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Add WaveFilePlayer, test files. Revised to lower WAV rates

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pull/16/merge
boblark 2 years ago
parent 5a9be45938
commit 624f14283e
12 changed files with 1032 additions and 0 deletions
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  1. 724
      AudioSDPlayer_F32.cpp
  2. 227
      AudioSDPlayer_F32.h
  3. 1
      OpenAudio_ArduinoLibrary.h
  4. 80
      examples/SDWavPlayer/SDWavPlayer.ino
  5. BIN
      utility/SDTEST1.WAV
  6. BIN
      utility/SDTEST2.WAV
  7. BIN
      utility/SDTEST3.WAV
  8. BIN
      utility/SDTEST4.WAV
  9. BIN
      utility/W9GR12.WAV
  10. BIN
      utility/W9GR24.WAV
  11. BIN
      utility/W9GR48.WAV
  12. BIN
      utility/W9GR6.WAV

724
AudioSDPlayer_F32.cpp
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/* Extended from Audio Library for Teensy which is
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice, development funding notice, and this permission
* notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*
* Extended by Chip Audette, OpenAudio, Dec 2019
* Converted to F32 and to variable audio block length
* The F32 conversion is under the MIT License. Use at your own risk.
*
* Further extensions to sub multiple WAV sample rates are copyright
* (c) 2023 Bob Larkin under the MIT License.
*/
#include <Arduino.h>
#include "AudioSDPlayer_F32.h"
#include "spi_interrupt.h"
#define STATE_DIRECT_8BIT_MONO 0 // playing mono at native sample rate
#define STATE_DIRECT_8BIT_STEREO 1 // playing stereo at native sample rate
#define STATE_DIRECT_16BIT_MONO 2 // playing mono at native sample rate
#define STATE_DIRECT_16BIT_STEREO 3 // playing stereo at native sample rate
#define STATE_CONVERT_8BIT_MONO 4 // playing mono, converting sample rate
#define STATE_CONVERT_8BIT_STEREO 5 // playing stereo, converting sample rate
#define STATE_CONVERT_16BIT_MONO 6 // playing mono, converting sample rate
#define STATE_CONVERT_16BIT_STEREO 7 // playing stereo, converting sample rate
#define STATE_PARSE1 8 // looking for 20 byte ID header
#define STATE_PARSE2 9 // looking for 16 byte format header
#define STATE_PARSE3 10 // looking for 8 byte data header
#define STATE_PARSE4 11 // ignoring unknown chunk after "fmt "
#define STATE_PARSE5 12 // ignoring unknown chunk before "fmt "
#define STATE_PAUSED 13
#define STATE_STOP 14
void AudioSDPlayer_F32::begin(void)
{
state = STATE_STOP;
state_play = STATE_STOP;
data_length = 0;
if (block_left_f32) {
AudioStream_F32::release(block_left_f32);
block_left_f32 = NULL;
}
if (block_right_f32) {
AudioStream_F32::release(block_right_f32);
block_right_f32 = NULL;
}
}
bool AudioSDPlayer_F32::play(const char *filename)
{
stop();
bool irq = false;
if (NVIC_IS_ENABLED(IRQ_SOFTWARE)) {
NVIC_DISABLE_IRQ(IRQ_SOFTWARE);
irq = true;
}
#if defined(HAS_KINETIS_SDHC)
if (!(SIM_SCGC3 & SIM_SCGC3_SDHC)) AudioStartUsingSPI();
#else
AudioStartUsingSPI();
#endif
wavfile = SD.open(filename);
if (!wavfile) {
#if defined(HAS_KINETIS_SDHC)
if (!(SIM_SCGC3 & SIM_SCGC3_SDHC)) AudioStopUsingSPI();
#else
AudioStopUsingSPI();
#endif
if (irq) NVIC_ENABLE_IRQ(IRQ_SOFTWARE);
return false;
}
buffer_length = 0;
buffer_offset = 0;
state_play = STATE_STOP;
data_length = 20;
header_offset = 0;
state = STATE_PARSE1;
if (irq) NVIC_ENABLE_IRQ(IRQ_SOFTWARE);
return true;
}
void AudioSDPlayer_F32::stop(void)
{
bool irq = false;
if (NVIC_IS_ENABLED(IRQ_SOFTWARE)) {
NVIC_DISABLE_IRQ(IRQ_SOFTWARE);
irq = true;
}
if (state != STATE_STOP) {
audio_block_f32_t *b1 = block_left_f32;
block_left_f32 = NULL;
audio_block_f32_t *b2 = block_right_f32;
block_right_f32 = NULL;
state = STATE_STOP;
if (b1) AudioStream_F32::release(b1);
if (b2) AudioStream_F32::release(b2);
wavfile.close();
#if defined(HAS_KINETIS_SDHC)
if (!(SIM_SCGC3 & SIM_SCGC3_SDHC)) AudioStopUsingSPI();
#else
AudioStopUsingSPI();
#endif
}
if (irq) NVIC_ENABLE_IRQ(IRQ_SOFTWARE);
}
void AudioSDPlayer_F32::togglePlayPause(void) {
// take no action if wave header is not parsed OR
// state is explicitly STATE_STOP
if(state_play >= 8 || state == STATE_STOP) return;
// toggle back and forth between state_play and STATE_PAUSED
if(state == state_play) {
state = STATE_PAUSED;
}
else if(state == STATE_PAUSED) {
state = state_play;
}
}
void AudioSDPlayer_F32::update(void)
{
int32_t n;
// only update if we're playing and not paused
if (state == STATE_STOP || state == STATE_PAUSED) return;
// allocate the audio blocks to transmit
block_left_f32 = AudioStream_F32::allocate_f32();
if (block_left_f32 == NULL) return;
if (state < 8 && (state & 1) == 1) {
// if we're playing stereo, allocate another
// block for the right channel output
block_right_f32 = AudioStream_F32::allocate_f32();
if (block_right_f32 == NULL) {
AudioStream_F32::release(block_left_f32);
return;
}
} else {
// if we're playing mono or just parsing
// the WAV file header, no right-side block
block_right_f32 = NULL;
}
block_offset = 0;
// is there buffered data?
n = buffer_length - buffer_offset;
if (n > 0) {
// Have buffered data. consume(n) returns true if audio transmitted.
if (consume(n)) return; // it was enough to transmit audio
}
// we only get to this point when buffer[512] is empty
if (state != STATE_STOP && wavfile.available()) {
// we can read more data from the file...
readagain:
buffer_length = wavfile.read(buffer, 512);
if (buffer_length == 0) goto end;
buffer_offset = 0;
bool parsing = (state >= 8);
bool txok = consume(buffer_length);
if (txok) {
if (state != STATE_STOP) return;
} else {
if (state != STATE_STOP) {
if (parsing && state < 8) goto readagain;
else goto cleanup;
}
}
}
end: // end of file reached or other reason to stop
wavfile.close();
#if defined(HAS_KINETIS_SDHC)
if (!(SIM_SCGC3 & SIM_SCGC3_SDHC)) AudioStopUsingSPI();
#else
AudioStopUsingSPI();
#endif
state_play = STATE_STOP;
state = STATE_STOP;
cleanup:
if (block_left_f32) {
if (block_offset > 0) {
for (uint32_t i=block_offset; i < audio_block_samples; i++) {
block_left_f32->data[i] = 0.0f;
}
transmit(block_left_f32, 0);
if (state < 8 && (state & 1) == 0) {
transmit(block_left_f32, 1);
}
}
AudioStream_F32::release(block_left_f32);
block_left_f32 = NULL;
}
if (block_right_f32) {
if (block_offset > 0) {
for (uint32_t i=block_offset; i < audio_block_samples; i++) {
block_right_f32->data[i] = 0.0f;
}
transmit(block_right_f32, 1);
}
AudioStream_F32::release(block_right_f32);
block_right_f32 = NULL;
}
}
// Consume already buffered WAV file data. Returns true if audio transmitted.
bool AudioSDPlayer_F32::consume(uint32_t size) {
uint32_t len;
uint8_t lsb, msb;
const uint8_t *p;
int16_t val_int16;
float32_t rateRatioF;
rateRatioF = (float32_t) pSampleSubMultiple->rateRatio;
p = buffer + buffer_offset;
start:
if (size == 0) return false;
/*
Serial.print("AudioSDPlayer_F32 consume, ");
Serial.print("size = ");
Serial.print(size);
Serial.print(", buffer_offset = ");
Serial.print(buffer_offset);
Serial.print(", data_length = ");
Serial.print(data_length);
Serial.print(", space = ");
Serial.print((audio_block_samples - block_offset) * 2);
Serial.print(", state = ");
Serial.println(state);
*/
switch (state) {
// parse wav file header, is this really a .wav file?
case STATE_PARSE1:
len = data_length;
if (size < len) len = size;
memcpy((uint8_t *)header + header_offset, p, len);
header_offset += len;
buffer_offset += len;
data_length -= len;
if (data_length > 0) return false;
// parse the header...
if (header[0] == 0x46464952 && header[2] == 0x45564157)
{
//Serial.println("is wav file");
if (header[3] == 0x20746D66)
{
// "fmt " header
if (header[4] < 16)
{
// WAV "fmt " info must be at least 16 bytes
break;
}
if (header[4] > sizeof(header))
{
// if such .wav files exist, increasing the
// size of header[] should accomodate them...
//Serial.println("WAVEFORMATEXTENSIBLE too long");
break;
}
//Serial.println("header ok");
header_offset = 0;
state = STATE_PARSE2;
}
else
{
// first chuck is something other than "fmt "
//Serial.print("skipping \"");
//Serial.printf("\" (%08X), ", __builtin_bswap32(header[3]));
//Serial.print(header[4]);
//Serial.println(" bytes");
header_offset = 12;
state = STATE_PARSE5;
}
p += len;
size -= len;
data_length = header[4];
goto start;
}
//Serial.println("unknown WAV header");
break;
// check & extract key audio parameters
case STATE_PARSE2:
len = data_length;
if (size < len) len = size;
memcpy((uint8_t *)header + header_offset, p, len);
header_offset += len;
buffer_offset += len;
data_length -= len;
if (data_length > 0) return false;
if (parse_format())
{
//Serial.println("audio format ok");
p += len;
size -= len;
data_length = 8;
header_offset = 0;
state = STATE_PARSE3;
goto start;
}
//Serial.println("unknown audio format");
break;
// find the data chunk
case STATE_PARSE3: // 10
len = data_length;
if (size < len) len = size;
memcpy((uint8_t *)header + header_offset, p, len);
header_offset += len;
buffer_offset += len;
data_length -= len;
if (data_length > 0) return false;
p += len;
size -= len;
data_length = header[1];
if (header[0] == 0x61746164)
{
// TODO: verify offset in file is an even number
// as required by WAV format. abort if odd. Code
// below will depend upon this and fail if not even.
leftover_bytes = 0;
state = state_play;
if (state & 1)
{
// if we're going to start stereo
// better allocate another output block
block_right_f32 = AudioStream_F32::allocate_f32();
if (!block_right_f32) return false;
}
total_length = data_length;
}
else
{
state = STATE_PARSE4;
}
goto start;
// ignore any extra unknown chunks (title & artist info)
case STATE_PARSE4: // 11
if (size < data_length)
{
data_length -= size;
buffer_offset += size;
return false;
}
p += data_length;
size -= data_length;
buffer_offset += data_length;
data_length = 8;
header_offset = 0;
state = STATE_PARSE3;
goto start;
// skip past "junk" data before "fmt " header
case STATE_PARSE5:
len = data_length;
if (size < len) len = size;
buffer_offset += len;
data_length -= len;
if (data_length > 0) return false;
p += len;
size -= len;
data_length = 8;
state = STATE_PARSE1;
goto start;
// playing mono at native sample rate
case STATE_DIRECT_8BIT_MONO:
return false;
// playing stereo at native sample rate
case STATE_DIRECT_8BIT_STEREO:
return false;
// Playing Mono at native sample rate ****** 16-BIT MONO ******
case STATE_DIRECT_16BIT_MONO:
if (size > data_length) // End of WAV file
size = data_length;
data_length -= size;
while (1)
{
if(zerosToSend > 0)
{
block_left_f32->data[block_offset++] = 0.0f; // Zeros for interpolation
zerosToSend--;
if (block_offset >= audio_block_samples)
{
if(pSampleSubMultiple->numCoeffs > 1 && // i.e., using FIR
pSampleSubMultiple->firBufferL )
{
arm_fir_f32(&fir_instL, block_left_f32->data,
block_left_f32->data, block_left_f32->length);
}
transmit(block_left_f32, 0); // Mono sends same to L&R
transmit(block_left_f32, 1);
AudioStream_F32::release(block_left_f32);
block_left_f32 = NULL;
data_length += size;
buffer_offset = p - buffer;
if (block_right_f32)
AudioStream_F32::release(block_right_f32);
if (data_length == 0)
state = STATE_STOP;
return true;
}
}
else // Not zeros, but data
{
lsb = *p++; // Little endian
msb = *p++;
size -= 2; // 2 bytes per word
// Convert to F32
val_int16 = (msb << 8) | lsb;
// Scale up by rateRatioF to account for zeros
block_left_f32->data[block_offset++] = rateRatioF*((float)val_int16)/(32768.0);
// For interpolation, each data point is followed by 0.0f's
zerosToSend = pSampleSubMultiple->rateRatio - 1; // 0, 1, 3, 7
if (block_offset >= audio_block_samples)
{
// The FIR update
if(pSampleSubMultiple->numCoeffs > 1 &&
pSampleSubMultiple->firBufferL )
{
arm_fir_f32(&fir_instL, block_left_f32->data,
block_left_f32->data, block_left_f32->length);
}
transmit(block_left_f32, 0); // Mono sends same to L&R
transmit(block_left_f32, 1);
AudioStream_F32::release(block_left_f32);
block_left_f32 = NULL;
data_length += size;
buffer_offset = p - buffer;
if (block_right_f32)
AudioStream_F32::release(block_right_f32);
if (data_length == 0)
state = STATE_STOP;
return true;
}
}
} // End while(1)
if (size == 0)
{
if (data_length == 0) break;
return false;
}
// End of file reached
if (block_offset > 0)
{
// TODO: fill remainder of last block with zero and transmit
}
state = STATE_STOP;
return false;
// Playing stereo at native sample rate ****** 16-BIT STEREO ******
case STATE_DIRECT_16BIT_STEREO:
if (size > data_length)
size = data_length;
data_length -= size;
if (leftover_bytes)
{
block_left_f32->data[block_offset] = header[0];
//PAH fix problem with left+right channels being swapped
//RSL Is this actually the CODEC L/R problem?
leftover_bytes = 0;
// goto right16; // RSL What is the deal???
}
while (1) {
if(zerosToSend > 0)
{
block_left_f32->data[block_offset] = 0.0f; // Zeros for interpolation
block_right_f32->data[block_offset++] = 0.0f;
zerosToSend--;
if (block_offset >= audio_block_samples)
{
if(pSampleSubMultiple->numCoeffs > 1 && // i.e., using FIR
pSampleSubMultiple->firBufferL)
{
arm_fir_f32(&fir_instL, block_left_f32->data,
block_left_f32->data, block_left_f32->length);
arm_fir_f32(&fir_instR, block_right_f32->data,
block_right_f32->data, block_right_f32->length);
}
transmit(block_left_f32, 0);
transmit(block_right_f32, 1);
AudioStream_F32::release(block_left_f32);
block_left_f32 = NULL;
data_length += size;
buffer_offset = p - buffer;
if (block_right_f32)
AudioStream_F32::release(block_right_f32);
if (data_length == 0)
state = STATE_STOP;
return true;
}
}
else // Not zeros, but data
{
lsb = *p++; // Little endian
msb = *p++;
size -= 2;
if (size == 0)
{
if (data_length == 0) break;
header[0] = (msb << 8) | lsb;
leftover_bytes = 2;
return false;
}
val_int16 = (int16_t)((msb << 8) | lsb);
//convert from int16 to float32 spanning +/-1.0
// Scale up by rateRatioF to account for zeros
block_left_f32->data[block_offset] = rateRatioF*((float)val_int16)/(32768.0);
// right16: See about 15 lines above
lsb = *p++;
msb = *p++;
size -= 2;
val_int16 = (int16_t)((msb << 8) | lsb);
// Convert from int16 to float32 spanning +/-1.0
// Scale up by rateRatioF to account for zeros
block_right_f32->data[block_offset++] = rateRatioF*((float)val_int16)/(32768.0);
// For stereo, the number of zeros to send refers to
// the number of *pairs* of zeros.
// For interpolation, each data point is followed by 0.0f's
zerosToSend = pSampleSubMultiple->rateRatio - 1; // 0, 1, 3, 7
if (block_offset >= audio_block_samples)
{
if(pSampleSubMultiple->numCoeffs > 1 && // i.e., using FIR
pSampleSubMultiple->firBufferL )
{
arm_fir_f32(&fir_instL, block_left_f32->data,
block_left_f32->data, block_left_f32->length);
arm_fir_f32(&fir_instR, block_right_f32->data,
block_right_f32->data, block_right_f32->length);
}
transmit(block_left_f32, 0);
AudioStream_F32::release(block_left_f32);
block_left_f32 = NULL;
transmit(block_right_f32, 1);
AudioStream_F32::release(block_right_f32);
block_right_f32 = NULL;
data_length += size;
buffer_offset = p - buffer;
if (data_length == 0) state = STATE_STOP;
return true;
}
if (size == 0)
{
if (data_length == 0) break;
leftover_bytes = 0;
return false;
}
} // Sending data, not zeros
// end of file reached
} // End while(1)
if (block_offset > 0)
{
// TODO: fill remainder of last block with zero and transmit
}
state = STATE_STOP;
return false;
// playing mono, converting sample rate
case STATE_CONVERT_8BIT_MONO :
return false;
// playing stereo, converting sample rate
case STATE_CONVERT_8BIT_STEREO:
return false;
// playing mono, converting sample rate
case STATE_CONVERT_16BIT_MONO:
return false;
// playing stereo, converting sample rate
case STATE_CONVERT_16BIT_STEREO:
return false;
// ignore any extra data after playing
// or anything following any error
case STATE_STOP:
return false;
// this is not supposed to happen!
//default:
//Serial.println("AudioSDPlayer_F32, unknown state");
}
state_play = STATE_STOP;
state = STATE_STOP;
return false;
}
bool AudioSDPlayer_F32::parse_format(void) {
uint8_t num = 0;
uint16_t format;
uint16_t channels;
uint32_t rate, b2m;
uint16_t bits;
format = header[0];
currentWavData.audio_format = header[0]; // uint16_t
//Serial.print(" format = ");
//Serial.println(format);
if (format != 1) return false;
rate = header[1];
currentWavData.sample_rate = header[1]; // uint32_t
Serial.print("WAV file sample rate = "); Serial.println(rate);
// b2m is used to determine playing time. We base it on the WAV
// file meta data. It is allowed to be played at a different rate
// but all we do is to make the info available via the
// struct currentWavData The INO needs to deal with differences.
// 4294967296000.0 = 2^32 * 1000
b2m = (uint32_t)((double)4294967296000.0 / (double)rate);
channels = header[0] >> 16;
currentWavData.num_channels = header[0] >> 16; // uint16_t
//Serial.print(" channels = ");
//Serial.println(channels);
if (channels == 1) { }
else if (channels == 2)
{
b2m >>= 1; // Divide b2m by 2
num |= 1;
}
else
return false;
bits = header[3] >> 16;
currentWavData.bits = header[3] >> 16; // uint16_t
//Serial.print(" bits = ");
//Serial.println(bits);
if (bits == 8) { }
else if (bits == 16)
{
b2m >>= 1; // Again divide b2m by 2
num |= 2;
}
else {return false;}
bytes2millis = b2m; // Transfer to global
Serial.print(" bytes2millis = "); Serial.println(b2m);
// we're not checking the byte rate and block align fields
// if they're not the expected values, all we could do is
// return false. Do any real wav files have unexpected
// values in these other fields?
state_play = num;
return true;
}
uint32_t AudioSDPlayer_F32::updateBytes2Millis(void) {
double b2m;
//account for sample rate
b2m = ((double)4294967296000.0 / ((double)sample_rate_Hz));
//account for channels
b2m = b2m / ((double)channels);
//account for bits per second
if (bits == 16)
b2m = b2m / 2;
else if (bits == 24)
b2m = b2m / 3; //can we handle 24 bits? I don't think that we can.
// if 8-bits, fall through
return bytes2millis = (uint32_t)b2m;
}
bool AudioSDPlayer_F32::isPlaying(void) {
uint8_t s = *(volatile uint8_t *)&state;
return (s < 8);
}
bool AudioSDPlayer_F32::isPaused(void) {
uint8_t s = *(volatile uint8_t *)&state;
return (s == STATE_PAUSED);
}
bool AudioSDPlayer_F32::isStopped(void) {
uint8_t s = *(volatile uint8_t *)&state;
return (s == STATE_STOP);
}
uint32_t AudioSDPlayer_F32::positionMillis(void) {
uint8_t s = *(volatile uint8_t *)&state;
if (s >= 8 && s != STATE_PAUSED) return 0;
uint32_t tlength = *(volatile uint32_t *)&total_length;
uint32_t dlength = *(volatile uint32_t *)&data_length;
uint32_t offset = tlength - dlength;
uint32_t b2m = *(volatile uint32_t *)&bytes2millis;
return ((uint64_t)offset * b2m) >> 32;
}
uint32_t AudioSDPlayer_F32::lengthMillis(void) {
uint8_t s = *(volatile uint8_t *)&state;
if (s >= 8 && s != STATE_PAUSED) return 0;
uint32_t tlength = *(volatile uint32_t *)&total_length;
uint32_t b2m = *(volatile uint32_t *)&bytes2millis;
return ((uint64_t)tlength * b2m) >> 32;
}

227
AudioSDPlayer_F32.h
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/* *** AudioSDPlayer_F32.h ***
*
* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.comaudio_block_samples
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice, development funding notice, and this permission
* notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/*
* Extended by Chip Audette, OpenAudio, Dec 2019
* Converted to F32 and to variable audio block length
* The F32 conversion is under the MIT License. Use at your own risk.
*/
/* WAV File Format
Bytes Meaning
1 - 4 “RIFF” Marks the file as a riff file. Characters are each 1 byte long.
5 - 8 File size (integer) Size of the overall file - 8 bytes, in bytes
(32-bit integer).
9 -12 “WAVE” File Type Header. For our purposes, it always equals “WAVE”.
13-16 “fmt " Format chunk marker. Includes trailing null
17-20 Length of format data as listed above, e.g., 16
21-22 Type of format (1 is PCM) - 2 byte integer
23-24 Number of Channels - 2 byte integer, e.g. 2
25-28 Sample Rate - 32 byte integer. Common values are 44100 (CD),
48000 (DAT). Sample Rate = Number of Samples per second, or Hertz.
29-32 176400 (Sample Rate * BitsPerSample * Channels) / 8.
33-34 4 (BitsPerSample * Channels) / 8.1 - 8 bit mono2 - 8 bit
stereo/16 bit mono4 - 16 bit stereo
35-36 16 Bits per sample or other
37-40 "data" Marks the beginning of the data section.
41-44 File size (data) Size of the data section.
*
Sample of WAV file start:
00000000 52494646 66EA6903 57415645 666D7420 RIFFf.i.WAVEfmt
00000010 10000000 01000200 44AC0000 10B10200 ........D.......
00000020 04001000 4C495354 3A000000 494E464F ....LIST:...INFO
00000030 494E414D 14000000 49205761 6E742054 INAM....I Want T
00000040 6F20436F 6D65204F 76657200 49415254 o Come Over.IART
00000050 12000000 4D656C69 73736120 45746865 ....Melissa Ethe
00000060 72696467 65006461 746100EA 69030100 ridge.data..i...
00000070 FEFF0300 FCFF0400 FDFF0200 0000FEFF ................
00000080 0300FDFF 0200FFFF 00000100 FEFF0300 ................
00000090 FDFF0300 FDFF0200 FFFF0100 0000FFFF ................
*/
/* *** SAMPLE RATES ***
* In the case of WAV files, there is a specified sample rate that is part
* of the header data. The file is a stream of numbers. If these are
* turned into voltages and played at the specified rate, everything will
* sound correct. For shorthand, we will call this the WAV rate.
*
* There is also a Teensy Sample Rate for the Audio system, set by things
* like the hardware clock for I2S. If this sample rate is the same as
* the WAV rate, life is simple and we process and output a sample
* with the WAV rate being achieved. A second case is for the WAV rate
* to be an integer sub-multiple of the Teensy Sample Rate. This would
* allow the wave file to run with a 12 ksps sample rate and the Teensy
* Sample Rate to be 48, or 96, ksps. There is a data interpolator
* included after the the data has been read from the file. This requires
* specification of the sub-multiple integer and a FIR filter to complete
* the interpolation operation. The structure sampleSubMultiple,
* provided by the .INO, communicates this design information from the INO.
*
* The third case is to have the Wave rate and the Teensy Sample rate
* related by a rational fraction. This would require a rate changet
* consisting of both a decimator and an interpolator. None of that is
* included in this class.
*/
#ifndef AudioSDPlayer_F32_h_
#define AudioSDPlayer_F32_h_
#include "Arduino.h"
#include "AudioSettings_F32.h"
#include "AudioStream_F32.h"
#include <SdFat.h> //included in Teensy install as of Teensyduino 1.54-bete3
// This communicates the info for running slow WAV file sample rates.
// This one is declared in the .INO
struct subMult {
uint16_t rateRatio; // Should be 1 for no rate change, else 2, 4, 8
uint16_t numCoeffs; // FIR filter
float32_t* firCoeffs; // FIR Filter Coeffs
float32_t* firBufferL; // pointer to 127 + numCoeffs float32_t, left ch
float32_t* firBufferR; // pointer to 127 + numCoeffs float32_t, right ch
};
// This communicates the important parameters of the WAV file. This is
// declared in AudioSDPlayer_F32 to provide data to the .INO.
struct wavData {
uint16_t audio_format; // Should be 1 for PCM
uint16_t num_channels; // 1 for mono, 2 for stereo
uint32_t sample_rate; // 44100, 48000, etc
uint16_t bits; // Number of bits per sample
};
class AudioSDPlayer_F32 : public AudioStream_F32
{
//GUI: inputs:0, outputs:2 //this line used for automatic generation of GUI nodes
public:
AudioSDPlayer_F32(void) :
AudioStream_F32(0, NULL), block_left_f32(NULL), block_right_f32(NULL)
{
begin();
}
AudioSDPlayer_F32(const AudioSettings_F32 &settings) :
AudioStream_F32(0, NULL), block_left_f32(NULL), block_right_f32(NULL)
{
setSampleRate_Hz(settings.sample_rate_Hz);
//setBlockSize(settings.audio_block_samples); // Always 128
begin();
}
void begin(void); //begins SD card
bool play(const char *filename);
void stop(void);
void togglePlayPause(void);
bool isPaused(void);
bool isStopped(void);
bool isPlaying(void);
uint32_t positionMillis(void);
uint32_t lengthMillis(void);
// Required when WAV file is at a sub-multiple rate of audio sampling rate
void setSubMult(subMult* pSampleSubMultipleStruct) {
if(pSampleSubMultipleStruct->rateRatio == 1 ||
pSampleSubMultipleStruct->rateRatio == 2 ||
pSampleSubMultipleStruct->rateRatio == 4 ||
pSampleSubMultipleStruct->rateRatio == 8)
{
pSampleSubMultiple = pSampleSubMultipleStruct;
if(pSampleSubMultiple->numCoeffs > 1 &&
pSampleSubMultiple->firBufferL )
{
arm_fir_init_f32(&fir_instL,
pSampleSubMultiple->numCoeffs,
(float32_t *)pSampleSubMultiple->firCoeffs,
(float32_t *)pSampleSubMultiple->firBufferL,
(uint32_t)audio_block_samples);
arm_fir_init_f32(&fir_instR,
pSampleSubMultiple->numCoeffs,
(float32_t *)pSampleSubMultiple->firCoeffs,
(float32_t *)pSampleSubMultiple->firBufferR,
(uint32_t)audio_block_samples);
}
}
else
Serial.println("Illegal sub-division multiple for WAV rate.");
}
// Provides basic meta-data about WAV file.
wavData* getCurrentWavData(void) {
return &currentWavData; // Pointer to structure
}
float32_t setSampleRate_Hz(float32_t fs_Hz) {
sample_rate_Hz = fs_Hz;
updateBytes2Millis();
return sample_rate_Hz;
}
virtual void update(void);
private:
File wavfile;
struct subMult* pSampleSubMultiple = &nEqOneTemp;
// Next is a dummy structure to divide by 1 when no INO structure
struct subMult nEqOneTemp = {1, 0, NULL, NULL, NULL};
arm_fir_instance_f32 fir_instL;
arm_fir_instance_f32 fir_instR;
struct wavData currentWavData = {1, 2, 44100, 16};
bool consume(uint32_t size);
bool parse_format(void);
uint32_t header[10]; // temporary storage of wav header data
uint32_t data_length; // number of bytes remaining in current section
uint32_t total_length; // number of audio data bytes in file
uint16_t channels = 1; //number of audio channels
uint16_t bits = 16; // number of bits per sample
uint32_t bytes2millis;
// Variables for audio library storage, float32_t
audio_block_f32_t *block_left_f32 = NULL;
audio_block_f32_t *block_right_f32 = NULL;
uint16_t block_offset; // how much data is in block_left & block_right
// Variables for buffering the WAV file read, uint8_t
uint8_t buffer[512]; // buffer one block of SD file data
uint16_t buffer_offset; // where we're at consuming "buffer"
uint16_t buffer_length; // how many data bytes are in "buffer" (512 until last read)
uint8_t header_offset; // number of bytes in header[]
// Variables to control the WAV file reading
uint8_t state;
uint8_t state_play;
uint8_t leftover_bytes;
// Variables for WAV file sampled at a sub rate from audio process
uint8_t zerosToSend = 0;
static unsigned long update_counter;
float sample_rate_Hz = ((float)AUDIO_SAMPLE_RATE_EXACT);
uint16_t audio_block_samples = AUDIO_BLOCK_SAMPLES;
uint32_t updateBytes2Millis(void);
//int32_t pctr = 0;
};
#endif

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OpenAudio_ArduinoLibrary.h
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#include "AudioLMSDenoiseNotch_F32.h" #include "AudioLMSDenoiseNotch_F32.h"
#include "AudioMixer_F32.h" #include "AudioMixer_F32.h"
#include "AudioMultiply_F32.h" #include "AudioMultiply_F32.h"
#include "AudioSDPlayer_F32.h"
#include "AudioSettings_F32.h" #include "AudioSettings_F32.h"
#include "AudioSpectralDenoise_F32.h" #include "AudioSpectralDenoise_F32.h"
#include "input_i2s_f32.h" #include "input_i2s_f32.h"

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examples/SDWavPlayer/SDWavPlayer.ino
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/*
* SDWavPlayer
*
* Created: Chip Audette, OpenAudio, Dec 2019
* Based On: WaveFilePlayer from Paul Stoffregen, PJRC, Teensy
*
* Play back a WAV file through the Typman.
*
* For access to WAV files, please visit https://www.pjrc.com/teensy/td_libs_AudioDataFiles.html.
*
*/
#include "OpenAudio_ArduinoLibrary.h"
#include "AudioSDPlayer_F32.h"
//set the sample rate and block size
const float sample_rate_Hz = 44100.0f;
const int audio_block_samples = 128; // Must be 128 for SD recording.
AudioSettings_F32 audio_settings(sample_rate_Hz, audio_block_samples);
//create audio objects
AudioSDPlayer_F32 audioSDPlayer(audio_settings);
AudioOutputI2S_F32 audioOutput(audio_settings);
//Tympan myTympan(TympanRev::E); //do TympanRev::D or TympanRev::E
//create audio connections
AudioConnection_F32 patchCord1(audioSDPlayer, 0, audioOutput, 0);
AudioConnection_F32 patchCord2(audioSDPlayer, 1, audioOutput, 1);
AudioControlSGTL5000 sgtl5000_1;
// Use these with the Teensy 4.x Rev D Audio Shield
#define SDCARD_CS_PIN 10
#define SDCARD_MOSI_PIN 11
#define SDCARD_SCK_PIN 13
void setup() {
Serial.begin(300); delay(1000);
Serial.print("### SDWavPlayer ###");
Serial.print("Sample Rate (Hz): "); Serial.println(audio_settings.sample_rate_Hz);
Serial.print("Audio Block Size (samples): "); Serial.println(audio_settings.audio_block_samples);
// Audio connections require memory to work.
AudioMemory_F32(20, audio_settings);
sgtl5000_1.enable();
SPI.setMOSI(SDCARD_MOSI_PIN);
SPI.setSCK(SDCARD_SCK_PIN);
if (!(SD.begin(SDCARD_CS_PIN))) {
// stop here, but print a message repetitively
while (1) {
Serial.println("*** Unable to access the SD card ***");
delay(1000);
}
}
//prepare SD player
audioSDPlayer.begin();
//finish setup
delay(2000); //stall a second
Serial.println("Setup complete.");
}
unsigned long end_millis = 0;
String filename = "SDTEST1.WAV";// filenames are always uppercase 8.3 format
void loop() {
/*
//service the audio player
if (!audioSDPlayer.isPlaying()) { //wait until previous play is done
//start playing audio
Serial.print("Starting audio player: ");
Serial.println(filename);
audioSDPlayer.play(filename);
}
*/
delay(500);
}

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