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- <p align="center">
- <b>Guitar and Bass Tuner Library</b><br>
- <b>Teensy 3.1 v2.0</b><br>
- </p>
-
- <img src="http://www.sciweavers.org/tex2img.php?eq=%20%5Ctau%20&bc=White&fc=Black&im=png&fs=12&ff=arev&edit=0" align="center" border="0" alt=" \tau " width="15" height="11" />
-
- >Software algorithm ([YIN]) for guitar and bass tuning using a Teensy Audio Library. This audio object's algorithm can be some what memory and processor hungry but will allow you to detect with fairly good accuracy the fundamental frequencies f<sub>o</sub> from electric guitars and basses.
- >>You can install this as a normal Arduino Library and will work with the Audio Library, no need to edit the Audio libraries source now.
-
- <!-- language: lang-none -->
- Hookup Guide - 1.2v DC Bias and High Pass Filter - No Amplification
-
- *--------------------------------------------------*
- | |
- | | ' | |
- *------------/\/\/\-------------* |' '| |
- | 47K | _|_'_|_ |
- | | |` ` ` `| |
- *---)|+--* | | ` ` ` | |
- | 10uF | | |` ` ` `| |
- TEENSY 3.1 | | | | ` ` ` | |
- _______________ *-/\/\/\-* | |` ` ` `| |
- |GND |_____| Vin| | 2.2K | | | ` ` ` | |
- |0 ---- AGND|<-* | | |` ` ` `| |
- |1 |````| 3.3V|>--/\/\/\--*--/\/\/\---* | | ` ` ` | |
- |2 |````| 23| 10K 47K | | |` ` ` `| |
- |3 ---- 22| | | | ` ` ` | |
- |4 |'| 21| | | \=====/ |
- |5 ------ 20| | | | :`| |
- |6 |::::::| 19| | REMOVE | | S`| |
- |7 |::::::| 18| | DC | | H`| |
- |8 |::::::| 17| | BIAS | | I`| |
- |9 ------A2/16|<---SIGNAL-1.2v-BIAS---*---+|(----* | E`|>--ANGD--*
- |10 --- 15| 1.2VDC 10uF | | L`|
- |11 |(`)| 14| | | D`|
- |12 --- 13| | | :`|
- --------------- | |===|
- | \_/
- | /T\
- | - I -
- *---<\ P /
- \_/
-
- >Many optimizations have been done to the [YIN] algorithm for frequencies between 29-360Hz.
- >>While its still using a brute force method ( n<sup>2</sup> ) for finding the fundamental frequency f<sub>o</sub>, it is tuned to skip certain <b>tau</b> values and focus mostly on frequencies found in the bass and guitar.
- >>>The input is double buffered so while you are processing one buffer it is filling the other to double throughput.
- >>>>There are a few parameters that can be adjusted to "dial in" the algorithm for better estimations. The defaults are what I found that have the best trade off for speed and accuracy.
-
- <h4>AudioTuner.h</h4>
-
- ```
- /****************************************************************/
- #define SAMPLE_RATE_DIVIDE_BY_1 1 // 44100 sample rate
- #define SAMPLE_RATE_DIVIDE_BY_2 2 // 22050 sample rate
- #define SAMPLE_RATE_DIVIDE_BY_4 4 // 11025 sample rate
- #define SAMPLE_RATE_DIVIDE_BY_8 8 // 5512.5 sample rate
- #define SAMPLE_RATE_DIVIDE_BY_16 16 // 2756.25 sample rate
- #define SAMPLE_RATE_DIVIDE_BY_32 32 // 1378.125 sample rate
- /****************************************************************
- * Safe to adjust these values below *
- ****************************************************************/
- // Adjust number of samples to collect in buffer here, also effects
- // convergence speed and resolution.
- #define NUM_SAMPLES 2048 // make a power of two
-
- // larger the divide-by, less resolution and lower the frequency for
- // a given number of samples that can be detected. Also effects
- // convergence speed.
- #define SAMPLE_SKIP SAMPLE_RATE_DIVIDE_BY_2
- /****************************************************************/
- ```
-
- ```
- SAMPLE_RATE_DIVIDE_BY_x --> This sets 'SAMPLE_SKIP' to pass on every (x) data point from the
- Audio Block being saved to the buffer, it determines the sample rate.
- ```
-
- ```
- NUM_SAMPLES --> This the size of each buffer, there two for double buffering.
- ```
- ```
- SAMPLE_SKIP --> This sets your sample window length and sampling rate. Sample Window Size is
- (NUM_SAMPLES * SAMPLE_SKIP) of the ~44100 samples every second. Sample Rate is
- (AUDIO_SAMPLE_RATE_EXACT / SAMPLE_SKIP).
- ```
-
- [YIN]:http://recherche.ircam.fr/equipes/pcm/cheveign/pss/2002_JASA_YIN.pdf
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