames_gen ;
63+
64+ int end_of_input ;
65+
66+ double src_ratio ;
67+} SRC_DATA ;
68+```
69+
70+The fields of this struct which must be filled in by the caller are:
71+
72+data_in
73+: A pointer to the input data samples.
74+
75+input_frames
76+: The number of frames of data pointed to by data_in.
77+
78+data_out
79+: A pointer to the output data samples.
80+
81+output_frames
82+: Maximum number of frames pointer to by data_out.
83+
84+src_ratio
85+: Equal to output_sample_rate / input_sample_rate.
86+
87+end_of_input
88+: Equal to 0 if more input data is available and 1 otherwise.
89+
90+Note that the data\_in and data\_out arrays may not overlap. If they do,
91+the library will return an error code.
92+
93+When the **src\_process** function returns **output\_frames\_gen** will
94+be set to the number of output frames generated and
95+**input\_frames\_used** will be set to the number of input frames
96+consumed to generate the provided number of output frames.
97+
98+The **src\_process** function returns non-zero if an error occurs. The
99+non-zero error return value can be decoded into a text string using the
100+function documented [here](api_misc.md#error-reporting).
101+
102+## Reset
103+
104+```c
105+int src_reset (SRC_STATE *state) ;
106+```
107+
108+The **src\_reset** function resets the internal state of the sample rate
109+converter object to the same state it had immediately after its creation
110+using **src\_new**. This should be called whenever a sample rate
111+converter is to be used on two separate, unrelated pieces of audio.
112+
113+## Clone
114+
115+```c
116+SRC_STATE* src_clone (SRC_STATE *state, int *error) ;
117+```
118+
119+The **src_clone** function creates a copy of the internal state of the sample
120+rate converter object. The output of the next call to **src\_process** will be
121+identical for both the original and cloned state (given the same **SRC_DATA**
122+input). This could be used to later resume sample rate conversion at a specific
123+location in a stream with the same state, which may be useful in real-time
124+applications.
125+
126+If an error occurs the function returns a NULL pointer and fills in the
127+error value pointed to by the **error** pointer supplied by the caller.
128+
129+## Set Ratio
130+
131+```c
132+int src_set_ratio (SRC_STATE *state, double new_ratio) ;
133+```
134+
135+When using the **src_process** or **src_callback_process** APIs and updating the
136+**src_ratio** field of the **SRC_STATE** struct, the library will try to
137+smoothly transition between the conversion ratio of the last call and the
138+conversion ratio of the current call.
139+
140+If the user want to bypass this smooth transition and achieve a step response in
141+the conversion ratio, the **src_set_ratio** function can be used to set the
142+starting conversion ratio of the next call to **src_process** or
143+**src_callback_process**.
144+
145+This function returns non-zero on error and the error return value can be
146+decoded into a text string using the function documented [here](api_misc.md#error-reporting).
docs/api_misc.md+143−0
@@ -0,0 +1,143 @@
1+---
2+layout: default
3+---
4+
5+# Miscellaneous API Documentation
6+
7+## Error Reporting
8+
9+Most of the API functions either return an integer error (ie **src_simple** and
10+**src_process**) or return an integer error value via an int pointer parameter
11+(**src_new**). These integer error values can be converted into a human readable
12+text strings by calling the function:
13+
14+```c
15+const char* src_strerror (int error) ;
16+```
17+
18+which will return an error string for valid error numbers, the string
19+\"No Error\" for an error value of zero or a NULL pointer if no error message
20+has been defined for that error value.
21+
22+## Converters
23+
24+Secret Rabbit Code has a number of different converters which can be selected
25+using the **converter_type** parameter when calling **src_simple** or
26+**src_new**. Currently, the five converters available are:
27+
28+```c
29+enum
30+{
31+ SRC_SINC_BEST_QUALITY = 0,
32+ SRC_SINC_MEDIUM_QUALITY = 1,
33+ SRC_SINC_FASTEST = 2,
34+ SRC_ZERO_ORDER_HOLD = 3,
35+ SRC_LINEAR = 4
36+} ;
37+```
38+
39+As new converters are added, they will given a number corresponding to the next
40+integer.
41+
42+The details of these converters are as follows:
43+
44+- **SRC_SINC_BEST_QUALITY** - This is a bandlimited interpolator derived from
45+ the mathematical **sinc** function and this is the highest quality sinc based
46+ converter, providing a worst case Signal-to-Noise Ratio (SNR) of 97 decibels
47+ (dB) at a bandwidth of 97%. All three SRC_SINC_* converters are based on the
48+ techniques of [Julius O. Smith](http://ccrma.stanford.edu/~jos/resample/)
49+ although this code was developed independantly.
50+- **SRC_SINC_MEDIUM_QUALITY** - This is another bandlimited interpolator much
51+ like the previous one. It has an SNR of 97dB and a bandwidth of 90%. The speed
52+ of the conversion is much faster than the previous one.
53+- **SRC_SINC_FASTEST** - This is the fastest bandlimited interpolator and has an
54+ SNR of 97dB and a bandwidth of 80%.
55+- **SRC_ZERO_ORDER_HOLD** - A Zero Order Hold converter (interpolated value is
56+ equal to the last value). The quality is poor but the conversion speed is
57+ blindlingly fast.
58+- **SRC_LINEAR** - A linear converter. Again the quality is poor, but the
59+ conversion speed is blindingly fast.
60+
61+There are two functions that give either a (text string) name or description for
62+each converter:
63+
64+```c
65+const char *src_get_name (int converter_type) ;
66+const char *src_get_description (int converter_type) ;
67+```
68+
69+The name will typically be a short string for use in a dialog box, while the
70+description string is longer.
71+
72+Both of these functions return a NULL pointer if there is no converter for the
73+given **converter_type** value. Since the converters have consecutive
74+**converter_type** values, the caller is easily able to figure out the number of
75+converters at run time. This enables a binary dynamically linked against an old
76+version of the library to know about converters from later versions of the
77+library as they become available.
78+
79+## SRC_DATA
80+
81+Both the simple and the full featured versions of the API use the **SRC_DATA**
82+struct to pass audio and control data into the sample rate converter. This
83+struct is defined as:
84+
85+```c
86+typedef struct
87+{ const float *data_in;
88+ float *data_out ;
89+
90+ long input_frames, output_frames ;
91+ long input_frames_used, output_frames_gen ;
92+
93+ int end_of_input ;
94+
95+ double src_ratio ;
96+} SRC_DATA ;
97+```
98+
99+The **data_in** pointer is used to pass audio data into the converter while the
100+**data_out** pointer supplies the converter with an array to hold the
101+converter's output. For a converter which has been configured for mulitchannel
102+operation, these pointers need to point to a single array of interleaved data.
103+
104+The **input_frames** and **output_frames** fields supply the converter with the
105+lengths of the arrays (in frames) pointed to by the **data_in** and **data_out**
106+pointers respectively. For monophinc data, these values would indicate the
107+length of the arrays while for multi channel data these values would be equal to
108+the the length of the array divided by the number of channels.
109+
110+The **end_of_input** field is only used when the sample rate converter is used
111+by calling the **src_process** function. In this case it should be set to zero
112+if more buffers are to be passed to the converter and 1 if the current buffer is
113+the last.
114+
115+Finally, the **src_ratio** field specifies the conversion ratio defined as the
116+output sample rate divided by the input sample rate. For a connected set of
117+buffers, this value can be varies on each call to **src_process** resulting in a
118+time varying sample rate conversion process. For time varying sample rate
119+conversions, the ratio will be linearly interpolated between the **src_ratio**
120+value of the previous call to **src_process** and the value for the current
121+call.
122+
123+The **input_frames_used** and **output_frames_gen** fields are set by the
124+converter to inform the caller of the number of frames consumed from the
125+**data_in** array and the number of frames generated in the **data_out** array
126+respectively. These values are for the current call to **src_process** only.
127+
128+## Auxillary Functions
129+
130+There are four auxillary functions for converting arrays of float data to and
131+from short or int data. These functions are defined as:
132+
133+```c
134+void src_short_to_float_array (const short *in, float *out, int len) ;
135+void src_float_to_short_array (const float *in, short *out, int len) ;
136+void src_int_to_float_array (const int *in, float *out, int len) ;
137+void src_float_to_int_array (const float *in, int *out, int len) ;
138+```
139+
140+The float data is assumed to be in the range [-1.0, 1.0] and it is automatically
141+scaled on the conversion to and from float. On the float to short/int conversion
142+path, any data values which would overflow the range of short/int data are
143+clipped.
docs/api_simple.md+71−0
@@ -0,0 +1,71 @@
1+---
2+layout: default
3+---
4+
5+# Simple API
6+
7+**Important Note:** The simple API is not designed to work on small chunks of a
8+larger piece of audio. If you attempt to use it this way you are doing it wrong
9+and will not get the results you want. For processing audio data in chunks you
10+**must** use the [full api](api_full.md) or the [callback based api](api_callback.md).
11+
12+The simple API consists of a single function:
13+
14+```c
15+int src_simple (SRC_DATA *data, int converter_type, int channels) ;
16+```
17+
18+The use of this function rather than the more fully featured API requires the
19+caller to know the total length of the input data before hand and that all input
20+and output data can be held in the system's memory at once. It also assumes that
21+there is a single constant ratio between input and output sample rates.
22+
23+Dealing with the easy stuff first, the **converter_type** parameter should be
24+one of the values defined in **samplerate.h** and documented [here](api_misc.md#converters)
25+while the **channels** parameter specifies the number of interleaved channels
26+that the sample rate converter is being asked to process (number of input
27+channels and output channels is always equal). There is no hard upper limit on
28+the number of channels; it is limited purely by the amount of memory available.
29+
30+The first parameter to **src_simple** is a pointer to an **SRC_DATA** struct
31+(more info [here](api_misc.md#src_data)) defined as follows:
32+
33+```c
34+typedef struct
35+{ const float *data_in;
36+ float *data_out ;
37+
38+ long input_frames, output_frames ;
39+ long input_frames_used, output_frames_gen ;
40+
41+ int end_of_input ;
42+
43+ double src_ratio ;
44+} SRC_DATA ;
45+```
46+
47+The fields of this struct which must be filled in by the caller are:
48+
49+data_in
50+: A pointer to the input data samples.
51+
52+input_frames
53+: The number of frames of data pointed to by data_in.
54+
55+data_out
56+: A pointer to the output data samples.
57+
58+output_frames
59+: Maximum number of frames pointer to by data_out.
60+
61+src_ratio
62+: Equal to output_sample_rate / input_sample_rate.
63+
64+When the **src_simple** function returns **output_frames_gen** will be set to
65+the number of output frames generated and **input_frames_used** will be set to
66+the number of input frames used to generate the provided number of output
67+frames.
68+
69+The **src_simple** function returns a non-zero value when an error occurs. See
70+[here](api_misc.md#error-reporting) for how to convert the error value into a
71+text string.
docs/bugs.md+37−0
@@ -0,0 +1,37 @@
1+---
2+layout: default
3+---
4+
5+# Bug Reporting
6+
7+If you are a user and have a problem using libsamplerate with another piece of
8+software, you should contact the author of that other software and get them to
9+explore their use of this library and possibly submit a bug report. If you are a
10+coder and think you have found a bug in libsamplerate then read on.
11+
12+Secret Rabbit Code is an extremely complex piece of code but I do think that it
13+is relatively bug free. In addition, the source code distribution includes a
14+comprehensive test suite for regression testing. This means it is extremely
15+unlikely that new bugs creep in when modifications are made to the code.
16+
17+SRC is also not the most simple library to use which means that I do get a
18+number of bug reports which turn out to be bugs in the user's program rather
19+than bugs in SRC. Up until now, I have investigated each bug report as
20+thoroughly as possible. Unfortunately, this chews up a lot of my time which
21+could otherwise be spent improving SRC, working on other Free Software or
22+spending time with my family.
23+
24+I have therefore decided, that I cannot investigate any bug report unless the
25+person reporting the problem can supply me with a short self contained test
26+program or a modification to one of the existing test programs in the tests/
27+directory of the source code distribution. The test program should meet the
28+following criteria:
29+
30+- Written in C or C++.
31+- Does not use any libraries or header files other than the ones which are
32+ standard for the relevant languages. (Of course libsamplerate can be used :-)).
33+- It is the minimal program which can adequately display the problem.
34+- It clearly displays the criteria for pass or fail.
35+
36+Supplying a good test program will maximize the speed with which your bug report
37+gets dealt with.
docs/download.md+15−0
@@ -0,0 +1,15 @@
1+---
2+layout: default
3+---
4+
5+# Download
6+
7+The latest version of Secret Rabbit Code is {{ site.version }}.
8+
9+Download it on [GitHub releases page](https://github.com/libsndfile/libsamplerate/releases/latest).
10+
11+Compiling libsamplerate is relatively easy. The INSTALL file in the top level
12+directory gives instructions on compiling and installing libsamplerate on
13+Unix-like systems (including MacOSX). For Win32 there are instructions in the
14+docs/ directory of the tarball. These instructions are mirrored
15+[here](win32.md).
docs/faq.md+193−0
@@ -0,0 +1,193 @@
1+---
2+layout: default
3+---
4+
5+# Frequently Asked Questions
6+
7+1. [Q1 : Is it normal for the output of libsamplerate to be louder than its input?](#Q001)
8+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)
9+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)
10+4. [Q4 : If I ran src\_simple (libsamplerate) on small chunks (160 frames) would that sound bad?](#Q004)
11+5. [Q5 : I\'m using libsamplerate but the high quality settings sound worse than the SRC\_LINEAR converter. Why?](#Q005)
12+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)
13+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)
14+
15+## Q1 : Is it normal for the output of libsamplerate to be louder than its input? {#Q001}
16+
17+The output of libsamplerate will be roughly the same volume as the input.
18+However, even if the input is strictly in the range (-1.0, 1.0), it is still
19+possible for the output to contain peak values outside this range.
20+
21+Consider four consecutive samples of [0.5 0.999 0.999 0.5]. If we are up
22+sampling by a factor of two we need to insert samples between each of the
23+existing samples. Its pretty obvious then, that the sample between the two 0.999
24+values should and will be bigger than 0.999.
25+
26+This means that anyone using libsamplerate should normalize its output before
27+doing things like saving the audio to a 16 bit WAV file.
28+
29+## 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}
30+
31+libsamplerate uses the pkg-config (man pkg-config) method of registering itself
32+with the host system. The best way of detecting its presence is using something
33+like this in configure.ac (or configure.in):
34+
35+ PKG_CHECK_MODULES(SAMPLERATE, samplerate >= 0.1.3,
36+ ac_cv_samplerate=1, ac_cv_samplerate=0)
37+
38+ AC_DEFINE_UNQUOTED([HAVE_SAMPLERATE],${ac_cv_samplerate},
39+ [Set to 1 if you have libsamplerate.])
40+
41+ AC_SUBST(SAMPLERATE_CFLAGS)
42+ AC_SUBST(SAMPLERATE_LIBS)
43+
44+This will automatically set the **SAMPLERATE_CFLAGS** and **SAMPLERATE_LIBS**
45+variables which can be used in Makefile.am or Makefile.in like this:
46+
47+ SAMPLERATE_CFLAGS = @SAMPLERATE_CFLAGS@
48+ SAMPLERATE_LIBS = @SAMPLERATE_LIBS@
49+
50+If you install libsamplerate from source, you will probably need to set the
51+**PKG_CONFIG_PATH** environment variable's suggested at the end of the
52+libsamplerate configure process. For instance on my system I get this:
53+
54+ -=-=-=-=-=-=-=-=-=-= Configuration Complete =-=-=-=-=-=-=-=-=-=-=-
55+
56+ Configuration summary :
57+
58+ Version : ..................... 0.1.3
59+ Enable debugging : ............ no
60+
61+ Tools :
62+
63+ Compiler is GCC : ............. yes
64+ GCC major version : ........... 3
65+
66+ Extra tools required for testing and examples :
67+
68+ Have FFTW : ................... yes
69+ Have libsndfile : ............. yes
70+ Have libefence : .............. no
71+
72+ Installation directories :
73+
74+ Library directory : ........... /usr/local/lib
75+ Program directory : ........... /usr/local/bin
76+ Pkgconfig directory : ......... /usr/local/lib/pkgconfig
77+
78+## 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}
79+
80+The short answer is that for the general case, no, you don't. The long answer is
81+that for some signals, with some converters, you will get very, very close.
82+
83+In order to resample correctly (ie using the **SRC_SINC_*** converters),
84+filtering needs to be applied, regardless of whether its upsampling or
85+downsampling. This filter needs to attenuate all frequencies above 0.5 times the
86+minimum of the source and destination sample rate (call this fshmin). Since the
87+filter needed to achieve full attenuation at this point, it has to start rolling
88+off a some frequency below this point. It is this rolloff of the very highest
89+frequencies which causes some of the loss.
90+
91+The other factor is that the filter itself can introduce transient artifacts
92+which causes the output to be different to the input.
93+
94+## Q4 : If I ran src_simple on small chunks (say 160 frames) would that sound bad? {#Q004}
95+
96+Well if you are after odd sound effects, it might sound OK. If you are after
97+high quality sample rate conversion you will be disappointed.
98+
99+The src_simple() was designed to provide a simple to use interface for people
100+who wanted to do sample rate conversion on say, a whole file all at once.
101+
102+## Q5 : I'm using libsamplerate but the high quality settings sound worse than the SRC_LINEAR converter. Why? {#Q005}
103+
104+There are two possible problems. Firstly, if you are using the src_simple()
105+function on successive blocks of a stream of samples, you will get bad results.
106+The src_simple() function is designed for use on a whole sound file, all at
107+once, not on contiguous segments of the same sound file. To fix the problem, you
108+need to move to the src_process() API or the callback based API.
109+
110+If you are already using the src_process() API or the callback based API and the
111+high quality settings sound worse than SRC_LINEAR, then you have other problems.
112+Read on for more debugging hints.
113+
114+All of the higher quality converters need to keep state while doing conversions
115+on segments of a large chunk of audio. This state information is kept inside the
116+private data pointed to by the SRC_STATE pointer returned by the src_new()
117+function. This means, that when you want to start doing sample rate conversion
118+on a stream of data, you should call src_new() to get a new SRC_STATE pointer
119+(or alternatively, call src_reset() on an existing SRC_STATE pointer). You
120+should then pass this SRC_STATE pointer to the src_process() function with each
121+new block of audio data. When you have completed the conversion, you can then
122+call src_delete() on the SRC_STATE pointer.
123+
124+If you are doing all of the above correctly, you need to examine your usage of
125+the values passed to src\_process() in the [SRC_DATA](api_misc.md#src_data)
126+struct. Specifically:
127+
128+- Check that input_frames and output_frames fields are being set in terms of
129+ frames (number of sample values times channels) instead of just the number of
130+ samples.
131+- Check that you are using the return values input_frames_used and
132+ output_frames_gen to update your source and destination pointers correctly.
133+- Check that you are updating the data_in and data_out pointers correctly for
134+ each successive call.
135+
136+While doing the above, it is probably useful to compare what you are doing to
137+what is done in the example programs in the examples/ directory of the source
138+code tarball.
139+
140+If you have done all of the above and are still having problems then its
141+probably time to email the author with the smallest chunk of code that
142+adequately demonstrates your problem. This chunk should not need to be any more
143+than 100 lines of code.
144+
145+## 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}
146+
147+The short answer is that there is a transport delay inside the converter itself.
148+Long answer follows.
149+
150+By way of example, the first time you call src_process() you might only get 1900
151+samples out. However, after that first call all subsequent calls will probably
152+get you about 2000 samples out for every 1000 samples you put in.
153+
154+The main problems people have with this transport delay is that they need to
155+read out an exact number of samples and the transport delay scews this up. The
156+best way to overcome this problem is to always supply more samples on the input
157+than is actually needed to create the required number of output samples. With
158+reference to the example above, if you always supply 1500 samples at the input,
159+you will always get 2000 samples at the output. You will always need to keep
160+track of the number of input frames used on each call to src_process() and deal
161+with these values appropriately.
162+
163+## 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}
164+
165+The short answer is no, the precision of the ratio is many orders of magnitude
166+more than is really needed.
167+
168+For the long answer, lets do come calculations. Firstly, the `src_ratio` field
169+is double precision floating point number which has [53 bits of precision](http://en.wikipedia.org/wiki/Double_precision).
170+
171+That means that the maximum error in your ratio converted to a double is one bit
172+in 2^53 which means the the double float value would be wrong by one sample
173+after 9007199254740992 samples have passed or wrong by more than half a sample
174+wrong after half that many (4503599627370496 samples) have passed.
175+
176+Now if for example our output sample rate is 96kHz then
177+
178+ 4503599627370496 samples at 96kHz is 46912496118 seconds
179+ 46912496118 seconds is 781874935 minutes
180+ 781874935 minutes is 13031248 hours
181+ 13031248 hours is 542968 days
182+ 542968 days is 1486 years
183+
184+So, after 1486 years, the input will be wrong by more than half of one sampling
185+period.
186+
187+All this assumes that the crystal oscillators uses to sample the audio stream is
188+perfect. This is not the case. According to [this web site](http://www.ieee-uffc.org/freqcontrol/quartz/vig/vigcomp.htm),
189+the accuracy of standard crystal oscillators (XO, TCXO, OCXO) is at best 1 in
190+100 million. The `src_ratio` is therefore 45035996 times more accurate than the
191+crystal clock source used to sample the original audio signal and any potential
192+problem with the `src_ratio` being a floating point number will be completely
193+swamped by sampling inaccuracies.
docs/foo_dsp_src9.png+0−0

Binary file not shown.

docs/history.md+34−0
@@ -0,0 +1,34 @@
1+---
2+layout: default
3+---
4+
5+# History
6+
7+- Version 0.0.0 (Oct 06 2002) First alpha release (friends and family only).
8+- Version 0.0.2 (Oct 10 2002) Second alpha release (fools rush in ....).
9+- ...
10+- Version 0.0.7 (Nov 17 2002) The last mono only version.
11+- Version 0.0.9 (Nov 21 2002) Full multi channel support. First pass at
12+ documentation. Still not ready for release.
13+- Version 0.0.11 (Nov 26 2002) Incorporated some suggestions from Conrad Parker.
14+- Version 0.0.12 (Nov 28 2002) First public release.
15+- Version 0.0.13 (Dec 03 2002) Fixes for MacOSX and Solaris.
16+- Version 0.0.14 (Jan 13 2003) Now compiles on Win32. Major code speedup. Minor
17+ bug fixes.
18+- Version 0.0.15 (May 02 2003) Minor bug fixes.
19+- Version 0.1.0 (Mar 14 2004) Add callback API, functions for short to float and
20+ float to short conversion. Minor bug fixes.
21+- Version 0.1.1 (Jul 17 2004) Callback API bug fix. Bugfix for aggressive
22+ gcc-3.4 optimisations.
23+- Version 0.1.2 (Sep 12 2004) Callback API reset bug fix.
24+- Version 0.1.3 (Mar 23 2008) Huge quality improvements to two best SINC based
25+ converters.
26+- Version 0.1.4 (Jul 02 2008) Fix segfault when using extremely low conversion
27+ ratios.
28+- Version 0.1.5 (Jan 11 2009) Optimisation resulting in dramatic throughput
29+ improvements ([See here.](http://www.mega-nerd.com/erikd/Blog/CodeHacking/SecretRabbitCode/rel_0_1_5.html)).
30+- Version 0.1.6 (Jan 27 2009) Minor bug fix in test suite (account for rounding
31+ error on x86_64).
32+- Version 0.1.7 (Feb 14 2009) Fix a segfault bug. Fix compilation under MSVC.
33+- Version 0.1.8 (Aug 15 2011) Minor bug fixes and updates.
34+- Version 0.1.9 (Sep 19 2016) Fix for a segfault. Relicense under BSD license.
docs/index.md+39−0
@@ -0,0 +1,39 @@
1+---
2+layout: default
3+title: Libsamplerate home page
4+---
5+
6+{: .indent_block}
7+*"Choosing a sample rate coverter wasn't easy. We ran numerous tests with Secret
8+Rabbit Code and other sample rate converters, then compared them all. In the
9+end, SRC outperformed the others, including some extremely well known and
10+popular software. We had one issue with SRC, but after emailing Erik, he got
11+back to us immediately with an answer. Choosing SRC was a no brainer."*
12+Ryan Smith, International Marketing Manager,
13+[Emersys Corp.](http://emersys.co.kr/), South Korea.
14+Product : [Maven3D](http://maven3d.com)
15+
16+**S**ecret **R**abbit **C**ode (aka libsamplerate) is a **S**ample **R**ate
17+**C**onverter for audio. One example of where such a thing would be useful is
18+converting audio from the CD sample rate of 44.1kHz to the 48kHz sample rate
19+used by DAT players.
20+
21+**SRC** is capable of arbitrary and time varying conversions; from downsampling
22+by a factor of 256 to upsampling by the same factor. Arbitrary in this case
23+means that the ratio of input and output sample rates can be an irrational
24+number. The conversion ratio can also vary with time for speeding up and slowing
25+down effects.
26+
27+**SRC** provides a small set of converters to allow quality to be traded off
28+against computation cost. The current best converter provides a signal-to-noise
29+ratio of 145dB with -3dB passband extending from DC to 96% of the theoretical
30+best bandwidth for a given pair of input and output sample rates.
31+
32+Since the library has few dependencies beyond that provided by the standard C
33+library, it should compile and work on just about any operating system. It is
34+known to work on Linux, MacOSX, [Win32](win32.md) and Solaris. With some
35+relatively minor hacking it should also be relatively easy to port it to
36+embedded systems and digital signal processors.
37+
38+In addition, the library comes with a comprehensive test suite which can
39+validate the performance of the library on new platforms.
docs/license.md+12−0
@@ -0,0 +1,12 @@
1+---
2+layout: default
3+---
4+
5+# License
6+
7+From its first release in 2002, Secret Rabbit Code was under a dual licensing
8+scheme where people could chose to use it under the terms of the GNU General
9+Public License or pay for a commercial use license.
10+
11+In 2016, thanks to a generous offer from Epic Games International, Secret Rabbit
12+Code was relicensed under the 2-clause BSD license.
docs/lists.md+17−0
@@ -0,0 +1,17 @@
1+---
2+layout: default
3+---
4+
5+# Mailing Lists
6+
7+There are currently two mailings lists for Secret Rabbit Code:
8+
9+- **src-announce\@mega-nerd.com**  
10+ [Subscribe](mailto:src-announce-request@mega-nerd.com?subject=subscribe)
11+
12+ A read only announcement list.
13+- **src\@mega-nerd.com**  
14+ [Subscribe](mailto:src-request@mega-nerd.com?subject=subscribe)
15+
16+ A general list which will also carry all the email from the announce
17+ list. Posting to this list is restricted to subscribers.
docs/paypal.png+0−0

Binary file not shown.

docs/quality.md+84−0
@@ -0,0 +1,84 @@
1+---
2+layout: default
3+---
4+
5+# SRC Quality
6+
7+**This document not yet complete.**
8+
9+When measuring the performance of a Sample Rate Converter, there are three
10+factors to consider:
11+
12+- **Signal-to-Noise Ratio** - a measure of how much noise the sample rate
13+ conversion process adds to the signal. This is measured in decibels (dB) and
14+ the higher this value the better. For most sample rate converters, the SNR
15+ will vary depending on the input signal and the ratio between input and output
16+ sample rates. The only valid comparison of SNR is between the worst case for
17+ each converter.
18+- **Bandwidth** - most sample rate converters attenuate high frequencies as part
19+ of their operation. Bandwidth can be measured by finding the frequency where
20+ the attenuation is 3dB and expressing that as a percentage of the full
21+ bandwidth at that sampling rate.
22+- **Speed** - the faster the better **:-)**.
23+
24+There are a number of sample rate converters available for downloading but I
25+will limit the comparison ot Secret Rabbit Code to the following:
26+
27+- [sndfile-resample](http://libsndfile.github.io/libsamplerate/download.html)
28+ which is a program (which uses libsamplerate) from the **examples/** directory
29+ of the Secret Rabbit Code source code distribution.
30+- [Resample](https://ccrma.stanford.edu/~jos/resample/) by Julius O Smiths which
31+ seems to have been the first high quality converter available as source code.
32+- [ResampAudio](http://www.tsp.ece.mcgill.ca/MMSP/Documents/Software/AFsp/ResampAudio.html)
33+ which is part of [Audio File Programs and Routines](http://www.tsp.ece.mcgill.ca/MMSP/Documents/Software/AFsp/AFsp.html)
34+ by Peter Kabal.
35+- [SoX](http://home.sprynet.com/~cbagwell/sox.html) which is maintained by Chris
36+ Bagwell. SoX is also able to perform some low quality sample rate conversions
37+ but these will not be investigated.
38+- [Shibatch](http://shibatch.sourceforge.net/) which seems to be a frequency
39+ domain sample rate converter. Unfortunately, this converter does not handle
40+ arbitrary conversion ratios and hence could not be properly compared to the
41+ other converters.
42+- [sr-convert](http://sr-convert.sourceforge.net/) is another converter which
43+ does not handle arbitrary conversion ratios.
44+
45+It should be noted that the first three converters above are based on the
46+algorithm by [Julius O. Smith](http://www-ccrma.stanford.edu/~jos/resample/)
47+which emulates the conversion of the digital signal to an analogue one and then
48+sampling the analogue signal at the new sample rate.
49+
50+## Methodology
51+
52+Measuring the SNR of a converter is relatively straight forward. Generate an
53+input signal consisting of a windowed sine wave, sample rate convert it and
54+measure the signal-to-noise ratio of the output signal. A typical length for the
55+original file is 30000 samples.
56+
57+The bandwidth of a sample rate converter is a little more difficult to measure.
58+Currently this is done by generating two short files containing a windowed sine
59+wave. The frequencies of the sine waves are 0.35 and 0.495 of the sample rate.
60+These file are then upsampled by a factor of 2 using the converter under test.
61+If the attenutaion of the lower frquency is less than 3dB and higher frequency
62+is more than 3dB, it is then possible to iteratively increase the lower
63+frequency and decrease the upper frequency keeping the -3dB point bracketed.
64+When the distance between the upper and lower frequency is sufficiently small,
65+it is possible to obtain a very accurate estimate of the -3dB frequency.
66+
67+The speed of a sample rate converter is easy to measure; simply perform a
68+conversion on a large file or a number of smaller files and time the conversion
69+process.
70+
71+The above measurement techniques are built into a test program which is
72+delivered with the Secret Rabbit Code source code distibution. This program is
73+able to test the first four of the above converters.
74+
75+## SoX
76+
77+SoX provides three methods of resampling; a linear interpolator, a polyphase
78+resampler and the Julius O. Smith simulated analogue filter method.
79+
80+## Shibatch
81+
82+Shibach
83+
84+**More Coming Soon.**
docs/win32.md+15−0
@@ -0,0 +1,15 @@
1+---
2+layout: default
3+---
4+
5+# Compiling on Win32
6+
7+You can use [CMake](https://cmake.org/) to generate Visual Studio project. The
8+configuration process is described [here](https://cmake.org/runningcmake/).
9+
10+The libsamplerate library itself does not require any dependencies, but if you
11+want to build examples and tests, you will need the [libsndfile](https://github.com/libsndfile/libsndfile)
12+and [FFTW](http://www.fftw.org/) libraries. The easiest way to install them is
13+to use a package manager, such as [Vcpkg](https://github.com/microsoft/vcpkg). Their README.md contains detailed
14+installation instructions for supported platforms. Libsamplerate requires the
15+`libsndfile` and `fftw3` packages.

Keyboard shortcuts

General

Show keyboard shortcuts
?
Focus the current search
/

Issues and pull requests

Open filters
f

Issues, pull requests and Inbox

Select next item
j↓
Select previous item
k↑
Open selected item
Enter
Dismiss filters or clear selection
Esc

Repository code

Go to file
t