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update to 1.9.7

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This commit is contained in:
willem
2023-01-04 14:01:20 +01:00
parent 6110583190
commit adbed6b207
56 changed files with 596 additions and 512 deletions
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10
src/AudioFileSourceFunction.cpp
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@@ -1,6 +1,6 @@
/*
AudioFileSourceFunction
Audio ouptut generator which can generate WAV file data from function
Audio output generator which can generate WAV file data from function
Copyright (C) 2021 Hideaki Tai
@@ -25,14 +25,14 @@ AudioFileSourceFunction::AudioFileSourceFunction(float sec, uint16_t channels, u
uint32_t len = uint32_t(sec * (float)bytes_per_sec);
// RIFF chunk
strncpy(wav_header.riff.chunk_id, "RIFF", 4);
memcpy(wav_header.riff.chunk_id, "RIFF", 4);
wav_header.riff.chunk_size = 4 // size of riff chunk w/o chunk_id and chunk_size
+ 8 + 16 // size of format chunk
+ 8 + len; // size of data chunk
strncpy(wav_header.riff.format, "WAVE", 4);
memcpy(wav_header.riff.format, "WAVE", 4);
// format chunk
strncpy(wav_header.format.chunk_id, "fmt ", 4);
memcpy(wav_header.format.chunk_id, "fmt ", 4);
wav_header.format.chunk_size = 16;
wav_header.format.format_tag = 0x0001; // PCM
wav_header.format.channels = channels;
@@ -42,7 +42,7 @@ AudioFileSourceFunction::AudioFileSourceFunction(float sec, uint16_t channels, u
wav_header.format.bits_per_sample = bits_per_sample;
// data chunk
strncpy(wav_header.data.chunk_id, "data", 4);
memcpy(wav_header.data.chunk_id, "data", 4);
wav_header.data.chunk_size = len;
funcs.reserve(channels);

2
src/AudioFileSourceFunction.h
View File

@@ -1,6 +1,6 @@
/*
AudioFileSourceFunction
Audio ouptut generator which can generate WAV file data from function
Audio output generator which can generate WAV file data from function
Copyright (C) 2021 Hideaki Tai

4
src/AudioFileSourceICYStream.cpp
View File

@@ -20,6 +20,9 @@
#if defined(ESP32) || defined(ESP8266)
#ifdef _GNU_SOURCE
#undef _GNU_SOURCE
#endif
#define _GNU_SOURCE
#include "AudioFileSourceICYStream.h"
@@ -47,6 +50,7 @@ bool AudioFileSourceICYStream::open(const char *url)
http.addHeader("Icy-MetaData", "1");
http.collectHeaders( hdr, 4 );
http.setReuse(true);
http.setFollowRedirects(HTTPC_FORCE_FOLLOW_REDIRECTS);
int code = http.GET();
if (code != HTTP_CODE_OK) {
http.end();

8
src/AudioFileSourceLittleFS.h
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@@ -18,10 +18,8 @@
along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef _AUDIOFILESOURCESPIFFS_H
#define _AUDIOFILESOURCESPIFFS_H
//#ifndef ESP32 // No LittleFS there, yet
#ifndef _AUDIOFILESOURCELITTLEFS_H
#define _AUDIOFILESOURCELITTLEFS_H
#include <Arduino.h>
#include <LittleFS.h>
@@ -37,7 +35,5 @@ class AudioFileSourceLittleFS : public AudioFileSourceFS
// Others are inherited from base
};
//#endif
#endif

3
src/AudioFileSourceSPIFFS.h
View File

@@ -27,6 +27,9 @@
#include "AudioFileSource.h"
#include "AudioFileSourceFS.h"
// Yes, I know SPIFFS is deprecated
#pragma GCC diagnostic ignored "-Wdeprecated-declarations"
class AudioFileSourceSPIFFS : public AudioFileSourceFS
{
public:

2
src/AudioFileSourceSPIRAMBuffer.cpp
View File

@@ -105,7 +105,7 @@ uint32_t AudioFileSourceSPIRAMBuffer::read(void *data, uint32_t len)
}
// Read up to the entire buffer from RAM
uint32_t toReadFromBuffer = std::min(len, writePtr - readPtr);
uint32_t toReadFromBuffer = std::min(len, (uint32_t)(writePtr - readPtr));
uint8_t *ptr = reinterpret_cast<uint8_t*>(data);
if (toReadFromBuffer > 0) {
#ifdef FAKERAM

5
src/AudioGeneratorFLAC.cpp
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@@ -62,6 +62,9 @@ bool AudioGeneratorFLAC::begin(AudioFileSource *source, AudioOutput *output)
output->begin();
running = true;
lastSample[0] = 0;
lastSample[1] = 0;
channels = 0;
return true;
}
@@ -71,7 +74,7 @@ bool AudioGeneratorFLAC::loop()
if (!running) goto done;
if (!output->ConsumeSample(lastSample)) goto done; // Try and send last buffered sample
if (channels && !output->ConsumeSample(lastSample)) goto done; // Try and send last buffered sample
do {
if (buffPtr == buffLen) {

14
src/AudioGeneratorMIDI.cpp
View File

@@ -1,6 +1,6 @@
/*
AudioGeneratorMIDI
Audio output generator that plays MIDI files using a SF2 SoundFont
Audio output generator that plays MIDI files using an SF2 SoundFont
Copyright (C) 2017 Earle F. Philhower, III
@@ -58,6 +58,10 @@
#include "AudioGeneratorMIDI.h"
#if __GNUC__ == 8
// Do not build, GCC8 has a compiler bug
#else // __GNUC__ == 8
#pragma GCC optimize ("O3")
#define TSF_NO_STDIO
@@ -172,7 +176,7 @@ unsigned int AudioGeneratorMIDI::buffer_int32 (int offset) {
unsigned long AudioGeneratorMIDI::get_varlen (int *ptr) {
/* Get a 1-4 byte variable-length value and adjust the pointer past it.
These are a succession of 7-bit values with a MSB bit of zero marking the end */
These are a succession of 7-bit values with an MSB bit of zero marking the end */
unsigned long val;
int i, byte;
@@ -331,7 +335,7 @@ note_off:
}
// Open file, parse headers, get ready tio process MIDI
// Open file, parse headers, get ready to process MIDI
void AudioGeneratorMIDI::PrepareMIDI(AudioFileSource *src)
{
MakeStreamFromAFS(src, &afsMIDI);
@@ -364,7 +368,7 @@ void AudioGeneratorMIDI::PrepareMIDI(AudioFileSource *src)
int AudioGeneratorMIDI::PlayMIDI()
{
/* Continue processing all tracks, in an order based on the simulated time.
This is not unlike multiway merging used for tape sorting algoritms in the 50's! */
This is not unlike multiway merging used for tape sorting algorithms in the 50's! */
do { /* while there are still track notes to process */
static struct track_status *trk;
@@ -637,3 +641,5 @@ void AudioGeneratorMIDI::MakeStreamFromAFS(AudioFileSource *src, tsf_stream *afs
afs->size = &afs_size;
}
#endif //__GNUC__ == 8

5
src/AudioGeneratorMIDI.h
View File

@@ -21,6 +21,10 @@
#ifndef _AUDIOGENERATORMIDI_H
#define _AUDIOGENERATORMIDI_H
#if __GNUC__ == 8
// Do not build, GCC8 has a compiler bug
#else // __GNUC__ == 8
#include "AudioGenerator.h"
#define TSF_NO_STDIO
@@ -176,6 +180,7 @@ class AudioGeneratorMIDI : public AudioGenerator
short samplesRendered[256];
};
#endif //__GNUC__ == 8
#endif

84
src/AudioGeneratorMOD.cpp
View File

@@ -69,6 +69,11 @@ bool AudioGeneratorMOD::stop()
free(FatBuffer.channels[i]);
FatBuffer.channels[i] = NULL;
}
if(running || ((file != NULL) && (file->isOpen() == true))) {
output->flush(); //flush I2S output buffer, if the player was actually running before.
}
if (file) file->close();
running = false;
output->stop();
@@ -124,7 +129,7 @@ bool AudioGeneratorMOD::begin(AudioFileSource *source, AudioOutput *out)
UpdateAmiga();
for (int i = 0; i < CHANNELS; i++) {
FatBuffer.channels[i] = reinterpret_cast<uint8_t*>(malloc(fatBufferSize));
FatBuffer.channels[i] = reinterpret_cast<uint8_t*>(calloc(fatBufferSize, 1));
if (!FatBuffer.channels[i]) {
stop();
return false;
@@ -232,6 +237,11 @@ bool AudioGeneratorMOD::LoadHeader()
Mod.numberOfChannels = (temp[0] - '0') * 10 + temp[1] - '0';
else
Mod.numberOfChannels = 4;
if (Mod.numberOfChannels > CHANNELS) {
audioLogger->printf("\nAudioGeneratorMOD::LoadHeader abort - too many channels (configured: %d, needed: %d)\n", CHANNELS, Mod.numberOfChannels);
return(false);
}
return true;
}
@@ -417,7 +427,7 @@ bool AudioGeneratorMOD::ProcessRow()
if (sampleNumber) {
Player.lastSampleNumber[channel] = sampleNumber - 1;
if (!(effectParameter == 0xE && effectParameterX == NOTEDELAY))
if (!(effectNumber == 0xE && effectParameterX == NOTEDELAY))
Player.volume[channel] = Mod.samples[Player.lastSampleNumber[channel]].volume;
}
@@ -565,7 +575,7 @@ bool AudioGeneratorMOD::ProcessRow()
Mixer.channelFrequency[channel] = Player.amiga / Player.lastAmigaPeriod[channel];
if (note != NONOTE)
Mixer.channelSampleOffset[channel] = sampleOffset << DIVIDER;
Mixer.channelSampleOffset[channel] = sampleOffset << FIXED_DIVIDER;
if (sampleNumber)
Mixer.channelSampleNumber[channel] = Player.lastSampleNumber[channel];
@@ -735,13 +745,14 @@ bool AudioGeneratorMOD::RunPlayer()
void AudioGeneratorMOD::GetSample(int16_t sample[2])
{
int16_t sumL;
int16_t sumR;
int32_t sumL;
int32_t sumR;
uint8_t channel;
uint32_t samplePointer;
int8_t current;
int8_t next;
int16_t out;
int32_t out32;
if (!running) return;
@@ -757,12 +768,12 @@ void AudioGeneratorMOD::GetSample(int16_t sample[2])
if (!Mixer.channelVolume[channel]) continue;
samplePointer = Mixer.sampleBegin[Mixer.channelSampleNumber[channel]] +
(Mixer.channelSampleOffset[channel] >> DIVIDER);
(Mixer.channelSampleOffset[channel] >> FIXED_DIVIDER);
if (Mixer.sampleLoopLength[Mixer.channelSampleNumber[channel]]) {
if (samplePointer >= Mixer.sampleLoopEnd[Mixer.channelSampleNumber[channel]]) {
Mixer.channelSampleOffset[channel] -= Mixer.sampleLoopLength[Mixer.channelSampleNumber[channel]] << DIVIDER;
Mixer.channelSampleOffset[channel] -= Mixer.sampleLoopLength[Mixer.channelSampleNumber[channel]] << FIXED_DIVIDER;
samplePointer -= Mixer.sampleLoopLength[Mixer.channelSampleNumber[channel]];
}
@@ -779,8 +790,8 @@ void AudioGeneratorMOD::GetSample(int16_t sample[2])
samplePointer >= FatBuffer.samplePointer[channel] + fatBufferSize - 1 ||
Mixer.channelSampleNumber[channel] != FatBuffer.channelSampleNumber[channel]) {
uint16_t toRead = Mixer.sampleEnd[Mixer.channelSampleNumber[channel]] - samplePointer + 1;
if (toRead > fatBufferSize) toRead = fatBufferSize;
uint32_t toRead = Mixer.sampleEnd[Mixer.channelSampleNumber[channel]] - samplePointer + 1;
if (toRead > (uint32_t)fatBufferSize) toRead = fatBufferSize;
if (!file->seek(samplePointer, SEEK_SET)) {
stop();
@@ -797,30 +808,55 @@ void AudioGeneratorMOD::GetSample(int16_t sample[2])
current = FatBuffer.channels[channel][(samplePointer - FatBuffer.samplePointer[channel]) /*& (FATBUFFERSIZE - 1)*/];
next = FatBuffer.channels[channel][(samplePointer + 1 - FatBuffer.samplePointer[channel]) /*& (FATBUFFERSIZE - 1)*/];
// preserve a few more bits from sample interpolation, by upscaling input values.
// This does (slightly) reduce quantization noise in higher frequencies, typically above 8kHz.
// Actually we could could even gain more bits, I was just not sure if more bits would cause overflows in other conputations.
int16_t current16 = (int16_t) current << 2;
int16_t next16 = (int16_t) next << 2;
out = current16;
out = current;
// Integer linear interpolation - only works correctly in 16bit
out += (next16 - current16) * (Mixer.channelSampleOffset[channel] & ((1 << FIXED_DIVIDER) - 1)) >> FIXED_DIVIDER;
// Integer linear interpolation
out += (next - current) * (Mixer.channelSampleOffset[channel] & ((1 << DIVIDER) - 1)) >> DIVIDER;
// Upscale to BITDEPTH
out <<= BITDEPTH - 8;
// Upscale to BITDEPTH, considering the we already gained two bits in the previous step
out32 = (int32_t)out << (BITDEPTH - 10);
// Channel volume
out = out * Mixer.channelVolume[channel] >> 6;
out32 = out32 * Mixer.channelVolume[channel] >> 6;
// Channel panning
sumL += out * min(128 - Mixer.channelPanning[channel], 64) >> 6;
sumR += out * min(Mixer.channelPanning[channel], 64) >> 6;
sumL += out32 * min(128 - Mixer.channelPanning[channel], 64) >> 6;
sumR += out32 * min(Mixer.channelPanning[channel], 64) >> 6;
}
// Downscale to BITDEPTH
sumL /= Mod.numberOfChannels;
sumR /= Mod.numberOfChannels;
// Downscale to BITDEPTH - a bit faster because the compiler can replaced division by constants with proper "right shift" + correct handling of sign bit
if (Mod.numberOfChannels <= 4) {
// up to 4 channels
sumL /= 4;
sumR /= 4;
} else {
if (Mod.numberOfChannels <= 6) {
// 5 or 6 channels - pre-multiply be 1.5, then divide by 8 -> same as division by 6
sumL = (sumL + (sumL/2)) / 8;
sumR = (sumR + (sumR/2)) / 8;
} else {
// 7,8, or more channels
sumL /= 8;
sumR /= 8;
}
}
// Fill the sound buffer with unsigned values
sample[AudioOutput::LEFTCHANNEL] = sumL + (1 << (BITDEPTH - 1));
sample[AudioOutput::RIGHTCHANNEL] = sumR + (1 << (BITDEPTH - 1));
// clip samples to 16bit (with saturation in case of overflow)
if(sumL <= INT16_MIN) sumL = INT16_MIN;
else if (sumL >= INT16_MAX) sumL = INT16_MAX;
if(sumR <= INT16_MIN) sumR = INT16_MIN;
else if (sumR >= INT16_MAX) sumR = INT16_MAX;
// Fill the sound buffer with signed values
sample[AudioOutput::LEFTCHANNEL] = sumL;
sample[AudioOutput::RIGHTCHANNEL] = sumR;
}
bool AudioGeneratorMOD::LoadMOD()

14
src/AudioGeneratorMOD.h
View File

@@ -53,19 +53,25 @@ class AudioGeneratorMOD : public AudioGenerator
protected:
int mixerTick;
enum {BITDEPTH = 15};
enum {BITDEPTH = 16};
int sampleRate;
int fatBufferSize; //(6*1024) // File system buffers per-CHANNEL (i.e. total mem required is 4 * FATBUFFERSIZE)
enum {DIVIDER = 10}; // Fixed-point mantissa used for integer arithmetic
enum {FIXED_DIVIDER = 10}; // Fixed-point mantissa used for integer arithmetic
int stereoSeparation; //STEREOSEPARATION = 32; // 0 (max) to 64 (mono)
bool usePAL;
// Hz = 7093789 / (amigaPeriod * 2) for PAL
// Hz = 7159091 / (amigaPeriod * 2) for NTSC
int AMIGA;
void UpdateAmiga() { AMIGA = ((usePAL?7159091:7093789) / 2 / sampleRate << DIVIDER); }
void UpdateAmiga() { AMIGA = ((usePAL?7159091:7093789) / 2 / sampleRate << FIXED_DIVIDER); }
#ifdef ESP8266 // Not sure if C3/C2 have RAM constraints, maybe add them here?
// support max 4 channels
enum {ROWS = 64, SAMPLES = 31, CHANNELS = 4, NONOTE = 0xFFFF, NONOTE8 = 0xff };
#else
// support max 8 channels
enum {ROWS = 64, SAMPLES = 31, CHANNELS = 8, NONOTE = 0xFFFF, NONOTE8 = 0xff };
#endif
typedef struct Sample {
uint16_t length;

5
src/AudioGeneratorRTTTL.cpp
View File

@@ -41,7 +41,10 @@ AudioGeneratorRTTTL::~AudioGeneratorRTTTL()
bool AudioGeneratorRTTTL::stop()
{
if (!running) return true;
if (!file || !output)
{
return false;
}
running = false;
output->stop();
return file->close();

479
src/AudioOutputI2S.cpp
View File

@@ -1,8 +1,7 @@
/*
AudioOutputI2S
Base class for I2S interface port
Copyright (C) 2017 Earle F. Philhower, III
This program is free software: you can redistribute it and/or modify
@@ -22,7 +21,9 @@
#include <Arduino.h>
#ifdef ESP32
#include "driver/i2s.h"
#elif defined(ARDUINO_ARCH_RP2040) || defined(ESP8266)
#elif defined(ARDUINO_ARCH_RP2040) || ARDUINO_ESP8266_MAJOR >= 3
#include <I2S.h>
#elif ARDUINO_ESP8266_MAJOR < 3
#include <i2s.h>
#endif
#include "AudioOutputI2S.h"
@@ -41,6 +42,7 @@ AudioOutputI2S::AudioOutputI2S(int port, int output_mode, int dma_buf_count, int
//set defaults
mono = false;
lsb_justified = false;
bps = 16;
channels = 2;
hertz = 44100;
@@ -50,37 +52,6 @@ AudioOutputI2S::AudioOutputI2S(int port, int output_mode, int dma_buf_count, int
SetGain(1.0);
}
bool AudioOutputI2S::SetPinout()
{
#ifdef ESP32
if (output_mode == INTERNAL_DAC || output_mode == INTERNAL_PDM)
return false; // Not allowed
i2s_pin_config_t pins = {
.bck_io_num = bclkPin,
.ws_io_num = wclkPin,
.data_out_num = doutPin,
.data_in_num = I2S_PIN_NO_CHANGE};
i2s_set_pin((i2s_port_t)portNo, &pins);
return true;
#else
(void)bclkPin;
(void)wclkPin;
(void)doutPin;
return false;
#endif
}
bool AudioOutputI2S::SetPinout(int bclk, int wclk, int dout)
{
bclkPin = bclk;
wclkPin = wclk;
doutPin = dout;
if (i2sOn)
return SetPinout();
return true;
}
#elif defined(ARDUINO_ARCH_RP2040)
AudioOutputI2S::AudioOutputI2S(long sampleRate, pin_size_t sck, pin_size_t data) {
i2sOn = false;
@@ -110,6 +81,38 @@ AudioOutputI2S::~AudioOutputI2S()
i2sOn = false;
}
bool AudioOutputI2S::SetPinout()
{
#ifdef ESP32
if (output_mode == INTERNAL_DAC || output_mode == INTERNAL_PDM)
return false; // Not allowed
i2s_pin_config_t pins = {
.mck_io_num = 0, // Unused
.bck_io_num = bclkPin,
.ws_io_num = wclkPin,
.data_out_num = doutPin,
.data_in_num = I2S_PIN_NO_CHANGE};
i2s_set_pin((i2s_port_t)portNo, &pins);
return true;
#else
(void)bclkPin;
(void)wclkPin;
(void)doutPin;
return false;
#endif
}
bool AudioOutputI2S::SetPinout(int bclk, int wclk, int dout)
{
bclkPin = bclk;
wclkPin = wclk;
doutPin = dout;
if (i2sOn)
return SetPinout();
return true;
}
bool AudioOutputI2S::SetRate(int hz)
{
// TODO - have a list of allowable rates from constructor, check them
@@ -121,7 +124,7 @@ bool AudioOutputI2S::SetRate(int hz)
#elif defined(ESP8266)
i2s_set_rate(AdjustI2SRate(hz));
#elif defined(ARDUINO_ARCH_RP2040)
I2S.setFrequency(hz);
i2s.setFrequency(hz);
#endif
}
return true;
@@ -147,6 +150,12 @@ bool AudioOutputI2S::SetOutputModeMono(bool mono)
return true;
}
bool AudioOutputI2S::SetLsbJustified(bool lsbJustified)
{
this->lsb_justified = lsbJustified;
return true;
}
bool AudioOutputI2S::begin(bool txDAC)
{
#ifdef ESP32
@@ -167,17 +176,45 @@ bool AudioOutputI2S::begin(bool txDAC)
i2s_mode_t mode = (i2s_mode_t)(I2S_MODE_MASTER | I2S_MODE_TX);
if (output_mode == INTERNAL_DAC)
{
#if CONFIG_IDF_TARGET_ESP32
mode = (i2s_mode_t)(mode | I2S_MODE_DAC_BUILT_IN);
#else
return false;
#endif
}
else if (output_mode == INTERNAL_PDM)
{
#if CONFIG_IDF_TARGET_ESP32
mode = (i2s_mode_t)(mode | I2S_MODE_PDM);
#else
return false;
#endif
}
i2s_comm_format_t comm_fmt = (i2s_comm_format_t)(I2S_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_MSB);
i2s_comm_format_t comm_fmt;
if (output_mode == INTERNAL_DAC)
{
comm_fmt = (i2s_comm_format_t)I2S_COMM_FORMAT_I2S_MSB;
#if ESP_IDF_VERSION >= ESP_IDF_VERSION_VAL(4, 2, 0)
comm_fmt = (i2s_comm_format_t) I2S_COMM_FORMAT_STAND_MSB;
#else
comm_fmt = (i2s_comm_format_t) I2S_COMM_FORMAT_I2S_MSB;
#endif
}
else if (lsb_justified)
{
#if ESP_IDF_VERSION >= ESP_IDF_VERSION_VAL(4, 2, 0)
comm_fmt = (i2s_comm_format_t) I2S_COMM_FORMAT_STAND_MSB;
#else
comm_fmt = (i2s_comm_format_t) (I2S_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_LSB);
#endif
}
else
{
#if ESP_IDF_VERSION >= ESP_IDF_VERSION_VAL(4, 2, 0)
comm_fmt = (i2s_comm_format_t) (I2S_COMM_FORMAT_STAND_I2S);
#else
comm_fmt = (i2s_comm_format_t) (I2S_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_MSB);
#endif
}
i2s_config_t i2s_config_dac = {
@@ -188,8 +225,12 @@ bool AudioOutputI2S::begin(bool txDAC)
.communication_format = comm_fmt,
.intr_alloc_flags = ESP_INTR_FLAG_LEVEL1, // lowest interrupt priority
.dma_buf_count = dma_buf_count,
.dma_buf_len = 64,
.use_apll = use_apll // Use audio PLL
.dma_buf_len = 128,
.use_apll = use_apll, // Use audio PLL
.tx_desc_auto_clear = true, // Silence on underflow
.fixed_mclk = 0, // Unused
.mclk_multiple = I2S_MCLK_MULTIPLE_DEFAULT, // Unused
.bits_per_chan = I2S_BITS_PER_CHAN_DEFAULT // Use bits per sample
};
audioLogger->printf("+%d %p\n", portNo, &i2s_config_dac);
if (i2s_driver_install((i2s_port_t)portNo, &i2s_config_dac, 0, NULL) != ESP_OK)
@@ -198,8 +239,12 @@ bool AudioOutputI2S::begin(bool txDAC)
}
if (output_mode == INTERNAL_DAC || output_mode == INTERNAL_PDM)
{
#if CONFIG_IDF_TARGET_ESP32
i2s_set_pin((i2s_port_t)portNo, NULL);
i2s_set_dac_mode(I2S_DAC_CHANNEL_BOTH_EN);
#else
return false;
#endif
}
else
{
@@ -230,9 +275,9 @@ bool AudioOutputI2S::begin(bool txDAC)
#elif defined(ARDUINO_ARCH_RP2040)
(void)txDAC;
if (!i2sOn) {
I2S.setBCLK(bclkPin);
I2S.setDOUT(doutPin);
I2S.begin(hertz);
i2s.setBCLK(bclkPin);
i2s.setDATA(doutPin);
i2s.begin(hertz);
}
#endif
i2sOn = true;
@@ -264,22 +309,24 @@ bool AudioOutputI2S::ConsumeSample(int16_t sample[2])
{
int16_t l = Amplify(ms[LEFTCHANNEL]) + 0x8000;
int16_t r = Amplify(ms[RIGHTCHANNEL]) + 0x8000;
s32 = (r << 16) | (l & 0xffff);
s32 = ((r & 0xffff) << 16) | (l & 0xffff);
}
else
{
s32 = ((Amplify(ms[RIGHTCHANNEL])) << 16) | (Amplify(ms[LEFTCHANNEL]) & 0xffff);
}
// Deprecated. Use i2s_write
//"i2s_write_bytes" has been removed in the ESP32 Arduino 2.0.0, use "i2s_write" instead.
// return i2s_write_bytes((i2s_port_t)portNo, (const char *)&s32, sizeof(uint32_t), 0);
size_t bytes_written;
i2s_write((i2s_port_t)portNo, (const char*)&s32, sizeof(uint32_t), &bytes_written, 0);
return bytes_written;
size_t i2s_bytes_written;
i2s_write((i2s_port_t)portNo, (const char*)&s32, sizeof(uint32_t), &i2s_bytes_written, 0);
return i2s_bytes_written;
#elif defined(ESP8266)
uint32_t s32 = ((Amplify(ms[RIGHTCHANNEL])) << 16) | (Amplify(ms[LEFTCHANNEL]) & 0xffff);
return i2s_write_sample_nb(s32); // If we can't store it, return false. OTW true
#elif defined(ARDUINO_ARCH_RP2040)
return !!I2S.write((void*)ms, 4);
uint32_t s32 = ((Amplify(ms[RIGHTCHANNEL])) << 16) | (Amplify(ms[LEFTCHANNEL]) & 0xffff);
return !!i2s.write((int32_t)s32, false);
#endif
}
@@ -287,7 +334,7 @@ void AudioOutputI2S::flush()
{
#ifdef ESP32
// makes sure that all stored DMA samples are consumed / played
int buffersize = 64 * this->dma_buf_count;
int buffersize = 128 * this->dma_buf_count;
int16_t samples[2] = {0x0, 0x0};
for (int i = 0; i < buffersize; i++)
{
@@ -297,7 +344,7 @@ void AudioOutputI2S::flush()
}
}
#elif defined(ARDUINO_ARCH_RP2040)
I2S.flush();
i2s.flush();
#endif
}
@@ -309,338 +356,8 @@ bool AudioOutputI2S::stop()
#ifdef ESP32
i2s_zero_dma_buffer((i2s_port_t)portNo);
#elif defined(ARDUINO_ARCH_RP2040)
I2S.end();
i2s.end();
#endif
i2sOn = false;
return true;
}
// /*
// AudioOutputI2S
// Base class for I2S interface port
// Copyright (C) 2017 Earle F. Philhower, III
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, either version 3 of the License, or
// (at your option) any later version.
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
// You should have received a copy of the GNU General Public License
// along with this program. If not, see <http://www.gnu.org/licenses/>.
// */
// #include <Arduino.h>
// #ifdef ESP32
// #include "driver/i2s.h"
// #if defined ESP_ARDUINO_VERSION_VAL
// #if (ESP_ARDUINO_VERSION >= ESP_ARDUINO_VERSION_VAL(2, 0, 0)
// #include "soc/soc_caps.h"
// #define ESP_V2
// #define I2S_FORMAT (I2S_COMM_FORMAT_STAND_I2S)
// #endif
// #else
// #define I2S_FORMAT (I2S_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_MSB)
// #endif
// #elif defined(ARDUINO_ARCH_RP2040) || defined(ESP8266)
// #include <i2s.h>
// #endif
// #include "AudioOutputI2S.h"
// #if defined(ESP32) || defined(ESP8266)
// AudioOutputI2S::AudioOutputI2S(int port, int output_mode, int dma_buf_count, int use_apll)
// {
// this->portNo = port;
// this->i2sOn = false;
// this->dma_buf_count = dma_buf_count;
// if (output_mode != EXTERNAL_I2S && output_mode != INTERNAL_DAC && output_mode != INTERNAL_PDM) {
// output_mode = EXTERNAL_I2S;
// }
// this->output_mode = output_mode;
// this->use_apll = use_apll;
// //set defaults
// mono = false;
// bps = 16;
// channels = 2;
// hertz = 44100;
// bclkPin = 26;
// wclkPin = 25;
// doutPin = 22;
// SetGain(1.0);
// }
// bool AudioOutputI2S::SetPinout()
// {
// #ifdef ESP32
// if (output_mode == INTERNAL_DAC || output_mode == INTERNAL_PDM)
// return false; // Not allowed
// i2s_pin_config_t pins = {
// .bck_io_num = bclkPin,
// .ws_io_num = wclkPin,
// .data_out_num = doutPin,
// .data_in_num = I2S_PIN_NO_CHANGE};
// i2s_set_pin((i2s_port_t)portNo, &pins);
// return true;
// #else
// (void)bclkPin;
// (void)wclkPin;
// (void)doutPin;
// return false;
// #endif
// }
// bool AudioOutputI2S::SetPinout(int bclk, int wclk, int dout)
// {
// bclkPin = bclk;
// wclkPin = wclk;
// doutPin = dout;
// if (i2sOn)
// return SetPinout();
// return true;
// }
// #elif defined(ARDUINO_ARCH_RP2040)
// AudioOutputI2S::AudioOutputI2S(long sampleRate, pin_size_t sck, pin_size_t data) {
// i2sOn = false;
// mono = false;
// bps = 16;
// channels = 2;
// hertz = sampleRate;
// bclkPin = sck;
// doutPin = data;
// SetGain(1.0);
// }
// #endif
// AudioOutputI2S::~AudioOutputI2S()
// {
// #ifdef ESP32
// if (i2sOn) {
// audioLogger->printf("UNINSTALL I2S\n");
// i2s_driver_uninstall((i2s_port_t)portNo); //stop & destroy i2s driver
// }
// #elif defined(ESP8266)
// if (i2sOn)
// i2s_end();
// #elif defined(ARDUINO_ARCH_RP2040)
// stop();
// #endif
// i2sOn = false;
// }
// bool AudioOutputI2S::SetRate(int hz)
// {
// // TODO - have a list of allowable rates from constructor, check them
// this->hertz = hz;
// if (i2sOn)
// {
// #ifdef ESP32
// i2s_set_sample_rates((i2s_port_t)portNo, AdjustI2SRate(hz));
// #elif defined(ESP8266)
// i2s_set_rate(AdjustI2SRate(hz));
// #elif defined(ARDUINO_ARCH_RP2040)
// I2S.setFrequency(hz);
// #endif
// }
// return true;
// }
// bool AudioOutputI2S::SetBitsPerSample(int bits)
// {
// if ( (bits != 16) && (bits != 8) ) return false;
// this->bps = bits;
// return true;
// }
// bool AudioOutputI2S::SetChannels(int channels)
// {
// if ( (channels < 1) || (channels > 2) ) return false;
// this->channels = channels;
// return true;
// }
// bool AudioOutputI2S::SetOutputModeMono(bool mono)
// {
// this->mono = mono;
// return true;
// }
// bool AudioOutputI2S::begin(bool txDAC)
// {
// #ifdef ESP32
// if (!i2sOn)
// {
// if (use_apll == APLL_AUTO)
// {
// // don't use audio pll on buggy rev0 chips
// use_apll = APLL_DISABLE;
// esp_chip_info_t out_info;
// esp_chip_info(&out_info);
// if (out_info.revision > 0)
// {
// use_apll = APLL_ENABLE;
// }
// }
// i2s_mode_t mode = (i2s_mode_t)(I2S_MODE_MASTER | I2S_MODE_TX);
// if (output_mode == INTERNAL_DAC)
// {
// #if SOC_I2S_SUPPORTS_DAC
// mode = (i2s_mode_t)(mode | I2S_MODE_DAC_BUILT_IN);
// #endif
// }
// else if (output_mode == INTERNAL_PDM)
// {
// mode = (i2s_mode_t)(mode | I2S_MODE_PDM);
// }
// i2s_comm_format_t comm_fmt = (i2s_comm_format_t)(I2S_FORMAT);
// if (output_mode == INTERNAL_DAC)
// {
// #if SOC_I2S_SUPPORTS_DAC
// comm_fmt = (i2s_comm_format_t)I2S_COMM_FORMAT_I2S_MSB;
// #endif
// }
// i2s_config_t i2s_config_dac = {
// .mode = mode,
// .sample_rate = 44100,
// .bits_per_sample = I2S_BITS_PER_SAMPLE_16BIT,
// .channel_format = I2S_CHANNEL_FMT_RIGHT_LEFT,
// .communication_format = comm_fmt,
// .intr_alloc_flags = ESP_INTR_FLAG_LEVEL1, // lowest interrupt priority
// .dma_buf_count = dma_buf_count,
// .dma_buf_len = 64,
// .use_apll = use_apll // Use audio PLL
// };
// audioLogger->printf("+%d %p\n", portNo, &i2s_config_dac);
// if (i2s_driver_install((i2s_port_t)portNo, &i2s_config_dac, 0, NULL) != ESP_OK)
// {
// audioLogger->println("ERROR: Unable to install I2S drives\n");
// }
// if (output_mode == INTERNAL_DAC || output_mode == INTERNAL_PDM)
// {
// #if SOC_I2S_SUPPORTS_DAC
// i2s_set_pin((i2s_port_t)portNo, NULL);
// i2s_set_dac_mode(I2S_DAC_CHANNEL_BOTH_EN);
// #endif
// }
// else
// {
// SetPinout();
// }
// i2s_zero_dma_buffer((i2s_port_t)portNo);
// }
// #elif defined(ESP8266)
// (void)dma_buf_count;
// (void)use_apll;
// if (!i2sOn)
// {
// orig_bck = READ_PERI_REG(PERIPHS_IO_MUX_MTDO_U);
// orig_ws = READ_PERI_REG(PERIPHS_IO_MUX_GPIO2_U);
// #ifdef I2S_HAS_BEGIN_RXTX_DRIVE_CLOCKS
// if (!i2s_rxtxdrive_begin(false, true, false, txDAC)) {
// return false;
// }
// #else
// if (!i2s_rxtx_begin(false, true)) {
// return false;
// }
// if (!txDAC) {
// audioLogger->printf_P(PSTR("I2SNoDAC: esp8266 arduino core should be upgraded to avoid conflicts with SPI\n"));
// }
// #endif
// }
// #elif defined(ARDUINO_ARCH_RP2040)
// (void)txDAC;
// if (!i2sOn) {
// I2S.setBCLK(bclkPin);
// I2S.setDOUT(doutPin);
// I2S.begin(hertz);
// }
// #endif
// i2sOn = true;
// SetRate(hertz); // Default
// return true;
// }
// bool AudioOutputI2S::ConsumeSample(int16_t sample[2])
// {
// //return if we haven't called ::begin yet
// if (!i2sOn)
// return false;
// int16_t ms[2];
// ms[0] = sample[0];
// ms[1] = sample[1];
// MakeSampleStereo16( ms );
// if (this->mono) {
// // Average the two samples and overwrite
// int32_t ttl = ms[LEFTCHANNEL] + ms[RIGHTCHANNEL];
// ms[LEFTCHANNEL] = ms[RIGHTCHANNEL] = (ttl>>1) & 0xffff;
// }
// #ifdef ESP32
// uint32_t s32;
// if (output_mode == INTERNAL_DAC)
// {
// int16_t l = Amplify(ms[LEFTCHANNEL]) + 0x8000;
// int16_t r = Amplify(ms[RIGHTCHANNEL]) + 0x8000;
// s32 = (r << 16) | (l & 0xffff);
// }
// else
// {
// s32 = ((Amplify(ms[RIGHTCHANNEL])) << 16) | (Amplify(ms[LEFTCHANNEL]) & 0xffff);
// }
// // Deprecated. Use i2s_write
// // return i2s_write_bytes((i2s_port_t)portNo, (const char *)&s32, sizeof(uint32_t), 0);
// size_t bytes_written;
// i2s_write((i2s_port_t)portNo, (const char*)&s32, sizeof(uint32_t), &bytes_written, 0);
// return bytes_written;
// #elif defined(ESP8266)
// uint32_t s32 = ((Amplify(ms[RIGHTCHANNEL])) << 16) | (Amplify(ms[LEFTCHANNEL]) & 0xffff);
// return i2s_write_sample_nb(s32); // If we can't store it, return false. OTW true
// #elif defined(ARDUINO_ARCH_RP2040)
// return !!I2S.write((void*)ms, 4);
// #endif
// }
// void AudioOutputI2S::flush()
// {
// #ifdef ESP32
// // makes sure that all stored DMA samples are consumed / played
// int buffersize = 64 * this->dma_buf_count;
// int16_t samples[2] = {0x0, 0x0};
// for (int i = 0; i < buffersize; i++)
// {
// while (!ConsumeSample(samples))
// {
// delay(10);
// }
// }
// #elif defined(ARDUINO_ARCH_RP2040)
// I2S.flush();
// #endif
// }
// bool AudioOutputI2S::stop()
// {
// if (!i2sOn)
// return false;
// #ifdef ESP32
// i2s_zero_dma_buffer((i2s_port_t)portNo);
// #elif defined(ARDUINO_ARCH_RP2040)
// I2S.end();
// #endif
// i2sOn = false;
// return true;
// }

13
src/AudioOutputI2S.h
View File

@@ -22,17 +22,22 @@
#include "AudioOutput.h"
#if defined(ARDUINO_ARCH_RP2040)
#include <Arduino.h>
#include <I2S.h>
#endif
class AudioOutputI2S : public AudioOutput
{
public:
#if defined(ESP32) || defined(ESP8266)
AudioOutputI2S(int port=0, int output_mode=EXTERNAL_I2S, int dma_buf_count = 8, int use_apll=APLL_DISABLE);
bool SetPinout(int bclkPin, int wclkPin, int doutPin);
enum : int { APLL_AUTO = -1, APLL_ENABLE = 1, APLL_DISABLE = 0 };
enum : int { EXTERNAL_I2S = 0, INTERNAL_DAC = 1, INTERNAL_PDM = 2 };
#elif defined(ARDUINO_ARCH_RP2040)
AudioOutputI2S(long sampleRate = 44100, pin_size_t sck = 26, pin_size_t data = 28);
#endif
bool SetPinout(int bclkPin, int wclkPin, int doutPin);
virtual ~AudioOutputI2S() override;
virtual bool SetRate(int hz) override;
virtual bool SetBitsPerSample(int bits) override;
@@ -44,6 +49,7 @@ class AudioOutputI2S : public AudioOutput
bool begin(bool txDAC);
bool SetOutputModeMono(bool mono); // Force mono output no matter the input
bool SetLsbJustified(bool lsbJustified); // Allow supporting non-I2S chips, e.g. PT8211
protected:
bool SetPinout();
@@ -51,6 +57,7 @@ class AudioOutputI2S : public AudioOutput
uint8_t portNo;
int output_mode;
bool mono;
int lsb_justified;
bool i2sOn;
int dma_buf_count;
int use_apll;
@@ -61,4 +68,8 @@ class AudioOutputI2S : public AudioOutput
uint8_t bclkPin;
uint8_t wclkPin;
uint8_t doutPin;
#if defined(ARDUINO_ARCH_RP2040)
I2S i2s;
#endif
};

46
src/AudioOutputI2SNoDAC.cpp
View File

@@ -1,7 +1,7 @@
/*
AudioOutputI2SNoDAC
Audio player using SW delta-sigma to generate "analog" on I2S data
Copyright (C) 2017 Earle F. Philhower, III
This program is free software: you can redistribute it and/or modify
@@ -21,12 +21,33 @@
#include <Arduino.h>
#ifdef ESP32
#include "driver/i2s.h"
#elif defined(ARDUINO_ARCH_RP2040) || defined(ESP8266)
#elif defined(ARDUINO_ARCH_RP2040) || ARDUINO_ESP8266_MAJOR >= 3
#include <I2S.h>
#elif ARDUINO_ESP8266_MAJOR < 3
#include <i2s.h>
#endif
#include "AudioOutputI2SNoDAC.h"
#if defined(ARDUINO_ARCH_RP2040)
//
// Create an alternate constructor for the RP2040. The AudioOutputI2S has an alternate
// constructor for the RP2040, so the code was passing port to the sampleRate and false to sck.
//
// AudioOutputI2S(long sampleRate = 44100, pin_size_t sck = 26, pin_size_t data = 28);
//
// So this new constructor adds the ability to pass both port and sck to the underlying class, but
// uses the same defaults in the AudioOutputI2S constructor.
//
AudioOutputI2SNoDAC::AudioOutputI2SNoDAC(int port, int sck) : AudioOutputI2S(44100, sck, port)
{
SetOversampling(32);
lastSamp = 0;
cumErr = 0;
}
#else
AudioOutputI2SNoDAC::AudioOutputI2SNoDAC(int port) : AudioOutputI2S(port, false)
{
SetOversampling(32);
@@ -36,7 +57,10 @@ AudioOutputI2SNoDAC::AudioOutputI2SNoDAC(int port) : AudioOutputI2S(port, false)
WRITE_PERI_REG(PERIPHS_IO_MUX_MTDO_U, orig_bck);
WRITE_PERI_REG(PERIPHS_IO_MUX_GPIO2_U, orig_ws);
#endif
}
#endif
AudioOutputI2SNoDAC::~AudioOutputI2SNoDAC()
{
@@ -66,7 +90,7 @@ void AudioOutputI2SNoDAC::DeltaSigma(int16_t sample[2], uint32_t dsBuff[8])
for (int j = 0; j < oversample32; j++) {
uint32_t bits = 0; // The bits we convert the sample into, MSB to go on the wire first
for (int i = 32; i > 0; i--) {
bits = bits << 1;
if (cumErr < 0) {
@@ -95,12 +119,11 @@ bool AudioOutputI2SNoDAC::ConsumeSample(int16_t sample[2])
// Either send complete pulse stream or nothing
#ifdef ESP32
// Deprecated. Use i2s_write
// if (!i2s_write_bytes((i2s_port_t)portNo, (const char *)dsBuff, sizeof(uint32_t) * (oversample/32), 0))
size_t bytes_written;
i2s_write((i2s_port_t)portNo, (const char *)dsBuff, sizeof(uint32_t) * (oversample/32), &bytes_written, 0);
if (!bytes_written)
size_t i2s_bytes_written;
i2s_write((i2s_port_t)portNo, (const char *)dsBuff, sizeof(uint32_t) * (oversample/32), &i2s_bytes_written, 0);
if (!i2s_bytes_written){
return false;
}
#elif defined(ESP8266)
if (!i2s_write_sample_nb(dsBuff[0])) return false; // No room at the inn
// At this point we've sent in first of possibly 8 32-bits, need to send
@@ -108,10 +131,9 @@ bool AudioOutputI2SNoDAC::ConsumeSample(int16_t sample[2])
for (int i = 32; i < oversample; i+=32)
i2s_write_sample( dsBuff[i / 32]);
#elif defined(ARDUINO_ARCH_RP2040)
int16_t *p = (int16_t *) dsBuff;
for (int i = 0; i < oversample / 16; i++) {
I2S.write(*(p++));
for (int i = 0; i < oversample / 32; i++) {
i2s.write((int32_t)dsBuff[i], true);
}
#endif
return true;
}
}

9
src/AudioOutputI2SNoDAC.h
View File

@@ -25,7 +25,16 @@
class AudioOutputI2SNoDAC : public AudioOutputI2S
{
public:
//
// Define a different constructor for the RP2040, as this class calls the constructor
// of the AudioOutputI2S which has an alternate constructor for the RP2040
//
#if defined(ARDUINO_ARCH_RP2040)
AudioOutputI2SNoDAC(int port = 28,int sck = 26);
#else
AudioOutputI2SNoDAC(int port = 0);
#endif
virtual ~AudioOutputI2SNoDAC() override;
virtual bool begin() override { return AudioOutputI2S::begin(false); }
virtual bool ConsumeSample(int16_t sample[2]) override;

17
src/AudioOutputSPDIF.cpp
View File

@@ -3,7 +3,7 @@
S/PDIF output via I2S
Needs transciever from CMOS level to either optical or coaxial interface
Needs transceiver from CMOS level to either optical or coaxial interface
See: https://www.epanorama.net/documents/audio/spdif.html
Original idea and sources:
@@ -94,11 +94,19 @@ AudioOutputSPDIF::AudioOutputSPDIF(int dout_pin, int port, int dma_buf_count)
.sample_rate = 88200, // 2 x sampling_rate
.bits_per_sample = I2S_BITS_PER_SAMPLE_32BIT, // 32bit words
.channel_format = I2S_CHANNEL_FMT_RIGHT_LEFT, // Right than left
.communication_format = (i2s_comm_format_t)(I2S_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_MSB),
#if ESP_IDF_VERSION >= ESP_IDF_VERSION_VAL(4, 2, 0)
.communication_format = (i2s_comm_format_t)(I2S_COMM_FORMAT_STAND_I2S),
#else
.communication_format = (i2s_comm_format_t)(I2S_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_MSB),
#endif
.intr_alloc_flags = ESP_INTR_FLAG_LEVEL1, // lowest interrupt priority
.dma_buf_count = dma_buf_count,
.dma_buf_len = DMA_BUF_SIZE_DEFAULT, // bigger buffers, reduces interrupts
.use_apll = true // Audio PLL is needed for low clock jitter
.use_apll = true, // Audio PLL is needed for low clock jitter
.tx_desc_auto_clear = true, // Silence on underflow
.fixed_mclk = 0, // Unused
.mclk_multiple = I2S_MCLK_MULTIPLE_DEFAULT, // Unused
.bits_per_chan = I2S_BITS_PER_CHAN_DEFAULT // Use bits per sample
};
if (i2s_driver_install((i2s_port_t)portNo, &i2s_config_spdif, 0, NULL) != ESP_OK) {
audioLogger->println(F("ERROR: Unable to install I2S drivers"));
@@ -143,6 +151,7 @@ bool AudioOutputSPDIF::SetPinout(int bclk, int wclk, int dout)
{
#if defined(ESP32)
i2s_pin_config_t pins = {
.mck_io_num = 0, // unused
.bck_io_num = bclk,
.ws_io_num = wclk,
.data_out_num = dout,
@@ -265,7 +274,7 @@ bool AudioOutputSPDIF::ConsumeSample(int16_t sample[2])
#if defined(ESP32)
// Assume DMA buffers are multiples of 16 bytes. Either we write all bytes or none.
uint32_t bytes_written;
size_t bytes_written;
esp_err_t ret = i2s_write((i2s_port_t)portNo, (const char*)&buf, 8 * channels, &bytes_written, 0);
// If we didn't write all bytes, return false early and do not increment frame_num
if ((ret != ESP_OK) || (bytes_written != (8 * channels))) return false;

2
src/AudioOutputSPDIF.h
View File

@@ -3,7 +3,7 @@
S/PDIF output via I2S
Needs transciever from CMOS level to either optical or coaxial interface
Needs transceiver from CMOS level to either optical or coaxial interface
See: https://www.epanorama.net/documents/audio/spdif.html
Original idea and sources:

10
src/AudioOutputSPIFFSWAV.cpp
View File

@@ -18,12 +18,18 @@
along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#if !defined(ARDUINO_ARCH_RP2040)
#include <Arduino.h>
#include <FS.h>
#ifdef ESP32
#include "SPIFFS.h"
#endif
// Yes, I know SPIFFS is deprecated
#pragma GCC diagnostic ignored "-Wdeprecated-declarations"
#include "AudioOutputSPIFFSWAV.h"
static const uint8_t wavHeaderTemplate[] PROGMEM = { // Hardcoded simple WAV header with 0xffffffff lengths all around
@@ -110,4 +116,6 @@ bool AudioOutputSPIFFSWAV::stop()
f.close();
return true;
}
#endif

3
src/AudioOutputSPIFFSWAV.h
View File

@@ -21,6 +21,8 @@
#ifndef _AUDIOOUTPUTSPIFFSWAV_H
#define _AUDIOOUTPUTSPIFFSWAV_H
#if !defined(ARDUINO_ARCH_RP2040)
#include <Arduino.h>
#include <FS.h>
@@ -43,3 +45,4 @@ class AudioOutputSPIFFSWAV : public AudioOutput
#endif
#endif

5
src/AudioOutputSTDIO.cpp
View File

@@ -54,6 +54,11 @@ bool AudioOutputSTDIO::begin()
bool AudioOutputSTDIO::ConsumeSample(int16_t sample[2])
{
static int avail = 100;
if (!(--avail)) {
avail = 100;
return false;
}
for (int i=0; i<channels; i++) {
if (bps == 8) {
uint8_t l = sample[i] & 0xff;

2
src/AudioOutputULP.cpp
View File

@@ -18,7 +18,7 @@
along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#ifdef ESP32
#if CONFIG_IDF_TARGET_ESP32 || CONFIG_IDF_TARGET_ESP32S2 || CONFIG_IDF_TARGET_ESP32S3
#include "AudioOutputULP.h"
#include <esp32/ulp.h>

8
src/driver/SinglePinI2SDriver.cpp
View File

@@ -3,7 +3,7 @@
ESP8266Audio I2S Minimal driver
Most of this code is taken and reworked from ESP8266 Arduino core,
which itsef is reworked from Espessif's I2S examples.
which itself is reworked from Espessif's I2S examples.
Original code is licensed under LGPL 2.1 or above
Reasons for rewrite:
@@ -196,7 +196,7 @@ void SinglePinI2SDriver::setDividers(uint8_t div1, uint8_t div2)
// Ensure dividers fit in bit fields
div1 &= I2SBDM;
div2 &= I2SCDM;
// trans master(active low), recv master(active_low), !bits mod(==16 bits/chanel), clear clock dividers
// trans master(active low), recv master(active_low), !bits mod(==16 bits/channel), clear clock dividers
I2SC &= ~(I2STSM | I2SRSM | (I2SBMM << I2SBM) | (I2SBDM << I2SBD) | (I2SCDM << I2SCD));
// I2SRF = Send/recv right channel first
// I2SMR = MSB recv/xmit first
@@ -249,7 +249,7 @@ void SinglePinI2SDriver::startI2S()
I2SFC |= I2SDE; // Enable DMA
// I2STXCMM, I2SRXCMM=0 => Dual channel mode, RX/TX CHAN_MOD=0
I2SCC &= ~((I2STXCMM << I2STXCM) | (I2SRXCMM << I2SRXCM));
// Set dividers to something resonable
// Set dividers to something reasonable
currentRate = 0;
setRate(44100);
// Start I2S peripheral
@@ -280,4 +280,4 @@ int SinglePinI2SDriver::getUnderflowCount()
// Global instance
SinglePinI2SDriver I2SDriver;
#endif // defined(ESP8266)
#endif // defined(ESP8266)

16
src/libflac/stream_decoder.c
View File

@@ -2199,14 +2199,14 @@ FLAC__bool read_frame_header_(FLAC__StreamDecoder *decoder)
* Three kinds of things can go wrong when reading the frame header:
* 1) We may have sync'ed incorrectly and not landed on a frame header.
* If we don't find a sync code, it can end up looking like we read
* a valid but unparseable header, until getting to the frame header
* a valid but unparsable header, until getting to the frame header
* CRC. Even then we could get a false positive on the CRC.
* 2) We may have sync'ed correctly but on an unparseable frame (from a
* 2) We may have sync'ed correctly but on an unparsable frame (from a
* future encoder).
* 3) We may be on a damaged frame which appears valid but unparseable.
* 3) We may be on a damaged frame which appears valid but unparsable.
*
* For all these reasons, we try and read a complete frame header as
* long as it seems valid, even if unparseable, up until the frame
* long as it seems valid, even if unparsable, up until the frame
* header CRC.
*/
@@ -2783,7 +2783,7 @@ FLAC__bool read_residual_partitioned_rice_(FLAC__StreamDecoder *decoder, uint32_
if(rice_parameter < pesc) {
partitioned_rice_contents->raw_bits[partition] = 0;
u = (partition_order == 0 || partition > 0)? partition_samples : partition_samples - predictor_order;
if(!FLAC__bitreader_read_rice_signed_block(decoder->private_->input, residual + sample, u, rice_parameter))
if(!FLAC__bitreader_read_rice_signed_block(decoder->private_->input, (int *)(residual + sample), u, rice_parameter))
return false; /* read_callback_ sets the state for us */
sample += u;
}
@@ -2792,7 +2792,7 @@ FLAC__bool read_residual_partitioned_rice_(FLAC__StreamDecoder *decoder, uint32_
return false; /* read_callback_ sets the state for us */
partitioned_rice_contents->raw_bits[partition] = rice_parameter;
for(u = (partition_order == 0 || partition > 0)? 0 : predictor_order; u < partition_samples; u++, sample++) {
if(!FLAC__bitreader_read_raw_int32(decoder->private_->input, &i, rice_parameter))
if(!FLAC__bitreader_read_raw_int32(decoder->private_->input, (FLAC__int32 *)&i, rice_parameter))
return false; /* read_callback_ sets the state for us */
residual[sample] = i;
}
@@ -2839,7 +2839,7 @@ FLAC__bool read_callback_(FLAC__byte buffer[], size_t *bytes, void *client_data)
* FLAC__STREAM_DECODER_UNPARSEABLE_STREAM and increment its
* unparseable_frame_count. But there is a remote possibility
* that it is properly synced at such a "future-codec frame",
* so to make sure, we wait to see many "unparseable" errors in
* so to make sure, we wait to see many "unparsable" errors in
* a row before bailing out.
*/
if(decoder->private_->is_seeking && decoder->private_->unparseable_frame_count > 20) {
@@ -3146,7 +3146,7 @@ FLAC__bool seek_to_absolute_sample_(FLAC__StreamDecoder *decoder, FLAC__uint64 s
return false;
}
/* Now we need to get a frame. First we need to reset our
* unparseable_frame_count; if we get too many unparseable
* unparseable_frame_count; if we get too many unparsable
* frames in a row, the read callback will return
* FLAC__STREAM_DECODER_READ_STATUS_ABORT, causing
* FLAC__stream_decoder_process_single() to return false.

2
src/libhelix-aac/assembly.h
View File

@@ -558,7 +558,7 @@ static __inline int CLZ(int x)
typedef union _U64 {
Word64 w64;
struct {
#ifdef __XTENSA__
#if defined(__XTENSA__) || defined (__riscv)
unsigned int lo32;
signed int hi32;
#else

5
src/libhelix-aac/sbrqmf.c
View File

@@ -222,7 +222,8 @@ static void PostMultiply64(int *fft1, int nSampsOut)
* Notes: this is carefully written to be efficient on ARM
* use the assembly code version in sbrqmfak.s when building for ARM!
**************************************************************************************/
#if (defined (__arm) && defined (__ARMCC_VERSION)) || (defined (_WIN32) && defined (_WIN32_WCE) && defined (ARM)) || (defined(__GNUC__) && defined(__arm__))
//TODO - ADD IN .S SOURCES SOMEHOW
#if 0 //(defined (__arm) && defined (__ARMCC_VERSION)) || (defined (_WIN32) && defined (_WIN32_WCE) && defined (ARM)) || (defined(__GNUC__) && defined(__arm__))
#ifdef __cplusplus
extern "C"
#endif
@@ -395,7 +396,7 @@ int QMFAnalysis(int *inbuf, int *delay, int *XBuf, int fBitsIn, int *delayIdx, i
* Notes: this is carefully written to be efficient on ARM
* use the assembly code version in sbrqmfsk.s when building for ARM!
**************************************************************************************/
#if (defined (__arm) && defined (__ARMCC_VERSION)) || (defined (_WIN32) && defined (_WIN32_WCE) && defined (ARM)) || (defined(__GNUC__) && defined(__arm__))
#if 0 //(defined (__arm) && defined (__ARMCC_VERSION)) || (defined (_WIN32) && defined (_WIN32_WCE) && defined (ARM)) || (defined(__GNUC__) && defined(__arm__))
#ifdef __cplusplus
extern "C"
#endif

2
src/libhelix-mp3/RPSL.txt
View File

@@ -201,7 +201,7 @@ and 3, above.
4.2 Compatible Source Licenses. Software modules that have been independently
developed without any use of Covered Code and which contain no portion of the
Covered Code, Modifications or other Derivative Works, but are used or combined
in any way wtih the Covered Code or any Derivative Work to form a larger
in any way with the Covered Code or any Derivative Work to form a larger
Derivative Work, are exempt from the conditions described in Section 4.1 but
only to the extent that: the software module, including any software that is
linked to, integrated with, or part of the same applications as, the software

4
src/libhelix-mp3/assembly.h
View File

@@ -40,7 +40,7 @@
*
* assembly.h - assembly language functions and prototypes for supported platforms
*
* - inline rountines with access to 64-bit multiply results
* - inline routines with access to 64-bit multiply results
* - x86 (_WIN32) and ARM (ARM_ADS, _WIN32_WCE) versions included
* - some inline functions are mix of asm and C for speed
* - some functions are in native asm files, so only the prototype is given here
@@ -241,7 +241,7 @@ static __inline int MULSHIFT32(int x, int y)
static __inline int FASTABS(int x)
{
int t=0; /*Really is not necessary to initialiaze only to avoid warning*/
int t=0; /*Really is not necessary to initialize only to avoid warning*/
__asm {
eor t, x, x, asr #31

2
src/libhelix-mp3/scalfact.c
View File

@@ -74,7 +74,7 @@ static const char SFLenTab[16][2] = {
* Return: none
*
* Notes: set order of short blocks to s[band][window] instead of s[window][band]
* so that we index through consectutive memory locations when unpacking
* so that we index through consecutive memory locations when unpacking
* (make sure dequantizer follows same convention)
* Illegal Intensity Position = 7 (always) for MPEG1 scale factors
**************************************************************************************/

1
src/libmad/layer3.c
View File

@@ -20,6 +20,7 @@
*/
#pragma GCC optimize ("O3")
#pragma GCC diagnostic ignored "-Wstrict-aliasing"
#include <pgmspace.h>
# include "config.h"

4
src/libogg/README.md
View File

@@ -7,7 +7,7 @@
Ogg project codecs use the Ogg bitstream format to arrange the raw,
compressed bitstream into a more robust, useful form. For example,
the Ogg bitstream makes seeking, time stamping and error recovery
possible, as well as mixing several sepearate, concurrent media
possible, as well as mixing several separate, concurrent media
streams into a single physical bitstream.
## What's here ##
@@ -18,7 +18,7 @@ use with Ogg bitstreams.
Directory:
- `src` The source for libogg, a BSD-license inplementation of the public domain Ogg bitstream format
- `src` The source for libogg, a BSD-license implementation of the public domain Ogg bitstream format
- `include` Library API headers

2
src/libtinysoundfont/README.ESP8266
View File

@@ -28,7 +28,7 @@ Even with the caching, it was found that SPIFFS, while having great
functionality, was horrbly slow. So I wrote a new "faster" ROM filesystem
called, surprisingly, FastROMFilesystem.
https://github.com/earlephilhower/ESP8266FastROMFS
If you are getting choppy playback, try this new filesytem or using a SD
If you are getting choppy playback, try this new filesystem or using a SD
card (not tested by myself, but it'd be hard top be slower than SPIFFS).
Simply going from SPIFFS to FastROMFilesystem took my testing of
FURELISE.MID and 1MGM.SF2 from 0.5x realtime to 2.5x (i.e. from unusable

4
src/libtinysoundfont/tsf.h
View File

@@ -104,7 +104,7 @@ TSFDEF tsf* tsf_load(struct tsf_stream* stream);
// Free the memory related to this tsf instance
TSFDEF void tsf_close(tsf* f);
// Stop all playing notes immediatly and reset all channel parameters
// Stop all playing notes immediately and reset all channel parameters
TSFDEF void tsf_reset(tsf* f);
// Returns the preset index from a bank and preset number, or -1 if it does not exist in the loaded SoundFont
@@ -201,7 +201,7 @@ TSFDEF void tsf_channel_set_tuning(tsf* f, int channel, float tuning);
TSFDEF void tsf_channel_note_on(tsf* f, int channel, int key, float vel);
TSFDEF void tsf_channel_note_off(tsf* f, int channel, int key);
TSFDEF void tsf_channel_note_off_all(tsf* f, int channel); //end with sustain and release
TSFDEF void tsf_channel_sounds_off_all(tsf* f, int channel); //end immediatly
TSFDEF void tsf_channel_sounds_off_all(tsf* f, int channel); //end immediately
// Apply a MIDI control change to the channel (not all controllers are supported!)
TSFDEF void tsf_channel_midi_control(tsf* f, int channel, int controller, int control_value);

2
src/opusfile/opusfile.c
View File

@@ -1422,7 +1422,7 @@ static int op_open_seekable2_impl(OggOpusFile *_of){
_of->end=sr[0].offset+sr[0].size;
if(OP_UNLIKELY(_of->end<data_offset)){free(sr); return OP_EBADLINK;}
/*Now enumerate the bitstream structure.*/
ret = op_bisect_forward_serialno(_of,data_offset,sr,sizeof(sr)/sizeof(*sr),
ret = op_bisect_forward_serialno(_of,data_offset,sr,64,
&_of->serialnos,&_of->nserialnos,&_of->cserialnos);
free(sr);
return ret;

4
src/opusfile/opusfile.h
View File

@@ -729,7 +729,7 @@ struct OpusServerInfo{
/**The software used by the origin server (Server).
This is <code>NULL</code> if there was no <code>Server</code> header.*/
char *server;
/**The media type of the entity sent to the recepient (Content-Type).
/**The media type of the entity sent to the recipient (Content-Type).
This is <code>NULL</code> if there was no <code>Content-Type</code>
header.*/
char *content_type;
@@ -1436,7 +1436,7 @@ void op_free(OggOpusFile *_of);
Some of these functions may be used successfully on the partially open
streams returned by op_test_callbacks() or one of the associated
convenience functions.
Their documention will indicate so explicitly.*/
Their documentation will indicate so explicitly.*/
/*@{*/
/**Returns whether or not the stream being read is seekable.

8
src/spiram-fast.h
View File

@@ -332,9 +332,9 @@ class ESP8266SPIRAM {
}
digitalWrite(csPin, HIGH);
pinMode(sck, SPECIAL);
pinMode(miso, SPECIAL);
pinMode(mosi, SPECIAL);
pinMode(sck, OUTPUT);
pinMode(miso, INPUT);
pinMode(mosi, OUTPUT);
pinMode(csPin, OUTPUT);
// Enable streaming read/write mode
@@ -364,4 +364,4 @@ class ESP8266SPIRAM {
#endif // ESP32
#endif
#endif