import esp audio lib

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSource.h

@@ -0,0 +1,51 @@
+/*
+  AudioFileSource
+  Base class of an input "file" to be used by AudioGenerator
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCE_H
+#define _AUDIOFILESOURCE_H
+
+#include <Arduino.h>
+#include "AudioStatus.h"
+
+class AudioFileSource
+{
+  public:
+    AudioFileSource() {};
+    virtual ~AudioFileSource() {};
+    virtual bool open(const char *filename) { (void)filename; return false; };
+    virtual uint32_t read(void *data, uint32_t len) { (void)data; (void)len; return 0; };
+    virtual uint32_t readNonBlock(void *data, uint32_t len) { return read(data, len); }; 
+    virtual bool seek(int32_t pos, int dir) { (void)pos; (void)dir; return false; };
+    virtual bool close() { return false; };
+    virtual bool isOpen() { return false; };
+    virtual uint32_t getSize() { return 0; };
+    virtual uint32_t getPos() { return 0; };
+    virtual bool loop() { return true; };
+
+  public:
+    virtual bool RegisterMetadataCB(AudioStatus::metadataCBFn fn, void *data) { return cb.RegisterMetadataCB(fn, data); }
+    virtual bool RegisterStatusCB(AudioStatus::statusCBFn fn, void *data) { return cb.RegisterStatusCB(fn, data); }
+
+  protected:
+    AudioStatus cb;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceBuffer.cpp

@@ -0,0 +1,190 @@
+/*
+  AudioFileSourceBuffer
+  Double-buffered file source using system RAM
+  
+  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>
+#include "AudioFileSourceBuffer.h"
+
+#pragma GCC optimize ("O3")
+
+AudioFileSourceBuffer::AudioFileSourceBuffer(AudioFileSource *source, uint32_t buffSizeBytes)
+{
+  buffSize = buffSizeBytes;
+  buffer = (uint8_t*)malloc(sizeof(uint8_t) * buffSize);
+  if (!buffer) audioLogger->printf_P(PSTR("Unable to allocate AudioFileSourceBuffer::buffer[]\n"));
+  deallocateBuffer = true;
+  writePtr = 0;
+  readPtr = 0;
+  src = source;
+  length = 0;
+  filled = false;
+}
+
+AudioFileSourceBuffer::AudioFileSourceBuffer(AudioFileSource *source, void *inBuff, uint32_t buffSizeBytes)
+{
+  buffSize = buffSizeBytes;
+  buffer = (uint8_t*)inBuff;
+  deallocateBuffer = false;
+  writePtr = 0;
+  readPtr = 0;
+  src = source;
+  length = 0;
+  filled = false;
+}
+
+AudioFileSourceBuffer::~AudioFileSourceBuffer()
+{
+  if (deallocateBuffer) free(buffer);
+  buffer = NULL;
+}
+
+bool AudioFileSourceBuffer::seek(int32_t pos, int dir)
+{
+  if(dir == SEEK_CUR && (readPtr+pos) < length) {
+    readPtr += pos;
+    return true;
+  } else {
+    // Invalidate
+    readPtr = 0;
+    writePtr = 0;
+    length = 0;
+    return src->seek(pos, dir);
+  }
+}
+
+bool AudioFileSourceBuffer::close()
+{
+  if (deallocateBuffer) free(buffer);
+  buffer = NULL;
+  return src->close();
+}
+
+bool AudioFileSourceBuffer::isOpen()
+{
+  return src->isOpen();
+}
+
+uint32_t AudioFileSourceBuffer::getSize()
+{
+  return src->getSize();
+}
+
+uint32_t AudioFileSourceBuffer::getPos()
+{
+  return src->getPos();
+}
+
+uint32_t AudioFileSourceBuffer::getFillLevel()
+{
+  return length;
+}
+
+uint32_t AudioFileSourceBuffer::read(void *data, uint32_t len)
+{
+  if (!buffer) return src->read(data, len);
+
+  uint32_t bytes = 0;
+  if (!filled) {
+    // Fill up completely before returning any data at all
+    cb.st(STATUS_FILLING, PSTR("Refilling buffer"));
+    length = src->read(buffer, buffSize);
+    writePtr = length % buffSize;
+    filled = true;
+  }
+
+  // Pull from buffer until we've got none left or we've satisfied the request
+  uint8_t *ptr = reinterpret_cast<uint8_t*>(data);
+  uint32_t toReadFromBuffer = (len < length) ? len : length;
+  if ( (toReadFromBuffer > 0) && (readPtr >= writePtr) ) {
+    uint32_t toReadToEnd = (toReadFromBuffer < (uint32_t)(buffSize - readPtr)) ? toReadFromBuffer : (buffSize - readPtr);
+    memcpy(ptr, &buffer[readPtr], toReadToEnd);
+    readPtr = (readPtr + toReadToEnd) % buffSize;
+    len -= toReadToEnd;
+    length -= toReadToEnd;
+    ptr += toReadToEnd;
+    bytes += toReadToEnd;
+    toReadFromBuffer -= toReadToEnd;
+  }
+  if (toReadFromBuffer > 0) { // We know RP < WP at this point
+    memcpy(ptr, &buffer[readPtr], toReadFromBuffer);
+    readPtr = (readPtr + toReadFromBuffer) % buffSize;
+    len -= toReadFromBuffer;
+    length -= toReadFromBuffer;
+    ptr += toReadFromBuffer;
+    bytes += toReadFromBuffer;
+    toReadFromBuffer -= toReadFromBuffer;
+  }
+
+  if (len) {
+    // Still need more, try direct read from src
+    bytes += src->read(ptr, len);
+    // We're out of buffered data, need to force a complete refill.  Thanks, @armSeb
+    readPtr = 0;
+    writePtr = 0;
+    length = 0;
+    filled = false;
+    cb.st(STATUS_UNDERFLOW, PSTR("Buffer underflow"));
+  }
+
+  fill();
+
+  return bytes;
+}
+
+void AudioFileSourceBuffer::fill()
+{
+  if (!buffer) return;
+
+  if (length < buffSize) {
+    // Now try and opportunistically fill the buffer
+    if (readPtr > writePtr) {
+      if (readPtr == writePtr+1) return;
+      uint32_t bytesAvailMid = readPtr - writePtr - 1;
+      int cnt = src->readNonBlock(&buffer[writePtr], bytesAvailMid);
+      length += cnt;
+      writePtr = (writePtr + cnt) % buffSize;
+      return;
+    }
+
+    if (buffSize > writePtr) {
+      uint32_t bytesAvailEnd = buffSize - writePtr;
+      int cnt = src->readNonBlock(&buffer[writePtr], bytesAvailEnd);
+      length += cnt;
+      writePtr = (writePtr + cnt) % buffSize;
+      if (cnt != (int)bytesAvailEnd) return;
+    }
+
+    if (readPtr > 1) {
+      uint32_t bytesAvailStart = readPtr - 1;
+      int cnt = src->readNonBlock(&buffer[writePtr], bytesAvailStart);
+      length += cnt;
+      writePtr = (writePtr + cnt) % buffSize;
+    }
+  }
+}
+
+
+
+bool AudioFileSourceBuffer::loop()
+{
+  if (!src->loop()) return false;
+  fill();
+  return true;
+}  
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceBuffer.h

@@ -0,0 +1,62 @@
+/*
+  AudioFileSourceBuffer
+  Double-buffered input file using system RAM
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCEBUFFER_H
+#define _AUDIOFILESOURCEBUFFER_H
+
+#include "AudioFileSource.h"
+
+
+class AudioFileSourceBuffer : public AudioFileSource
+{
+  public:
+    AudioFileSourceBuffer(AudioFileSource *in, uint32_t bufferBytes);
+    AudioFileSourceBuffer(AudioFileSource *in, void *buffer, uint32_t bufferBytes); // Pre-allocated buffer by app
+    virtual ~AudioFileSourceBuffer() override;
+    
+    virtual uint32_t read(void *data, uint32_t len) override;
+    virtual bool seek(int32_t pos, int dir) override;
+    virtual bool close() override;
+    virtual bool isOpen() override;
+    virtual uint32_t getSize() override;
+    virtual uint32_t getPos() override;
+    virtual bool loop() override;
+
+    virtual uint32_t getFillLevel();
+
+    enum { STATUS_FILLING=2, STATUS_UNDERFLOW };
+
+  private:
+    virtual void fill();
+
+  private:
+    AudioFileSource *src;
+    uint32_t buffSize;
+    uint8_t *buffer;
+    bool deallocateBuffer;
+    uint32_t writePtr;
+    uint32_t readPtr;
+    uint32_t length;
+    bool filled;
+};
+
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceFATFS.h

@@ -0,0 +1,64 @@
+/*
+  AudioFileSourceFS
+  Input Arduion "file" to be used by AudioGenerator
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCEFATFS_H
+#define _AUDIOFILESOURCEFATFS_H
+
+#ifdef ESP32
+
+#include <Arduino.h>
+#include <FS.h>
+#include <FFat.h>
+
+#include "AudioFileSource.h"
+#include "AudioFileSourceFS.h"
+
+/*
+ AudioFileSource for FAT filesystem.
+ */
+class AudioFileSourceFATFS : public AudioFileSourceFS
+{
+  public:
+    AudioFileSourceFATFS() : AudioFileSourceFS(FFat) {};
+    AudioFileSourceFATFS(const char *filename) : AudioFileSourceFS(FFat) {
+      // We call open() ourselves because calling AudioFileSourceFS(FFat, filename)
+      // would call the parent open() and we do not want that
+      open(filename);
+    };
+
+    virtual bool open(const char *filename) override {
+      // make sure that the FATFS filesystem has been mounted
+      if (!FFat.begin()) {
+        audioLogger->printf_P(PSTR("Unable to initialize FATFS filesystem\n"));
+        return false;
+      } else {
+        // now that the fielsystem has been mounted, we can call the regular parent open() function
+        return AudioFileSourceFS::open(filename);
+      }
+    };
+
+    // Others are inherited from base
+};
+
+#endif
+
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceFS.cpp

@@ -0,0 +1,73 @@
+/*
+  AudioFileSourceFS
+  Input "file" to be used by AudioGenerator
+  
+  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 "AudioFileSourceFS.h"
+#ifdef ESP32
+#include "SPIFFS.h"
+#endif
+
+AudioFileSourceFS::AudioFileSourceFS(FS &fs, const char *filename)
+{
+  filesystem = &fs;
+  open(filename);
+}
+
+bool AudioFileSourceFS::open(const char *filename)
+{
+#ifndef ESP32
+  filesystem->begin();
+#endif
+  f = filesystem->open(filename, "r");
+  return f;
+}
+
+AudioFileSourceFS::~AudioFileSourceFS()
+{
+  if (f) f.close();
+}
+
+uint32_t AudioFileSourceFS::read(void *data, uint32_t len)
+{
+  return f.read(reinterpret_cast<uint8_t*>(data), len);
+}
+
+bool AudioFileSourceFS::seek(int32_t pos, int dir)
+{
+  return f.seek(pos, (dir==SEEK_SET)?SeekSet:(dir==SEEK_CUR)?SeekCur:SeekEnd);
+}
+
+bool AudioFileSourceFS::close()
+{
+  f.close();
+  return true;
+}
+
+bool AudioFileSourceFS::isOpen()
+{
+  return f?true:false;
+}
+
+uint32_t AudioFileSourceFS::getSize()
+{
+  if (!f) return 0;
+  return f.size();
+}
+
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceFS.h

@@ -0,0 +1,51 @@
+/*
+  AudioFileSourceFS
+  Input Arduion "file" to be used by AudioGenerator
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCEFS_H
+#define _AUDIOFILESOURCEFS_H
+
+#include <Arduino.h>
+#include <FS.h>
+
+#include "AudioFileSource.h"
+
+class AudioFileSourceFS : public AudioFileSource
+{
+  public:
+    AudioFileSourceFS(fs::FS &fs) { filesystem = &fs; }
+    AudioFileSourceFS(fs::FS &fs, const char *filename);
+    virtual ~AudioFileSourceFS() override;
+    
+    virtual bool open(const char *filename) override;
+    virtual uint32_t read(void *data, uint32_t len) override;
+    virtual bool seek(int32_t pos, int dir) override;
+    virtual bool close() override;
+    virtual bool isOpen() override;
+    virtual uint32_t getSize() override;
+    virtual uint32_t getPos() override { if (!f) return 0; else return f.position(); };
+
+  private:
+    fs::FS *filesystem;
+    fs::File f;
+};
+
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceFunction.cpp

@@ -0,0 +1,148 @@
+/*
+  AudioFileSourceFunction
+  Audio ouptut generator which can generate WAV file data from function
+
+  Copyright (C) 2021  Hideaki Tai
+
+  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 "AudioFileSourceFunction.h"
+
+AudioFileSourceFunction::AudioFileSourceFunction(float sec, uint16_t channels, uint32_t sample_per_sec, uint16_t bits_per_sample) {
+  uint32_t bytes_per_sec = sample_per_sec * channels * bits_per_sample / 8;
+  uint32_t len = uint32_t(sec * (float)bytes_per_sec);
+
+  // RIFF chunk
+  strncpy(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);
+
+  // format chunk
+  strncpy(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;
+  wav_header.format.sample_per_sec = sample_per_sec;
+  wav_header.format.avg_bytes_per_sec = bytes_per_sec;
+  wav_header.format.block_align = channels * bits_per_sample / 8;
+  wav_header.format.bits_per_sample = bits_per_sample;
+
+  // data chunk
+  strncpy(wav_header.data.chunk_id, "data", 4);
+  wav_header.data.chunk_size = len;
+
+  funcs.reserve(channels);
+  pos = 0;
+  size = sizeof(WavHeader) + len;
+  is_ready = false;
+  is_unique = false;
+}
+
+AudioFileSourceFunction::~AudioFileSourceFunction() {
+  close();
+}
+
+uint32_t AudioFileSourceFunction::read(void* data, uint32_t len) {
+  // callback size must be 1 or equal to channels
+  if (!is_ready)
+    return 0;
+
+  uint8_t* d = reinterpret_cast<uint8_t*>(data);
+  uint32_t i = 0;
+  while (i < len) {
+    uint32_t p = pos + i;
+    if (p < sizeof(WavHeader)) {
+      // header bytes
+      d[i] = wav_header.bytes[p];
+      i += 1;
+    } else {
+      // data bytes
+      float time = (float)(p - sizeof(WavHeader)) / (float)wav_header.format.avg_bytes_per_sec;
+      float v = funcs[0](time);
+      for (size_t ch = 0; ch < wav_header.format.channels; ++ch) {
+        if (!is_unique && ch > 0)
+          v = funcs[ch](time);
+
+        switch (wav_header.format.bits_per_sample) {
+          case 8: {
+            Uint8AndInt8 vs {int8_t(v * (float)0x7F)};
+            d[i] = vs.u;
+            break;
+          }
+          case 32: {
+            Uint8AndInt32 vs {int32_t(v * (float)0x7FFFFFFF)};
+            d[i + 0] = vs.u[0];
+            d[i + 1] = vs.u[1];
+            d[i + 2] = vs.u[2];
+            d[i + 3] = vs.u[3];
+            break;
+          }
+          case 16:
+          default: {
+            Uint8AndInt16 vs {int16_t(v * (float)0x7FFF)};
+            d[i + 0] = vs.u[0];
+            d[i + 1] = vs.u[1];
+            break;
+          }
+        }
+      }
+      i += wav_header.format.block_align;
+    }
+  }
+  pos += i;
+  return (pos >= size) ? 0 : i;
+}
+
+bool AudioFileSourceFunction::seek(int32_t pos, int dir) {
+  if (dir == SEEK_SET) {
+    if (pos < 0 || (uint32_t)pos >= size)
+      return false;
+    this->pos = pos;
+  } else if (dir == SEEK_CUR) {
+    int32_t p = (int32_t)this->pos + pos;
+    if (p < 0 || (uint32_t)p >= size)
+      return false;
+    this->pos = p;
+  } else {
+    int32_t p = (int32_t)this->size + pos;
+    if (p < 0 || (uint32_t)p >= size)
+      return false;
+    this->pos = p;
+  }
+  return true;
+}
+
+bool AudioFileSourceFunction::close() {
+  funcs.clear();
+  pos = 0;
+  size = 0;
+  is_ready = false;
+  is_unique = false;
+  return true;
+}
+
+bool AudioFileSourceFunction::isOpen() {
+  return is_ready;
+}
+
+uint32_t AudioFileSourceFunction::getSize() {
+  return size;
+}
+
+uint32_t AudioFileSourceFunction::getPos() {
+  return pos;
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceFunction.h

@@ -0,0 +1,119 @@
+/*
+  AudioFileSourceFunction
+  Audio ouptut generator which can generate WAV file data from function
+
+  Copyright (C) 2021  Hideaki Tai
+
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCEFUNCTION_H
+#define _AUDIOFILESOURCEFUNCTION_H
+
+#include <Arduino.h>
+#include <vector>
+#include <functional>
+
+#include "AudioFileSource.h"
+
+class AudioFileSourceFunction : public AudioFileSource {
+  union WavHeader {
+    struct {
+      // RIFF chunk
+      struct {
+        char chunk_id[4];     // "RIFF"
+        uint32_t chunk_size;  // 4 + (8 + sizeof(format_chunk)(16)) + (8 + sizeof(data_chunk))
+        char format[4];       // "WAVE"
+      } riff;
+      // format chunk
+      struct {
+        char chunk_id[4];            // "fmt "
+        uint32_t chunk_size;         // 16
+        uint16_t format_tag;         // 1: PCM
+        uint16_t channels;           // 1: MONO, 2: STEREO
+        uint32_t sample_per_sec;     // 8000, 11025, 22050, 44100, 48000
+        uint32_t avg_bytes_per_sec;  // sample_per_sec * channels * bits_per_sample / 8
+        uint16_t block_align;        // channels * bits_per_sample / 8
+        uint16_t bits_per_sample;    // 8, 16, 32
+      } format;
+      // data chunk
+      struct {
+        char chunk_id[4];     // "data"
+        uint32_t chunk_size;  // num_samples * channels * bytes_per_sample
+        // audio data follows here...
+      } data;
+    };
+    uint8_t bytes[44];
+  } wav_header;
+
+  union Uint8AndInt8 {
+    int8_t i;
+    uint8_t u;
+  };
+
+  union Uint8AndInt16 {
+    int16_t i;
+    uint8_t u[2];
+  };
+
+  union Uint8AndInt32 {
+    int32_t i;
+    uint8_t u[4];
+  };
+
+  using callback_t = std::function<float(float)>;
+  std::vector<callback_t> funcs;
+  uint32_t pos;
+  uint32_t size;
+  bool is_ready;
+  bool is_unique;
+
+public:
+  AudioFileSourceFunction(float sec, uint16_t channels = 1, uint32_t sample_per_sec = 8000, uint16_t bits_per_sample = 16);
+  virtual ~AudioFileSourceFunction() override;
+
+  template <typename F, typename... Fs>
+  bool addAudioGenerators(const F& f, Fs&&... fs) {
+    funcs.emplace_back(f);
+    return addAudioGenerators(std::forward<Fs>(fs)...);
+  }
+  bool addAudioGenerators() {
+    funcs.shrink_to_fit();
+    if (funcs.size() == 1) {
+      is_ready = true;
+      is_unique = true;
+      return true;
+    } else if (funcs.size() == wav_header.format.channels) {
+      is_ready = true;
+      is_unique = false;
+      return true;
+    } else {
+      is_ready = false;
+      is_unique = false;
+      funcs.clear();
+      return false;
+    }
+  }
+
+  virtual uint32_t read(void* data, uint32_t len) override;
+  virtual bool seek(int32_t pos, int dir) override;
+
+  virtual bool close() override;
+  virtual bool isOpen() override;
+
+  virtual uint32_t getSize() override;
+  virtual uint32_t getPos() override;
+};
+
+#endif  // _AUDIOFILESOURCEFUNCTION_H

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceHTTPStream.cpp

@@ -0,0 +1,158 @@
+/*
+  AudioFileSourceHTTPStream
+  Streaming HTTP source
+
+  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/>.
+*/
+
+#if defined(ESP32) || defined(ESP8266)
+
+#include "AudioFileSourceHTTPStream.h"
+
+AudioFileSourceHTTPStream::AudioFileSourceHTTPStream()
+{
+  pos = 0;
+  reconnectTries = 0;
+  saveURL[0] = 0;
+}
+
+AudioFileSourceHTTPStream::AudioFileSourceHTTPStream(const char *url)
+{
+  saveURL[0] = 0;
+  reconnectTries = 0;
+  open(url);
+}
+
+bool AudioFileSourceHTTPStream::open(const char *url)
+{
+  pos = 0;
+  http.begin(client, url);
+  http.setReuse(true);
+#ifndef ESP32
+  http.setFollowRedirects(HTTPC_FORCE_FOLLOW_REDIRECTS);
+#endif
+  int code = http.GET();
+  if (code != HTTP_CODE_OK) {
+    http.end();
+    cb.st(STATUS_HTTPFAIL, PSTR("Can't open HTTP request"));
+    return false;
+  }
+  size = http.getSize();
+  strncpy(saveURL, url, sizeof(saveURL));
+  saveURL[sizeof(saveURL)-1] = 0;
+  return true;
+}
+
+AudioFileSourceHTTPStream::~AudioFileSourceHTTPStream()
+{
+  http.end();
+}
+
+uint32_t AudioFileSourceHTTPStream::read(void *data, uint32_t len)
+{
+  if (data==NULL) {
+    audioLogger->printf_P(PSTR("ERROR! AudioFileSourceHTTPStream::read passed NULL data\n"));
+    return 0;
+  }
+  return readInternal(data, len, false);
+}
+
+uint32_t AudioFileSourceHTTPStream::readNonBlock(void *data, uint32_t len)
+{
+  if (data==NULL) {
+    audioLogger->printf_P(PSTR("ERROR! AudioFileSourceHTTPStream::readNonBlock passed NULL data\n"));
+    return 0;
+  }
+  return readInternal(data, len, true);
+}
+
+uint32_t AudioFileSourceHTTPStream::readInternal(void *data, uint32_t len, bool nonBlock)
+{
+retry:
+  if (!http.connected()) {
+    cb.st(STATUS_DISCONNECTED, PSTR("Stream disconnected"));
+    http.end();
+    for (int i = 0; i < reconnectTries; i++) {
+      char buff[64];
+      sprintf_P(buff, PSTR("Attempting to reconnect, try %d"), i);
+      cb.st(STATUS_RECONNECTING, buff);
+      delay(reconnectDelayMs);
+      if (open(saveURL)) {
+        cb.st(STATUS_RECONNECTED, PSTR("Stream reconnected"));
+        break;
+      }
+    }
+    if (!http.connected()) {
+      cb.st(STATUS_DISCONNECTED, PSTR("Unable to reconnect"));
+      return 0;
+    }
+  }
+  if ((size > 0) && (pos >= size)) return 0;
+
+  WiFiClient *stream = http.getStreamPtr();
+
+  // Can't read past EOF...
+  if ( (size > 0) && (len > (uint32_t)(pos - size)) ) len = pos - size;
+
+  if (!nonBlock) {
+    int start = millis();
+    while ((stream->available() < (int)len) && (millis() - start < 500)) yield();
+  }
+
+  size_t avail = stream->available();
+  if (!nonBlock && !avail) {
+    cb.st(STATUS_NODATA, PSTR("No stream data available"));
+    http.end();
+    goto retry;
+  }
+  if (avail == 0) return 0;
+  if (avail < len) len = avail;
+
+  int read = stream->read(reinterpret_cast<uint8_t*>(data), len);
+  pos += read;
+  return read;
+}
+
+bool AudioFileSourceHTTPStream::seek(int32_t pos, int dir)
+{
+  audioLogger->printf_P(PSTR("ERROR! AudioFileSourceHTTPStream::seek not implemented!"));
+  (void) pos;
+  (void) dir;
+  return false;
+}
+
+bool AudioFileSourceHTTPStream::close()
+{
+  http.end();
+  return true;
+}
+
+bool AudioFileSourceHTTPStream::isOpen()
+{
+  return http.connected();
+}
+
+uint32_t AudioFileSourceHTTPStream::getSize()
+{
+  return size;
+}
+
+uint32_t AudioFileSourceHTTPStream::getPos()
+{
+  return pos;
+}
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceHTTPStream.h

@@ -0,0 +1,67 @@
+/*
+  AudioFileSourceHTTPStream
+  Connect to a HTTP based streaming service
+  
+  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/>.
+*/
+
+#if defined(ESP32) || defined(ESP8266)
+#pragma once
+
+#include <Arduino.h>
+#ifdef ESP32
+  #include <HTTPClient.h>
+#else
+  #include <ESP8266HTTPClient.h>
+#endif
+#include "AudioFileSource.h"
+
+class AudioFileSourceHTTPStream : public AudioFileSource
+{
+  friend class AudioFileSourceICYStream;
+
+  public:
+    AudioFileSourceHTTPStream();
+    AudioFileSourceHTTPStream(const char *url);
+    virtual ~AudioFileSourceHTTPStream() override;
+    
+    virtual bool open(const char *url) override;
+    virtual uint32_t read(void *data, uint32_t len) override;
+    virtual uint32_t readNonBlock(void *data, uint32_t len) override;
+    virtual bool seek(int32_t pos, int dir) override;
+    virtual bool close() override;
+    virtual bool isOpen() override;
+    virtual uint32_t getSize() override;
+    virtual uint32_t getPos() override;
+    bool SetReconnect(int tries, int delayms) { reconnectTries = tries; reconnectDelayMs = delayms; return true; }
+    void useHTTP10 () { http.useHTTP10(true); }
+
+    enum { STATUS_HTTPFAIL=2, STATUS_DISCONNECTED, STATUS_RECONNECTING, STATUS_RECONNECTED, STATUS_NODATA };
+
+  private:
+    virtual uint32_t readInternal(void *data, uint32_t len, bool nonBlock);
+    WiFiClient client;
+    HTTPClient http;
+    int pos;
+    int size;
+    int reconnectTries;
+    int reconnectDelayMs;
+    char saveURL[128];
+};
+
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceICYStream.cpp

@@ -0,0 +1,222 @@
+/*
+  AudioFileSourceICYStream
+  Streaming Shoutcast ICY source
+  
+  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/>.
+*/
+
+#if defined(ESP32) || defined(ESP8266)
+
+#define _GNU_SOURCE
+
+#include "AudioFileSourceICYStream.h"
+#include <string.h>
+
+AudioFileSourceICYStream::AudioFileSourceICYStream()
+{
+  pos = 0;
+  reconnectTries = 0;
+  saveURL[0] = 0;
+}
+
+AudioFileSourceICYStream::AudioFileSourceICYStream(const char *url)
+{
+  saveURL[0] = 0;
+  reconnectTries = 0;
+  open(url);
+}
+
+bool AudioFileSourceICYStream::open(const char *url)
+{
+  static const char *hdr[] = { "icy-metaint", "icy-name", "icy-genre", "icy-br" };
+  pos = 0;
+  http.begin(client, url);
+  http.addHeader("Icy-MetaData", "1");
+  http.collectHeaders( hdr, 4 );
+  http.setReuse(true);
+  int code = http.GET();
+  if (code != HTTP_CODE_OK) {
+    http.end();
+    cb.st(STATUS_HTTPFAIL, PSTR("Can't open HTTP request"));
+    return false;
+  }
+  if (http.hasHeader(hdr[0])) {
+    String ret = http.header(hdr[0]);
+    icyMetaInt = ret.toInt();
+  } else {
+    icyMetaInt = 0;
+  }
+  if (http.hasHeader(hdr[1])) {
+    String ret = http.header(hdr[1]);
+//    cb.md("SiteName", false, ret.c_str());
+  }
+  if (http.hasHeader(hdr[2])) {
+    String ret = http.header(hdr[2]);
+//    cb.md("Genre", false, ret.c_str());
+  }
+  if (http.hasHeader(hdr[3])) {
+    String ret = http.header(hdr[3]);
+//    cb.md("Bitrate", false, ret.c_str());
+  }
+
+  icyByteCount = 0;
+  size = http.getSize();
+  strncpy(saveURL, url, sizeof(saveURL));
+  saveURL[sizeof(saveURL)-1] = 0;
+  return true;
+}
+
+AudioFileSourceICYStream::~AudioFileSourceICYStream()
+{
+  http.end();
+}
+
+uint32_t AudioFileSourceICYStream::readInternal(void *data, uint32_t len, bool nonBlock)
+{
+  // Ensure we can't possibly read 2 ICY headers in a single go #355
+  if (icyMetaInt > 1) {
+    len = std::min((int)(icyMetaInt >> 1), (int)len);
+  }
+retry:
+  if (!http.connected()) {
+    cb.st(STATUS_DISCONNECTED, PSTR("Stream disconnected"));
+    http.end();
+    for (int i = 0; i < reconnectTries; i++) {
+      char buff[64];
+      sprintf_P(buff, PSTR("Attempting to reconnect, try %d"), i);
+      cb.st(STATUS_RECONNECTING, buff);
+      delay(reconnectDelayMs);
+      if (open(saveURL)) {
+        cb.st(STATUS_RECONNECTED, PSTR("Stream reconnected"));
+        break;
+      }
+    }
+    if (!http.connected()) {
+      cb.st(STATUS_DISCONNECTED, PSTR("Unable to reconnect"));
+      return 0;
+    }
+  }
+  if ((size > 0) && (pos >= size)) return 0;
+
+  WiFiClient *stream = http.getStreamPtr();
+
+  // Can't read past EOF...
+  if ( (size > 0) && (len > (uint32_t)(pos - size)) ) len = pos - size;
+
+  if (!nonBlock) {
+    int start = millis();
+    while ((stream->available() < (int)len) && (millis() - start < 500)) yield();
+  }
+
+  size_t avail = stream->available();
+  if (!nonBlock && !avail) {
+    cb.st(STATUS_NODATA, PSTR("No stream data available"));
+    http.end();
+    goto retry;
+  }
+  if (avail == 0) return 0;
+  if (avail < len) len = avail;
+
+  int read = 0;
+  int ret = 0;
+  // If the read would hit an ICY block, split it up...
+  if (((int)(icyByteCount + len) > (int)icyMetaInt) && (icyMetaInt > 0)) {
+    int beforeIcy = icyMetaInt - icyByteCount;
+    if (beforeIcy > 0) {
+      ret = stream->read(reinterpret_cast<uint8_t*>(data), beforeIcy);
+      if (ret < 0) ret = 0;
+      read += ret;
+      pos += ret;
+      len -= ret;
+      data = (void *)(reinterpret_cast<char*>(data) + ret);
+      icyByteCount += ret;
+      if (ret != beforeIcy) return read; // Partial read
+    }
+
+    // ICY MD handling
+    int mdSize;
+    uint8_t c;
+    int mdret = stream->read(&c, 1);
+    if (mdret==0) return read;
+    mdSize = c * 16;
+    if ((mdret == 1) && (mdSize > 0)) {
+      // This is going to get ugly fast.
+      char icyBuff[256 + 16 + 1];
+      char *readInto = icyBuff + 16;
+      memset(icyBuff, 0, 16); // Ensure no residual matches occur
+      while (mdSize) {
+        int toRead = mdSize > 256 ? 256 : mdSize;
+        int ret = stream->read((uint8_t*)readInto, toRead);
+        if (ret < 0) return read;
+        if (ret == 0) { delay(1); continue; }
+        mdSize -= ret;
+        // At this point we have 0...15 = last 15 chars read from prior read plus new data
+        int end = 16 + ret; // The last byte of valid data
+        char *header = (char *)memmem((void*)icyBuff, end, (void*)"StreamTitle=", 12);
+        if (!header) {
+          // No match, so move the last 16 bytes back to the start and continue
+          memmove(icyBuff, icyBuff+end-16, 16);
+          delay(1);
+	  continue;
+        }
+        // Found header, now move it to the front
+        int lastValidByte = end - (header -icyBuff) + 1;
+        memmove(icyBuff, header, lastValidByte);
+        // Now fill the buffer to the end with read data
+        while (mdSize && lastValidByte < 255) {
+          int toRead = mdSize > (256 - lastValidByte) ? (256 - lastValidByte) : mdSize;
+          ret = stream->read((uint8_t*)icyBuff + lastValidByte, toRead);
+          if (ret==-1) return read; // error
+          if (ret == 0) { delay(1); continue; }
+          mdSize -= ret;
+          lastValidByte += ret;
+        }
+        // Buffer now contains StreamTitle=....., parse it
+        char *p = icyBuff+12;
+        if (*p=='\'' || *p== '"' ) {
+          char closing[] = { *p, ';', '\0' };
+          char *psz = strstr( p+1, closing );
+          if( !psz ) psz = strchr( &icyBuff[13], ';' );
+          if( psz ) *psz = '\0';
+          p++;
+        } else {
+          char *psz = strchr( p, ';' );
+          if( psz ) *psz = '\0';
+        }
+        cb.md("StreamTitle", false, p);
+
+        // Now skip rest of MD block
+        while (mdSize) {
+          int toRead = mdSize > 256 ? 256 : mdSize;
+          ret = stream->read((uint8_t*)icyBuff, toRead);
+          if (ret < 0) return read;
+          if (ret == 0) { delay(1); continue; }
+          mdSize -= ret;
+        }
+      }
+    }
+    icyByteCount = 0;
+  }
+
+  ret = stream->read(reinterpret_cast<uint8_t*>(data), len);
+  if (ret < 0) ret = 0;
+  read += ret;
+  pos += ret;
+  icyByteCount += ret;
+  return read;
+}
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceICYStream.h

@@ -0,0 +1,48 @@
+/*
+  AudioFileSourceHTTPStream
+  Connect to a HTTP based streaming service
+  
+  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/>.
+*/
+
+#if defined(ESP32) || defined(ESP8266)
+#pragma once
+
+#include <Arduino.h>
+#ifdef ESP32
+  #include <HTTPClient.h>
+#else
+  #include <ESP8266HTTPClient.h>
+#endif
+
+#include "AudioFileSourceHTTPStream.h"
+
+class AudioFileSourceICYStream : public AudioFileSourceHTTPStream
+{
+  public:
+    AudioFileSourceICYStream();
+    AudioFileSourceICYStream(const char *url);
+    virtual ~AudioFileSourceICYStream() override;
+    
+    virtual bool open(const char *url) override;
+
+  private:
+    virtual uint32_t readInternal(void *data, uint32_t len, bool nonBlock) override;
+    int icyMetaInt;
+    int icyByteCount;
+};
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceID3.cpp

@@ -0,0 +1,280 @@
+/*
+  AudioFileSourceID3
+  ID3 filter that extracts any ID3 fields and sends to CB function
+  
+  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 "AudioFileSourceID3.h"
+
+// Handle unsync operation in ID3 with custom class
+class AudioFileSourceUnsync : public AudioFileSource
+{
+  public:
+    AudioFileSourceUnsync(AudioFileSource *src, int len, bool unsync);
+    virtual ~AudioFileSourceUnsync() override;
+    virtual uint32_t read(void *data, uint32_t len) override;
+
+    int getByte();
+    bool eof();
+
+  private:
+    AudioFileSource *src;
+    int remaining;
+    bool unsync;
+    int savedByte;
+};
+
+AudioFileSourceUnsync::AudioFileSourceUnsync(AudioFileSource *src, int len, bool unsync)
+{
+  this->src = src;
+  this->remaining = len;
+  this->unsync = unsync;
+  this->savedByte = -1;
+}
+
+AudioFileSourceUnsync::~AudioFileSourceUnsync()
+{
+}
+
+uint32_t AudioFileSourceUnsync::read(void *data, uint32_t len)
+{
+  uint32_t bytes = 0;
+  uint8_t *ptr = reinterpret_cast<uint8_t*>(data);
+
+  // This is only used during ID3 parsing, so no need to optimize here...
+  while (len--) {
+    int b = getByte();
+    if (b >= 0) {
+      *(ptr++) = (uint8_t)b;
+      bytes++;
+    }
+  }
+  return bytes;
+}
+
+int AudioFileSourceUnsync::getByte()
+{
+  // If we're not unsync, just read.
+  if (!unsync) {
+    uint8_t c;
+    if (!remaining) return -1;
+    remaining--;
+    if (1 != src->read(&c, 1)) return -1;
+    return c;
+  }
+
+  // If we've saved a pre-read character, return it immediately
+  if (savedByte >= 0) {
+    int s = savedByte;
+    savedByte = -1;
+    return s;
+  }
+
+  if (remaining <= 0) {
+    return -1;
+  } else if (remaining == 1) {
+    remaining--;
+    uint8_t c;
+    if (1 != src->read(&c, 1)) return -1;
+    else return c;
+  } else {
+    uint8_t c;
+    remaining--;
+    if (1 != src->read(&c, 1)) return -1;
+    if (c != 0xff) {
+      return c;
+    }
+    // Saw 0xff, check next byte.  If 0 then eat it, OTW return the 0xff
+    uint8_t d;
+    remaining--;
+    if (1 != src->read(&d, 1)) return c;
+    if (d != 0x00) {
+      savedByte = d;
+    }
+    return c;
+  }
+}
+
+bool AudioFileSourceUnsync::eof()
+{
+  if (remaining<=0) return true;
+  else return false;
+}
+
+
+
+AudioFileSourceID3::AudioFileSourceID3(AudioFileSource *src)
+{
+  this->src = src;
+  this->checked = false;
+}
+
+AudioFileSourceID3::~AudioFileSourceID3()
+{
+}
+
+uint32_t AudioFileSourceID3::read(void *data, uint32_t len)
+{
+  int rev = 0;
+
+  if (checked) {
+    return src->read(data, len);
+  }
+  checked = true;
+  // <10 bytes initial read, not enough space to check header
+  if (len<10) return src->read(data, len);
+
+  uint8_t *buff = reinterpret_cast<uint8_t*>(data);
+  int ret = src->read(data, 10);
+  if (ret<10) return ret;
+
+  if ((buff[0]!='I') || (buff[1]!='D') || (buff[2]!='3') || (buff[3]>0x04) || (buff[3]<0x02) || (buff[4]!=0)) {
+    cb.md("eof", false, "id3");
+    return 10 + src->read(buff+10, len-10);
+  }
+
+  rev = buff[3];
+  bool unsync = false;
+  bool exthdr = false;
+
+  switch(rev) {
+    case 2:
+      unsync = (buff[5] & 0x80);
+      exthdr = false;
+      break;
+    case 3:
+    case 4:
+      unsync = (buff[5] & 0x80);
+      exthdr = (buff[5] & 0x40);
+      break;
+  };
+
+  int id3Size = buff[6];
+  id3Size = id3Size << 7;
+  id3Size |= buff[7];
+  id3Size = id3Size << 7;
+  id3Size |= buff[8];
+  id3Size = id3Size << 7;
+  id3Size |= buff[9];
+  // Every read from now may be unsync'd
+  AudioFileSourceUnsync id3(src, id3Size, unsync);
+
+  if (exthdr) {
+    int ehsz = (id3.getByte()<<24) | (id3.getByte()<<16) | (id3.getByte()<<8) | (id3.getByte());
+    for (int j=0; j<ehsz-4; j++) id3.getByte(); // Throw it away
+  }
+
+  do {
+    unsigned char frameid[4];
+    int framesize;
+    bool compressed;
+
+    frameid[0] = id3.getByte();
+    frameid[1] = id3.getByte();
+    frameid[2] = id3.getByte();
+    if (rev==2) frameid[3] = 0;
+    else frameid[3] = id3.getByte();
+
+    if (frameid[0]==0 && frameid[1]==0 && frameid[2]==0 && frameid[3]==0) {
+      // We're in padding
+      while (!id3.eof()) {
+        id3.getByte();
+      }
+    } else {
+      if (rev==2) {
+        framesize = (id3.getByte()<<16) | (id3.getByte()<<8) | (id3.getByte());
+        compressed = false;
+      } else {
+        framesize = (id3.getByte()<<24) | (id3.getByte()<<16) | (id3.getByte()<<8) | (id3.getByte());
+        id3.getByte(); // skip 1st flag
+        compressed = id3.getByte()&0x80;
+      }
+      if (compressed) {
+        int decompsize = (id3.getByte()<<24) | (id3.getByte()<<16) | (id3.getByte()<<8) | (id3.getByte());
+        // TODO - add libz decompression, for now ignore this one...
+        (void)decompsize;
+        for (int j=0; j<framesize; j++)
+          id3.getByte();
+      }
+
+      // Read the value and send to callback
+      char value[64];
+      uint32_t i;
+      bool isUnicode = (id3.getByte()==1) ? true : false;
+      for (i=0; i<(uint32_t)framesize-1; i++) {
+        if (i<sizeof(value)-1) value[i] = id3.getByte();
+        else (void)id3.getByte();
+      }
+      value[i<sizeof(value)-1?i:sizeof(value)-1] = 0; // Terminate the string...
+      if ( (frameid[0]=='T' && frameid[1]=='A' && frameid[2]=='L' && frameid[3] == 'B' ) ||
+           (frameid[0]=='T' && frameid[1]=='A' && frameid[2]=='L' && rev==2) ) {
+        cb.md("Album", isUnicode, value);
+      } else if ( (frameid[0]=='T' && frameid[1]=='I' && frameid[2]=='T' && frameid[3] == '2') ||
+                  (frameid[0]=='T' && frameid[1]=='T' && frameid[2]=='2' && rev==2) ) {
+        cb.md("Title", isUnicode, value);
+      } else if ( (frameid[0]=='T' && frameid[1]=='P' && frameid[2]=='E' && frameid[3] == '1') ||
+                  (frameid[0]=='T' && frameid[1]=='P' && frameid[2]=='1' && rev==2) ) {
+        cb.md("Performer", isUnicode, value);
+      } else if ( (frameid[0]=='T' && frameid[1]=='Y' && frameid[2]=='E' && frameid[3] == 'R') ||
+                  (frameid[0]=='T' && frameid[1]=='Y' && frameid[2]=='E' && rev==2) ) {
+        cb.md("Year", isUnicode, value);
+      } else if ( (frameid[0]=='T' && frameid[1]=='R' && frameid[2]=='C' && frameid[3] == 'K') ||
+                  (frameid[0]=='T' && frameid[1]=='R' && frameid[2]=='K' && rev==2) ) {
+        cb.md("track", isUnicode, value);
+      } else if ( (frameid[0]=='T' && frameid[1]=='P' && frameid[2]=='O' && frameid[3] == 'S') ||
+                  (frameid[0]=='T' && frameid[1]=='P' && frameid[2]=='A' && rev==2) ) {
+        cb.md("Set", isUnicode, value);
+      } else if ( (frameid[0]=='P' && frameid[1]=='O' && frameid[2]=='P' && frameid[3] == 'M') ||
+                  (frameid[0]=='P' && frameid[1]=='O' && frameid[2]=='P' && rev==2) ) {
+        cb.md("Popularimeter", isUnicode, value);
+      } else if ( (frameid[0]=='T' && frameid[1]=='C' && frameid[2]=='M' && frameid[3] == 'P') ) {
+        cb.md("Compilation", isUnicode, value);
+      }
+    }
+  } while (!id3.eof());
+
+  // use callback function to signal end of tags and beginning of content.
+  cb.md("eof", false, "id3");
+
+  // All ID3 processing done, return to main caller
+  return src->read(data, len);
+}
+
+bool AudioFileSourceID3::seek(int32_t pos, int dir)
+{
+  return src->seek(pos, dir);
+}
+
+bool AudioFileSourceID3::close()
+{
+  return src->close();
+}
+
+bool AudioFileSourceID3::isOpen()
+{
+  return src->isOpen();
+}
+
+uint32_t AudioFileSourceID3::getSize()
+{
+  return src->getSize();
+}
+
+uint32_t AudioFileSourceID3::getPos()
+{
+  return src->getPos();
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceID3.h

@@ -0,0 +1,48 @@
+/*
+  AudioFileSourceID3
+  ID3 filter that extracts any ID3 fields and sends to CB function
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCEID3_H
+#define _AUDIOFILESOURCEID3_H
+
+#include <Arduino.h>
+
+#include "AudioFileSource.h"
+
+class AudioFileSourceID3 : public AudioFileSource
+{
+  public:
+    AudioFileSourceID3(AudioFileSource *src);
+    virtual ~AudioFileSourceID3() override;
+    
+    virtual uint32_t read(void *data, uint32_t len) override;
+    virtual bool seek(int32_t pos, int dir) override;
+    virtual bool close() override;
+    virtual bool isOpen() override;
+    virtual uint32_t getSize() override;
+    virtual uint32_t getPos() override;
+
+  private:
+    AudioFileSource *src;
+    bool checked;
+};
+
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceLittleFS.h

@@ -0,0 +1,43 @@
+/*
+  AudioFileSourceFS
+  Input Arduion "file" to be used by AudioGenerator
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCESPIFFS_H
+#define _AUDIOFILESOURCESPIFFS_H
+
+#ifndef ESP32 // No LittleFS there, yet
+
+#include <Arduino.h>
+#include <LittleFS.h>
+
+#include "AudioFileSource.h"
+#include "AudioFileSourceFS.h"
+
+class AudioFileSourceLittleFS : public AudioFileSourceFS
+{
+  public:
+    AudioFileSourceLittleFS() : AudioFileSourceFS(LittleFS) { };
+    AudioFileSourceLittleFS(const char *filename) : AudioFileSourceFS(LittleFS, filename) {};
+    // Others are inherited from base
+};
+
+#endif
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourcePROGMEM.cpp

@@ -0,0 +1,99 @@
+/*
+  AudioFileSourcePROGMEM
+  Store a "file" as a PROGMEM array and use it as audio source data
+  
+  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 "AudioFileSourcePROGMEM.h"
+
+AudioFileSourcePROGMEM::AudioFileSourcePROGMEM()
+{
+  opened = false;
+  progmemData = NULL;
+  progmemLen = 0;
+  filePointer = 0;
+}
+
+AudioFileSourcePROGMEM::AudioFileSourcePROGMEM(const void *data, uint32_t len)
+{
+  open(data, len);
+}
+
+AudioFileSourcePROGMEM::~AudioFileSourcePROGMEM()
+{
+}
+
+bool AudioFileSourcePROGMEM::open(const void *data, uint32_t len)
+{
+  if (!data || !len) return false;
+
+  opened = true;
+  progmemData = data;
+  progmemLen = len;
+  filePointer = 0;
+  return true;
+}
+
+uint32_t AudioFileSourcePROGMEM::getSize()
+{
+  if (!opened) return 0;
+  return progmemLen;
+}
+
+bool AudioFileSourcePROGMEM::isOpen()
+{
+  return opened;
+}
+
+bool AudioFileSourcePROGMEM::close()
+{
+  opened = false;
+  progmemData = NULL;
+  progmemLen = 0;
+  filePointer = 0;
+  return true;
+}  
+
+bool AudioFileSourcePROGMEM::seek(int32_t pos, int dir)
+{
+  if (!opened) return false;
+  uint32_t newPtr;
+  switch (dir) {
+    case SEEK_SET: newPtr = pos; break;
+    case SEEK_CUR: newPtr = filePointer + pos; break;
+    case SEEK_END: newPtr = progmemLen - pos; break;
+    default: return false;
+  }
+  if (newPtr > progmemLen) return false;
+  filePointer = newPtr;
+  return true;
+}
+
+uint32_t AudioFileSourcePROGMEM::read(void *data, uint32_t len)
+{
+  if (!opened) return 0;
+  if (filePointer >= progmemLen) return 0;
+
+  uint32_t toRead = progmemLen - filePointer;
+  if (toRead > len) toRead = len;
+
+  memcpy_P(data, reinterpret_cast<const uint8_t*>(progmemData)+filePointer, toRead);
+  filePointer += toRead;
+  return toRead;
+}
+
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourcePROGMEM.h

@@ -0,0 +1,49 @@
+/*
+  AudioFileSourcePROGMEM
+  Store a "file" as a PROGMEM array and use it as audio source data
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCEPROGMEM_H
+#define _AUDIOFILESOURCEPROGMEM_H
+
+#include "AudioFileSource.h"
+
+class AudioFileSourcePROGMEM : public AudioFileSource
+{
+  public:
+    AudioFileSourcePROGMEM();
+    AudioFileSourcePROGMEM(const void *data, uint32_t len);
+    virtual ~AudioFileSourcePROGMEM() override;
+    virtual uint32_t read(void *data, uint32_t len) override;
+    virtual bool seek(int32_t pos, int dir) override;
+    virtual bool close() override;
+    virtual bool isOpen() override;
+    virtual uint32_t getSize() override;
+    virtual uint32_t getPos() override { if (!opened) return 0; else return filePointer; };
+
+    bool open(const void *data, uint32_t len);
+
+  private:
+    bool opened;
+    const void *progmemData;
+    uint32_t progmemLen;
+    uint32_t filePointer;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceSD.cpp

@@ -0,0 +1,78 @@
+/*
+  AudioFileSourceSPIFFS
+  Input SD card "file" to be used by AudioGenerator
+  
+  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 "AudioFileSourceSD.h"
+
+AudioFileSourceSD::AudioFileSourceSD()
+{
+}
+
+AudioFileSourceSD::AudioFileSourceSD(const char *filename)
+{
+  open(filename);
+}
+
+bool AudioFileSourceSD::open(const char *filename)
+{
+  f = SD.open(filename, FILE_READ);
+  return f;
+}
+
+AudioFileSourceSD::~AudioFileSourceSD()
+{
+  if (f) f.close();
+}
+
+uint32_t AudioFileSourceSD::read(void *data, uint32_t len)
+{
+  return f.read(reinterpret_cast<uint8_t*>(data), len);
+}
+
+bool AudioFileSourceSD::seek(int32_t pos, int dir)
+{
+  if (!f) return false;
+  if (dir==SEEK_SET) return f.seek(pos);
+  else if (dir==SEEK_CUR) return f.seek(f.position() + pos);
+  else if (dir==SEEK_END) return f.seek(f.size() + pos);
+  return false;
+}
+
+bool AudioFileSourceSD::close()
+{
+  f.close();
+  return true;
+}
+
+bool AudioFileSourceSD::isOpen()
+{
+  return f?true:false;
+}
+
+uint32_t AudioFileSourceSD::getSize()
+{
+  if (!f) return 0;
+  return f.size();
+}
+
+uint32_t AudioFileSourceSD::getPos()
+{
+  if (!f) return 0;
+  return f.position();
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceSD.h

@@ -0,0 +1,49 @@
+/*
+  AudioFileSourceSPIFFS
+  Input SD card "file" to be used by AudioGenerator
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCESD_H
+#define _AUDIOFILESOURCESD_H
+
+#include "AudioFileSource.h"
+#include <SD.h>
+
+
+class AudioFileSourceSD : public AudioFileSource
+{
+  public:
+    AudioFileSourceSD();
+    AudioFileSourceSD(const char *filename);
+    virtual ~AudioFileSourceSD() override;
+    
+    virtual bool open(const char *filename) override;
+    virtual uint32_t read(void *data, uint32_t len) override;
+    virtual bool seek(int32_t pos, int dir) override;
+    virtual bool close() override;
+    virtual bool isOpen() override;
+    virtual uint32_t getSize() override;
+    virtual uint32_t getPos() override;
+
+  private:
+    File f;
+};
+
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceSPIFFS.h

@@ -0,0 +1,40 @@
+/*
+  AudioFileSourceFS
+  Input Arduion "file" to be used by AudioGenerator
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCESPIFFS_H
+#define _AUDIOFILESOURCESPIFFS_H
+
+#include <Arduino.h>
+#include <FS.h>
+
+#include "AudioFileSource.h"
+#include "AudioFileSourceFS.h"
+
+class AudioFileSourceSPIFFS : public AudioFileSourceFS
+{
+  public:
+    AudioFileSourceSPIFFS() : AudioFileSourceFS(SPIFFS) { };
+    AudioFileSourceSPIFFS(const char *filename) : AudioFileSourceFS(SPIFFS, filename) {};
+    // Others are inherited from base
+};
+
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceSPIRAMBuffer.cpp

@@ -0,0 +1,171 @@
+/*
+  AudioFileSourceSPIRAMBuffer
+  Buffered file source in external SPI RAM
+
+  Copyright (C) 2017  Sebastien Decourriere
+  Based on AudioFileSourceBuffer class from Earle F. Philhower, III
+
+  Copyright (C) 2020  Earle F. Philhower, III
+  Rewritten for speed and functionality
+
+  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/>.
+*/
+
+#if defined(ESP32) || defined(ESP8266)
+
+#include <Arduino.h>
+#include "AudioFileSourceSPIRAMBuffer.h"
+
+#pragma GCC optimize ("O3")
+
+AudioFileSourceSPIRAMBuffer::AudioFileSourceSPIRAMBuffer(AudioFileSource *source, uint8_t csPin, uint32_t buffSizeBytes)
+{
+    ram.begin(40, csPin);
+    ramSize = buffSizeBytes;
+    writePtr = 0;
+    readPtr = 0;
+    filled = false;
+    src = source;
+    audioLogger->printf_P(PSTR("SPI RAM buffer size: %u Bytes\n"), ramSize);
+}
+
+AudioFileSourceSPIRAMBuffer::~AudioFileSourceSPIRAMBuffer()
+{
+    ram.end();
+}
+
+bool AudioFileSourceSPIRAMBuffer::seek(int32_t pos, int dir)
+{
+    // Invalidate
+    readPtr = 0;
+    writePtr = 0;
+    filled = false;
+    return src->seek(pos, dir);
+}
+
+bool AudioFileSourceSPIRAMBuffer::close()
+{
+    return src->close();
+}
+
+bool AudioFileSourceSPIRAMBuffer::isOpen()
+{
+    return src->isOpen();
+}
+
+uint32_t AudioFileSourceSPIRAMBuffer::getSize()
+{
+    return src->getSize();
+}
+
+uint32_t AudioFileSourceSPIRAMBuffer::getPos()
+{
+    return src->getPos() - (writePtr - readPtr);
+}
+
+uint32_t AudioFileSourceSPIRAMBuffer::read(void *data, uint32_t len)
+{
+    uint32_t bytes = 0;
+
+    // Check if the buffer isn't empty, otherwise we try to fill completely
+    if (!filled) {
+        cb.st(999, PSTR("Filling buffer..."));
+        uint8_t buffer[256];
+        writePtr = 0;
+        readPtr = 0;
+        // Fill up completely before returning any data at all
+        do {
+            int toRead = std::min(ramSize - (writePtr - readPtr), sizeof(buffer));
+            int length = src->read(buffer, toRead);
+            if (length > 0) {
+#ifdef FAKERAM
+                for (size_t i=0; i<length; i++) fakeRAM[(i+writePtr)%ramSize] = buffer[i];
+#else
+                ram.writeBytes(writePtr % ramSize, buffer, length);
+#endif
+                writePtr += length;
+            } else {
+                // EOF, break out of read loop
+                break;
+            }
+        } while ((writePtr - readPtr) < ramSize);
+        filled = true;
+        cb.st(999, PSTR("Buffer filled..."));
+    }
+
+    // Read up to the entire buffer from RAM
+    uint32_t toReadFromBuffer = std::min(len, writePtr - readPtr);
+    uint8_t *ptr = reinterpret_cast<uint8_t*>(data);
+    if (toReadFromBuffer > 0) {
+#ifdef FAKERAM
+        for (size_t i=0; i<toReadFromBuffer; i++) ptr[i] = fakeRAM[(i+readPtr)%ramSize];
+#else
+        ram.readBytes(readPtr % ramSize, ptr, toReadFromBuffer);
+#endif
+        readPtr += toReadFromBuffer;
+        ptr += toReadFromBuffer;
+        bytes += toReadFromBuffer;
+        len -= toReadFromBuffer;
+    }
+
+    // If len>0 there is no data left in buffer and we try to read more directly from source.
+    // Then, we trigger a complete buffer refill
+    if (len) {
+        bytes += src->read(data, len);
+        filled = false;
+    }
+    return bytes;
+}
+
+void AudioFileSourceSPIRAMBuffer::fill()
+{
+    // Make sure the buffer is pre-filled before make partial fill.
+    if (!filled) return;
+
+    for (auto i=0; i<5; i++) {
+        // Make sure there is at least buffer size free in RAM
+        uint8_t buffer[128];
+        if ((ramSize - (writePtr - readPtr)) < sizeof(buffer)) {
+            return;
+        }
+
+        int cnt = src->readNonBlock(buffer, sizeof(buffer));
+        if (cnt) {
+#ifdef FAKERAM
+            for (size_t i=0; i<cnt; i++) fakeRAM[(i+writePtr)%ramSize] = buffer[i];
+#else
+            ram.writeBytes(writePtr % ramSize, buffer, cnt);
+#endif
+            writePtr += cnt;
+        }
+    }
+}
+
+bool AudioFileSourceSPIRAMBuffer::loop()
+{
+    static uint32_t last = 0;
+    if (!src->loop()) return false;
+    fill();
+    if ((ESP.getCycleCount() - last) > microsecondsToClockCycles(1000000)) {
+        last = ESP.getCycleCount();
+            char str[65];
+            memset(str, '#', 64);
+            str[64] = 0;
+            str[((writePtr - readPtr) * 64)/ramSize] = 0;
+            cb.st(((writePtr - readPtr) * 100)/ramSize, str);
+    }
+    return true;
+}
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceSPIRAMBuffer.h

@@ -0,0 +1,65 @@
+/*
+  AudioFileSourceSPIRAMBuffer
+  Buffered file source in external SPI RAM
+
+  Copyright (C) 2017  Sebastien Decourriere
+  Based on AudioFileSourceBuffer class from 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/>.
+*/
+
+#if defined(ESP32) || defined(ESP8266)
+#pragma once
+
+#include "AudioFileSource.h"
+#include <SPI.h>
+#include "spiram-fast.h"
+//#define FAKERAM
+// #define SPIBUF_DEBUG
+
+class AudioFileSourceSPIRAMBuffer : public AudioFileSource
+{
+  public:
+#ifdef FAKERAM
+    AudioFileSourceSPIRAMBuffer(AudioFileSource *in, uint8_t csPin = 15, uint32_t bufferBytes = 2048);
+#else
+    AudioFileSourceSPIRAMBuffer(AudioFileSource *in, uint8_t csPin = 15, uint32_t bufferBytes = 128*1024);
+#endif
+    virtual ~AudioFileSourceSPIRAMBuffer() override;
+
+    virtual uint32_t read(void *data, uint32_t len) override;
+    virtual bool seek(int32_t pos, int dir) override;
+    virtual bool close() override;
+    virtual bool isOpen() override;
+    virtual uint32_t getSize() override;
+    virtual uint32_t getPos() override;
+    virtual bool loop() override;
+
+  private:
+    virtual void fill();
+
+  private:
+    AudioFileSource *src;
+    ESP8266SPIRAM ram;
+    size_t ramSize;
+    size_t writePtr;
+    size_t readPtr;
+    bool filled;
+
+#ifdef FAKERAM
+    char fakeRAM[2048];
+#endif
+};
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceSTDIO.cpp

@@ -0,0 +1,98 @@
+/*
+  AudioFileSourceSTDIO
+  Input STDIO "file" to be used by AudioGenerator
+  Only for hot-based testing
+  
+  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>
+#ifndef ARDUINO
+#include <time.h>
+
+#include "AudioFileSourceSTDIO.h"
+
+AudioFileSourceSTDIO::AudioFileSourceSTDIO()
+{
+  f = NULL;
+  srand(time(NULL));
+}
+
+AudioFileSourceSTDIO::AudioFileSourceSTDIO(const char *filename)
+{
+  open(filename);
+}
+
+bool AudioFileSourceSTDIO::open(const char *filename)
+{
+  f = fopen(filename, "rb");
+  return f;
+}
+
+AudioFileSourceSTDIO::~AudioFileSourceSTDIO()
+{
+  if (f) fclose(f);
+  f = NULL;
+}
+
+uint32_t AudioFileSourceSTDIO::read(void *data, uint32_t len)
+{
+//  if (rand() % 100 == 69) { // Give 0 data 1%
+//    printf("0 read\n");
+//    len = 0;
+//  } else if (rand() % 100 == 1) { // Give short reads 1%
+//    printf("0 read\n");
+//    len = 0;
+//  }
+  int ret = fread(reinterpret_cast<uint8_t*>(data), 1, len, f);
+//  if (ret && rand() % 100 < 5 ) {
+//    // We're really mean...throw bad data in the mix
+//    printf("bad data\n");
+//    for (int i=0; i<100; i++)
+//      *(reinterpret_cast<uint8_t*>(data) + (rand() % ret)) = rand();
+//  }
+  return ret;
+}
+
+bool AudioFileSourceSTDIO::seek(int32_t pos, int dir)
+{
+  return fseek(f, pos, dir) == 0;
+}
+
+bool AudioFileSourceSTDIO::close()
+{
+  fclose(f);
+  f = NULL;
+  return true;
+}
+
+bool AudioFileSourceSTDIO::isOpen()
+{
+  return f?true:false;
+}
+
+uint32_t AudioFileSourceSTDIO::getSize()
+{
+  if (!f) return 0;
+  uint32_t p = ftell(f);
+  fseek(f, 0, SEEK_END);
+  uint32_t len = ftell(f);
+  fseek(f, p, SEEK_SET);
+  return len;
+}
+
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileSourceSTDIO.h

@@ -0,0 +1,53 @@
+/*
+  AudioFileSourceSTDIO
+  Input SPIFFS "file" to be used by AudioGenerator
+  Only for host-based testing, not Arduino
+  
+  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/>.
+*/
+
+#ifndef _AUDIOFILESOURCESTDIO_H
+#define _AUDIOFILESOURCESTDIO_H
+
+#include <Arduino.h>
+
+#ifndef ARDUINO
+
+#include "AudioFileSource.h"
+
+class AudioFileSourceSTDIO : public AudioFileSource
+{
+  public:
+    AudioFileSourceSTDIO();
+    AudioFileSourceSTDIO(const char *filename);
+    virtual ~AudioFileSourceSTDIO() override;
+    
+    virtual bool open(const char *filename) override;
+    virtual uint32_t read(void *data, uint32_t len) override;
+    virtual bool seek(int32_t pos, int dir) override;
+    virtual bool close() override;
+    virtual bool isOpen() override;
+    virtual uint32_t getSize() override;
+    virtual uint32_t getPos() override { if (!f) return 0; else return (uint32_t)ftell(f); };
+
+  private:
+    FILE *f;
+};
+
+#endif // !ARDUINO
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileStream.cpp

@@ -0,0 +1,61 @@
+#include <Arduino.h>
+#include "AudioFileStream.h"
+
+
+AudioFileStream::AudioFileStream(AudioFileSource *source, int definedLen)
+{
+  src = source;
+  len = definedLen;
+  ptr = 0;
+  saved = -1;
+}
+
+AudioFileStream::~AudioFileStream()
+{
+  // If there's a defined len, read until we're empty
+  if (len) {
+    while (ptr++ < len) (void)read();
+  }
+}
+
+
+int AudioFileStream::available()
+{
+  if (saved >= 0) return 1;
+  else if (len) return ptr - len;
+  else if (src->getSize()) return (src->getPos() - src->getSize());
+  else return 1;
+}
+
+int AudioFileStream::read()
+{
+  uint8_t c;
+  int r;
+  if (ptr >= len) return -1;
+  ptr++;
+  if (saved >= 0) {
+    c = (uint8_t)saved;
+    saved = -1;
+    r = 1;
+  } else {
+    r = src->read(&c, 1);
+  }
+  if (r != 1) return -1;
+  return (int)c;
+}
+
+int AudioFileStream::peek()
+{
+  uint8_t c;
+  if ((ptr+1) >= len) return -1;
+  if (saved >= 0) return saved;
+  int r = src->read(&c, 1);
+  if (r<1) return -1;
+  saved = c;
+  return saved;
+}
+
+void AudioFileStream::flush()
+{
+  /* noop? */
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioFileStream.h

@@ -0,0 +1,48 @@
+/*
+  AudioFileStream
+  Convert an AudioFileSource* to a Stream*
+  
+  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/>.
+*/
+
+#ifndef AUDIOFILESTREAM_H
+#define AUDIOFILESTREAM_H
+
+#include <Arduino.h>
+#include "AudioFileSource.h"
+
+class AudioFileStream : public Stream
+{
+public:
+  AudioFileStream(AudioFileSource *source, int definedLen);
+  virtual ~AudioFileStream();
+
+public:
+  // Stream interface - see the Arduino library documentation.
+  virtual int available() override;
+  virtual int read() override;
+  virtual int peek() override;
+  virtual void flush() override;
+  virtual size_t write(uint8_t x) override { (void)x; return 0; };
+
+private:
+  AudioFileSource *src;
+  int saved;
+  int len;
+  int ptr;
+};
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGenerator.h

@@ -0,0 +1,54 @@
+/*
+  AudioGenerator
+  Base class of an audio output generator
+  
+  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/>.
+*/
+
+#ifndef _AUDIOGENERATOR_H
+#define _AUDIOGENERATOR_H
+
+#include <Arduino.h>
+#include "AudioStatus.h"
+#include "AudioFileSource.h"
+#include "AudioOutput.h"
+
+class AudioGenerator
+{
+  public:
+    AudioGenerator() { lastSample[0] = 0; lastSample[1] = 0; };
+    virtual ~AudioGenerator() {};
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) { (void)source; (void)output; return false; };
+    virtual bool loop() { return false; };
+    virtual bool stop() { return false; };
+    virtual bool isRunning() { return false;};
+    virtual void desync () { };
+
+  public:
+    virtual bool RegisterMetadataCB(AudioStatus::metadataCBFn fn, void *data) { return cb.RegisterMetadataCB(fn, data); }
+    virtual bool RegisterStatusCB(AudioStatus::statusCBFn fn, void *data) { return cb.RegisterStatusCB(fn, data); }
+
+  protected:
+    bool running;
+    AudioFileSource *file;
+    AudioOutput *output;
+    int16_t lastSample[2];
+
+  protected:
+    AudioStatus cb;
+};
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorAAC.cpp

@@ -0,0 +1,219 @@
+/*
+  AudioGeneratorAAC
+  Audio output generator using the Helix AAC decoder
+  
+  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/>.
+*/
+
+#pragma GCC optimize ("O3")
+
+#include "AudioGeneratorAAC.h"
+
+AudioGeneratorAAC::AudioGeneratorAAC()
+{
+  preallocateSpace = NULL;
+  preallocateSize = 0;
+
+  running = false;
+  file = NULL;
+  output = NULL;
+
+  buff = (uint8_t*)malloc(buffLen);
+  outSample = (int16_t*)malloc(1024 * 2 * sizeof(uint16_t));
+  if (!buff || !outSample) {
+    audioLogger->printf_P(PSTR("ERROR: Out of memory in AAC\n"));
+    Serial.flush();
+  }
+
+  hAACDecoder = AACInitDecoder();
+  if (!hAACDecoder) {
+    audioLogger->printf_P(PSTR("Out of memory error! hAACDecoder==NULL\n"));
+    Serial.flush();
+  }
+
+  buffValid = 0;
+  lastFrameEnd = 0;
+  validSamples = 0;
+  curSample = 0;
+  lastRate = 0;
+  lastChannels = 0;
+}
+
+AudioGeneratorAAC::AudioGeneratorAAC(void *preallocateData, int preallocateSz)
+{
+  preallocateSpace = preallocateData;
+  preallocateSize = preallocateSz;
+
+  running = false;
+  file = NULL;
+  output = NULL;
+
+  uint8_t *p = (uint8_t*)preallocateSpace;
+  buff = (uint8_t*) p;
+  p += (buffLen + 7) & ~7;
+  outSample = (int16_t*) p;
+  p += (1024 * 2 * sizeof(int16_t) + 7) & ~7;
+  int used = p - (uint8_t*)preallocateSpace;
+  int availSpace = preallocateSize - used;
+  if (availSpace < 0 ) {
+    audioLogger->printf_P(PSTR("ERROR: Out of memory in AAC\n"));
+  }
+
+  hAACDecoder = AACInitDecoderPre(p, availSpace);
+  if (!hAACDecoder) {
+    audioLogger->printf_P(PSTR("Out of memory error! hAACDecoder==NULL\n"));
+    Serial.flush();
+  }
+  buffValid = 0;
+  lastFrameEnd = 0;
+  validSamples = 0;
+  curSample = 0;
+  lastRate = 0;
+  lastChannels = 0;
+}
+
+
+
+AudioGeneratorAAC::~AudioGeneratorAAC()
+{
+  if (!preallocateSpace) {
+    AACFreeDecoder(hAACDecoder);
+    free(buff);
+    free(outSample);
+  }
+}
+
+bool AudioGeneratorAAC::stop()
+{
+  running = false;
+  output->stop();
+  return file->close();
+}
+
+bool AudioGeneratorAAC::isRunning()
+{
+  return running;
+}
+
+bool AudioGeneratorAAC::FillBufferWithValidFrame()
+{
+  buff[0] = 0; // Destroy any existing sync word @ 0
+  int nextSync;
+  do {
+    nextSync = AACFindSyncWord(buff + lastFrameEnd, buffValid - lastFrameEnd);
+    if (nextSync >= 0) nextSync += lastFrameEnd;
+    lastFrameEnd = 0;
+    if (nextSync == -1) {
+      if (buffValid && buff[buffValid-1]==0xff) { // Could be 1st half of syncword, preserve it...
+        buff[0] = 0xff;
+        buffValid = file->read(buff+1, buffLen-1);
+        if (buffValid==0) return false; // No data available, EOF
+      } else { // Try a whole new buffer
+        buffValid = file->read(buff, buffLen-1);
+        if (buffValid==0) return false; // No data available, EOF
+      }
+    }
+  } while (nextSync == -1);
+
+  // Move the frame to start at offset 0 in the buffer
+  buffValid -= nextSync; // Throw out prior to nextSync
+  memmove(buff, buff+nextSync, buffValid);
+
+  // We have a sync word at 0 now, try and fill remainder of buffer
+  buffValid += file->read(buff + buffValid, buffLen - buffValid);
+
+  return true;
+}
+
+bool AudioGeneratorAAC::loop()
+{
+  if (!running) goto done; // Nothing to do here!
+
+  // If we've got data, try and pump it out...
+  while (validSamples) {
+    if (lastChannels == 1) {
+       lastSample[0] = outSample[curSample];
+       lastSample[1] = outSample[curSample];
+    } else {
+      lastSample[0] = outSample[curSample*2];
+      lastSample[1] = outSample[curSample*2 + 1];
+    }
+    if (!output->ConsumeSample(lastSample)) goto done; // Can't send, but no error detected
+    validSamples--;
+    curSample++;
+  }
+
+  // No samples available, need to decode a new frame
+  if (FillBufferWithValidFrame()) {
+    // buff[0] start of frame, decode it...
+    unsigned char *inBuff = reinterpret_cast<unsigned char *>(buff);
+    int bytesLeft = buffValid;
+    int ret = AACDecode(hAACDecoder, &inBuff, &bytesLeft, outSample);
+    if (ret) {
+      // Error, skip the frame...
+      char buff[48];
+      sprintf_P(buff, PSTR("AAC decode error %d"), ret);
+      cb.st(ret, buff);
+    } else {
+      lastFrameEnd = buffValid - bytesLeft;
+      AACFrameInfo fi;
+      AACGetLastFrameInfo(hAACDecoder, &fi);
+      if ((int)fi.sampRateOut != (int)lastRate) {
+        output->SetRate(fi.sampRateOut);
+        lastRate = fi.sampRateOut;
+      }
+      if (fi.nChans != lastChannels) {
+        output->SetChannels(fi.nChans);
+        lastChannels = fi.nChans;
+      }
+      curSample = 0;
+      validSamples = fi.outputSamps / lastChannels;
+    }
+  } else {
+    running = false; // No more data, we're done here...
+  }
+
+done:
+  file->loop();
+  output->loop();
+
+  return running;
+}
+
+bool AudioGeneratorAAC::begin(AudioFileSource *source, AudioOutput *output)
+{
+  if (!source) return false;
+  file = source;
+  if (!output) return false;
+  this->output = output;
+  if (!file->isOpen()) return false; // Error
+
+  output->begin();
+  
+  // AAC always comes out at 16 bits
+  output->SetBitsPerSample(16);
+ 
+
+  memset(buff, 0, buffLen);
+  memset(outSample, 0, 1024*2*sizeof(int16_t));
+
+ 
+  running = true;
+  
+  return true;
+}
+
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorAAC.h

@@ -0,0 +1,64 @@
+/*
+  AudioGeneratorAAC
+  Audio output generator using the Helix AAC decoder
+  
+  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/>.
+*/
+
+#ifndef _AUDIOGENERATORAAC_H
+#define _AUDIOGENERATORAAC_H
+
+#include "AudioGenerator.h"
+#include "libhelix-aac/aacdec.h"
+
+class AudioGeneratorAAC : public AudioGenerator
+{
+  public:
+    AudioGeneratorAAC();
+    AudioGeneratorAAC(void *preallocateData, int preallocateSize);
+    virtual ~AudioGeneratorAAC() override;
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override;
+
+  protected:
+    void *preallocateSpace;
+    int preallocateSize;
+
+    // Helix AAC decoder
+    HAACDecoder hAACDecoder;
+
+    // Input buffering
+    const int buffLen = 1600;
+    uint8_t *buff; //[1600]; // File buffer required to store at least a whole compressed frame
+    int16_t buffValid;
+    int16_t lastFrameEnd;
+    bool FillBufferWithValidFrame(); // Read until we get a valid syncword and min(feof, 2048) butes in the buffer
+
+    // Output buffering
+    int16_t *outSample; //[1024 * 2]; // Interleaved L/R
+    int16_t validSamples;
+    int16_t curSample;
+
+    // Each frame may change this if they're very strange, I guess
+    unsigned int lastRate;
+    int lastChannels;
+
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorFLAC.cpp

@@ -0,0 +1,199 @@
+/*
+  AudioGeneratorFLAC
+  Audio output generator that plays FLAC audio files
+    
+  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 <AudioGeneratorFLAC.h>
+
+AudioGeneratorFLAC::AudioGeneratorFLAC()
+{
+  flac = NULL;
+  channels = 0;
+  sampleRate = 0;
+  bitsPerSample = 0;
+  buff[0] = NULL;
+  buff[1] = NULL;
+  buffPtr = 0;
+  buffLen = 0;
+  running = false;
+}
+
+AudioGeneratorFLAC::~AudioGeneratorFLAC()
+{
+  if (flac)
+    FLAC__stream_decoder_delete(flac);
+  flac = NULL;
+}
+
+bool AudioGeneratorFLAC::begin(AudioFileSource *source, AudioOutput *output)
+{
+  if (!source) return false;
+  file = source;
+  if (!output) return false;
+  this->output = output;
+  if (!file->isOpen()) return false; // Error
+
+  flac = FLAC__stream_decoder_new();
+  if (!flac) return false;
+  
+  (void)FLAC__stream_decoder_set_md5_checking(flac, false);
+
+  FLAC__StreamDecoderInitStatus ret = FLAC__stream_decoder_init_stream(flac, _read_cb, _seek_cb, _tell_cb, _length_cb, _eof_cb, _write_cb, _metadata_cb, _error_cb, reinterpret_cast<void*>(this) );
+  if (ret != FLAC__STREAM_DECODER_INIT_STATUS_OK) {
+    FLAC__stream_decoder_delete(flac);
+    flac = NULL;
+    return false;
+  }
+
+  output->begin();
+  running = true;
+  return true;
+}
+
+bool AudioGeneratorFLAC::loop()
+{
+  FLAC__bool ret;
+
+  if (!running) goto done;
+
+  if (!output->ConsumeSample(lastSample)) goto done; // Try and send last buffered sample
+
+  do {
+    if (buffPtr == buffLen) {
+      ret = FLAC__stream_decoder_process_single(flac);
+      if (!ret) {
+        running = false;
+        goto done;
+      } else {
+        // We might be done...
+        if (FLAC__stream_decoder_get_state(flac)==FLAC__STREAM_DECODER_END_OF_STREAM) {
+          running = false;
+          goto done;
+        }
+        unsigned newsr = FLAC__stream_decoder_get_sample_rate(flac);
+        unsigned newch = FLAC__stream_decoder_get_channels(flac);
+        unsigned newbps = FLAC__stream_decoder_get_bits_per_sample(flac);
+        if (newsr != sampleRate) output->SetRate(sampleRate = newsr);
+        if (newch != channels) output->SetChannels(channels = newch);
+        if (newbps != bitsPerSample) output->SetBitsPerSample( bitsPerSample = newbps);
+      }
+    }
+
+    // Check for some weird case where above didn't give any data
+    if (buffPtr == buffLen) {
+      goto done; // At some point the flac better error and we'll return 
+    }
+    if (bitsPerSample <= 16) {
+      lastSample[AudioOutput::LEFTCHANNEL] = buff[0][buffPtr] & 0xffff; 
+      if (channels==2) lastSample[AudioOutput::RIGHTCHANNEL] = buff[1][buffPtr] & 0xffff; 
+      else lastSample[AudioOutput::RIGHTCHANNEL] = lastSample[AudioOutput::LEFTCHANNEL];
+    } else if (bitsPerSample <= 24) {
+      lastSample[AudioOutput::LEFTCHANNEL] = (buff[0][buffPtr]>>8) & 0xffff; 
+      if (channels==2) lastSample[AudioOutput::RIGHTCHANNEL] = (buff[1][buffPtr]>>8) & 0xffff; 
+      else lastSample[AudioOutput::RIGHTCHANNEL] = lastSample[AudioOutput::LEFTCHANNEL];
+    } else {
+      lastSample[AudioOutput::LEFTCHANNEL] = (buff[0][buffPtr]>>16) & 0xffff; 
+      if (channels==2) lastSample[AudioOutput::RIGHTCHANNEL] = (buff[1][buffPtr]>>16) & 0xffff; 
+      else lastSample[AudioOutput::RIGHTCHANNEL] = lastSample[AudioOutput::LEFTCHANNEL];
+    }
+    buffPtr++;
+  } while (running && output->ConsumeSample(lastSample));
+
+done:
+  file->loop();
+  output->loop();
+
+  return running;
+}
+
+bool AudioGeneratorFLAC::stop()
+{
+  if (flac)
+    FLAC__stream_decoder_delete(flac);
+  flac = NULL;
+  running = false;
+  output->stop();
+  return true;
+}
+
+bool AudioGeneratorFLAC::isRunning()
+{
+  return running;
+}
+
+
+
+FLAC__StreamDecoderReadStatus AudioGeneratorFLAC::read_cb(const FLAC__StreamDecoder *decoder, FLAC__byte buffer[], size_t *bytes)
+{
+  (void) decoder;
+  if (*bytes==0) return FLAC__STREAM_DECODER_READ_STATUS_ABORT;
+  *bytes = file->read(buffer, sizeof(FLAC__byte) * (*bytes));
+  if (*bytes==0) return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
+  return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
+}
+FLAC__StreamDecoderSeekStatus AudioGeneratorFLAC::seek_cb(const FLAC__StreamDecoder *decoder, FLAC__uint64 absolute_byte_offset)
+{
+  (void) decoder;
+  if (!file->seek((int32_t)absolute_byte_offset, 0)) return FLAC__STREAM_DECODER_SEEK_STATUS_ERROR;
+  return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
+}
+FLAC__StreamDecoderTellStatus AudioGeneratorFLAC::tell_cb(const FLAC__StreamDecoder *decoder, FLAC__uint64 *absolute_byte_offset)
+{
+  (void) decoder;
+  *absolute_byte_offset = file->getPos();
+  return FLAC__STREAM_DECODER_TELL_STATUS_OK;
+}
+
+FLAC__StreamDecoderLengthStatus AudioGeneratorFLAC::length_cb(const FLAC__StreamDecoder *decoder, FLAC__uint64 *stream_length)
+{
+  (void) decoder;
+  *stream_length = file->getSize();
+  return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
+}
+FLAC__bool AudioGeneratorFLAC::eof_cb(const FLAC__StreamDecoder *decoder)
+{
+  (void) decoder;
+  if (file->getPos() >= file->getSize()) return true;
+  return false;
+}
+FLAC__StreamDecoderWriteStatus AudioGeneratorFLAC::write_cb(const FLAC__StreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 *const buffer[])
+{
+  (void) decoder;
+  // Hackish warning here.  FLAC sends the buffer but doesn't free it until the next call to decode_frame, so we stash
+  // the pointers here and use it in our loop() instead of memcpy()'ing into yet another buffer.
+  buffLen = frame->header.blocksize;
+  buff[0] = (const int *)buffer[0];
+  if (frame->header.channels>1) buff[1] = (const int *)buffer[1];
+  else buff[1] = (const int *)buffer[0];
+  buffPtr = 0;
+  return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+}
+void AudioGeneratorFLAC::metadata_cb(const FLAC__StreamDecoder *decoder, const FLAC__StreamMetadata *metadata)
+{
+  (void) decoder;
+  (void) metadata;
+  audioLogger->printf_P(PSTR("Metadata\n"));
+}
+char AudioGeneratorFLAC::error_cb_str[64];
+void AudioGeneratorFLAC::error_cb(const FLAC__StreamDecoder *decoder, FLAC__StreamDecoderErrorStatus status)
+{
+  (void) decoder;
+  strncpy_P(error_cb_str, FLAC__StreamDecoderErrorStatusString[status], 64);
+  cb.st((int)status, error_cb_str);
+}
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorFLAC.h

@@ -0,0 +1,89 @@
+/*
+  AudioGeneratorFLAC
+  Audio output generator that plays FLAC audio files
+    
+  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/>.
+*/
+
+#ifndef _AUDIOGENERATORFLAC_H
+#define _AUDIOGENERATORFLAC_H
+
+#include <AudioGenerator.h>
+extern "C" {
+    #include "libflac/FLAC/stream_decoder.h"
+};
+
+class AudioGeneratorFLAC : public AudioGenerator
+{
+  public:
+    AudioGeneratorFLAC();
+    virtual ~AudioGeneratorFLAC() override;
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override;
+
+  protected:
+    // FLAC info
+    uint16_t channels;
+    uint32_t sampleRate;
+    uint16_t bitsPerSample;
+
+    // We need to buffer some data in-RAM to avoid doing 1000s of small reads
+    const int *buff[2];
+    uint16_t buffPtr;
+    uint16_t buffLen;
+    FLAC__StreamDecoder *flac;
+
+    // FLAC callbacks, need static functions to bounce into c++ from c
+    static FLAC__StreamDecoderReadStatus _read_cb(const FLAC__StreamDecoder *decoder, FLAC__byte buffer[], size_t *bytes, void *client_data) {
+      return static_cast<AudioGeneratorFLAC*>(client_data)->read_cb(decoder, buffer, bytes);
+    };
+    static FLAC__StreamDecoderSeekStatus _seek_cb(const FLAC__StreamDecoder *decoder, FLAC__uint64 absolute_byte_offset, void *client_data) {
+      return static_cast<AudioGeneratorFLAC*>(client_data)->seek_cb(decoder, absolute_byte_offset);
+    };
+    static FLAC__StreamDecoderTellStatus _tell_cb(const FLAC__StreamDecoder *decoder, FLAC__uint64 *absolute_byte_offset, void *client_data) {
+      return static_cast<AudioGeneratorFLAC*>(client_data)->tell_cb(decoder, absolute_byte_offset);
+    };
+    static FLAC__StreamDecoderLengthStatus _length_cb(const FLAC__StreamDecoder *decoder, FLAC__uint64 *stream_length, void *client_data) {
+      return static_cast<AudioGeneratorFLAC*>(client_data)->length_cb(decoder, stream_length);
+    };
+    static FLAC__bool _eof_cb(const FLAC__StreamDecoder *decoder, void *client_data) {
+      return static_cast<AudioGeneratorFLAC*>(client_data)->eof_cb(decoder);
+    };
+    static FLAC__StreamDecoderWriteStatus _write_cb(const FLAC__StreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 *const buffer[], void *client_data) {
+      return static_cast<AudioGeneratorFLAC*>(client_data)->write_cb(decoder, frame, buffer);
+    };
+    static void _metadata_cb(const FLAC__StreamDecoder *decoder, const FLAC__StreamMetadata *metadata, void *client_data) {
+      static_cast<AudioGeneratorFLAC*>(client_data)->metadata_cb(decoder, metadata);
+    };
+    static void _error_cb(const FLAC__StreamDecoder *decoder, FLAC__StreamDecoderErrorStatus status, void *client_data) {
+      static_cast<AudioGeneratorFLAC*>(client_data)->error_cb(decoder, status);
+    };
+    // Actual FLAC callbacks
+    FLAC__StreamDecoderReadStatus read_cb(const FLAC__StreamDecoder *decoder, FLAC__byte buffer[], size_t *bytes);
+    FLAC__StreamDecoderSeekStatus seek_cb(const FLAC__StreamDecoder *decoder, FLAC__uint64 absolute_byte_offset);
+    FLAC__StreamDecoderTellStatus tell_cb(const FLAC__StreamDecoder *decoder, FLAC__uint64 *absolute_byte_offset);
+    FLAC__StreamDecoderLengthStatus length_cb(const FLAC__StreamDecoder *decoder, FLAC__uint64 *stream_length);
+    FLAC__bool eof_cb(const FLAC__StreamDecoder *decoder);
+    FLAC__StreamDecoderWriteStatus write_cb(const FLAC__StreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 *const buffer[]);
+    void metadata_cb(const FLAC__StreamDecoder *decoder, const FLAC__StreamMetadata *metadata);
+    static char error_cb_str[64];
+    void error_cb(const FLAC__StreamDecoder *decoder, FLAC__StreamDecoderErrorStatus status);
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorMIDI.cpp

@@ -0,0 +1,639 @@
+/*
+  AudioGeneratorMIDI
+  Audio output generator that plays MIDI files using a SF2 SoundFont
+    
+  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/>.
+*/
+
+/*
+  The MIDI processing engine is a heavily modified version of MIDITONES,
+  by Len Shustek, https://github.com/LenShustek/miditones .
+  Whereas MIDITONES original simply parsed a file beforehand to a byte
+  stream to be played by another program, this does the parsing and
+  playback in real-time.
+
+   Here's his original header/readme w/MIT license, which is subsumed by the
+   GPL license of the ESP8266Audio project.
+*/
+
+/***************************************************************************
+
+   MIDITONES: Convert a MIDI file into a simple bytestream of notes
+
+  -------------------------------------------------------------------------
+  The MIT License (MIT)
+  Copyright (c) 2011,2013,2015,2016, Len Shustek
+
+  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 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.
+**************************************************************************/
+
+
+#include "AudioGeneratorMIDI.h"
+
+#pragma GCC optimize ("O3")
+
+#define TSF_NO_STDIO
+#define TSF_IMPLEMENTATION
+#include "libtinysoundfont/tsf.h"
+
+/****************  utility routines  **********************/
+
+/* announce a fatal MIDI file format error */
+
+void AudioGeneratorMIDI::midi_error(const char *msg, int curpos)
+{
+  cb.st(curpos, msg);
+#if 0
+  int ptr;
+  audioLogger->printf("---> MIDI file error at position %04X (%d): %s\n", (uint16_t) curpos, (uint16_t) curpos, msg);
+  /* print some bytes surrounding the error */
+  ptr = curpos - 16;
+  if (ptr < 0) ptr = 0;
+  buffer.seek( buffer.data, ptr );
+  for (int i = 0; i < 32; i++) {
+    char c;
+    buffer.read (buffer.data, &c, 1);
+    audioLogger->printf((ptr + i) == curpos ? " [%02X]  " : "%02X ", (int) c & 0xff);
+  }
+  audioLogger->printf("\n");
+#endif
+  running = false;
+}
+
+/* check that we have a specified number of bytes left in the buffer */
+
+void AudioGeneratorMIDI::chk_bufdata (int ptr, unsigned long int len) {
+  if ((unsigned) (ptr + len) > buflen)
+    midi_error ("data missing", ptr);
+}
+
+/* fetch big-endian numbers */
+
+uint16_t AudioGeneratorMIDI::rev_short (uint16_t val) {
+  return ((val & 0xff) << 8) | ((val >> 8) & 0xff);
+}
+
+uint32_t AudioGeneratorMIDI::rev_long (uint32_t val) {
+  return (((rev_short ((uint16_t) val) & 0xffff) << 16) |
+          (rev_short ((uint16_t) (val >> 16)) & 0xffff));
+}
+
+/**************  process the MIDI file header  *****************/
+
+void AudioGeneratorMIDI::process_header (void) {
+  struct midi_header hdr;
+  unsigned int time_division;
+
+  chk_bufdata (hdrptr, sizeof (struct midi_header));
+  buffer.seek (buffer.data, hdrptr);
+  buffer.read (buffer.data, &hdr, sizeof (hdr));
+  if (!charcmp ((char *) hdr.MThd, "MThd"))
+    midi_error ("Missing 'MThd'", hdrptr);
+  num_tracks = rev_short (hdr.number_of_tracks);
+  time_division = rev_short (hdr.time_division);
+  if (time_division < 0x8000)
+    ticks_per_beat = time_division;
+  else
+    ticks_per_beat = ((time_division >> 8) & 0x7f) /* SMTE frames/sec */ *(time_division & 0xff);     /* ticks/SMTE frame */
+  hdrptr += rev_long (hdr.header_size) + 8;    /* point past header to track header, presumably. */
+  return;
+}
+
+
+/****************  Process a MIDI track header *******************/
+
+void AudioGeneratorMIDI::start_track (int tracknum) {
+  struct track_header hdr;
+  unsigned long tracklen;
+
+  chk_bufdata (hdrptr, sizeof (struct track_header));
+  buffer.seek (buffer.data, hdrptr);
+  buffer.read (buffer.data, &hdr, sizeof (hdr));
+  if (!charcmp ((char *) (hdr.MTrk), "MTrk"))
+    midi_error ("Missing 'MTrk'", hdrptr);
+  tracklen = rev_long (hdr.track_size);
+  hdrptr += sizeof (struct track_header);      /* point past header */
+  chk_bufdata (hdrptr, tracklen);
+  track[tracknum].trkptr = hdrptr;
+  hdrptr += tracklen;          /* point to the start of the next track */
+  track[tracknum].trkend = hdrptr;     /* the point past the end of the track */
+}
+
+unsigned char AudioGeneratorMIDI::buffer_byte (int offset) {
+  unsigned char c;
+  buffer.seek (buffer.data, offset);
+  buffer.read (buffer.data, &c, 1);
+  return c;
+}
+
+unsigned short AudioGeneratorMIDI::buffer_short (int offset) {
+  unsigned short s;
+  buffer.seek (buffer.data, offset);
+  buffer.read (buffer.data, &s, sizeof (short));
+  return s;
+}
+
+unsigned int AudioGeneratorMIDI::buffer_int32 (int offset) {
+  uint32_t i;
+  buffer.seek (buffer.data, offset);
+  buffer.read (buffer.data, &i, sizeof (i));
+  return i;
+}
+
+/* Get a MIDI-style variable-length integer */
+
+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 */
+
+  unsigned long val;
+  int i, byte;
+
+  val = 0;
+  for (i = 0; i < 4; ++i) {
+    byte = buffer_byte ((*ptr)++);
+    val = (val << 7) | (byte & 0x7f);
+    if (!(byte & 0x80))
+      return val;
+  }
+  return val;
+}
+
+
+/***************  Process the MIDI track data  ***************************/
+
+/* Skip in the track for the next "note on", "note off" or "set tempo" command,
+  then record that information in the track status block and return. */
+
+void AudioGeneratorMIDI::find_note (int tracknum) {
+  unsigned long int delta_time;
+  int event, chan;
+  int note, velocity, controller, pressure, pitchbend, instrument;
+  int meta_cmd, meta_length;
+  unsigned long int sysex_length;
+  struct track_status *t;
+  const char *tag;
+
+  /* process events */
+
+  t = &track[tracknum];        /* our track status structure */
+  while (t->trkptr < t->trkend) {
+
+    delta_time = get_varlen (&t->trkptr);
+    t->time += delta_time;
+    if (buffer_byte (t->trkptr) < 0x80)
+      event = t->last_event; /* using "running status": same event as before */
+    else {                    /* otherwise get new "status" (event type) */
+      event = buffer_byte (t->trkptr++);
+    }
+    if (event == 0xff) {      /* meta-event */
+      meta_cmd = buffer_byte (t->trkptr++);
+      meta_length = get_varlen(&t->trkptr);
+      switch (meta_cmd) {
+        case 0x00:
+          break;
+        case 0x01:
+          tag = "description";
+          goto show_text;
+        case 0x02:
+          tag = "copyright";
+          goto show_text;
+        case 0x03:
+          tag = "track name";
+          goto show_text;
+        case 0x04:
+          tag = "instrument name";
+          goto show_text;
+        case 0x05:
+          tag = "lyric";
+          goto show_text;
+        case 0x06:
+          tag = "marked point";
+          goto show_text;
+        case 0x07:
+          tag = "cue point";
+show_text:
+          break;
+        case 0x20:
+          break;
+        case 0x2f:
+          break;
+        case 0x51:            /* tempo: 3 byte big-endian integer! */
+          t->cmd = CMD_TEMPO;
+          t->tempo = rev_long (buffer_int32 (t->trkptr - 1)) & 0xffffffL;
+          t->trkptr += meta_length;
+          return;
+        case 0x54:
+          break;
+        case 0x58:
+          break;
+        case 0x59:
+          break;
+        case 0x7f:
+          tag = "sequencer data";
+          goto show_hex;
+        default:              /* unknown meta command */
+          tag = "???";
+show_hex:
+          break;
+      }
+      t->trkptr += meta_length;
+    }
+
+    else if (event < 0x80)
+      midi_error ("Unknown MIDI event type", t->trkptr);
+
+    else {
+      if (event < 0xf0)
+        t->last_event = event;      // remember "running status" if not meta or sysex event
+      chan = event & 0xf;
+      t->chan = chan;
+      switch (event >> 4) {
+        case 0x8:
+          t->note = buffer_byte (t->trkptr++);
+          velocity = buffer_byte (t->trkptr++);
+note_off:
+          t->cmd = CMD_STOPNOTE;
+          return;             /* stop processing and return */
+        case 0x9:
+          t->note = buffer_byte (t->trkptr++);
+          velocity = buffer_byte (t->trkptr++);
+          if (velocity == 0)  /* some scores use note-on with zero velocity for off! */
+            goto note_off;
+          t->velocity = velocity;
+          t->cmd = CMD_PLAYNOTE;
+          return;             /* stop processing and return */
+        case 0xa:
+          note = buffer_byte (t->trkptr++);
+          velocity = buffer_byte (t->trkptr++);
+          break;
+        case 0xb:
+          controller = buffer_byte (t->trkptr++);
+          velocity = buffer_byte (t->trkptr++);
+          break;
+        case 0xc:
+          instrument = buffer_byte (t->trkptr++);
+          midi_chan_instrument[chan] = instrument;    // record new instrument for this channel
+          break;
+        case 0xd:
+          pressure = buffer_byte (t->trkptr++);
+          break;
+        case 0xe:
+          pitchbend = buffer_byte (t->trkptr) | (buffer_byte (t->trkptr + 1) << 7);
+          t->trkptr += 2;
+          break;
+        case 0xf:
+          sysex_length = get_varlen (&t->trkptr);
+          t->trkptr += sysex_length;
+          break;
+        default:
+          midi_error ("Unknown MIDI command", t->trkptr);
+      }
+    }
+  }
+  t->cmd = CMD_TRACKDONE;      /* no more notes to process */
+  ++tracks_done;
+
+  // Remove unused warnings..maybe some day we'll look at these
+  (void)note;
+  (void)controller;
+  (void)pressure;
+  (void)pitchbend;
+  (void)tag;
+}
+
+
+// Open file, parse headers, get ready tio process MIDI
+void AudioGeneratorMIDI::PrepareMIDI(AudioFileSource *src)
+{
+  MakeStreamFromAFS(src, &afsMIDI);
+  tsf_stream_wrap_cached(&afsMIDI, 32, 64, &buffer);
+  buflen = buffer.size (buffer.data);
+
+  /* process the MIDI file header */
+
+  hdrptr = buffer.tell (buffer.data);  /* pointer to file and track headers */
+  process_header ();
+  printf ("  Processing %d tracks.\n", num_tracks);
+  if (num_tracks > MAX_TRACKS)
+    midi_error ("Too many tracks", buffer.tell (buffer.data));
+
+  /* initialize processing of all the tracks */
+
+  for (tracknum = 0; tracknum < num_tracks; ++tracknum) {
+    start_track (tracknum);   /* process the track header */
+    find_note (tracknum);     /* position to the first note on/off */
+  }
+
+  notes_skipped = 0;
+  tracknum = 0;
+  earliest_tracknum = 0;
+  earliest_time = 0;
+}
+
+// Parses the note on/offs until we are ready to render some more samples.  Then return the
+// total number of samples to render before we need to be called again
+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! */
+
+  do {                         /* while there are still track notes to process */
+    static struct track_status *trk;
+    static struct tonegen_status *tg;
+    static int tgnum;
+    static int count_tracks;
+    static unsigned long delta_time, delta_msec;
+
+    /* Find the track with the earliest event time,
+       and output a delay command if time has advanced.
+
+       A potential improvement: If there are multiple tracks with the same time,
+       first do the ones with STOPNOTE as the next command, if any.  That would
+       help avoid running out of tone generators.  In practice, though, most MIDI
+       files do all the STOPNOTEs first anyway, so it won't have much effect.
+    */
+
+    earliest_time = 0x7fffffff;
+
+    /* Usually we start with the track after the one we did last time (tracknum),
+       so that if we run out of tone generators, we have been fair to all the tracks.
+       The alternate "strategy1" says we always start with track 0, which means
+       that we favor early tracks over later ones when there aren't enough tone generators.
+    */
+
+    count_tracks = num_tracks;
+    do {
+      if (++tracknum >= num_tracks)
+        tracknum = 0;
+      trk = &track[tracknum];
+      if (trk->cmd != CMD_TRACKDONE && trk->time < earliest_time) {
+        earliest_time = trk->time;
+        earliest_tracknum = tracknum;
+      }
+    } while (--count_tracks);
+
+    tracknum = earliest_tracknum;     /* the track we picked */
+    trk = &track[tracknum];
+    if (earliest_time < timenow)
+      midi_error ("INTERNAL: time went backwards", trk->trkptr);
+
+    /* If time has advanced, output a "delay" command */
+
+    delta_time = earliest_time - timenow;
+    if (delta_time) {
+      /* Convert ticks to milliseconds based on the current tempo */
+      unsigned long long temp;
+      temp = ((unsigned long long) delta_time * tempo) / ticks_per_beat;
+      delta_msec = temp / 1000;      // get around LCC compiler bug
+      if (delta_msec > 0x7fff)
+        midi_error ("INTERNAL: time delta too big", trk->trkptr);
+      int samples = (((int) delta_msec) * freq) / 1000;
+      timenow = earliest_time;
+      return samples;
+    }
+    timenow = earliest_time;
+
+    /*  If this track event is "set tempo", just change the global tempo.
+       That affects how we generate "delay" commands. */
+
+    if (trk->cmd == CMD_TEMPO) {
+      tempo = trk->tempo;
+      find_note (tracknum);
+    }
+
+    /*  If this track event is "stop note", process it and all subsequent "stop notes" for this track
+       that are happening at the same time. Doing so frees up as many tone generators as possible.  */
+
+    else if (trk->cmd == CMD_STOPNOTE)
+      do {
+        // stop a note
+        for (tgnum = 0; tgnum < num_tonegens; ++tgnum) {    /* find which generator is playing it */
+          tg = &tonegen[tgnum];
+          if (tg->playing && tg->track == tracknum && tg->note == trk->note) {
+            tsf_note_off (g_tsf, tg->instrument, tg->note);
+            tg->playing = false;
+            trk->tonegens[tgnum] = false;
+          }
+        }
+        find_note (tracknum);       // use up the note
+      } while (trk->cmd == CMD_STOPNOTE && trk->time == timenow);
+
+    /*  If this track event is "start note", process only it.
+       Don't do more than one, so we allow other tracks their chance at grabbing tone generators. */
+
+    else if (trk->cmd == CMD_PLAYNOTE) {
+      bool foundgen = false;
+      /* if not, then try for any free tone generator */
+      if (!foundgen)
+        for (tgnum = 0; tgnum < num_tonegens; ++tgnum) {
+          tg = &tonegen[tgnum];
+          if (!tg->playing) {
+            foundgen = true;
+            break;
+          }
+        }
+      if (foundgen) {
+        if (tgnum + 1 > num_tonegens_used)
+          num_tonegens_used = tgnum + 1;
+        tg->playing = true;
+        tg->track = tracknum;
+        tg->note = trk->note;
+        trk->tonegens[tgnum] = true;
+        trk->preferred_tonegen = tgnum;
+        if (tg->instrument != midi_chan_instrument[trk->chan]) {    /* new instrument for this generator */
+          tg->instrument = midi_chan_instrument[trk->chan];
+        }
+        tsf_note_on (g_tsf, tg->instrument, tg->note, trk->velocity / 127.0); // velocity = 0...127
+      } else {
+        ++notes_skipped;
+      }
+      find_note (tracknum);     // use up the note
+    }
+  }
+  while (tracks_done < num_tracks);
+  return -1; // EOF
+}
+
+
+void AudioGeneratorMIDI::StopMIDI()
+{
+
+  buffer.close(buffer.data);
+  tsf_close(g_tsf);
+  printf ("  %s %d tone generators were used.\n",
+          num_tonegens_used < num_tonegens ? "Only" : "All", num_tonegens_used);
+  if (notes_skipped)
+    printf
+    ("  %d notes were skipped because there weren't enough tone generators.\n", notes_skipped);
+
+  printf ("  Done.\n");
+}
+
+
+bool AudioGeneratorMIDI::begin(AudioFileSource *src, AudioOutput *out)
+{
+  // Clear out status variables
+  for (int i=0; i<MAX_TONEGENS; i++) memset(&tonegen[i], 0, sizeof(struct tonegen_status));
+  for (int i=0; i<MAX_TRACKS; i++) memset(&track[i], 0, sizeof(struct track_status));
+  memset(midi_chan_instrument, 0, sizeof(midi_chan_instrument));
+
+  g_tsf = tsf_load(&afsSF2);
+  if (!g_tsf) return false;
+  tsf_set_output (g_tsf, TSF_MONO, freq, -10 /* dB gain -10 */ );
+
+  if (!out->SetRate( freq )) return false;
+  if (!out->SetBitsPerSample( 16 )) return false;
+  if (!out->SetChannels( 1 )) return false;
+  if (!out->begin()) return false;
+
+  output = out;
+  file = src;
+
+  running = true;
+
+  PrepareMIDI(src);
+
+  samplesToPlay = 0;
+  numSamplesRendered = 0;
+  sentSamplesRendered = 0;
+  
+  sawEOF = false;
+  return running;
+}
+
+
+bool AudioGeneratorMIDI::loop()
+{
+  static int c = 0;
+
+  if (!running) goto done; // Nothing to do here!
+
+  // First, try and push in the stored sample.  If we can't, then punt and try later
+  if (!output->ConsumeSample(lastSample)) goto done; // Can't send, but no error detected
+
+  // Try and stuff the buffer one sample at a time
+  do {
+    c++;
+    if (c%44100 == 0) yield();
+
+play:
+
+    if (sentSamplesRendered < numSamplesRendered) {
+      lastSample[AudioOutput::LEFTCHANNEL] = samplesRendered[sentSamplesRendered];
+      lastSample[AudioOutput::RIGHTCHANNEL] = samplesRendered[sentSamplesRendered];
+      sentSamplesRendered++;
+    } else if (samplesToPlay) {
+      numSamplesRendered = sizeof(samplesRendered)/sizeof(samplesRendered[0]);
+      if ((int)samplesToPlay < (int)(sizeof(samplesRendered)/sizeof(samplesRendered[0]))) numSamplesRendered = samplesToPlay;
+      tsf_render_short_fast(g_tsf, samplesRendered, numSamplesRendered, 0);
+      lastSample[AudioOutput::LEFTCHANNEL] = samplesRendered[0];
+      lastSample[AudioOutput::RIGHTCHANNEL] = samplesRendered[0];
+      sentSamplesRendered = 1;
+      samplesToPlay -= numSamplesRendered;
+    } else {
+      numSamplesRendered = 0;
+      sentSamplesRendered = 0;
+      if (sawEOF) {
+        running = false;
+      } else {
+        samplesToPlay = PlayMIDI();
+        if (samplesToPlay == -1) {
+            sawEOF = true;
+            samplesToPlay = freq / 2;
+        }
+        goto play;
+      }
+    }
+  } while (running && output->ConsumeSample(lastSample));
+
+done:
+  file->loop();
+  output->loop();
+
+  return running;
+}
+ 
+bool AudioGeneratorMIDI::stop()
+{
+  StopMIDI();
+  output->stop();
+  return true;
+}
+
+
+int AudioGeneratorMIDI::afs_read(void *data, void *ptr, unsigned int size)
+{
+  AudioFileSource *s = reinterpret_cast<AudioFileSource *>(data);
+  return s->read(ptr, size);
+}
+
+int AudioGeneratorMIDI::afs_tell(void *data)
+{
+  AudioFileSource *s = reinterpret_cast<AudioFileSource *>(data);
+  return s->getPos();
+}
+
+int AudioGeneratorMIDI::afs_skip(void *data, unsigned int count)
+{
+  AudioFileSource *s = reinterpret_cast<AudioFileSource *>(data);
+  return s->seek(count, SEEK_CUR);
+}
+
+int AudioGeneratorMIDI::afs_seek(void *data, unsigned int pos)
+{
+  AudioFileSource *s = reinterpret_cast<AudioFileSource *>(data);
+  return s->seek(pos, SEEK_SET);
+}
+
+int AudioGeneratorMIDI::afs_close(void *data)
+{
+  AudioFileSource *s = reinterpret_cast<AudioFileSource *>(data);
+  return s->close();
+}
+
+int AudioGeneratorMIDI::afs_size(void *data)
+{
+  AudioFileSource *s = reinterpret_cast<AudioFileSource *>(data);
+  return s->getSize();
+}
+
+void AudioGeneratorMIDI::MakeStreamFromAFS(AudioFileSource *src, tsf_stream *afs)
+{
+  afs->data = reinterpret_cast<void*>(src);
+  afs->read = &afs_read;
+  afs->tell = &afs_tell;
+  afs->skip = &afs_skip;
+  afs->seek = &afs_seek;
+  afs->close = &afs_close;
+  afs->size = &afs_size;
+}
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorMIDI.h

@@ -0,0 +1,181 @@
+/*
+  AudioGeneratorMIDI
+  Audio output generator that plays MIDI files using a SF2 SoundFont
+    
+  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/>.
+*/
+
+#ifndef _AUDIOGENERATORMIDI_H
+#define _AUDIOGENERATORMIDI_H
+
+#include "AudioGenerator.h"
+
+#define TSF_NO_STDIO
+#include "libtinysoundfont/tsf.h"
+
+class AudioGeneratorMIDI : public AudioGenerator
+{
+  public:
+    AudioGeneratorMIDI() { freq=44100; running = false; };
+    virtual ~AudioGeneratorMIDI() override {};
+    bool SetSoundfont(AudioFileSource *newsf2) {
+      if (isRunning()) return false;
+      sf2 = newsf2;
+      MakeStreamFromAFS(sf2, &afsSF2);
+      return true;
+    }
+    bool SetSampleRate(int newfreq) {
+      if (isRunning()) return false;
+      freq = newfreq;
+      return true;
+    }
+    virtual bool begin(AudioFileSource *mid, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override { return running; };
+
+  private:
+    int freq;
+    tsf *g_tsf;
+    struct tsf_stream buffer;
+    struct tsf_stream afsMIDI;
+    struct tsf_stream afsSF2;
+    AudioFileSource *sf2;
+    AudioFileSource *midi;
+
+  protected:
+    struct midi_header {
+      int8_t MThd[4];
+      uint32_t header_size;
+      uint16_t format_type;
+      uint16_t number_of_tracks;
+      uint16_t time_division;
+    };
+
+    struct track_header {
+      int8_t MTrk[4];
+      uint32_t track_size;
+    };
+
+    enum { MAX_TONEGENS = 32,         /* max tone generators: tones we can play simultaneously */
+           MAX_TRACKS = 24
+         };         /* max number of MIDI tracks we will process */
+
+    int hdrptr;
+    unsigned long buflen;
+    int num_tracks;
+    int tracks_done = 0;
+    int num_tonegens = MAX_TONEGENS;
+    int num_tonegens_used = 0;
+    unsigned int ticks_per_beat = 240;
+    unsigned long timenow = 0;
+    unsigned long tempo;            /* current tempo in usec/qnote */
+    // State needed for PlayMID()
+    int notes_skipped = 0;
+    int tracknum = 0;
+    int earliest_tracknum = 0;
+    unsigned long earliest_time = 0;
+
+    struct tonegen_status {         /* current status of a tone generator */
+      bool playing;                /* is it playing? */
+      char track;                   /* if so, which track is the note from? */
+      char note;                    /* what note is playing? */
+      char instrument;              /* what instrument? */
+    } tonegen[MAX_TONEGENS];
+
+    struct track_status {           /* current processing point of a MIDI track */
+      int trkptr;                  /* ptr to the next note change */
+      int trkend;                  /* ptr past the end of the track */
+      unsigned long time;          /* what time we're at in the score */
+      unsigned long tempo;         /* the tempo last set, in usec per qnote */
+      unsigned int preferred_tonegen;      /* for strategy2, try to use this generator */
+      unsigned char cmd;           /* CMD_xxxx next to do */
+      unsigned char note;          /* for which note */
+      unsigned char chan;          /* from which channel it was */
+      unsigned char velocity;      /* the current volume */
+      unsigned char last_event;    /* the last event, for MIDI's "running status" */
+      bool tonegens[MAX_TONEGENS]; /* which tone generators our notes are playing on */
+    } track[MAX_TRACKS];
+
+    int midi_chan_instrument[16];   /* which instrument is currently being played on each channel */
+
+    /* output bytestream commands, which are also stored in track_status.cmd */
+    enum { CMD_PLAYNOTE   = 0x90,    /* play a note: low nibble is generator #, note is next byte */
+           CMD_STOPNOTE   = 0x80,    /* stop a note: low nibble is generator # */
+           CMD_INSTRUMENT = 0xc0,    /* change instrument; low nibble is generator #, instrument is next byte */
+           CMD_RESTART    = 0xe0,    /* restart the score from the beginning */
+           CMD_STOP       = 0xf0,    /* stop playing */
+           CMD_TEMPO      = 0xFE,    /* tempo in usec per quarter note ("beat") */
+           CMD_TRACKDONE  = 0xFF
+         };   /* no more data left in this track */
+
+
+
+    /* portable string length */
+    int strlength (const char *str) {
+      int i;
+      for (i = 0; str[i] != '\0'; ++i);
+      return i;
+    }
+
+
+    /* match a constant character sequence */
+
+    int charcmp (const char *buf, const char *match) {
+      int len, i;
+      len = strlength (match);
+      for (i = 0; i < len; ++i)
+        if (buf[i] != match[i])
+          return 0;
+      return 1;
+    }
+
+    unsigned char buffer_byte (int offset);
+    unsigned short buffer_short (int offset);
+    unsigned int buffer_int32 (int offset);
+
+    void midi_error (const char *msg, int curpos);
+    void chk_bufdata (int ptr, unsigned long int len);
+    uint16_t rev_short (uint16_t val);
+    uint32_t rev_long (uint32_t val);
+    void process_header (void);
+    void start_track (int tracknum);
+
+    unsigned long get_varlen (int *ptr);
+    void find_note (int tracknum);
+    void PrepareMIDI(AudioFileSource *src);
+    int PlayMIDI();
+    void StopMIDI();
+
+    // tsf_stream <-> AudioFileSource
+    static int afs_read(void *data, void *ptr, unsigned int size);
+    static int afs_tell(void *data);
+    static int afs_skip(void *data, unsigned int count);
+    static int afs_seek(void *data, unsigned int pos);
+    static int afs_close(void *data);
+    static int afs_size(void *data);
+    void MakeStreamFromAFS(AudioFileSource *src, tsf_stream *afs);
+
+    int samplesToPlay;
+    bool sawEOF;
+    int numSamplesRendered;
+    int sentSamplesRendered ;
+    short samplesRendered[256];
+};
+
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorMOD.cpp

@@ -0,0 +1,876 @@
+/*
+  AudioGeneratorMOD
+  Audio output generator that plays Amiga MOD tracker files
+    
+  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/>.
+*/
+#define PGM_READ_UNALIGNED 0
+
+#include "AudioGeneratorMOD.h"
+
+/* 
+   Ported/hacked out from STELLARPLAYER by Ronen K.
+   http://mobile4dev.blogspot.com/2012/11/stellaris-launchpad-mod-player.html
+   A version exists in GitHub at https://github.com/steveway/stellarplayer
+   and also at https://github.com/MikesModz/StellarPlayer
+   Both which were themselves a port of the PIC32 MOD player
+   https://www.youtube.com/watch?v=i3Yl0TISQBE (seems to no longer be available.)
+
+   Most changes involved reducing memory usage by changing data structures,
+   moving constants to PROGMEM and minor tweaks to allow non pow2 buffer sizes.
+*/
+
+#pragma GCC optimize ("O3")
+
+#define NOTE(r, c) (Player.currentPattern.note8[r][c]==NONOTE8?NONOTE:8*Player.currentPattern.note8[r][c])
+
+#ifndef min
+#define min(X,Y) ((X) < (Y) ? (X) : (Y))
+#endif
+
+AudioGeneratorMOD::AudioGeneratorMOD()
+{
+  sampleRate = 44100;
+  fatBufferSize = 6 * 1024;
+  stereoSeparation = 32;
+  mixerTick = 0;
+  usePAL = false;
+  UpdateAmiga();
+  running = false;
+  file = NULL;
+  output = NULL;
+}
+
+AudioGeneratorMOD::~AudioGeneratorMOD()
+{
+  // Free any remaining buffers
+  for (int i = 0; i < CHANNELS; i++) {
+    FatBuffer.channels[i] = NULL;
+  }
+}
+
+bool AudioGeneratorMOD::stop()
+{
+  // We may be stopping because of allocation failures, so always deallocate
+  for (int i = 0; i < CHANNELS; i++) {
+    free(FatBuffer.channels[i]);
+    FatBuffer.channels[i] = NULL;
+  }
+  if (file) file->close();
+  running = false;
+  output->stop();
+  return true;
+}
+
+bool AudioGeneratorMOD::loop()
+{
+  if (!running) goto done; // Easy-peasy
+
+  // First, try and push in the stored sample.  If we can't, then punt and try later
+  if (!output->ConsumeSample(lastSample)) goto done; // FIFO full, wait...
+
+  // Now advance enough times to fill the i2s buffer
+  do {
+    if (mixerTick == 0) {
+      running = RunPlayer();
+      if (!running) {
+        stop();
+        goto done;
+      }
+      mixerTick = Player.samplesPerTick;
+    }
+    GetSample( lastSample );
+    mixerTick--;
+  } while (output->ConsumeSample(lastSample));
+
+done:
+  file->loop();
+  output->loop();
+
+  // We'll be left with one sample still in our buffer because it couldn't fit in the FIFO
+  return running;
+}
+
+bool AudioGeneratorMOD::begin(AudioFileSource *source, AudioOutput *out)
+{
+  if (running) stop();
+  
+  if (!source) return false;
+  file = source;
+  if (!out) return false;
+  output = out;
+  
+  if (!file->isOpen()) return false; // Can't read the file!
+
+  // Set the output values properly
+  if (!output->SetRate(sampleRate)) return false;
+  if (!output->SetBitsPerSample(16)) return false;
+  if (!output->SetChannels(2)) return false;
+  if (!output->begin()) return false;
+
+  UpdateAmiga();
+
+  for (int i = 0; i < CHANNELS; i++) {
+    FatBuffer.channels[i] = reinterpret_cast<uint8_t*>(malloc(fatBufferSize));
+    if (!FatBuffer.channels[i]) {
+      stop();
+      return false;
+    }
+  }
+  if (!LoadMOD()) {
+    stop();
+    return false;
+  }
+  running = true;
+  return true;
+}
+
+// Sorted Amiga periods
+static const uint16_t amigaPeriods[296] PROGMEM = {
+  907, 900, 894, 887, 881, 875, 868, 862, //  -8 to -1
+  856, 850, 844, 838, 832, 826, 820, 814, // C-1 to +7
+  808, 802, 796, 791, 785, 779, 774, 768, // C#1 to +7
+  762, 757, 752, 746, 741, 736, 730, 725, // D-1 to +7
+  720, 715, 709, 704, 699, 694, 689, 684, // D#1 to +7
+  678, 675, 670, 665, 660, 655, 651, 646, // E-1 to +7
+  640, 636, 632, 628, 623, 619, 614, 610, // F-1 to +7
+  604, 601, 597, 592, 588, 584, 580, 575, // F#1 to +7
+  570, 567, 563, 559, 555, 551, 547, 543, // G-1 to +7
+  538, 535, 532, 528, 524, 520, 516, 513, // G#1 to +7
+  508, 505, 502, 498, 494, 491, 487, 484, // A-1 to +7
+  480, 477, 474, 470, 467, 463, 460, 457, // A#1 to +7
+  453, 450, 447, 444, 441, 437, 434, 431, // B-1 to +7
+  428, 425, 422, 419, 416, 413, 410, 407, // C-2 to +7
+  404, 401, 398, 395, 392, 390, 387, 384, // C#2 to +7
+  381, 379, 376, 373, 370, 368, 365, 363, // D-2 to +7
+  360, 357, 355, 352, 350, 347, 345, 342, // D#2 to +7
+  339, 337, 335, 332, 330, 328, 325, 323, // E-2 to +7
+  320, 318, 316, 314, 312, 309, 307, 305, // F-2 to +7
+  302, 300, 298, 296, 294, 292, 290, 288, // F#2 to +7
+  285, 284, 282, 280, 278, 276, 274, 272, // G-2 to +7
+  269, 268, 266, 264, 262, 260, 258, 256, // G#2 to +7
+  254, 253, 251, 249, 247, 245, 244, 242, // A-2 to +7
+  240, 238, 237, 235, 233, 232, 230, 228, // A#2 to +7
+  226, 225, 223, 222, 220, 219, 217, 216, // B-2 to +7
+  214, 212, 211, 209, 208, 206, 205, 203, // C-3 to +7
+  202, 200, 199, 198, 196, 195, 193, 192, // C#3 to +7
+  190, 189, 188, 187, 185, 184, 183, 181, // D-3 to +7
+  180, 179, 177, 176, 175, 174, 172, 171, // D#3 to +7
+  170, 169, 167, 166, 165, 164, 163, 161, // E-3 to +7
+  160, 159, 158, 157, 156, 155, 154, 152, // F-3 to +7
+  151, 150, 149, 148, 147, 146, 145, 144, // F#3 to +7
+  143, 142, 141, 140, 139, 138, 137, 136, // G-3 to +7
+  135, 134, 133, 132, 131, 130, 129, 128, // G#3 to +7
+  127, 126, 125, 125, 123, 123, 122, 121, // A-3 to +7
+  120, 119, 118, 118, 117, 116, 115, 114, // A#3 to +7
+  113, 113, 112, 111, 110, 109, 109, 108  // B-3 to +7
+};
+#define ReadAmigaPeriods(a) (uint16_t)pgm_read_word(amigaPeriods + (a))
+
+static const uint8_t sine[64] PROGMEM = {
+  0,  24,  49,  74,  97, 120, 141, 161,
+  180, 197, 212, 224, 235, 244, 250, 253,
+  255, 253, 250, 244, 235, 224, 212, 197,
+  180, 161, 141, 120,  97,  74,  49,  24
+};
+#define ReadSine(a) pgm_read_byte(sine + (a))
+
+
+static inline uint16_t MakeWord(uint8_t h, uint8_t l) { return h << 8 | l; }
+
+bool AudioGeneratorMOD::LoadHeader()
+{
+  uint8_t i;
+  uint8_t temp[4];
+  uint8_t junk[22];
+
+  if (20 != file->read(/*Mod.name*/junk, 20)) return false; // Skip MOD name
+  for (i = 0; i < SAMPLES; i++) {
+    if (22 != file->read(junk /*Mod.samples[i].name*/, 22)) return false; // Skip sample name
+    if (2 != file->read(temp, 2)) return false;
+    Mod.samples[i].length = MakeWord(temp[0], temp[1]) * 2;
+    if (1 != file->read(reinterpret_cast<uint8_t*>(&Mod.samples[i].fineTune), 1)) return false;
+    if (Mod.samples[i].fineTune > 7) Mod.samples[i].fineTune -= 16;
+    if (1 != file->read(&Mod.samples[i].volume, 1)) return false;
+    if (2 != file->read(temp, 2)) return false;
+    Mod.samples[i].loopBegin = MakeWord(temp[0], temp[1]) * 2;
+    if (2 != file->read(temp, 2)) return false;
+    Mod.samples[i].loopLength = MakeWord(temp[0], temp[1]) * 2;
+    if (Mod.samples[i].loopBegin + Mod.samples[i].loopLength > Mod.samples[i].length)
+      Mod.samples[i].loopLength = Mod.samples[i].length - Mod.samples[i].loopBegin;
+  }
+
+  if (1 != file->read(&Mod.songLength, 1)) return false;
+  if (1 != file->read(temp, 1)) return false; // Discard this byte
+
+  Mod.numberOfPatterns = 0;
+  for (i = 0; i < 128; i++) {
+    if (1 != file->read(&Mod.order[i], 1)) return false;
+    if (Mod.order[i] > Mod.numberOfPatterns)
+      Mod.numberOfPatterns = Mod.order[i];
+  }
+  Mod.numberOfPatterns++;
+
+  // Offset 1080
+  if (4 != file->read(temp, 4)) return false;;
+  if (!strncmp(reinterpret_cast<const char*>(temp + 1), "CHN", 3))
+    Mod.numberOfChannels = temp[0] - '0';
+  else if (!strncmp(reinterpret_cast<const char*>(temp + 2), "CH", 2))
+    Mod.numberOfChannels = (temp[0] - '0') * 10 + temp[1] - '0';
+  else
+    Mod.numberOfChannels = 4;
+
+  return true;
+}
+
+void AudioGeneratorMOD::LoadSamples()
+{
+  uint8_t i;
+  uint32_t fileOffset = 1084 + Mod.numberOfPatterns * ROWS * Mod.numberOfChannels * 4 - 1;
+
+  for (i = 0; i < SAMPLES; i++) {
+
+    if (Mod.samples[i].length) {
+      Mixer.sampleBegin[i] = fileOffset;
+      Mixer.sampleEnd[i] = fileOffset + Mod.samples[i].length;
+      if (Mod.samples[i].loopLength > 2) {
+        Mixer.sampleloopBegin[i] = fileOffset + Mod.samples[i].loopBegin;
+        Mixer.sampleLoopLength[i] = Mod.samples[i].loopLength;
+        Mixer.sampleLoopEnd[i] = Mixer.sampleloopBegin[i] + Mixer.sampleLoopLength[i];
+      } else {
+        Mixer.sampleloopBegin[i] = 0;
+        Mixer.sampleLoopLength[i] = 0;
+        Mixer.sampleLoopEnd[i] = 0;
+      }
+      fileOffset += Mod.samples[i].length;
+    }
+
+  }
+
+}
+
+bool AudioGeneratorMOD::LoadPattern(uint8_t pattern)
+{
+  uint8_t row;
+  uint8_t channel;
+  uint8_t i;
+  uint8_t temp[4];
+  uint16_t amigaPeriod;
+
+  if (!file->seek(1084 + pattern * ROWS * Mod.numberOfChannels * 4, SEEK_SET)) return false;
+
+  for (row = 0; row < ROWS; row++) {
+    for (channel = 0; channel < Mod.numberOfChannels; channel++) {
+
+      if (4 != file->read(temp, 4)) return false;
+
+      Player.currentPattern.sampleNumber[row][channel] = (temp[0] & 0xF0) + (temp[2] >> 4);
+
+      amigaPeriod = ((temp[0] & 0xF) << 8) + temp[1];
+      //   Player.currentPattern.note[row][channel] = NONOTE;
+      Player.currentPattern.note8[row][channel] = NONOTE8;
+      for (i = 1; i < 37; i++)
+        if (amigaPeriod > ReadAmigaPeriods(i * 8) - 3 &&
+            amigaPeriod < ReadAmigaPeriods(i * 8) + 3)
+          Player.currentPattern.note8[row][channel] = i;
+
+      Player.currentPattern.effectNumber[row][channel] = temp[2] & 0xF;
+      Player.currentPattern.effectParameter[row][channel] = temp[3];
+    }
+  }
+
+  return true;
+}
+
+void AudioGeneratorMOD::Portamento(uint8_t channel)
+{
+  if (Player.lastAmigaPeriod[channel] < Player.portamentoNote[channel]) {
+    Player.lastAmigaPeriod[channel] += Player.portamentoSpeed[channel];
+    if (Player.lastAmigaPeriod[channel] > Player.portamentoNote[channel])
+      Player.lastAmigaPeriod[channel] = Player.portamentoNote[channel];
+  }
+  if (Player.lastAmigaPeriod[channel] > Player.portamentoNote[channel]) {
+    Player.lastAmigaPeriod[channel] -= Player.portamentoSpeed[channel];
+    if (Player.lastAmigaPeriod[channel] < Player.portamentoNote[channel])
+      Player.lastAmigaPeriod[channel] = Player.portamentoNote[channel];
+  }
+  Mixer.channelFrequency[channel] = Player.amiga / Player.lastAmigaPeriod[channel];
+}
+
+void AudioGeneratorMOD::Vibrato(uint8_t channel)
+{
+  uint16_t delta;
+  uint16_t temp;
+
+  temp = Player.vibratoPos[channel] & 31;
+
+  switch (Player.waveControl[channel] & 3) {
+    case 0:
+      delta = ReadSine(temp);
+      break;
+    case 1:
+      temp <<= 3;
+      if (Player.vibratoPos[channel] < 0)
+        temp = 255 - temp;
+      delta = temp;
+      break;
+    case 2:
+      delta = 255;
+      break;
+    case 3:
+      delta = rand() & 255;
+      break;
+  }
+
+  delta *= Player.vibratoDepth[channel];
+  delta >>= 7;
+
+  if (Player.vibratoPos[channel] >= 0)
+    Mixer.channelFrequency[channel] = Player.amiga / (Player.lastAmigaPeriod[channel] + delta);
+  else
+    Mixer.channelFrequency[channel] = Player.amiga / (Player.lastAmigaPeriod[channel] - delta);
+
+  Player.vibratoPos[channel] += Player.vibratoSpeed[channel];
+  if (Player.vibratoPos[channel] > 31) Player.vibratoPos[channel] -= 64;
+}
+
+void AudioGeneratorMOD::Tremolo(uint8_t channel)
+{
+  uint16_t delta;
+  uint16_t temp;
+
+  temp = Player.tremoloPos[channel] & 31;
+
+  switch (Player.waveControl[channel] & 3) {
+    case 0:
+      delta = ReadSine(temp);
+      break;
+    case 1:
+      temp <<= 3;
+      if (Player.tremoloPos[channel] < 0)
+        temp = 255 - temp;
+      delta = temp;
+      break;
+    case 2:
+      delta = 255;
+      break;
+    case 3:
+      delta = rand() & 255;
+      break;
+  }
+
+  delta *= Player.tremoloDepth[channel];
+  delta >>= 6;
+
+  if (Player.tremoloPos[channel] >= 0) {
+    if (Player.volume[channel] + delta > 64) delta = 64 - Player.volume[channel];
+    Mixer.channelVolume[channel] = Player.volume[channel] + delta;
+  } else {
+    if (Player.volume[channel] - delta < 0) delta = Player.volume[channel];
+    Mixer.channelVolume[channel] = Player.volume[channel] - delta;
+  }
+
+  Player.tremoloPos[channel] += Player.tremoloSpeed[channel];
+  if (Player.tremoloPos[channel] > 31) Player.tremoloPos[channel] -= 64;
+}
+
+bool AudioGeneratorMOD::ProcessRow()
+{
+  bool jumpFlag;
+  bool breakFlag;
+  uint8_t channel;
+  uint8_t sampleNumber;
+  uint16_t note;
+  uint8_t effectNumber;
+  uint8_t effectParameter;
+  uint8_t effectParameterX;
+  uint8_t effectParameterY;
+  uint16_t sampleOffset;
+
+  if (!running) return false;
+
+  Player.lastRow = Player.row++;
+  jumpFlag = false;
+  breakFlag = false;
+  for (channel = 0; channel < Mod.numberOfChannels; channel++) {
+
+    sampleNumber = Player.currentPattern.sampleNumber[Player.lastRow][channel];
+    note = NOTE(Player.lastRow, channel);
+    effectNumber = Player.currentPattern.effectNumber[Player.lastRow][channel];
+    effectParameter = Player.currentPattern.effectParameter[Player.lastRow][channel];
+    effectParameterX = effectParameter >> 4;
+    effectParameterY = effectParameter & 0xF;
+    sampleOffset = 0;
+
+    if (sampleNumber) {
+      Player.lastSampleNumber[channel] = sampleNumber - 1;
+      if (!(effectParameter == 0xE && effectParameterX == NOTEDELAY))
+        Player.volume[channel] = Mod.samples[Player.lastSampleNumber[channel]].volume;
+    }
+
+    if (note != NONOTE) {
+      Player.lastNote[channel] = note;
+      Player.amigaPeriod[channel] = ReadAmigaPeriods(note + Mod.samples[Player.lastSampleNumber[channel]].fineTune);
+
+      if (effectNumber != TONEPORTAMENTO && effectNumber != PORTAMENTOVOLUMESLIDE)
+        Player.lastAmigaPeriod[channel] = Player.amigaPeriod[channel];
+
+      if (!(Player.waveControl[channel] & 0x80)) Player.vibratoPos[channel] = 0;
+      if (!(Player.waveControl[channel] & 0x08)) Player.tremoloPos[channel] = 0;
+    }
+
+    switch (effectNumber) {
+      case TONEPORTAMENTO:
+        if (effectParameter) Player.portamentoSpeed[channel] = effectParameter;
+        Player.portamentoNote[channel] = Player.amigaPeriod[channel];
+        note = NONOTE;
+        break;
+
+      case VIBRATO:
+        if (effectParameterX) Player.vibratoSpeed[channel] = effectParameterX;
+        if (effectParameterY) Player.vibratoDepth[channel] = effectParameterY;
+        break;
+
+      case PORTAMENTOVOLUMESLIDE:
+        Player.portamentoNote[channel] = Player.amigaPeriod[channel];
+        note = NONOTE;
+        break;
+
+      case TREMOLO:
+        if (effectParameterX) Player.tremoloSpeed[channel] = effectParameterX;
+        if (effectParameterY) Player.tremoloDepth[channel] = effectParameterY;
+        break;
+
+      case SETCHANNELPANNING:
+        Mixer.channelPanning[channel] = effectParameter >> 1;
+        break;
+
+      case SETSAMPLEOFFSET:
+        sampleOffset = effectParameter << 8;
+        if (sampleOffset > Mod.samples[Player.lastSampleNumber[channel]].length)
+          sampleOffset = Mod.samples[Player.lastSampleNumber[channel]].length;
+        break;
+
+      case JUMPTOORDER:
+        Player.orderIndex = effectParameter;
+        if (Player.orderIndex >= Mod.songLength)
+          Player.orderIndex = 0;
+        Player.row = 0;
+        jumpFlag = true;
+        break;
+
+      case SETVOLUME:
+        if (effectParameter > 64) Player.volume[channel] = 64;
+        else Player.volume[channel] = effectParameter;
+        break;
+
+      case BREAKPATTERNTOROW:
+        Player.row = effectParameterX * 10 + effectParameterY;
+        if (Player.row >= ROWS)
+          Player.row = 0;
+        if (!jumpFlag && !breakFlag) {
+          Player.orderIndex++;
+          if (Player.orderIndex >= Mod.songLength)
+            Player.orderIndex = 0;
+        }
+        breakFlag = true;
+        break;
+
+      case 0xE:
+        switch (effectParameterX) {
+          case FINEPORTAMENTOUP:
+            Player.lastAmigaPeriod[channel] -= effectParameterY;
+            break;
+
+          case FINEPORTAMENTODOWN:
+            Player.lastAmigaPeriod[channel] += effectParameterY;
+            break;
+
+          case SETVIBRATOWAVEFORM:
+            Player.waveControl[channel] &= 0xF0;
+            Player.waveControl[channel] |= effectParameterY;
+            break;
+
+          case SETFINETUNE:
+            Mod.samples[Player.lastSampleNumber[channel]].fineTune = effectParameterY;
+            if (Mod.samples[Player.lastSampleNumber[channel]].fineTune > 7)
+              Mod.samples[Player.lastSampleNumber[channel]].fineTune -= 16;
+            break;
+
+          case PATTERNLOOP:
+            if (effectParameterY) {
+              if (Player.patternLoopCount[channel])
+                Player.patternLoopCount[channel]--;
+              else
+                Player.patternLoopCount[channel] = effectParameterY;
+              if (Player.patternLoopCount[channel])
+                Player.row = Player.patternLoopRow[channel] - 1;
+            } else
+              Player.patternLoopRow[channel] = Player.row;
+            break;
+
+          case SETTREMOLOWAVEFORM:
+            Player.waveControl[channel] &= 0xF;
+            Player.waveControl[channel] |= effectParameterY << 4;
+            break;
+
+          case FINEVOLUMESLIDEUP:
+            Player.volume[channel] += effectParameterY;
+            if (Player.volume[channel] > 64) Player.volume[channel] = 64;
+            break;
+
+          case FINEVOLUMESLIDEDOWN:
+            Player.volume[channel] -= effectParameterY;
+            if (Player.volume[channel] < 0) Player.volume[channel] = 0;
+            break;
+
+          case NOTECUT:
+            note = NONOTE;
+            break;
+
+          case PATTERNDELAY:
+            Player.patternDelay = effectParameterY;
+            break;
+
+          case INVERTLOOP:
+
+            break;
+        }
+        break;
+
+      case SETSPEED:
+        if (effectParameter < 0x20)
+          Player.speed = effectParameter;
+        else
+          Player.samplesPerTick = sampleRate / (2 * effectParameter / 5);
+        break;
+    }
+
+    if (note != NONOTE || (Player.lastAmigaPeriod[channel] &&
+        effectNumber != VIBRATO && effectNumber != VIBRATOVOLUMESLIDE &&
+        !(effectNumber == 0xE && effectParameterX == NOTEDELAY)))
+      Mixer.channelFrequency[channel] = Player.amiga / Player.lastAmigaPeriod[channel];
+
+    if (note != NONOTE)
+      Mixer.channelSampleOffset[channel] = sampleOffset << DIVIDER;
+
+    if (sampleNumber)
+      Mixer.channelSampleNumber[channel] = Player.lastSampleNumber[channel];
+
+    if (effectNumber != TREMOLO)
+      Mixer.channelVolume[channel] = Player.volume[channel];
+
+  }
+  return true;
+}
+
+bool AudioGeneratorMOD::ProcessTick()
+{
+  uint8_t channel;
+  uint8_t sampleNumber;
+  uint16_t note;
+  uint8_t effectNumber;
+  uint8_t effectParameter;
+  uint8_t effectParameterX;
+  uint8_t effectParameterY;
+  uint16_t tempNote;
+
+  if (!running) return false;
+
+  for (channel = 0; channel < Mod.numberOfChannels; channel++) {
+
+    if (Player.lastAmigaPeriod[channel]) {
+
+      sampleNumber = Player.currentPattern.sampleNumber[Player.lastRow][channel];
+      //   note = Player.currentPattern.note[Player.lastRow][channel];
+      note = NOTE(Player.lastRow, channel);
+      effectNumber = Player.currentPattern.effectNumber[Player.lastRow][channel];
+      effectParameter = Player.currentPattern.effectParameter[Player.lastRow][channel];
+      effectParameterX = effectParameter >> 4;
+      effectParameterY = effectParameter & 0xF;
+
+      switch (effectNumber) {
+        case ARPEGGIO:
+          if (effectParameter)
+            switch (Player.tick % 3) {
+              case 0:
+                Mixer.channelFrequency[channel] = Player.amiga / Player.lastAmigaPeriod[channel];
+                break;
+              case 1:
+                tempNote = Player.lastNote[channel] + effectParameterX * 8 + Mod.samples[Player.lastSampleNumber[channel]].fineTune;
+                if (tempNote < 296) Mixer.channelFrequency[channel] = Player.amiga / ReadAmigaPeriods(tempNote);
+                break;
+              case 2:
+                tempNote = Player.lastNote[channel] + effectParameterY * 8 + Mod.samples[Player.lastSampleNumber[channel]].fineTune;
+                if (tempNote < 296) Mixer.channelFrequency[channel] = Player.amiga / ReadAmigaPeriods(tempNote);
+                break;
+            }
+          break;
+
+        case PORTAMENTOUP:
+          Player.lastAmigaPeriod[channel] -= effectParameter;
+          if (Player.lastAmigaPeriod[channel] < 113) Player.lastAmigaPeriod[channel] = 113;
+          Mixer.channelFrequency[channel] = Player.amiga / Player.lastAmigaPeriod[channel];
+          break;
+
+        case PORTAMENTODOWN:
+          Player.lastAmigaPeriod[channel] += effectParameter;
+          if (Player.lastAmigaPeriod[channel] > 856) Player.lastAmigaPeriod[channel] = 856;
+          Mixer.channelFrequency[channel] = Player.amiga / Player.lastAmigaPeriod[channel];
+          break;
+
+        case TONEPORTAMENTO:
+          Portamento(channel);
+          break;
+
+        case VIBRATO:
+          Vibrato(channel);
+          break;
+
+        case PORTAMENTOVOLUMESLIDE:
+          Portamento(channel);
+          Player.volume[channel] += effectParameterX - effectParameterY;
+          if (Player.volume[channel] < 0) Player.volume[channel] = 0;
+          else if (Player.volume[channel] > 64) Player.volume[channel] = 64;
+          Mixer.channelVolume[channel] = Player.volume[channel];
+          break;
+
+        case VIBRATOVOLUMESLIDE:
+          Vibrato(channel);
+          Player.volume[channel] += effectParameterX - effectParameterY;
+          if (Player.volume[channel] < 0) Player.volume[channel] = 0;
+          else if (Player.volume[channel] > 64) Player.volume[channel] = 64;
+          Mixer.channelVolume[channel] = Player.volume[channel];
+          break;
+
+        case TREMOLO:
+          Tremolo(channel);
+          break;
+
+        case VOLUMESLIDE:
+          Player.volume[channel] += effectParameterX - effectParameterY;
+          if (Player.volume[channel] < 0) Player.volume[channel] = 0;
+          else if (Player.volume[channel] > 64) Player.volume[channel] = 64;
+          Mixer.channelVolume[channel] = Player.volume[channel];
+          break;
+
+        case 0xE:
+          switch (effectParameterX) {
+            case RETRIGGERNOTE:
+              if (!effectParameterY) break;
+              if (!(Player.tick % effectParameterY)) {
+                Mixer.channelSampleOffset[channel] = 0;
+              }
+              break;
+
+            case NOTECUT:
+              if (Player.tick == effectParameterY)
+                Mixer.channelVolume[channel] = Player.volume[channel] = 0;
+              break;
+
+            case NOTEDELAY:
+              if (Player.tick == effectParameterY) {
+                if (sampleNumber) Player.volume[channel] = Mod.samples[Player.lastSampleNumber[channel]].volume;
+                if (note != NONOTE) Mixer.channelSampleOffset[channel] = 0;
+                Mixer.channelFrequency[channel] = Player.amiga / Player.lastAmigaPeriod[channel];
+                Mixer.channelVolume[channel] = Player.volume[channel];
+              }
+              break;
+          }
+          break;
+      }
+
+    }
+
+  }
+  return true;
+}
+
+bool AudioGeneratorMOD::RunPlayer()
+{
+  if (!running) return false;
+
+  if (Player.tick == Player.speed) {
+    Player.tick = 0;
+
+    if (Player.row == ROWS) {
+      Player.orderIndex++;
+      if (Player.orderIndex == Mod.songLength)
+      {
+        //Player.orderIndex = 0;
+        // No loop, just say we're done!
+        return false;
+      }
+      Player.row = 0;
+    }
+
+    if (Player.patternDelay) {
+      Player.patternDelay--;
+    } else {
+      if (Player.orderIndex != Player.oldOrderIndex)
+        if (!LoadPattern(Mod.order[Player.orderIndex])) return false;
+      Player.oldOrderIndex = Player.orderIndex;
+      if (!ProcessRow()) return false;
+    }
+
+  } else {
+    if (!ProcessTick()) return false;
+  }
+  Player.tick++;
+  return true;
+}
+
+void AudioGeneratorMOD::GetSample(int16_t sample[2])
+{
+  int16_t sumL;
+  int16_t sumR;
+  uint8_t channel;
+  uint32_t samplePointer;
+  int8_t current;
+  int8_t next;
+  int16_t out;
+
+  if (!running) return;
+
+  sumL = 0;
+  sumR = 0;
+  for (channel = 0; channel < Mod.numberOfChannels; channel++) {
+
+    if (!Mixer.channelFrequency[channel] ||
+        !Mod.samples[Mixer.channelSampleNumber[channel]].length) continue;
+
+    Mixer.channelSampleOffset[channel] += Mixer.channelFrequency[channel];
+
+    if (!Mixer.channelVolume[channel]) continue;
+
+    samplePointer = Mixer.sampleBegin[Mixer.channelSampleNumber[channel]] +
+                    (Mixer.channelSampleOffset[channel] >> DIVIDER);
+
+    if (Mixer.sampleLoopLength[Mixer.channelSampleNumber[channel]]) {
+
+      if (samplePointer >= Mixer.sampleLoopEnd[Mixer.channelSampleNumber[channel]]) {
+        Mixer.channelSampleOffset[channel] -= Mixer.sampleLoopLength[Mixer.channelSampleNumber[channel]] << DIVIDER;
+        samplePointer -= Mixer.sampleLoopLength[Mixer.channelSampleNumber[channel]];
+      }
+
+    } else {
+
+      if (samplePointer >= Mixer.sampleEnd[Mixer.channelSampleNumber[channel]]) {
+        Mixer.channelFrequency[channel] = 0;
+        samplePointer = Mixer.sampleEnd[Mixer.channelSampleNumber[channel]];
+      }
+
+    }
+
+    if (samplePointer < FatBuffer.samplePointer[channel] ||
+        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;
+
+      if (!file->seek(samplePointer, SEEK_SET)) {
+        stop();
+        return;
+      }
+      if (toRead != file->read(FatBuffer.channels[channel], toRead)) {
+        stop();
+        return;
+      }
+
+      FatBuffer.samplePointer[channel] = samplePointer;
+      FatBuffer.channelSampleNumber[channel] = Mixer.channelSampleNumber[channel];
+    }
+
+    current = FatBuffer.channels[channel][(samplePointer - FatBuffer.samplePointer[channel]) /*& (FATBUFFERSIZE - 1)*/];
+    next = FatBuffer.channels[channel][(samplePointer + 1 - FatBuffer.samplePointer[channel]) /*& (FATBUFFERSIZE - 1)*/];
+
+    out = current;
+
+    // Integer linear interpolation
+    out += (next - current) * (Mixer.channelSampleOffset[channel] & ((1 << DIVIDER) - 1)) >> DIVIDER;
+
+    // Upscale to BITDEPTH
+    out <<= BITDEPTH - 8;
+
+    // Channel volume
+    out = out * Mixer.channelVolume[channel] >> 6;
+
+    // Channel panning
+    sumL += out * min(128 - Mixer.channelPanning[channel], 64) >> 6;
+    sumR += out * min(Mixer.channelPanning[channel], 64) >> 6;
+  }
+
+  // Downscale to BITDEPTH
+  sumL /= Mod.numberOfChannels;
+  sumR /= Mod.numberOfChannels;
+
+  // Fill the sound buffer with unsigned values
+  sample[AudioOutput::LEFTCHANNEL] = sumL + (1 << (BITDEPTH - 1));
+  sample[AudioOutput::RIGHTCHANNEL] = sumR + (1 << (BITDEPTH - 1));
+}
+
+bool AudioGeneratorMOD::LoadMOD()
+{
+  uint8_t channel;
+
+  if (!LoadHeader()) return false;
+  LoadSamples();
+
+  Player.amiga = AMIGA;
+  Player.samplesPerTick = sampleRate / (2 * 125 / 5); // Hz = 2 * BPM / 5
+  Player.speed = 6;
+  Player.tick = Player.speed;
+  Player.row = 0;
+
+  Player.orderIndex = 0;
+  Player.oldOrderIndex = 0xFF;
+  Player.patternDelay = 0;
+
+  for (channel = 0; channel < Mod.numberOfChannels; channel++) {
+    Player.patternLoopCount[channel] = 0;
+    Player.patternLoopRow[channel] = 0;
+
+    Player.lastAmigaPeriod[channel] = 0;
+
+    Player.waveControl[channel] = 0;
+
+    Player.vibratoSpeed[channel] = 0;
+    Player.vibratoDepth[channel] = 0;
+    Player.vibratoPos[channel] = 0;
+
+    Player.tremoloSpeed[channel] = 0;
+    Player.tremoloDepth[channel] = 0;
+    Player.tremoloPos[channel] = 0;
+
+    FatBuffer.samplePointer[channel] = 0;
+    FatBuffer.channelSampleNumber[channel] = 0xFF;
+
+    Mixer.channelSampleOffset[channel] = 0;
+    Mixer.channelFrequency[channel] = 0;
+    Mixer.channelVolume[channel] = 0;
+    switch (channel % 4) {
+      case 0:
+      case 3:
+        Mixer.channelPanning[channel] = stereoSeparation;
+        break;
+      default:
+        Mixer.channelPanning[channel] = 128 - stereoSeparation;
+    }
+  }
+  return true;
+}
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorMOD.h

@@ -0,0 +1,168 @@
+/*
+  AudioGeneratorMOD
+  Audio output generator that plays Amiga MOD tracker files
+    
+  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/>.
+*/
+
+#ifndef _AUDIOGENERATORMOD_H
+#define _AUDIOGENERATORMOD_H
+
+#include "AudioGenerator.h"
+
+class AudioGeneratorMOD : public AudioGenerator
+{
+  public:
+    AudioGeneratorMOD();
+    virtual ~AudioGeneratorMOD() override;
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override { return running; }
+    bool SetSampleRate(int hz) { if (running || (hz < 1) || (hz > 96000) ) return false; sampleRate = hz; return true; }
+    bool SetBufferSize(int sz) { if (running || (sz < 1) ) return false; fatBufferSize = sz; return true; }
+    bool SetStereoSeparation(int sep) { if (running || (sep<0) || (sep>64)) return false; stereoSeparation = sep; return true; }
+    bool SetPAL(bool use) { if (running) return false; usePAL = use; return true; }
+
+  protected:
+    bool LoadMOD();
+    bool LoadHeader();
+    void GetSample(int16_t sample[2]);
+    bool RunPlayer();
+    void LoadSamples();
+    bool LoadPattern(uint8_t pattern);
+    bool ProcessTick();
+    bool ProcessRow();
+    void Tremolo(uint8_t channel);
+    void Portamento(uint8_t channel);
+    void Vibrato(uint8_t channel);
+
+
+  protected:
+    int mixerTick;
+    enum {BITDEPTH = 15};
+    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
+    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); }
+    
+    enum {ROWS = 64, SAMPLES = 31, CHANNELS = 4, NONOTE = 0xFFFF, NONOTE8 = 0xff };
+
+    typedef struct Sample {
+      uint16_t length;
+      int8_t fineTune;
+      uint8_t volume;
+      uint16_t loopBegin;
+      uint16_t loopLength;
+    } Sample;
+    
+    typedef struct mod {
+      Sample samples[SAMPLES];
+      uint8_t songLength;
+      uint8_t numberOfPatterns;
+      uint8_t order[128];
+      uint8_t numberOfChannels;
+    } mod;
+    
+    // Save 256 bytes by storing raw note values, unpack with macro NOTE
+    typedef struct Pattern {
+      uint8_t sampleNumber[ROWS][CHANNELS];
+      uint8_t note8[ROWS][CHANNELS];
+      uint8_t effectNumber[ROWS][CHANNELS];
+      uint8_t effectParameter[ROWS][CHANNELS];
+    } Pattern;
+    
+    typedef struct player {
+      Pattern currentPattern;
+    
+      uint32_t amiga;
+      uint16_t samplesPerTick;
+      uint8_t speed;
+      uint8_t tick;
+      uint8_t row;
+      uint8_t lastRow;
+    
+      uint8_t orderIndex;
+      uint8_t oldOrderIndex;
+      uint8_t patternDelay;
+      uint8_t patternLoopCount[CHANNELS];
+      uint8_t patternLoopRow[CHANNELS];
+    
+      uint8_t lastSampleNumber[CHANNELS];
+      int8_t volume[CHANNELS];
+      uint16_t lastNote[CHANNELS];
+      uint16_t amigaPeriod[CHANNELS];
+      int16_t lastAmigaPeriod[CHANNELS];
+    
+      uint16_t portamentoNote[CHANNELS];
+      uint8_t portamentoSpeed[CHANNELS];
+    
+      uint8_t waveControl[CHANNELS];
+    
+      uint8_t vibratoSpeed[CHANNELS];
+      uint8_t vibratoDepth[CHANNELS];
+      int8_t vibratoPos[CHANNELS];
+    
+      uint8_t tremoloSpeed[CHANNELS];
+      uint8_t tremoloDepth[CHANNELS];
+      int8_t tremoloPos[CHANNELS];
+    } player;
+    
+    typedef struct mixer {
+      uint32_t sampleBegin[SAMPLES];
+      uint32_t sampleEnd[SAMPLES];
+      uint32_t sampleloopBegin[SAMPLES];
+      uint16_t sampleLoopLength[SAMPLES];
+      uint32_t sampleLoopEnd[SAMPLES];
+    
+      uint8_t channelSampleNumber[CHANNELS];
+      uint32_t channelSampleOffset[CHANNELS];
+      uint16_t channelFrequency[CHANNELS];
+      uint8_t channelVolume[CHANNELS];
+      uint8_t channelPanning[CHANNELS];
+    } mixer;
+    
+    typedef struct fatBuffer {
+      uint8_t *channels[CHANNELS]; // Make dynamically allocated [FATBUFFERSIZE];
+      uint32_t samplePointer[CHANNELS];
+      uint8_t channelSampleNumber[CHANNELS];
+    } fatBuffer;
+
+    // Effects
+    typedef enum { ARPEGGIO = 0, PORTAMENTOUP, PORTAMENTODOWN, TONEPORTAMENTO, VIBRATO, PORTAMENTOVOLUMESLIDE,
+                   VIBRATOVOLUMESLIDE, TREMOLO, SETCHANNELPANNING, SETSAMPLEOFFSET, VOLUMESLIDE, JUMPTOORDER,
+                   SETVOLUME, BREAKPATTERNTOROW, ESUBSET, SETSPEED } EffectsValues;
+    
+    // 0xE subset
+    typedef enum { SETFILTER = 0, FINEPORTAMENTOUP, FINEPORTAMENTODOWN, GLISSANDOCONTROL, SETVIBRATOWAVEFORM,
+                   SETFINETUNE, PATTERNLOOP, SETTREMOLOWAVEFORM, SUBEFFECT8, RETRIGGERNOTE, FINEVOLUMESLIDEUP,
+                   FINEVOLUMESLIDEDOWN, NOTECUT, NOTEDELAY, PATTERNDELAY, INVERTLOOP } Effect08Subvalues;
+    
+    // Our state lives here...
+    player Player;
+    mod Mod;
+    mixer Mixer;
+    fatBuffer FatBuffer;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorMP3.cpp

@@ -0,0 +1,412 @@
+/*
+  AudioGeneratorMP3
+  Wrap libmad MP3 library to play audio
+  
+  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 "AudioGeneratorMP3.h"
+
+AudioGeneratorMP3::AudioGeneratorMP3()
+{
+  running = false;
+  file = NULL;
+  output = NULL;
+  buff = NULL;
+  nsCountMax = 1152/32;
+  madInitted = false;
+}
+
+AudioGeneratorMP3::AudioGeneratorMP3(void *space, int size): preallocateSpace(space), preallocateSize(size)
+{
+  running = false;
+  file = NULL;
+  output = NULL;
+  buff = NULL;
+  nsCountMax = 1152/32;
+  madInitted = false;
+}
+
+AudioGeneratorMP3::AudioGeneratorMP3(void *buff, int buffSize, void *stream, int streamSize, void *frame, int frameSize, void *synth, int synthSize):
+    preallocateSpace(buff), preallocateSize(buffSize),
+    preallocateStreamSpace(stream), preallocateStreamSize(streamSize),
+    preallocateFrameSpace(frame), preallocateFrameSize(frameSize),
+    preallocateSynthSpace(synth), preallocateSynthSize(synthSize)
+{
+  running = false;
+  file = NULL;
+  output = NULL;
+  buff = NULL;
+  nsCountMax = 1152/32;
+  madInitted = false;
+}
+
+AudioGeneratorMP3::~AudioGeneratorMP3()
+{
+  if (!preallocateSpace) {
+    free(buff);
+    free(synth);
+    free(frame);
+    free(stream);
+  } 
+}
+
+
+bool AudioGeneratorMP3::stop()
+{
+  if (madInitted) {
+    mad_synth_finish(synth);
+    mad_frame_finish(frame);
+    mad_stream_finish(stream);
+    madInitted = false;
+  }
+
+  if (!preallocateSpace) {
+    free(buff);
+    free(synth);
+    free(frame);
+    free(stream);
+  }
+
+  buff = NULL;
+  synth = NULL;
+  frame = NULL;
+  stream = NULL;
+
+  running = false;
+  output->stop();
+  return file->close();
+}
+
+bool AudioGeneratorMP3::isRunning()
+{
+  return running;
+}
+
+enum mad_flow AudioGeneratorMP3::ErrorToFlow()
+{
+  char err[64];
+  char errLine[128];
+
+  // Special case - eat "lost sync @ byte 0" as it always occurs and is not really correct....it never had sync!
+  if ((lastReadPos==0) && (stream->error==MAD_ERROR_LOSTSYNC)) return MAD_FLOW_CONTINUE;
+
+  strcpy_P(err, mad_stream_errorstr(stream));
+  snprintf_P(errLine, sizeof(errLine), PSTR("Decoding error '%s' at byte offset %d"),
+           err, (stream->this_frame - buff) + lastReadPos);
+  yield(); // Something bad happened anyway, ensure WiFi gets some time, too
+  cb.st(stream->error, errLine);
+  return MAD_FLOW_CONTINUE;
+}
+
+enum mad_flow AudioGeneratorMP3::Input()
+{
+  int unused = 0;
+
+  if (stream->next_frame) {
+    unused = lastBuffLen - (stream->next_frame - buff);
+    if (unused < 0) {
+      desync();
+      unused = 0;
+    } else {
+      memmove(buff, stream->next_frame, unused);
+    }
+    stream->next_frame = NULL;
+  }
+
+  if (unused == lastBuffLen) {
+    // Something wicked this way came, throw it all out and try again
+    unused = 0;
+  }
+
+  lastReadPos = file->getPos() - unused;
+  int len = buffLen - unused;
+  len = file->read(buff + unused, len);
+  if ((len == 0)  && (unused == 0)) {
+    // Can't read any from the file, and we don't have anything left.  It's done....
+    return MAD_FLOW_STOP;
+  }
+  if (len < 0) {
+    desync();
+    unused = 0;
+  }
+
+  lastBuffLen = len + unused;
+  mad_stream_buffer(stream, buff, lastBuffLen);
+
+  return MAD_FLOW_CONTINUE;
+}
+
+void AudioGeneratorMP3::desync ()
+{
+    audioLogger->printf_P(PSTR("MP3:desync\n"));
+    if (stream) {
+        stream->next_frame = nullptr;
+        stream->this_frame = nullptr;
+        stream->sync = 0;
+    }
+    lastBuffLen = 0;
+}
+
+bool AudioGeneratorMP3::DecodeNextFrame()
+{
+  if (mad_frame_decode(frame, stream) == -1) {
+    ErrorToFlow(); // Always returns CONTINUE
+    return false;
+  }
+  nsCountMax  = MAD_NSBSAMPLES(&frame->header);
+  return true;
+}
+
+bool AudioGeneratorMP3::GetOneSample(int16_t sample[2])
+{
+  if (synth->pcm.samplerate != lastRate) {
+    output->SetRate(synth->pcm.samplerate);
+    lastRate = synth->pcm.samplerate;
+  }
+  if (synth->pcm.channels != lastChannels) {
+    output->SetChannels(synth->pcm.channels);
+    lastChannels = synth->pcm.channels;
+  }
+    
+  // If we're here, we have one decoded frame and sent 0 or more samples out
+  if (samplePtr < synth->pcm.length) {
+    sample[AudioOutput::LEFTCHANNEL ] = synth->pcm.samples[0][samplePtr];
+    sample[AudioOutput::RIGHTCHANNEL] = synth->pcm.samples[1][samplePtr];
+    samplePtr++;
+  } else {
+    samplePtr = 0;
+    
+    switch ( mad_synth_frame_onens(synth, frame, nsCount++) ) {
+        case MAD_FLOW_STOP:
+        case MAD_FLOW_BREAK: audioLogger->printf_P(PSTR("msf1ns failed\n"));
+          return false; // Either way we're done
+        default:
+          break; // Do nothing
+    }
+    // for IGNORE and CONTINUE, just play what we have now
+    sample[AudioOutput::LEFTCHANNEL ] = synth->pcm.samples[0][samplePtr];
+    sample[AudioOutput::RIGHTCHANNEL] = synth->pcm.samples[1][samplePtr];
+    samplePtr++;
+  }
+  return true;
+}
+
+
+bool AudioGeneratorMP3::loop()
+{
+  if (!running) goto done; // Nothing to do here!
+
+  // First, try and push in the stored sample.  If we can't, then punt and try later
+  if (!output->ConsumeSample(lastSample)) goto done; // Can't send, but no error detected
+
+  // Try and stuff the buffer one sample at a time
+  do
+  {
+    // Decode next frame if we're beyond the existing generated data
+    if ( (samplePtr >= synth->pcm.length) && (nsCount >= nsCountMax) ) {
+retry:
+      if (Input() == MAD_FLOW_STOP) {
+        return false;
+      }
+
+      if (!DecodeNextFrame()) {
+        if (stream->error == MAD_ERROR_BUFLEN) {
+          // randomly seeking can lead to endless
+          // and unrecoverable "MAD_ERROR_BUFLEN" loop
+          audioLogger->printf_P(PSTR("MP3:ERROR_BUFLEN %d\n"), unrecoverable);
+          if (++unrecoverable >= 3) {
+            unrecoverable = 0;
+            stop();
+            return running;
+          }
+        } else {
+          unrecoverable = 0;
+        }
+        goto retry;
+      }
+      samplePtr = 9999;
+      nsCount = 0;
+    }
+
+    if (!GetOneSample(lastSample)) {
+      audioLogger->printf_P(PSTR("G1S failed\n"));
+      running = false;
+      goto done;
+    }
+  } while (running && output->ConsumeSample(lastSample));
+
+done:
+  file->loop();
+  output->loop();
+
+  return running;
+}
+
+
+
+bool AudioGeneratorMP3::begin(AudioFileSource *source, AudioOutput *output)
+{
+  if (!source)  return false;
+  file = source;
+  if (!output) return false;
+  this->output = output;
+  if (!file->isOpen()) {
+    audioLogger->printf_P(PSTR("MP3 source file not open\n"));
+    return false; // Error
+  }
+
+  // Reset error count from previous file
+  unrecoverable = 0;
+
+  output->SetBitsPerSample(16); // Constant for MP3 decoder
+  output->SetChannels(2);
+
+  if (!output->begin()) return false;
+
+  // Where we are in generating one frame's data, set to invalid so we will run loop on first getsample()
+  samplePtr = 9999;
+  nsCount = 9999;
+  lastRate = 0;
+  lastChannels = 0;
+  lastReadPos = 0;
+  lastBuffLen = 0;
+
+  // Allocate all large memory chunks
+  if (preallocateStreamSize + preallocateFrameSize + preallocateSynthSize) {
+    if (preallocateSize >= preAllocBuffSize() &&
+        preallocateStreamSize >= preAllocStreamSize() &&
+        preallocateFrameSize >= preAllocFrameSize() &&
+        preallocateSynthSize >= preAllocSynthSize()) {
+      buff = reinterpret_cast<unsigned char *>(preallocateSpace);
+      stream = reinterpret_cast<struct mad_stream *>(preallocateStreamSpace);
+      frame = reinterpret_cast<struct mad_frame *>(preallocateFrameSpace);
+      synth = reinterpret_cast<struct mad_synth *>(preallocateSynthSpace);
+    }
+    else {
+      audioLogger->printf_P("OOM error in MP3:  Want %d/%d/%d/%d bytes, have %d/%d/%d/%d bytes preallocated.\n",
+          preAllocBuffSize(), preAllocStreamSize(), preAllocFrameSize(), preAllocSynthSize(),
+          preallocateSize, preallocateStreamSize, preallocateFrameSize, preallocateSynthSize);
+      return false;
+    }
+  } else if (preallocateSpace) {
+    uint8_t *p = reinterpret_cast<uint8_t *>(preallocateSpace);
+    buff = reinterpret_cast<unsigned char *>(p);
+    p += preAllocBuffSize();
+    stream = reinterpret_cast<struct mad_stream *>(p);
+    p += preAllocStreamSize();
+    frame = reinterpret_cast<struct mad_frame *>(p);
+    p += preAllocFrameSize();
+    synth = reinterpret_cast<struct mad_synth *>(p);
+    p += preAllocSynthSize();
+    int neededBytes = p - reinterpret_cast<uint8_t *>(preallocateSpace);
+    if (neededBytes > preallocateSize) {
+      audioLogger->printf_P("OOM error in MP3:  Want %d bytes, have %d bytes preallocated.\n", neededBytes, preallocateSize);
+      return false;
+    }
+  } else {
+    buff = reinterpret_cast<unsigned char *>(malloc(buffLen));
+    stream = reinterpret_cast<struct mad_stream *>(malloc(sizeof(struct mad_stream)));
+    frame = reinterpret_cast<struct mad_frame *>(malloc(sizeof(struct mad_frame)));
+    synth = reinterpret_cast<struct mad_synth *>(malloc(sizeof(struct mad_synth)));
+    if (!buff || !stream || !frame || !synth) {
+      free(buff);
+      free(stream);
+      free(frame);
+      free(synth);
+      buff = NULL;
+      stream = NULL;
+      frame = NULL;
+      synth = NULL;
+      return false;
+    }
+  }
+ 
+  mad_stream_init(stream);
+  mad_frame_init(frame);
+  mad_synth_init(synth);
+  synth->pcm.length = 0;
+  mad_stream_options(stream, 0); // TODO - add options support
+  madInitted = true;
+ 
+  running = true;
+  return true;
+}
+
+// The following are helper routines for use in libmad to check stack/heap free
+// and to determine if there's enough stack space to allocate some blocks there
+// instead of precious heap.
+
+#undef stack
+extern "C" {
+#ifdef ESP32
+  //TODO - add ESP32 checks
+  void stack(const char *s, const char *t, int i)
+  {
+  }
+  int stackfree()
+  {
+    return 8192;
+  }
+#elif defined(ESP8266) && !defined(CORE_MOCK)
+  #include <cont.h>
+  extern cont_t g_cont;
+
+  void stack(const char *s, const char *t, int i)
+  {
+    (void) t;
+    (void) i;
+    register uint32_t *sp asm("a1");
+    int freestack = 4 * (sp - g_cont.stack);
+    int freeheap = ESP.getFreeHeap();
+    if ((freestack < 512) || (freeheap < 5120)) {
+      static int laststack, lastheap;
+      if (laststack!=freestack|| lastheap !=freeheap) {
+        audioLogger->printf_P(PSTR("%s: FREESTACK=%d, FREEHEAP=%d\n"), s, /*t, i,*/ freestack, /*cont_get_free_stack(&g_cont),*/ freeheap);
+      }
+      if (freestack < 256) {
+        audioLogger->printf_P(PSTR("out of stack!\n"));
+      }
+      if (freeheap < 1024) {
+        audioLogger->printf_P(PSTR("out of heap!\n"));
+      }
+      Serial.flush();
+      laststack = freestack;
+      lastheap = freeheap;
+    }
+  }
+
+  int stackfree()
+  {
+    register uint32_t *sp asm("a1");
+    int freestack = 4 * (sp - g_cont.stack);
+    return freestack;
+  }
+#else
+  void stack(const char *s, const char *t, int i)
+  {
+    (void) s;
+    (void) t;
+    (void) i;
+  }
+  int stackfree()
+  {
+    return 8192;
+  }
+#endif
+}
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorMP3.h

@@ -0,0 +1,84 @@
+/*
+  AudioGeneratorMP3
+  Wrap libmad MP3 library to play audio
+    
+  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/>.
+*/
+
+#ifndef _AUDIOGENERATORMP3_H
+#define _AUDIOGENERATORMP3_H
+
+#include "AudioGenerator.h"
+#include "libmad/config.h"
+#include "libmad/mad.h"
+
+class AudioGeneratorMP3 : public AudioGenerator
+{
+  public:
+    AudioGeneratorMP3();
+    AudioGeneratorMP3(void *preallocateSpace, int preallocateSize);
+    AudioGeneratorMP3(void *buff, int buffSize, void *stream, int streamSize, void *frame, int frameSize, void *synth, int synthSize);
+    virtual ~AudioGeneratorMP3() override;
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override;
+    virtual void desync () override;
+
+    static constexpr int preAllocSize () { return preAllocBuffSize() + preAllocStreamSize() + preAllocFrameSize() + preAllocSynthSize(); }
+    static constexpr int preAllocBuffSize () { return ((buffLen + 7) & ~7); }
+    static constexpr int preAllocStreamSize () { return ((sizeof(struct mad_stream) + 7) & ~7); }
+    static constexpr int preAllocFrameSize () { return (sizeof(struct mad_frame) + 7) & ~7; }
+    static constexpr int preAllocSynthSize () { return (sizeof(struct mad_synth) + 7) & ~7; }
+
+  protected:   
+    void *preallocateSpace = nullptr;
+    int preallocateSize = 0;
+    void *preallocateStreamSpace = nullptr;
+    int preallocateStreamSize = 0;
+    void *preallocateFrameSpace = nullptr;
+    int preallocateFrameSize = 0;
+    void *preallocateSynthSpace = nullptr;
+    int preallocateSynthSize = 0;
+
+    static constexpr int buffLen = 0x600; // Slightly larger than largest MP3 frame
+    unsigned char *buff;
+    int lastReadPos;
+    int lastBuffLen;
+    unsigned int lastRate;
+    int lastChannels;
+    
+    // Decoding bits
+    bool madInitted;
+    struct mad_stream *stream;
+    struct mad_frame *frame;
+    struct mad_synth *synth;
+    int samplePtr;
+    int nsCount;
+    int nsCountMax;
+
+    // The internal helpers
+    enum mad_flow ErrorToFlow();
+    enum mad_flow Input();
+    bool DecodeNextFrame();
+    bool GetOneSample(int16_t sample[2]);
+
+  private:
+    int unrecoverable = 0;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorMP3a.cpp

@@ -0,0 +1,162 @@
+/*
+  AudioGeneratorMP3
+  Audio output generator using the Helix MP3 decoder
+  
+  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/>.
+*/
+
+#pragma GCC optimize ("O3")
+
+#include "AudioGeneratorMP3a.h"
+
+
+AudioGeneratorMP3a::AudioGeneratorMP3a()
+{
+  running = false;
+  file = NULL;
+  output = NULL;
+  hMP3Decoder = MP3InitDecoder();
+  if (!hMP3Decoder) {
+    audioLogger->printf_P(PSTR("Out of memory error! hMP3Decoder==NULL\n"));
+    Serial.flush();
+  }
+  // For sanity's sake...
+  memset(buff, 0, sizeof(buff));
+  memset(outSample, 0, sizeof(outSample));
+  buffValid = 0;
+  lastFrameEnd = 0;
+  validSamples = 0;
+  curSample = 0;
+  lastRate = 0;
+  lastChannels = 0;
+}
+
+AudioGeneratorMP3a::~AudioGeneratorMP3a()
+{
+  MP3FreeDecoder(hMP3Decoder);
+}
+
+bool AudioGeneratorMP3a::stop()
+{
+  if (!running) return true;
+  running = false;
+  output->stop();
+  return file->close();
+}
+
+bool AudioGeneratorMP3a::isRunning()
+{
+  return running;
+}
+
+bool AudioGeneratorMP3a::FillBufferWithValidFrame()
+{
+  buff[0] = 0; // Destroy any existing sync word @ 0
+  int nextSync;
+  do {
+    nextSync = MP3FindSyncWord(buff + lastFrameEnd, buffValid - lastFrameEnd);
+    if (nextSync >= 0) nextSync += lastFrameEnd;
+    lastFrameEnd = 0;
+    if (nextSync == -1) {
+      if (buff[buffValid-1]==0xff) { // Could be 1st half of syncword, preserve it...
+        buff[0] = 0xff;
+        buffValid = file->read(buff+1, sizeof(buff)-1);
+        if (buffValid==0) return false; // No data available, EOF
+      } else { // Try a whole new buffer
+        buffValid = file->read(buff, sizeof(buff));
+        if (buffValid==0) return false; // No data available, EOF
+      }
+    }
+  } while (nextSync == -1);
+
+  // Move the frame to start at offset 0 in the buffer
+  buffValid -= nextSync; // Throw out prior to nextSync
+  memmove(buff, buff+nextSync, buffValid);
+
+  // We have a sync word at 0 now, try and fill remainder of buffer
+  buffValid += file->read(buff + buffValid, sizeof(buff) - buffValid);
+
+  return true;
+}
+
+bool AudioGeneratorMP3a::loop()
+{
+  if (!running) goto done; // Nothing to do here!
+
+  // If we've got data, try and pump it out...
+  while (validSamples) {
+    lastSample[0] = outSample[curSample*2];
+    lastSample[1] = outSample[curSample*2 + 1];
+    if (!output->ConsumeSample(lastSample)) goto done; // Can't send, but no error detected
+    validSamples--;
+    curSample++;
+  }
+
+  // No samples available, need to decode a new frame
+  if (FillBufferWithValidFrame()) {
+    // buff[0] start of frame, decode it...
+    unsigned char *inBuff = reinterpret_cast<unsigned char *>(buff);
+    int bytesLeft = buffValid;
+    int ret = MP3Decode(hMP3Decoder, &inBuff, &bytesLeft, outSample, 0);
+   if (ret) {
+      // Error, skip the frame...
+      char buff[48];
+      sprintf(buff, "MP3 decode error %d", ret);
+      cb.st(ret, buff);
+    } else {
+      lastFrameEnd = buffValid - bytesLeft;
+      MP3FrameInfo fi;
+      MP3GetLastFrameInfo(hMP3Decoder, &fi);
+      if ((int)fi.samprate!= (int)lastRate) {
+        output->SetRate(fi.samprate);
+        lastRate = fi.samprate;
+      }
+      if (fi.nChans != lastChannels) {
+        output->SetChannels(fi.nChans);
+        lastChannels = fi.nChans;
+      }
+      curSample = 0;
+      validSamples = fi.outputSamps / lastChannels;
+    }
+  } else {
+    running = false; // No more data, we're done here...
+  }
+
+done:
+  file->loop();
+  output->loop();
+
+  return running;
+}
+
+bool AudioGeneratorMP3a::begin(AudioFileSource *source, AudioOutput *output)
+{
+  if (!source) return false;
+  file = source;
+  if (!output) return false;
+  this->output = output;
+  if (!file->isOpen()) return false; // Error
+
+  output->begin();
+  
+  // AAC always comes out at 16 bits
+  output->SetBitsPerSample(16);
+  
+  running = true;
+  
+  return true;
+}
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorMP3a.h

@@ -0,0 +1,57 @@
+/*
+  AudioGeneratorMP3
+  Audio output generator using the Helix MP3 decoder
+  
+  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/>.
+*/
+
+#ifndef _AUDIOGENERATORMP3A_H
+#define _AUDIOGENERATORMP3A_H
+
+#include "AudioGenerator.h"
+#include "libhelix-mp3/mp3dec.h"
+
+class AudioGeneratorMP3a : public AudioGenerator
+{
+  public:
+    AudioGeneratorMP3a();
+    virtual ~AudioGeneratorMP3a() override;
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override;
+
+  protected:
+    // Helix MP3 decoder
+    HMP3Decoder hMP3Decoder;
+
+    // Input buffering
+    uint8_t buff[1600]; // File buffer required to store at least a whole compressed frame
+    int16_t buffValid;
+    int16_t lastFrameEnd;
+    bool FillBufferWithValidFrame(); // Read until we get a valid syncword and min(feof, 2048) butes in the buffer
+
+    // Output buffering
+    int16_t outSample[1152 * 2]; // Interleaved L/R
+    int16_t validSamples;
+    int16_t curSample;
+
+    // Each frame may change this if they're very strange, I guess
+    unsigned int lastRate;
+    int lastChannels;
+};
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorOpus.cpp

@@ -0,0 +1,142 @@
+/*
+  AudioGeneratorOpus
+  Audio output generator that plays Opus audio files
+    
+  Copyright (C) 2020  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 <AudioGeneratorOpus.h>
+
+AudioGeneratorOpus::AudioGeneratorOpus()
+{
+  of = nullptr;
+  buff = nullptr;
+  buffPtr = 0;
+  buffLen = 0;
+  running = false;
+}
+
+AudioGeneratorOpus::~AudioGeneratorOpus()
+{
+  if (of) op_free(of);
+  of = nullptr;
+  free(buff);
+  buff = nullptr;
+}
+
+#define OPUS_BUFF 1024
+
+bool AudioGeneratorOpus::begin(AudioFileSource *source, AudioOutput *output)
+{
+  buff = (int16_t*)malloc(OPUS_BUFF * sizeof(int16_t));
+  if (!buff) return false;
+
+  if (!source) return false;
+  file = source;
+  if (!output) return false;
+  this->output = output;
+  if (!file->isOpen()) return false; // Error
+
+  of = op_open_callbacks((void*)this, &cb, nullptr, 0, nullptr);
+  if (!of) return false;
+
+  prev_li = -1;
+  lastSample[0] = 0;
+  lastSample[1] = 0;
+
+  buffPtr = 0;
+  buffLen = 0;
+
+  output->begin();
+
+  // These are fixed by Opus
+  output->SetRate(48000);
+  output->SetBitsPerSample(16);
+  output->SetChannels(2);
+
+  running = true;
+  return true;
+}
+
+bool AudioGeneratorOpus::loop()
+{
+
+  if (!running) goto done;
+
+  if (!output->ConsumeSample(lastSample)) goto done; // Try and send last buffered sample
+
+  do {
+    if (buffPtr == buffLen) {
+      int ret = op_read_stereo(of, (opus_int16 *)buff, OPUS_BUFF);
+      if (ret == OP_HOLE) {
+        // fprintf(stderr,"\nHole detected! Corrupt file segment?\n");
+        continue;
+      } else if (ret <= 0) {
+        running = false;
+        goto done;
+      }
+     buffPtr = 0;
+     buffLen = ret * 2;
+    }
+
+    lastSample[AudioOutput::LEFTCHANNEL] = buff[buffPtr] & 0xffff; 
+    lastSample[AudioOutput::RIGHTCHANNEL] = buff[buffPtr+1] & 0xffff; 
+    buffPtr += 2;
+  } while (running && output->ConsumeSample(lastSample));
+
+done:
+  file->loop();
+  output->loop();
+
+  return running;
+}
+
+bool AudioGeneratorOpus::stop()
+{
+  if (of) op_free(of);
+  of = nullptr;
+  free(buff);
+  buff = nullptr;
+  running = false;
+  output->stop();
+  return true;
+}
+
+bool AudioGeneratorOpus::isRunning()
+{
+  return running;
+}
+
+int AudioGeneratorOpus::read_cb(unsigned char *_ptr, int _nbytes) {
+  if (_nbytes == 0) return 0;
+  _nbytes = file->read(_ptr, _nbytes);
+  if (_nbytes == 0) return -1;
+  return _nbytes;
+}
+
+int AudioGeneratorOpus::seek_cb(opus_int64 _offset, int _whence) {
+  if (!file->seek((int32_t)_offset, _whence)) return -1;
+  return 0;
+}
+
+opus_int64 AudioGeneratorOpus::tell_cb() {
+  return file->getPos();
+}
+
+int AudioGeneratorOpus::close_cb() {
+  // NO OP, we close in main loop
+  return 0;
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorOpus.h

@@ -0,0 +1,70 @@
+/*
+  AudioGeneratorOpus
+  Audio output generator that plays Opus audio files
+    
+  Copyright (C) 2020  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/>.
+*/
+
+#ifndef _AUDIOGENERATOROPUS_H
+#define _AUDIOGENERATOROPUS_H
+
+#include <AudioGenerator.h>
+//#include "libopus/opus.h"
+#include "opusfile/opusfile.h"
+
+class AudioGeneratorOpus : public AudioGenerator
+{
+  public:
+    AudioGeneratorOpus();
+    virtual ~AudioGeneratorOpus() override;
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override;
+
+  protected:
+    // Opus callbacks, need static functions to bounce into C++ from C
+    static int OPUS_read(void *_stream, unsigned char *_ptr, int _nbytes) {
+      return static_cast<AudioGeneratorOpus*>(_stream)->read_cb(_ptr, _nbytes);
+    }
+    static int OPUS_seek(void *_stream, opus_int64 _offset, int _whence) {
+      return static_cast<AudioGeneratorOpus*>(_stream)->seek_cb(_offset, _whence);
+    }
+    static opus_int64 OPUS_tell(void *_stream) {
+      return static_cast<AudioGeneratorOpus*>(_stream)->tell_cb();
+    }
+    static int OPUS_close(void *_stream) {
+      return static_cast<AudioGeneratorOpus*>(_stream)->close_cb();
+    }
+
+    // Actual Opus callbacks
+    int read_cb(unsigned char *_ptr, int _nbytes);
+    int seek_cb(opus_int64 _offset, int _whence);
+    opus_int64 tell_cb();
+    int close_cb();
+
+  private:
+    OpusFileCallbacks cb = {OPUS_read, OPUS_seek, OPUS_tell, OPUS_close};
+    OggOpusFile *of;
+    int prev_li; // To detect changes in streams
+
+    int16_t *buff;
+    uint32_t buffPtr;
+    uint32_t buffLen;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorRTTTL.cpp

@@ -0,0 +1,292 @@
+/*
+  AudioGeneratorRTTTL
+  Audio output generator that plays RTTTL (Nokia ringtone)
+
+  Based on the Rtttl Arduino library by James BM, https://github.com/spicajames/Rtttl
+  Based on the gist from Daniel Hall https://gist.github.com/smarthall/1618800
+  
+  Copyright (C) 2018  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 "AudioGeneratorRTTTL.h"
+
+AudioGeneratorRTTTL::AudioGeneratorRTTTL()
+{
+  running = false;
+  file = NULL;
+  output = NULL;
+  rate = 22050;
+  buff = nullptr;
+  ptr = 0;
+}
+
+AudioGeneratorRTTTL::~AudioGeneratorRTTTL()
+{
+  free(buff);
+}
+
+bool AudioGeneratorRTTTL::stop()
+{
+  if (!running) return true;
+  running = false;
+  output->stop();
+  return file->close();
+}
+
+bool AudioGeneratorRTTTL::isRunning()
+{
+  return running;
+}
+
+bool AudioGeneratorRTTTL::loop()
+{
+  if (!running) goto done; // Nothing to do here!
+
+  // Load the next note, if we've hit the end of the last one
+  if (samplesSent == ttlSamples) {
+    if (!GetNextNote()) {
+      running = false;
+      goto done;
+    }
+    samplesSent = 0;
+  }
+  
+  // Try and send out the remainder of the existing note, one per loop()
+  if (ttlSamplesPerWaveFP10 == 0) { // Mute
+    int16_t mute[2] = {0, 0};
+    while ((samplesSent < ttlSamples) && output->ConsumeSample(mute)) {
+      samplesSent++;
+    }
+  } else {
+    while (samplesSent < ttlSamples) {
+      int samplesSentFP10 = samplesSent << 10;
+      int rem = samplesSentFP10 % ttlSamplesPerWaveFP10;
+      int16_t val = (rem > ttlSamplesPerWaveFP10/2) ? 8192:-8192;
+      int16_t s[2] = { val, val };
+      if (!output->ConsumeSample(s)) goto done;
+      samplesSent++;
+    }
+  }
+
+done:
+  file->loop();
+  output->loop();
+
+  return running;
+}
+
+bool AudioGeneratorRTTTL::SkipWhitespace()
+{
+  while ((ptr < len) && (buff[ptr] == ' ')) ptr++;
+  return ptr < len;
+}
+
+bool AudioGeneratorRTTTL::ReadInt(int *dest)
+{
+  if (ptr >= len) return false;
+
+  SkipWhitespace();
+  if (ptr >= len) return false;
+  if ((buff[ptr] <'0') || (buff[ptr] > '9')) return false;
+
+  int t = 0;
+  while ((buff[ptr] >= '0') && (buff[ptr] <='9')) {
+    t = (t * 10) + (buff[ptr] - '0');
+    ptr++;
+  }
+  *dest = t;
+  return true;
+}
+  
+
+bool AudioGeneratorRTTTL::ParseHeader()
+{
+  // Skip the title
+  while ((ptr < len) && (buff[ptr] != ':')) ptr++;
+  if (ptr >= len) return false;
+  if (buff[ptr++] != ':') return false;
+  if (!SkipWhitespace()) return false;
+  if ((buff[ptr] != 'd') && (buff[ptr] != 'D')) return false;
+  ptr++;
+  if (!SkipWhitespace()) return false;
+  if (buff[ptr++] != '=') return false;
+  if (!ReadInt(&defaultDuration)) return false;
+  if (!SkipWhitespace()) return false;
+  if (buff[ptr++] != ',') return false;
+
+  if (!SkipWhitespace()) return false;
+  if ((buff[ptr] != 'o') && (buff[ptr] != 'O')) return false;
+  ptr++;
+  if (!SkipWhitespace()) return false;
+  if (buff[ptr++] != '=') return false;
+  if (!ReadInt(&defaultOctave)) return false;
+  if (!SkipWhitespace()) return false;
+  if (buff[ptr++] != ',') return false;
+
+  int bpm;
+  if (!SkipWhitespace()) return false;
+  if ((buff[ptr] != 'b') && (buff[ptr] != 'B')) return false;
+  ptr++;
+  if (!SkipWhitespace()) return false;
+  if (buff[ptr++] != '=') return false;
+  if (!ReadInt(&bpm)) return false;
+  if (!SkipWhitespace()) return false;
+  if (buff[ptr++] != ':') return false;
+
+  wholeNoteMS = (60 * 1000 * 4) / bpm;
+
+  return true;
+}
+
+#define NOTE_C4  262
+#define NOTE_CS4 277
+#define NOTE_D4  294
+#define NOTE_DS4 311
+#define NOTE_E4  330
+#define NOTE_F4  349
+#define NOTE_FS4 370
+#define NOTE_G4  392
+#define NOTE_GS4 415
+#define NOTE_A4  440
+#define NOTE_AS4 466
+#define NOTE_B4  494
+#define NOTE_C5  523
+#define NOTE_CS5 554
+#define NOTE_D5  587
+#define NOTE_DS5 622
+#define NOTE_E5  659
+#define NOTE_F5  698
+#define NOTE_FS5 740
+#define NOTE_G5  784
+#define NOTE_GS5 831
+#define NOTE_A5  880
+#define NOTE_AS5 932
+#define NOTE_B5  988
+#define NOTE_C6  1047
+#define NOTE_CS6 1109
+#define NOTE_D6  1175
+#define NOTE_DS6 1245
+#define NOTE_E6  1319
+#define NOTE_F6  1397
+#define NOTE_FS6 1480
+#define NOTE_G6  1568
+#define NOTE_GS6 1661
+#define NOTE_A6  1760
+#define NOTE_AS6 1865
+#define NOTE_B6  1976
+#define NOTE_C7  2093
+#define NOTE_CS7 2217
+#define NOTE_D7  2349
+#define NOTE_DS7 2489
+#define NOTE_E7  2637
+#define NOTE_F7  2794
+#define NOTE_FS7 2960
+#define NOTE_G7  3136
+#define NOTE_GS7 3322
+#define NOTE_A7  3520
+#define NOTE_AS7 3729
+#define NOTE_B7 3951
+static int notes[49] = { 0,
+  NOTE_C4, NOTE_CS4, NOTE_D4, NOTE_DS4, NOTE_E4, NOTE_F4, NOTE_FS4, NOTE_G4, NOTE_GS4, NOTE_A4, NOTE_AS4, NOTE_B4,
+  NOTE_C5, NOTE_CS5, NOTE_D5, NOTE_DS5, NOTE_E5, NOTE_F5, NOTE_FS5, NOTE_G5, NOTE_GS5, NOTE_A5, NOTE_AS5, NOTE_B5,
+  NOTE_C6, NOTE_CS6, NOTE_D6, NOTE_DS6, NOTE_E6, NOTE_F6, NOTE_FS6, NOTE_G6, NOTE_GS6, NOTE_A6, NOTE_AS6, NOTE_B6,
+  NOTE_C7, NOTE_CS7, NOTE_D7, NOTE_DS7, NOTE_E7, NOTE_F7, NOTE_FS7, NOTE_G7, NOTE_GS7, NOTE_A7, NOTE_AS7, NOTE_B7 };
+
+bool AudioGeneratorRTTTL::GetNextNote()
+{
+  int dur, note, scale;
+  if (ptr >= len) return false;
+
+  if (!ReadInt(&dur)) {
+    dur = defaultDuration;
+  }
+  dur = wholeNoteMS / dur;
+
+  if (ptr >= len) return false;
+  note = 0;
+  switch (buff[ptr++]) {
+    case 'c': case 'C': note = 1; break;
+    case 'd': case 'D': note = 3; break;
+    case 'e': case 'E': note = 5; break;
+    case 'f': case 'F': note = 6; break;
+    case 'g': case 'G': note = 8; break;
+    case 'a': case 'A': note = 10; break;
+    case 'b': case 'B': note = 12; break;
+    case 'p': case 'P': note = 0; break;
+    default: return false;
+  }
+  if ((ptr < len) && (buff[ptr] == '#')) {
+    ptr++;
+    note++;
+  }
+  if ((ptr < len) && (buff[ptr] == '.')) {
+    ptr++;
+    dur += dur / 2;
+  }
+  if (!ReadInt(&scale)) {
+    scale = defaultOctave;
+  }
+  // Eat any trailing whitespace and comma
+  SkipWhitespace();
+  if ((ptr < len) && (buff[ptr]==',')) {
+    ptr++;
+  }
+
+  if (scale < 4) scale = 4;
+  if (scale > 7) scale = 7;
+  if (note) {
+    int freq = notes[(scale - 4) * 12 + note];
+    // Convert from frequency in Hz to high and low samples in fixed point
+    ttlSamplesPerWaveFP10 = (rate << 10) / freq;
+  } else {
+    ttlSamplesPerWaveFP10 = 0;
+  }
+  ttlSamples = (rate * dur ) / 1000;
+
+  //audioLogger->printf("%d %d %d %d %d\n", dur, note, scale, ttlSamplesPerWaveFP10, ttlSamples );
+
+  return true;
+}
+
+bool AudioGeneratorRTTTL::begin(AudioFileSource *source, AudioOutput *output)
+{
+  if (!source) return false;
+  file = source;
+  if (!output) return false;
+  this->output = output;
+  if (!file->isOpen()) return false; // Error
+  
+  len = file->getSize();
+  buff = (char *)malloc(len);
+  if (!buff) return false;
+  if (file->read(buff, len) != (uint32_t)len) return false;
+
+  ptr = 0;
+  samplesSent = 0;
+  ttlSamples = 0;
+
+  if (!ParseHeader()) return false;
+
+  if (!output->SetRate( rate )) return false;
+  if (!output->SetBitsPerSample( 16 )) return false;
+  if (!output->SetChannels( 2 )) return false;
+  if (!output->begin()) return false;
+
+  running = true;
+  
+  return true;
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorRTTTL.h

@@ -0,0 +1,66 @@
+/*
+  AudioGeneratorRTTTL
+  Audio output generator that plays RTTTL (Nokia ringtones)
+
+  Based on the Rtttl Arduino library by James BM, https://github.com/spicajames/Rtttl
+  Based on the gist from Daniel Hall https://gist.github.com/smarthall/1618800
+    
+  Copyright (C) 2018  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/>.
+*/
+
+#ifndef _AUDIOGENERATORRTTTL_H
+#define _AUDIOGENERATORRTTTL_H
+
+#include "AudioGenerator.h"
+
+class AudioGeneratorRTTTL : public AudioGenerator
+{
+  public:
+    AudioGeneratorRTTTL();
+    virtual ~AudioGeneratorRTTTL() override;
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override;
+    void SetRate(uint16_t hz) { rate = hz; }
+
+  private:
+    bool SkipWhitespace();
+    bool ReadInt(int *dest);
+    bool ParseHeader();
+    bool GetNextNote();
+    
+  protected:
+    uint16_t rate;
+
+    // We copy the entire tiny song to a buffer for easier access
+    char *buff;
+    int len;
+    int ptr;
+
+    // Song-global settings
+    int defaultDuration;
+    int defaultOctave;
+    int wholeNoteMS;
+
+    // The note we're currently playing
+    int ttlSamplesPerWaveFP10;
+    int ttlSamples;
+    int samplesSent;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorTalkie.cpp

@@ -0,0 +1,302 @@
+/*
+  AudioGeneratorTalkie
+  Audio output generator that speaks using the LPC code in old TI speech chips
+  Output is locked at 8khz as that's that the hardcoded LPC coefficients are built around
+
+  Based on the Talkie Arduino library by Peter Knight, https://github.com/going-digital/Talkie
+    
+  Copyright (C) 2020 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 "AudioGeneratorTalkie.h"
+
+AudioGeneratorTalkie::AudioGeneratorTalkie()
+{
+  running = false;
+  lastFrame = false;
+  file = nullptr;
+  output = nullptr;
+  buff = nullptr;
+}
+
+AudioGeneratorTalkie::~AudioGeneratorTalkie()
+{
+  free(buff);
+}
+
+bool AudioGeneratorTalkie::say(const uint8_t *data, size_t len, bool async) {
+  // Finish saying anything in the pipe
+  while (running) {
+    loop();
+    delay(0);
+  }
+  buff = (uint8_t*)realloc(buff, len);
+  if (!buff) return false;
+  memcpy_P(buff, data, len);
+
+  // Reset the interpreter to the start of the stream
+  ptrAddr = buff;
+  ptrBit = 0;
+  frameLeft = 0;
+  lastFrame = false;
+
+  running = true;
+
+  if (!async) {
+    // Finish saying anything in the pipe
+    while (running) {
+      loop();
+      delay(0);
+    }
+  }
+
+  return true;
+}
+
+bool AudioGeneratorTalkie::begin(AudioFileSource *source, AudioOutput *output)
+{
+  if (!output) return false;
+  this->output = output;
+  if (source) {
+    file = source;
+    if (!file->isOpen()) return false; // Error
+    auto len = file->getSize();
+    uint8_t *temp = (uint8_t *)malloc(len);
+    if (!temp) return false;
+    if (file->read(temp, len) != (uint32_t)len) return false;
+    say(temp, len);
+    free(temp);
+  } else {
+    // Reset the interpreter to the start of the stream
+    ptrAddr = buff;
+    ptrBit = 0;
+    frameLeft = 0;
+    running = false;
+  }
+  
+  if (!output->SetRate( 8000 )) return false;
+  if (!output->SetBitsPerSample( 16 )) return false;
+  if (!output->SetChannels( 2 )) return false;
+  if (!output->begin()) return false;
+
+
+
+  return true;
+}
+
+
+bool AudioGeneratorTalkie::stop()
+{
+  if (!running) return true;
+  running = false;
+  output->stop();
+  return file ? file->close() : true;
+}
+
+bool AudioGeneratorTalkie::isRunning()
+{
+  return running;
+}
+
+bool AudioGeneratorTalkie::loop()
+{
+  if (!running) goto done; // Nothing to do here!
+
+  if (!frameLeft) {
+    if (lastFrame) {
+      running = false;
+      goto done;
+    }
+    lastFrame = genOneFrame();
+  }
+
+  if (frameLeft) {
+    for ( ; frameLeft; frameLeft--) {
+      auto res = genOneSample();
+      int16_t r[2] = {res, res};
+      if (!output->ConsumeSample(r)) break;
+    }
+  }
+  
+done:
+  if (file) file->loop();
+  output->loop();
+
+  return running;
+}
+
+// The ROMs used with the TI speech were serial, not byte wide.
+// Here's a handy routine to flip ROM data which is usually reversed.
+uint8_t AudioGeneratorTalkie::rev(uint8_t a)
+{
+  // 76543210
+  a = (a>>4) | (a<<4); // Swap in groups of 4
+  // 32107654
+  a = ((a & 0xcc)>>2) | ((a & 0x33)<<2); // Swap in groups of 2
+  // 10325476
+  a = ((a & 0xaa)>>1) | ((a & 0x55)<<1); // Swap bit pairs
+  // 01234567
+  return a;
+}
+
+
+uint8_t AudioGeneratorTalkie::getBits(uint8_t bits) {
+	uint8_t value;
+	uint16_t data;
+	data = rev(ptrAddr[0])<<8;
+	if (ptrBit+bits > 8) {
+		data |= rev(ptrAddr[1]);
+	}
+	data <<= ptrBit;
+	value = data >> (16-bits);
+	ptrBit += bits;
+	if (ptrBit >= 8) {
+		ptrBit -= 8;
+		ptrAddr++;
+	}
+	return value;
+}
+
+
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wnarrowing"
+// Constant LPC coefficient tables
+static const uint8_t tmsEnergy[0x10]  = {0x00,0x02,0x03,0x04,0x05,0x07,0x0a,0x0f,0x14,0x20,0x29,0x39,0x51,0x72,0xa1,0xff};
+static const uint8_t tmsPeriod[0x40]  = {0x00,0x10,0x11,0x12,0x13,0x14,0x15,0x16,0x17,0x18,0x19,0x1A,0x1B,0x1C,0x1D,0x1E,0x1F,0x20,0x21,0x22,0x23,0x24,0x25,0x26,0x27,0x28,0x29,0x2A,0x2B,0x2D,0x2F,0x31,0x33,0x35,0x36,0x39,0x3B,0x3D,0x3F,0x42,0x45,0x47,0x49,0x4D,0x4F,0x51,0x55,0x57,0x5C,0x5F,0x63,0x66,0x6A,0x6E,0x73,0x77,0x7B,0x80,0x85,0x8A,0x8F,0x95,0x9A,0xA0};
+static const int16_t tmsK1[0x20]      = {0x82C0,0x8380,0x83C0,0x8440,0x84C0,0x8540,0x8600,0x8780,0x8880,0x8980,0x8AC0,0x8C00,0x8D40,0x8F00,0x90C0,0x92C0,0x9900,0xA140,0xAB80,0xB840,0xC740,0xD8C0,0xEBC0,0x0000,0x1440,0x2740,0x38C0,0x47C0,0x5480,0x5EC0,0x6700,0x6D40};
+static const int16_t tmsK2[0x20]      = {0xAE00,0xB480,0xBB80,0xC340,0xCB80,0xD440,0xDDC0,0xE780,0xF180,0xFBC0,0x0600,0x1040,0x1A40,0x2400,0x2D40,0x3600,0x3E40,0x45C0,0x4CC0,0x5300,0x5880,0x5DC0,0x6240,0x6640,0x69C0,0x6CC0,0x6F80,0x71C0,0x73C0,0x7580,0x7700,0x7E80};
+static const int8_t tmsK3[0x10]       = {0x92,0x9F,0xAD,0xBA,0xC8,0xD5,0xE3,0xF0,0xFE,0x0B,0x19,0x26,0x34,0x41,0x4F,0x5C};
+static const int8_t tmsK4[0x10]       = {0xAE,0xBC,0xCA,0xD8,0xE6,0xF4,0x01,0x0F,0x1D,0x2B,0x39,0x47,0x55,0x63,0x71,0x7E};
+static const int8_t tmsK5[0x10]       = {0xAE,0xBA,0xC5,0xD1,0xDD,0xE8,0xF4,0xFF,0x0B,0x17,0x22,0x2E,0x39,0x45,0x51,0x5C};
+static const int8_t tmsK6[0x10]       = {0xC0,0xCB,0xD6,0xE1,0xEC,0xF7,0x03,0x0E,0x19,0x24,0x2F,0x3A,0x45,0x50,0x5B,0x66};
+static const int8_t tmsK7[0x10]       = {0xB3,0xBF,0xCB,0xD7,0xE3,0xEF,0xFB,0x07,0x13,0x1F,0x2B,0x37,0x43,0x4F,0x5A,0x66};
+static const int8_t tmsK8[0x08]       = {0xC0,0xD8,0xF0,0x07,0x1F,0x37,0x4F,0x66};
+static const int8_t tmsK9[0x08]       = {0xC0,0xD4,0xE8,0xFC,0x10,0x25,0x39,0x4D};
+static const int8_t tmsK10[0x08]      = {0xCD,0xDF,0xF1,0x04,0x16,0x20,0x3B,0x4D};
+
+// The chirp we active the filter using
+static const int8_t chirp[] = {0x00,0x2a,0xd4,0x32,0xb2,0x12,0x25,0x14,0x02,0xe1,0xc5,0x02,0x5f,0x5a,0x05,0x0f,0x26,0xfc,0xa5,0xa5,0xd6,0xdd,0xdc,0xfc,0x25,0x2b,0x22,0x21,0x0f,0xff,0xf8,0xee,0xed,0xef,0xf7,0xf6,0xfa,0x00,0x03,0x02,0x01};
+#pragma GCC diagnostic pop
+
+
+bool AudioGeneratorTalkie::genOneFrame() {
+  uint8_t energy;
+  uint8_t repeat;
+
+  // Read speech data, processing the variable size frames.
+
+  energy = getBits(4);
+  if (energy == 0) {
+    // Energy = 0: rest frame
+    synthEnergy = 0;
+  } else if (energy == 0xf) {
+    // Energy = 15: stop frame. Silence the synthesiser.
+    synthEnergy = 0;
+    synthK1 = 0;
+    synthK2 = 0;
+    synthK3 = 0;
+    synthK4 = 0;
+    synthK5 = 0;
+    synthK6 = 0;
+    synthK7 = 0;
+    synthK8 = 0;
+    synthK9 = 0;
+    synthK10 = 0;
+  } else {
+    synthEnergy = tmsEnergy[energy];
+    repeat = getBits(1);
+    synthPeriod = tmsPeriod[getBits(6)];
+    // A repeat frame uses the last coefficients
+    if (!repeat) {
+      // All frames use the first 4 coefficients
+      synthK1 = tmsK1[getBits(5)];
+      synthK2 = tmsK2[getBits(5)];
+      synthK3 = tmsK3[getBits(4)];
+      synthK4 = tmsK4[getBits(4)];
+      if (synthPeriod) {
+        // Voiced frames use 6 extra coefficients.
+        synthK5 = tmsK5[getBits(4)];
+        synthK6 = tmsK6[getBits(4)];
+        synthK7 = tmsK7[getBits(4)];
+        synthK8 = tmsK8[getBits(3)];
+        synthK9 = tmsK9[getBits(3)];
+        synthK10 = tmsK10[getBits(3)];
+      }
+    }
+  }
+
+  frameLeft = 8000 / 40;
+  
+  return (energy == 0xf); // Last frame will return true
+}
+
+int16_t AudioGeneratorTalkie::genOneSample()
+{
+  static uint8_t periodCounter;
+  static int16_t x0,x1,x2,x3,x4,x5,x6,x7,x8,x9;
+  int16_t u0,u1,u2,u3,u4,u5,u6,u7,u8,u9,u10;
+
+  if (synthPeriod) {
+    // Voiced source
+    if (periodCounter < synthPeriod) {
+      periodCounter++;
+    } else {
+      periodCounter = 0;
+    }
+    if (periodCounter < sizeof(chirp)) {
+      u10 = ((chirp[periodCounter]) * (uint32_t) synthEnergy) >> 8;
+    } else {
+      u10 = 0;
+    }
+  } else {
+    // Unvoiced source
+    static uint16_t synthRand = 1;
+    synthRand = (synthRand >> 1) ^ ((synthRand & 1) ? 0xB800 : 0);
+    u10 = (synthRand & 1) ? synthEnergy : -synthEnergy;
+  }
+  // Lattice filter forward path
+  u9 = u10 - (((int16_t)synthK10*x9) >> 7);
+  u8 = u9 - (((int16_t)synthK9*x8) >> 7);
+  u7 = u8 - (((int16_t)synthK8*x7) >> 7);
+  u6 = u7 - (((int16_t)synthK7*x6) >> 7);
+  u5 = u6 - (((int16_t)synthK6*x5) >> 7);
+  u4 = u5 - (((int16_t)synthK5*x4) >> 7);
+  u3 = u4 - (((int16_t)synthK4*x3) >> 7);
+  u2 = u3 - (((int16_t)synthK3*x2) >> 7);
+  u1 = u2 - (((int32_t)synthK2*x1) >> 15);
+  u0 = u1 - (((int32_t)synthK1*x0) >> 15);
+
+  // Output clamp
+  if (u0 > 511) u0 = 511;
+  if (u0 < -512) u0 = -512;
+  
+  // Lattice filter reverse path
+  x9 = x8 + (((int16_t)synthK9*u8) >> 7);
+  x8 = x7 + (((int16_t)synthK8*u7) >> 7);
+  x7 = x6 + (((int16_t)synthK7*u6) >> 7);
+  x6 = x5 + (((int16_t)synthK6*u5) >> 7);
+  x5 = x4 + (((int16_t)synthK5*u4) >> 7);
+  x4 = x3 + (((int16_t)synthK4*u3) >> 7);
+  x3 = x2 + (((int16_t)synthK3*u2) >> 7);
+  x2 = x1 + (((int32_t)synthK2*u1) >> 15);
+  x1 = x0 + (((int32_t)synthK1*u0) >> 15);
+  x0 = u0;
+
+  uint16_t v = u0; // 10 bits
+  v <<= 6; // Now full 16
+  return v;
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorTalkie.h

@@ -0,0 +1,66 @@
+/*
+  AudioGeneratorTalkie
+  Audio output generator that speaks using the LPC code in old TI speech chips
+  Output is locked at 8khz as that's that the hardcoded LPC coefficients are built around
+
+  Based on the Talkie Arduino library by Peter Knight, https://github.com/going-digital/Talkie
+    
+  Copyright (C) 2020 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/>.
+*/
+
+#ifndef _AUDIOGENERATORTALKIE_H
+#define _AUDIOGENERATORTALKIE_H
+
+#include "AudioGenerator.h"
+
+class AudioGeneratorTalkie : public AudioGenerator
+{
+  public:
+    AudioGeneratorTalkie();
+    virtual ~AudioGeneratorTalkie() override;
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override;
+    bool say(const uint8_t *data, size_t len, bool async = false);
+    
+  protected:
+    // The data stream we're playing
+    uint8_t *buff;
+    
+    // Codeword stream handlers
+    uint8_t *ptrAddr;
+    uint8_t ptrBit;
+
+    bool    lastFrame;
+    bool    genOneFrame(); // Fill up one frame's worth of data, returns if this is the last frame
+    int16_t genOneSample(); // Generate one sample of a frame
+    
+    // Utilities
+    uint8_t rev(uint8_t a);
+    uint8_t getBits(uint8_t bits);
+
+    // Synthesizer state
+    uint8_t  synthPeriod;
+    uint16_t synthEnergy;
+    int16_t  synthK1, synthK2;
+    int8_t   synthK3, synthK4, synthK5, synthK6, synthK7, synthK8, synthK9, synthK10;
+    
+    int frameLeft;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorWAV.cpp

@@ -0,0 +1,316 @@
+/*
+  AudioGeneratorWAV
+  Audio output generator that reads 8 and 16-bit WAV files
+  
+  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 "AudioGeneratorWAV.h"
+
+AudioGeneratorWAV::AudioGeneratorWAV()
+{
+  running = false;
+  file = NULL;
+  output = NULL;
+  buffSize = 128;
+  buff = NULL;
+  buffPtr = 0;
+  buffLen = 0;
+}
+
+AudioGeneratorWAV::~AudioGeneratorWAV()
+{
+  free(buff);
+  buff = NULL;
+}
+
+bool AudioGeneratorWAV::stop()
+{
+  if (!running) return true;
+  running = false;
+  free(buff);
+  buff = NULL;
+  output->stop();
+  return file->close();
+}
+
+bool AudioGeneratorWAV::isRunning()
+{
+  return running;
+}
+
+
+// Handle buffered reading, reload each time we run out of data
+bool AudioGeneratorWAV::GetBufferedData(int bytes, void *dest)
+{
+  if (!running) return false; // Nothing to do here!
+  uint8_t *p = reinterpret_cast<uint8_t*>(dest);
+  while (bytes--) {
+    // Potentially load next batch of data...
+    if (buffPtr >= buffLen) {
+      buffPtr = 0;
+      uint32_t toRead = availBytes > buffSize ? buffSize : availBytes;
+      buffLen = file->read( buff, toRead );
+      availBytes -= buffLen;
+    }
+    if (buffPtr >= buffLen)
+      return false; // No data left!
+    *(p++) = buff[buffPtr++];
+  }
+  return true;
+}
+
+bool AudioGeneratorWAV::loop()
+{
+  if (!running) goto done; // Nothing to do here!
+
+  // First, try and push in the stored sample.  If we can't, then punt and try later
+  if (!output->ConsumeSample(lastSample)) goto done; // Can't send, but no error detected
+
+  // Try and stuff the buffer one sample at a time
+  do
+  {
+    if (bitsPerSample == 8) {
+      uint8_t l, r;
+      if (!GetBufferedData(1, &l)) stop();
+      if (channels == 2) {
+        if (!GetBufferedData(1, &r)) stop();
+      } else {
+        r = 0;
+      }
+      lastSample[AudioOutput::LEFTCHANNEL] = l;
+      lastSample[AudioOutput::RIGHTCHANNEL] = r;
+    } else if (bitsPerSample == 16) {
+      if (!GetBufferedData(2, &lastSample[AudioOutput::LEFTCHANNEL])) stop();
+      if (channels == 2) {
+        if (!GetBufferedData(2, &lastSample[AudioOutput::RIGHTCHANNEL])) stop();
+      } else {
+        lastSample[AudioOutput::RIGHTCHANNEL] = 0;
+      }
+    }
+  } while (running && output->ConsumeSample(lastSample));
+
+done:
+  file->loop();
+  output->loop();
+
+  return running;
+}
+
+
+bool AudioGeneratorWAV::ReadWAVInfo()
+{
+  uint32_t u32;
+  uint16_t u16;
+  int toSkip;
+
+  // WAV specification document:
+  // https://www.aelius.com/njh/wavemetatools/doc/riffmci.pdf
+
+  // Header == "RIFF"
+  if (!ReadU32(&u32)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+  if (u32 != 0x46464952) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: cannot read WAV, invalid RIFF header, got: %08X \n"), (uint32_t) u32);
+    return false;
+  }
+
+  // Skip ChunkSize
+  if (!ReadU32(&u32)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+
+  // Format == "WAVE"
+  if (!ReadU32(&u32)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+  if (u32 != 0x45564157) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: cannot read WAV, invalid WAVE header, got: %08X \n"), (uint32_t) u32);
+    return false;
+  }
+
+  // there might be JUNK or PAD - ignore it by continuing reading until we get to "fmt "
+  while (1) {
+    if (!ReadU32(&u32)) {
+      Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+      return false;
+    };
+    if (u32 == 0x20746d66) break; // 'fmt '
+  };
+
+  // subchunk size
+  if (!ReadU32(&u32)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+  if (u32 == 16) { toSkip = 0; }
+  else if (u32 == 18) { toSkip = 18 - 16; }
+  else if (u32 == 40) { toSkip = 40 - 16; }
+  else {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: cannot read WAV, appears not to be standard PCM \n"));
+    return false;
+  } // we only do standard PCM
+
+  // AudioFormat
+  if (!ReadU16(&u16)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+  if (u16 != 1) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: cannot read WAV, AudioFormat appears not to be standard PCM \n"));
+    return false;
+  } // we only do standard PCM
+
+  // NumChannels
+  if (!ReadU16(&channels)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+  if ((channels<1) || (channels>2)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: cannot read WAV, only mono and stereo are supported \n"));
+    return false;
+  } // Mono or stereo support only
+
+  // SampleRate
+  if (!ReadU32(&sampleRate)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+  if (sampleRate < 1) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: cannot read WAV, unknown sample rate \n"));
+    return false;
+  }  // Weird rate, punt.  Will need to check w/DAC to see if supported
+
+  // Ignore byterate and blockalign
+  if (!ReadU32(&u32)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+  if (!ReadU16(&u16)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+
+  // Bits per sample
+  if (!ReadU16(&bitsPerSample)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+  if ((bitsPerSample!=8) && (bitsPerSample != 16)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: cannot read WAV, only 8 or 16 bits is supported \n"));
+    return false;
+  }  // Only 8 or 16 bits
+
+  // Skip any extra header
+  while (toSkip) {
+    uint8_t ign;
+    if (!ReadU8(&ign)) {
+      Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+      return false;
+    };
+    toSkip--;
+  }
+
+  // look for data subchunk
+  do {
+    // id == "data"
+    if (!ReadU32(&u32)) {
+      Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+      return false;
+    };
+    if (u32 == 0x61746164) break; // "data"
+    // Skip size, read until end of chunk
+    if (!ReadU32(&u32)) {
+      Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+      return false;
+    };
+    if(!file->seek(u32, SEEK_CUR)) {
+      Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data, seek failed\n"));
+      return false;
+    }
+  } while (1);
+  if (!file->isOpen()) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: cannot read WAV, file is not open\n"));
+    return false;
+  };
+
+  // Skip size, read until end of file...
+  if (!ReadU32(&u32)) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: failed to read WAV data\n"));
+    return false;
+  };
+  availBytes = u32;
+
+  // Now set up the buffer or fail
+  buff = reinterpret_cast<uint8_t *>(malloc(buffSize));
+  if (!buff) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::ReadWAVInfo: cannot read WAV, failed to set up buffer \n"));
+    return false;
+  };
+  buffPtr = 0;
+  buffLen = 0;
+
+  return true;
+}
+
+bool AudioGeneratorWAV::begin(AudioFileSource *source, AudioOutput *output)
+{
+  if (!source) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::begin: failed: invalid source\n"));
+    return false;
+  }
+  file = source;
+  if (!output) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::begin: invalid output\n"));
+    return false;
+  }
+  this->output = output;
+  if (!file->isOpen()) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::begin: file not open\n"));
+    return false;
+  } // Error
+
+  if (!ReadWAVInfo()) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::begin: failed during ReadWAVInfo\n"));
+    return false;
+  }
+
+  if (!output->SetRate( sampleRate )) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::begin: failed to SetRate in output\n"));
+    return false;
+  }
+  if (!output->SetBitsPerSample( bitsPerSample )) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::begin: failed to SetBitsPerSample in output\n"));
+    return false;
+  }
+  if (!output->SetChannels( channels )) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::begin: failed to SetChannels in output\n"));
+    return false;
+  }
+  if (!output->begin()) {
+    Serial.printf_P(PSTR("AudioGeneratorWAV::begin: output's begin did not return true\n"));
+    return false;
+  }
+
+  running = true;
+
+  return true;
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioGeneratorWAV.h

@@ -0,0 +1,61 @@
+/*
+  AudioGeneratorWAV
+  Audio output generator that reads 8 and 16-bit WAV files
+    
+  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/>.
+*/
+
+#ifndef _AUDIOGENERATORWAV_H
+#define _AUDIOGENERATORWAV_H
+
+#include "AudioGenerator.h"
+
+class AudioGeneratorWAV : public AudioGenerator
+{
+  public:
+    AudioGeneratorWAV();
+    virtual ~AudioGeneratorWAV() override;
+    virtual bool begin(AudioFileSource *source, AudioOutput *output) override;
+    virtual bool loop() override;
+    virtual bool stop() override;
+    virtual bool isRunning() override;
+    void SetBufferSize(int sz) { buffSize = sz; }
+
+  private:
+    bool ReadU32(uint32_t *dest) { return file->read(reinterpret_cast<uint8_t*>(dest), 4); }
+    bool ReadU16(uint16_t *dest) { return file->read(reinterpret_cast<uint8_t*>(dest), 2); }
+    bool ReadU8(uint8_t *dest) { return file->read(reinterpret_cast<uint8_t*>(dest), 1); }
+    bool GetBufferedData(int bytes, void *dest);
+    bool ReadWAVInfo();
+
+    
+  protected:
+    // WAV info
+    uint16_t channels;
+    uint32_t sampleRate;
+    uint16_t bitsPerSample;
+    
+    uint32_t availBytes;
+
+    // We need to buffer some data in-RAM to avoid doing 1000s of small reads
+    uint32_t buffSize;
+    uint8_t *buff;
+    uint16_t buffPtr;
+    uint16_t buffLen;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioLogger.cpp

@@ -0,0 +1,5 @@
+
+#include "AudioLogger.h"
+
+DevNullOut silencedLogger;
+Print* audioLogger = &silencedLogger;

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioLogger.h

@@ -0,0 +1,19 @@
+
+#include <Arduino.h>
+
+#ifndef _AUDIOLOGGER_H
+#define _AUDIOLOGGER_H
+
+class DevNullOut: public Print
+{
+public:
+    virtual size_t write(uint8_t) { return 1; }
+};
+
+extern DevNullOut silencedLogger;
+
+// Global `audioLogger` is initialized to &silencedLogger
+// It can be initialized anytime to &Serial or any other Print:: derivative instance.
+extern Print* audioLogger;
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutput.h

@@ -0,0 +1,85 @@
+/*
+  AudioOutput
+  Base class of an audio output player
+  
+  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/>.
+*/
+
+#ifndef _AUDIOOUTPUT_H
+#define _AUDIOOUTPUT_H
+
+#include <Arduino.h>
+#include "AudioStatus.h"
+
+class AudioOutput
+{
+  public:
+    AudioOutput() { };
+    virtual ~AudioOutput() {};
+    virtual bool SetRate(int hz) { hertz = hz; return true; }
+    virtual bool SetBitsPerSample(int bits) { bps = bits; return true; }
+    virtual bool SetChannels(int chan) { channels = chan; return true; }
+    virtual bool SetGain(float f) { if (f>4.0) f = 4.0; if (f<0.0) f=0.0; gainF2P6 = (uint8_t)(f*(1<<6)); return true; }
+    virtual bool begin() { return false; };
+    typedef enum { LEFTCHANNEL=0, RIGHTCHANNEL=1 } SampleIndex;
+    virtual bool ConsumeSample(int16_t sample[2]) { (void)sample; return false; }
+    virtual uint16_t ConsumeSamples(int16_t *samples, uint16_t count)
+    {
+      for (uint16_t i=0; i<count; i++) {
+        if (!ConsumeSample(samples)) return i;
+        samples += 2;
+      }
+      return count;
+    }
+    virtual bool stop() { return false; }
+    virtual void flush() { return; }
+    virtual bool loop() { return true; }
+
+  public:
+    virtual bool RegisterMetadataCB(AudioStatus::metadataCBFn fn, void *data) { return cb.RegisterMetadataCB(fn, data); }
+    virtual bool RegisterStatusCB(AudioStatus::statusCBFn fn, void *data) { return cb.RegisterStatusCB(fn, data); }
+
+  protected:
+    void MakeSampleStereo16(int16_t sample[2]) {
+      // Mono to "stereo" conversion
+      if (channels == 1)
+        sample[RIGHTCHANNEL] = sample[LEFTCHANNEL];
+      if (bps == 8) {
+        // Upsample from unsigned 8 bits to signed 16 bits
+        sample[LEFTCHANNEL] = (((int16_t)(sample[LEFTCHANNEL]&0xff)) - 128) << 8;
+        sample[RIGHTCHANNEL] = (((int16_t)(sample[RIGHTCHANNEL]&0xff)) - 128) << 8;
+      }
+    };
+
+    inline int16_t Amplify(int16_t s) {
+      int32_t v = (s * gainF2P6)>>6;
+      if (v < -32767) return -32767;
+      else if (v > 32767) return 32767;
+      else return (int16_t)(v&0xffff);
+    }
+
+  protected:
+    uint16_t hertz;
+    uint8_t bps;
+    uint8_t channels;
+    uint8_t gainF2P6; // Fixed point 2.6
+
+  protected:
+    AudioStatus cb;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputBuffer.cpp

@@ -0,0 +1,89 @@
+/*
+  AudioOutputBuffer
+  Adds additional bufferspace to the output chain
+  
+  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>
+#include "AudioOutputBuffer.h"
+
+AudioOutputBuffer::AudioOutputBuffer(int buffSizeSamples, AudioOutput *dest)
+{
+  buffSize = buffSizeSamples;
+  leftSample = (int16_t*)malloc(sizeof(int16_t) * buffSize);
+  rightSample = (int16_t*)malloc(sizeof(int16_t) * buffSize);
+  writePtr = 0;
+  readPtr = 0;
+  sink = dest;
+}
+
+AudioOutputBuffer::~AudioOutputBuffer()
+{
+  free(leftSample);
+  free(rightSample);
+}
+
+bool AudioOutputBuffer::SetRate(int hz)
+{
+  return sink->SetRate(hz);
+}
+
+bool AudioOutputBuffer::SetBitsPerSample(int bits)
+{
+  return sink->SetBitsPerSample(bits);
+}
+
+bool AudioOutputBuffer::SetChannels(int channels)
+{
+  return sink->SetChannels(channels);
+}
+
+bool AudioOutputBuffer::begin()
+{
+  filled = false;
+  return sink->begin();
+}
+
+bool AudioOutputBuffer::ConsumeSample(int16_t sample[2])
+{
+  // First, try and fill I2S...
+  if (filled) {
+    while (readPtr != writePtr) {
+      int16_t s[2] = {leftSample[readPtr], rightSample[readPtr]};
+      if (!sink->ConsumeSample(s)) break; // Can't stuff any more in I2S...
+      readPtr = (readPtr + 1) % buffSize;
+    }
+  }
+
+  // Now, do we have space for a new sample?
+  int nextWritePtr = (writePtr + 1) % buffSize;
+  if (nextWritePtr == readPtr) {
+    filled = true;
+    return false;
+  }
+  leftSample[writePtr] = sample[LEFTCHANNEL];
+  rightSample[writePtr] = sample[RIGHTCHANNEL];
+  writePtr = nextWritePtr;
+  return true;
+}
+
+bool AudioOutputBuffer::stop()
+{
+  return sink->stop();
+}
+
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputBuffer.h

@@ -0,0 +1,49 @@
+/*
+  AudioOutputBuffer
+  Adds additional bufferspace to the output chain
+  
+  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/>.
+*/
+
+#ifndef _AUDIOOUTPUTBUFFER_H
+#define _AUDIOOUTPUTBUFFER_H
+
+#include "AudioOutput.h"
+
+class AudioOutputBuffer : public AudioOutput
+{
+  public:
+    AudioOutputBuffer(int bufferSizeSamples, AudioOutput *dest);
+    virtual ~AudioOutputBuffer() override;
+    virtual bool SetRate(int hz) override;
+    virtual bool SetBitsPerSample(int bits) override;
+    virtual bool SetChannels(int channels) override;
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+    
+  protected:
+    AudioOutput *sink;
+    int buffSize;
+    int16_t *leftSample;
+    int16_t *rightSample;
+    int writePtr;
+    int readPtr;
+    bool filled;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputFilterBiquad.cpp

@@ -0,0 +1,245 @@
+/*
+  AudioOutputFilterBiquad
+  Implements a Biquad filter
+
+  Copyright (C) 2012  Nigel Redmon
+  Copyright (C) 2021  William Bérubé
+
+  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>
+#include "AudioOutputFilterBiquad.h"
+
+AudioOutputFilterBiquad::AudioOutputFilterBiquad(AudioOutput *sink)
+{
+  this->sink = sink;
+  
+  type = bq_type_lowpass;
+  a0 = 1.0;
+  a1 = a2 = b1 = b2 = 0.0;
+  Fc = 0.50;
+  Q = 0.707;
+  peakGain = 0.0;
+  z1 = z2 = 0.0;
+}
+
+AudioOutputFilterBiquad::AudioOutputFilterBiquad(int type, float Fc, float Q, float peakGain, AudioOutput *sink)
+{
+  this->sink = sink;
+  
+  SetBiquad(type, Fc, Q, peakGain);
+  z1 = z2 = 0.0;
+}
+
+AudioOutputFilterBiquad::~AudioOutputFilterBiquad() {}
+
+bool AudioOutputFilterBiquad::SetRate(int hz)
+{
+  return sink->SetRate(hz);
+}
+
+bool AudioOutputFilterBiquad::SetBitsPerSample(int bits)
+{
+  return sink->SetBitsPerSample(bits);
+}
+
+bool AudioOutputFilterBiquad::SetChannels(int channels)
+{
+  return sink->SetChannels(channels);
+}
+
+bool AudioOutputFilterBiquad::SetGain(float gain)
+{
+  return sink->SetGain(gain);
+}
+
+void AudioOutputFilterBiquad::SetType(int type)
+{
+  this->type = type;
+  CalcBiquad();
+}
+
+void AudioOutputFilterBiquad::SetFc(float Fc)
+{
+  this->Fc = Fc;
+  CalcBiquad();
+}
+
+void AudioOutputFilterBiquad::SetQ(float Q)
+{
+  this->Q = Q;
+  CalcBiquad();
+}
+
+void AudioOutputFilterBiquad::SetPeakGain(float peakGain)
+{
+  this->peakGain = peakGain;
+  CalcBiquad();
+}
+
+void AudioOutputFilterBiquad::SetBiquad(int type, float Fc, float Q, float peakGain)
+{
+  this->type = type;
+  this->Fc = Fc;
+  this->Q = Q;
+  this->peakGain = peakGain;
+  CalcBiquad();
+}
+
+void AudioOutputFilterBiquad::CalcBiquad()
+{
+  float norm;
+  float V = pow(10, fabs(peakGain) / 20.0);
+  float K = tan(M_PI * Fc);
+
+  switch (this->type) {
+    case bq_type_lowpass:
+      norm = 1 / (1 + K / Q + K * K);
+      a0 = K * K * norm;
+      a1 = 2 * a0;
+      a2 = a0;
+      b1 = 2 * (K * K - 1) * norm;
+      b2 = (1 - K / Q + K * K) * norm;
+      break;
+      
+    case bq_type_highpass:
+      norm = 1 / (1 + K / Q + K * K);
+      a0 = 1 * norm;
+      a1 = -2 * a0;
+      a2 = a0;
+      b1 = 2 * (K * K - 1) * norm;
+      b2 = (1 - K / Q + K * K) * norm;
+      break;
+      
+    case bq_type_bandpass:
+      norm = 1 / (1 + K / Q + K * K);
+      a0 = K / Q * norm;
+      a1 = 0;
+      a2 = -a0;
+      b1 = 2 * (K * K - 1) * norm;
+      b2 = (1 - K / Q + K * K) * norm;
+      break;
+      
+    case bq_type_notch:
+      norm = 1 / (1 + K / Q + K * K);
+      a0 = (1 + K * K) * norm;
+      a1 = 2 * (K * K - 1) * norm;
+      a2 = a0;
+      b1 = a1;
+      b2 = (1 - K / Q + K * K) * norm;
+      break;
+      
+    case bq_type_peak:
+      if (peakGain >= 0) {    // boost
+        norm = 1 / (1 + 1/Q * K + K * K);
+        a0 = (1 + V/Q * K + K * K) * norm;
+        a1 = 2 * (K * K - 1) * norm;
+        a2 = (1 - V/Q * K + K * K) * norm;
+        b1 = a1;
+        b2 = (1 - 1/Q * K + K * K) * norm;
+      } else {    // cut
+        norm = 1 / (1 + V/Q * K + K * K);
+        a0 = (1 + 1/Q * K + K * K) * norm;
+        a1 = 2 * (K * K - 1) * norm;
+        a2 = (1 - 1/Q * K + K * K) * norm;
+        b1 = a1;
+        b2 = (1 - V/Q * K + K * K) * norm;
+      }
+      break;
+
+    case bq_type_lowshelf:
+      if (peakGain >= 0) {    // boost
+        norm = 1 / (1 + sqrt(2) * K + K * K);
+        a0 = (1 + sqrt(2*V) * K + V * K * K) * norm;
+        a1 = 2 * (V * K * K - 1) * norm;
+        a2 = (1 - sqrt(2*V) * K + V * K * K) * norm;
+        b1 = 2 * (K * K - 1) * norm;
+        b2 = (1 - sqrt(2) * K + K * K) * norm;
+      }
+      else {    // cut
+        norm = 1 / (1 + sqrt(2*V) * K + V * K * K);
+        a0 = (1 + sqrt(2) * K + K * K) * norm;
+        a1 = 2 * (K * K - 1) * norm;
+        a2 = (1 - sqrt(2) * K + K * K) * norm;
+        b1 = 2 * (V * K * K - 1) * norm;
+        b2 = (1 - sqrt(2*V) * K + V * K * K) * norm;
+      }
+      break;
+
+    case bq_type_highshelf:
+      if (peakGain >= 0) {    // boost
+        norm = 1 / (1 + sqrt(2) * K + K * K);
+        a0 = (V + sqrt(2*V) * K + K * K) * norm;
+        a1 = 2 * (K * K - V) * norm;
+        a2 = (V - sqrt(2*V) * K + K * K) * norm;
+        b1 = 2 * (K * K - 1) * norm;
+        b2 = (1 - sqrt(2) * K + K * K) * norm;
+      }
+      else {    // cut
+        norm = 1 / (V + sqrt(2*V) * K + K * K);
+        a0 = (1 + sqrt(2) * K + K * K) * norm;
+        a1 = 2 * (K * K - 1) * norm;
+        a2 = (1 - sqrt(2) * K + K * K) * norm;
+        b1 = 2 * (K * K - V) * norm;
+        b2 = (V - sqrt(2*V) * K + K * K) * norm;
+      }
+      break;
+  }
+
+    i_a0 = a0 * BQ_DECAL;
+    i_a1 = a1 * BQ_DECAL;
+    i_a2 = a2 * BQ_DECAL;
+
+    i_b1 = b1 * BQ_DECAL;
+    i_b2 = b2 * BQ_DECAL;
+
+    i_lz1 = i_rz1 = z1 * BQ_DECAL;
+    i_lz2 = i_rz2 = z2 * BQ_DECAL;
+
+    i_Fc = Fc * BQ_DECAL;
+    i_Q = Q * BQ_DECAL;
+    i_peakGain = peakGain * BQ_DECAL;
+}
+
+bool AudioOutputFilterBiquad::begin()
+{
+  return sink->begin();
+}
+
+bool AudioOutputFilterBiquad::ConsumeSample(int16_t sample[2])
+{
+
+  int32_t leftSample = (sample[LEFTCHANNEL] << BQ_SHIFT) / 2;
+  int32_t rightSample = (sample[RIGHTCHANNEL] << BQ_SHIFT) / 2;
+
+  int64_t leftOutput = ((leftSample * i_a0) >> BQ_SHIFT) + i_lz1;
+  i_lz1 = ((leftSample * i_a1) >> BQ_SHIFT) + i_lz2 - ((i_b1 * leftOutput) >> BQ_SHIFT);
+  i_lz2 = ((leftSample * i_a2) >> BQ_SHIFT) - ((i_b2 * leftOutput) >> BQ_SHIFT);
+
+  int64_t rightOutput = ((rightSample * i_a0) >> BQ_SHIFT) + i_rz1;
+  i_rz1 = ((rightSample * i_a1) >> BQ_SHIFT) + i_rz2 - ((i_b1 * rightOutput) >> BQ_SHIFT);
+  i_rz2 = ((rightSample * i_a2) >> BQ_SHIFT) - ((i_b2 * rightOutput) >> BQ_SHIFT);
+  
+  int16_t out[2];
+  out[LEFTCHANNEL] = (int16_t)(leftOutput >> BQ_SHIFT);
+  out[RIGHTCHANNEL] = (int16_t)(rightOutput >> BQ_SHIFT);
+
+  return sink->ConsumeSample(out);
+}
+
+bool AudioOutputFilterBiquad::stop()
+{
+  return sink->stop();
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputFilterBiquad.h

@@ -0,0 +1,80 @@
+/*
+  AudioOutputFilterBiquad
+  Implements a Biquad filter
+  
+  Copyright (C) 2012  Nigel Redmon
+  Copyright (C) 2021  William Bérubé
+
+  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/>.
+*/
+
+#ifndef _AudioOutputFilterBiquad_H
+#define _AudioOutputFilterBiquad_H
+
+#include "AudioOutput.h"
+
+#define      BQ_SHIFT 16
+#define      BQ_DECAL 65536
+
+enum {
+    bq_type_lowpass = 0,
+    bq_type_highpass,
+    bq_type_bandpass,
+    bq_type_notch,
+    bq_type_peak,
+    bq_type_lowshelf,
+    bq_type_highshelf
+};
+
+class AudioOutputFilterBiquad : public AudioOutput
+{
+  public:
+    AudioOutputFilterBiquad(AudioOutput *sink);
+    AudioOutputFilterBiquad(int type, float Fc, float Q, float peakGain, AudioOutput *sink);
+    virtual ~AudioOutputFilterBiquad() override;
+    virtual bool SetRate(int hz) override;
+    virtual bool SetBitsPerSample(int bits) override;
+    virtual bool SetChannels(int chan) override;
+    virtual bool SetGain(float f) override;
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+
+  private:
+    void SetType(int type);
+    void SetFc(float Fc);
+    void SetQ(float Q);
+    void SetPeakGain(float peakGain);
+    void SetBiquad(int type, float Fc, float Q, float peakGain);
+
+  protected:
+    AudioOutput *sink;
+    int buffSize;
+    int16_t *leftSample;
+    int16_t *rightSample;
+    int writePtr;
+    int readPtr;
+    bool filled;
+    int type;
+    void CalcBiquad();
+    int64_t i_a0, i_a1, i_a2, i_b1, i_b2;
+    int64_t i_Fc, i_Q, i_peakGain;
+    int64_t i_lz1, i_lz2, i_rz1, i_rz2;
+    float a0, a1, a2, b1, b2;
+    float Fc, Q, peakGain;
+    float z1, z2;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputFilterDecimate.cpp

@@ -0,0 +1,112 @@
+/*
+  AudioOutputFilter
+  Implements a user-defined FIR on a passthrough
+
+  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>
+#include "AudioOutputFilterDecimate.h"
+
+AudioOutputFilterDecimate::AudioOutputFilterDecimate(uint8_t taps, const int16_t *tap, int num, int den, AudioOutput *sink)
+{
+  this->sink = sink;
+
+  // The filter state.  Passed in TAPS must be available throughout object lifetime
+  this->taps = taps;
+  this->tap = (int16_t*)malloc(sizeof(int16_t) * taps);
+  memcpy_P(this->tap, tap, sizeof(int16_t) * taps);
+  this->hist[0] = (int16_t*)malloc(sizeof(int16_t) * taps);
+  memset(this->hist[0], 0, sizeof(int16_t) * taps);
+  this->hist[1] = (int16_t*)malloc(sizeof(int16_t) * taps);
+  memset(this->hist[1], 0, sizeof(int16_t) * taps);
+  this->idx = 0;
+
+  // Decimator numerator and denominator with an error signal.  Not great, but fast and simple
+  this->num = num;
+  this->den = den;
+  this->err = 0;
+}
+
+AudioOutputFilterDecimate::~AudioOutputFilterDecimate()
+{
+  free(hist[1]);
+  free(hist[0]);
+  free(tap);
+}
+
+bool AudioOutputFilterDecimate::SetRate(int hz)
+{
+  // Modify input frequency to account for decimation
+  hz *= den;
+  hz /= num;
+  return sink->SetRate(hz);
+}
+
+bool AudioOutputFilterDecimate::SetBitsPerSample(int bits)
+{
+  return sink->SetBitsPerSample(bits);
+}
+
+bool AudioOutputFilterDecimate::SetChannels(int channels)
+{
+  return sink->SetChannels(channels);
+}
+
+bool AudioOutputFilterDecimate::SetGain(float gain)
+{
+  return sink->SetGain(gain);
+}
+
+bool AudioOutputFilterDecimate::begin()
+{
+  return sink->begin();
+}
+
+bool AudioOutputFilterDecimate::ConsumeSample(int16_t sample[2])
+{
+  // Store the data samples in history always
+  hist[LEFTCHANNEL][idx] = sample[LEFTCHANNEL];
+  hist[RIGHTCHANNEL][idx] = sample[RIGHTCHANNEL];
+  idx++;
+  if (idx == taps) idx = 0;
+
+  // Only output if the error signal says we're ready to decimate.  This simplistic way might give some aliasing noise
+  err += num;
+  if (err >= den) {
+    err -= den;
+    // Need to output a sample, so actually calculate the filter at this point in time
+    // Smarter might actually shift the history by the fractional remainder or take two filters and interpolate
+    int32_t accL = 0;
+    int32_t accR = 0;
+    int index = idx;
+    for (size_t i=0; i < taps; i++) {
+      index = index != 0 ? index-1 : taps-1;
+      accL += (int32_t)hist[LEFTCHANNEL][index] * tap[i];
+      accR += (int32_t)hist[RIGHTCHANNEL][index] * tap[i];
+    };
+    int16_t out[2];
+    out[LEFTCHANNEL] = accL >> 16;
+    out[RIGHTCHANNEL] = accR >> 16;
+    return sink->ConsumeSample(out);
+  }
+  return true; // Nothing to do here...
+}
+
+bool AudioOutputFilterDecimate::stop()
+{
+  return sink->stop();
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputFilterDecimate.h

@@ -0,0 +1,51 @@
+/*
+  AudioOutputFilterDecimate
+  Implements a user-defined FIR on a passthrough w/rational decimation
+  
+  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/>.
+*/
+
+#ifndef _AUDIOOUTPUTFILTERDECIMATE_H
+#define _AUDIOOUTPUTFILTERDECIMATE_H
+
+#include "AudioOutput.h"
+
+class AudioOutputFilterDecimate : public AudioOutput
+{
+  public:
+    AudioOutputFilterDecimate(uint8_t taps, const int16_t *tap, int num, int den, AudioOutput *sink);
+    virtual ~AudioOutputFilterDecimate() override;
+    virtual bool SetRate(int hz) override;
+    virtual bool SetBitsPerSample(int bits) override;
+    virtual bool SetChannels(int chan) override;
+    virtual bool SetGain(float f) override;
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+
+  protected:
+    AudioOutput *sink;
+    uint8_t taps;
+    int16_t *tap;
+    int16_t *hist[2];
+    int idx;
+    int num;
+    int den;
+    int err;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputI2S.cpp

@@ -0,0 +1,313 @@
+/*
+  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"
+#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)
+      {
+        mode = (i2s_mode_t)(mode | I2S_MODE_DAC_BUILT_IN);
+      }
+      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_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_MSB);
+      if (output_mode == INTERNAL_DAC)
+      {
+        comm_fmt = (i2s_comm_format_t)I2S_COMM_FORMAT_I2S_MSB;
+      }
+
+      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)
+      {
+        i2s_set_pin((i2s_port_t)portNo, NULL);
+        i2s_set_dac_mode(I2S_DAC_CHANNEL_BOTH_EN);
+      }
+      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);
+    }
+    return i2s_write_bytes((i2s_port_t)portNo, (const char *)&s32, sizeof(uint32_t), 0);
+  #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;
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputI2S.h

@@ -0,0 +1,69 @@
+/*
+  AudioOutputI2S
+  Base class for an I2S output 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/>.
+*/
+
+#pragma once
+
+#include "AudioOutput.h"
+
+#if defined(__SAMD51__)
+  #include "I2C.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
+    virtual ~AudioOutputI2S() override;
+    virtual bool SetRate(int hz) override;
+    virtual bool SetBitsPerSample(int bits) override;
+    virtual bool SetChannels(int channels) override;
+    virtual bool begin() override { return begin(true); }
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual void flush() override;
+    virtual bool stop() override;
+    
+    bool begin(bool txDAC);
+    bool SetOutputModeMono(bool mono);  // Force mono output no matter the input
+
+  protected:
+    bool SetPinout();
+    virtual int AdjustI2SRate(int hz) { return hz; }
+    uint8_t portNo;
+    int output_mode;
+    bool mono;
+    bool i2sOn;
+    int dma_buf_count;
+    int use_apll;
+    // We can restore the old values and free up these pins when in NoDAC mode
+    uint32_t orig_bck;
+    uint32_t orig_ws;
+    
+    uint8_t bclkPin;
+    uint8_t wclkPin;
+    uint8_t doutPin;
+};

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputI2SNoDAC.cpp

@@ -0,0 +1,113 @@
+/*
+  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
+  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"
+#elif defined(ARDUINO_ARCH_RP2040) || defined(ESP8266)
+  #include <I2S.h>
+#endif
+#include "AudioOutputI2SNoDAC.h"
+
+
+AudioOutputI2SNoDAC::AudioOutputI2SNoDAC(int port) : AudioOutputI2S(port, false)
+{
+  SetOversampling(32);
+  lastSamp = 0;
+  cumErr = 0;
+#ifdef ESP8266
+  WRITE_PERI_REG(PERIPHS_IO_MUX_MTDO_U, orig_bck);
+  WRITE_PERI_REG(PERIPHS_IO_MUX_GPIO2_U, orig_ws);
+#endif
+}
+
+AudioOutputI2SNoDAC::~AudioOutputI2SNoDAC()
+{
+}
+
+bool AudioOutputI2SNoDAC::SetOversampling(int os) {
+  if (os % 32) return false;  // Only Nx32 oversampling supported
+  if (os > 256) return false; // Don't be silly now!
+  if (os < 32) return false;  // Nothing under 32 allowed
+
+  oversample = os;
+  return SetRate(hertz);
+}
+
+void AudioOutputI2SNoDAC::DeltaSigma(int16_t sample[2], uint32_t dsBuff[8])
+{
+  // Not shift 8 because addition takes care of one mult x 2
+  int32_t sum = (((int32_t)sample[0]) + ((int32_t)sample[1])) >> 1;
+  fixed24p8_t newSamp = ( (int32_t)Amplify(sum) ) << 8;
+
+  int oversample32 = oversample / 32;
+  // How much the comparison signal changes each oversample step
+  fixed24p8_t diffPerStep = (newSamp - lastSamp) >> (4 + oversample32);
+
+  // Don't need lastSamp anymore, store this one for next round
+  lastSamp = newSamp;
+
+  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) {
+        bits |= 1;
+        cumErr += fixedPosValue - newSamp;
+      } else {
+        // Bits[0] = 0 handled already by left shift
+        cumErr -= fixedPosValue + newSamp;
+      }
+      newSamp += diffPerStep; // Move the reference signal towards destination
+    }
+    dsBuff[j] = bits;
+  }
+}
+
+bool AudioOutputI2SNoDAC::ConsumeSample(int16_t sample[2])
+{
+  int16_t ms[2];
+  ms[0] = sample[0];
+  ms[1] = sample[1];
+  MakeSampleStereo16( ms );
+
+  // Make delta-sigma filled buffer
+  uint32_t dsBuff[8];
+  DeltaSigma(ms, dsBuff);
+
+  // Either send complete pulse stream or nothing
+#ifdef ESP32
+  if (!i2s_write_bytes((i2s_port_t)portNo, (const char *)dsBuff, sizeof(uint32_t) * (oversample/32), 0))
+    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
+  // remaining ones even if they block.
+  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++));
+  }
+#endif
+  return true;
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputI2SNoDAC.h

@@ -0,0 +1,43 @@
+/*
+  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
+  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/>.
+*/
+
+#pragma once
+
+#include "AudioOutputI2S.h"
+
+class AudioOutputI2SNoDAC : public AudioOutputI2S
+{
+  public:
+    AudioOutputI2SNoDAC(int port = 0);
+    virtual ~AudioOutputI2SNoDAC() override;
+    virtual bool begin() override { return AudioOutputI2S::begin(false); }
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    
+    bool SetOversampling(int os);
+    
+  protected:
+    virtual int AdjustI2SRate(int hz) override { return hz * oversample/32; }
+    uint8_t oversample;
+    void DeltaSigma(int16_t sample[2], uint32_t dsBuff[4]);
+    typedef int32_t fixed24p8_t;
+    enum {fixedPosValue=0x007fff00}; /* 24.8 of max-signed-int */
+    fixed24p8_t lastSamp; // Last sample value
+    fixed24p8_t cumErr;   // Running cumulative error since time began
+};

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputI2S_samd51.cpp

@@ -0,0 +1,313 @@
+/*
+  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"
+#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)
+      {
+        mode = (i2s_mode_t)(mode | I2S_MODE_DAC_BUILT_IN);
+      }
+      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_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_MSB);
+      if (output_mode == INTERNAL_DAC)
+      {
+        comm_fmt = (i2s_comm_format_t)I2S_COMM_FORMAT_I2S_MSB;
+      }
+
+      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)
+      {
+        i2s_set_pin((i2s_port_t)portNo, NULL);
+        i2s_set_dac_mode(I2S_DAC_CHANNEL_BOTH_EN);
+      }
+      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);
+    }
+    return i2s_write_bytes((i2s_port_t)portNo, (const char *)&s32, sizeof(uint32_t), 0);
+  #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;
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputI2S_samd51.h

@@ -0,0 +1,64 @@
+/*
+  AudioOutputI2S
+  Base class for an I2S output 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/>.
+*/
+
+#pragma once
+
+#include "AudioOutput.h"
+
+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
+    virtual ~AudioOutputI2S() override;
+    virtual bool SetRate(int hz) override;
+    virtual bool SetBitsPerSample(int bits) override;
+    virtual bool SetChannels(int channels) override;
+    virtual bool begin() override { return begin(true); }
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual void flush() override;
+    virtual bool stop() override;
+    
+    bool begin(bool txDAC);
+    bool SetOutputModeMono(bool mono);  // Force mono output no matter the input
+
+  protected:
+    bool SetPinout();
+    virtual int AdjustI2SRate(int hz) { return hz; }
+    uint8_t portNo;
+    int output_mode;
+    bool mono;
+    bool i2sOn;
+    int dma_buf_count;
+    int use_apll;
+    // We can restore the old values and free up these pins when in NoDAC mode
+    uint32_t orig_bck;
+    uint32_t orig_ws;
+    
+    uint8_t bclkPin;
+    uint8_t wclkPin;
+    uint8_t doutPin;
+};

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputMixer.cpp

@@ -0,0 +1,244 @@
+/*
+  AudioOutputMixer
+  Simple mixer which can combine multiple inputs to a single output stream
+  
+  Copyright (C) 2018  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>
+#include "AudioOutputMixer.h"
+
+AudioOutputMixerStub::AudioOutputMixerStub(AudioOutputMixer *sink, int id) : AudioOutput()
+{
+  this->id = id;
+  this->parent = sink;
+  SetGain(1.0);
+}
+
+AudioOutputMixerStub::~AudioOutputMixerStub()
+{
+  parent->RemoveInput(id);
+}
+
+bool AudioOutputMixerStub::SetRate(int hz)
+{
+  return parent->SetRate(hz, id);
+}
+
+bool AudioOutputMixerStub::SetBitsPerSample(int bits)
+{
+  return parent->SetBitsPerSample(bits, id);
+}
+
+bool AudioOutputMixerStub::SetChannels(int channels)
+{
+  return parent->SetChannels(channels, id);
+}
+
+bool AudioOutputMixerStub::begin()
+{
+  return parent->begin(id);
+}
+
+bool AudioOutputMixerStub::ConsumeSample(int16_t sample[2])
+{
+  int16_t amp[2];
+  amp[LEFTCHANNEL] = Amplify(sample[LEFTCHANNEL]);
+  amp[RIGHTCHANNEL] = Amplify(sample[RIGHTCHANNEL]);
+  return parent->ConsumeSample(amp, id);
+}
+
+bool AudioOutputMixerStub::stop()
+{
+  return parent->stop(id);
+}
+
+
+
+AudioOutputMixer::AudioOutputMixer(int buffSizeSamples, AudioOutput *dest) : AudioOutput()
+{
+  buffSize = buffSizeSamples;
+  leftAccum = (int32_t*)calloc(sizeof(int32_t), buffSize);
+  rightAccum = (int32_t*)calloc(sizeof(int32_t), buffSize);
+  for (int i=0; i<maxStubs; i++) {
+    stubAllocated[i] = false;
+    stubRunning[i] = false;
+    writePtr[i] = 0;
+  }
+  readPtr = 0;
+  sink = dest;
+  sinkStarted = false;
+}
+
+AudioOutputMixer::~AudioOutputMixer()
+{
+  free(leftAccum);
+  free(rightAccum);
+}
+
+
+// Most "standard" interfaces should fail, only MixerStub should be able to talk to us
+bool AudioOutputMixer::SetRate(int hz)
+{
+  (void) hz;
+  return false;
+}
+
+bool AudioOutputMixer::SetBitsPerSample(int bits)
+{
+  (void) bits;
+  return false;
+}
+
+bool AudioOutputMixer::SetChannels(int channels)
+{
+  (void) channels;
+  return false;
+}
+
+bool AudioOutputMixer::ConsumeSample(int16_t sample[2])
+{
+  (void) sample;
+  return false;
+}
+
+bool AudioOutputMixer::begin()
+{
+  return false;
+}
+
+bool AudioOutputMixer::stop()
+{
+  return false;
+}
+
+
+// TODO - actually ensure all samples are same speed, size, channels, rate
+bool AudioOutputMixer::SetRate(int hz, int id)
+{
+  (void) id;
+  return sink->SetRate(hz);
+}
+
+bool AudioOutputMixer::SetBitsPerSample(int bits, int id)
+{
+  (void) id;
+  return sink->SetBitsPerSample(bits);
+}
+
+bool AudioOutputMixer::SetChannels(int channels, int id)
+{
+  (void) id;
+  return sink->SetChannels(channels);
+}
+
+bool AudioOutputMixer::begin(int id)
+{
+  stubRunning[id] = true;
+
+  if (!sinkStarted) {
+    sinkStarted = true;
+    return sink->begin();
+  } else {
+    return true;
+  }
+}
+  
+AudioOutputMixerStub *AudioOutputMixer::NewInput()
+{
+  for (int i=0; i<maxStubs; i++) {
+    if (!stubAllocated[i]) {
+      stubAllocated[i] = true;
+      stubRunning[i] = false;
+      writePtr[i] = readPtr; // TODO - should it be 1 before readPtr?
+      AudioOutputMixerStub *stub = new AudioOutputMixerStub(this, i);
+      return stub;
+    }
+  }
+  return nullptr;
+}
+
+void AudioOutputMixer::RemoveInput(int id)
+{
+  stubAllocated[id] = false;
+  stubRunning[id] = false;
+}
+
+bool AudioOutputMixer::loop()
+{
+  // First, try and fill I2S...
+  // This is not optimal, but algorithmically should work fine
+  bool avail;
+  do {
+    avail = true;
+    for (int i=0; i<maxStubs && avail; i++) {
+      if (stubRunning[i] && writePtr[i] == readPtr) {
+        avail = false;  // The read pointer is touching an active writer, can't advance
+      }
+    }
+    if (avail) {
+      int16_t s[2];
+      if (leftAccum[readPtr] > 32767) {
+        s[LEFTCHANNEL] = 32767;
+      } else if (leftAccum[readPtr] < -32767) {
+        s[LEFTCHANNEL] = -32767;
+      } else {
+        s[LEFTCHANNEL] = leftAccum[readPtr];
+      }
+      if (rightAccum[readPtr] > 32767) {
+        s[RIGHTCHANNEL] = 32767;
+      } else if (rightAccum[readPtr] < -32767) {
+        s[RIGHTCHANNEL] = -32767;
+      } else {
+        s[RIGHTCHANNEL] = rightAccum[readPtr];
+      }
+//      s[LEFTCHANNEL] = Amplify(s[LEFTCHANNEL]);
+//      s[RIGHTCHANNEL] = Amplify(s[RIGHTCHANNEL]);
+      if (!sink->ConsumeSample(s)) {
+        break; // Can't stuff any more in I2S...
+      }
+      // Clear the accums and advance the pointer to next potential sample
+      leftAccum[readPtr] = 0;
+      rightAccum[readPtr] = 0;
+      readPtr = (readPtr + 1) % buffSize;
+    }
+  } while (avail);
+  return true;
+}
+
+bool AudioOutputMixer::ConsumeSample(int16_t sample[2], int id)
+{
+  loop(); // Send any pre-existing, completed I2S data we can fit
+
+  // Now, do we have space for a new sample?
+  int nextWritePtr = (writePtr[id] + 1) % buffSize;
+  if (nextWritePtr == readPtr) {
+    return false;
+  }
+
+  leftAccum[writePtr[id]] += sample[LEFTCHANNEL];
+  rightAccum[writePtr[id]] += sample[RIGHTCHANNEL];
+  writePtr[id] = nextWritePtr;
+  return true;
+}
+
+bool AudioOutputMixer::stop(int id)
+{
+  stubRunning[id] = false;
+  return true;
+}
+
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputMixer.h

@@ -0,0 +1,88 @@
+/*
+  AudioOutputMixer
+  Simple mixer which can combine multiple inputs to a single output stream
+  
+  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/>.
+*/
+
+#ifndef _AUDIOOUTPUTMIXER_H
+#define _AUDIOOUTPUTMIXER_H
+
+#include "AudioOutput.h"
+
+class AudioOutputMixer;
+
+
+// The output stub exported by the mixer for use by the generator
+class AudioOutputMixerStub : public AudioOutput
+{
+  public:
+    AudioOutputMixerStub(AudioOutputMixer *sink, int id);
+    virtual ~AudioOutputMixerStub() override;
+    virtual bool SetRate(int hz) override;
+    virtual bool SetBitsPerSample(int bits) override;
+    virtual bool SetChannels(int channels) override;
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+
+  protected:
+    AudioOutputMixer *parent;
+    int id;
+};
+
+// Single mixer object per output
+class AudioOutputMixer : public AudioOutput
+{
+  public:
+    AudioOutputMixer(int samples, AudioOutput *sink);
+    virtual ~AudioOutputMixer() override;
+    virtual bool SetRate(int hz) override;
+    virtual bool SetBitsPerSample(int bits) override;
+    virtual bool SetChannels(int channels) override;
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+    virtual bool loop() override; // Send all existing samples we can to I2S
+
+    AudioOutputMixerStub *NewInput(); // Get a new stub to pass to a generator
+
+  // Stub called functions
+  friend class AudioOutputMixerStub;
+  private:
+    void RemoveInput(int id);
+    bool SetRate(int hz, int id);
+    bool SetBitsPerSample(int bits, int id);
+    bool SetChannels(int channels, int id);
+    bool begin(int id);
+    bool ConsumeSample(int16_t sample[2], int id);
+    bool stop(int id);
+
+  protected:
+    enum { maxStubs = 8 };
+    AudioOutput *sink;
+    bool sinkStarted;
+    int16_t buffSize;
+    int32_t *leftAccum;
+    int32_t *rightAccum;
+    bool stubAllocated[maxStubs];
+    bool stubRunning[maxStubs];
+    int16_t writePtr[maxStubs]; // Array of pointers for allocated stubs
+    int16_t readPtr;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputNull.h

@@ -0,0 +1,45 @@
+/*
+  AudioOutput
+  Base class of an audio output player
+  
+  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/>.
+*/
+
+#ifndef _AUDIOOUTPUTNULL_H
+#define _AUDIOOUTPUTNULL_H
+
+#include "AudioOutput.h"
+
+class AudioOutputNull : public AudioOutput 
+{
+  public:
+    AudioOutputNull() {};
+    ~AudioOutputNull() {};
+    virtual bool begin() { samples = 0; startms = millis(); return true; }
+    virtual bool ConsumeSample(int16_t sample[2]) { (void)sample; samples++; return true; }
+    virtual bool stop() { endms = millis(); return true; };
+    unsigned long GetMilliseconds() { return endms - startms; }
+    int GetSamples() { return samples; }
+    int GetFrequency() { return hertz; }
+
+  protected:
+    unsigned long startms;
+    unsigned long endms;
+    int samples;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputSPDIF.cpp

@@ -0,0 +1,291 @@
+/*
+  AudioOutputSPDIF
+  
+  S/PDIF output via I2S
+  
+  Needs transciever from CMOS level to either optical or coaxial interface
+  See: https://www.epanorama.net/documents/audio/spdif.html
+
+  Original idea and sources: 
+    Forum thread dicussing implementation
+      https://forum.pjrc.com/threads/28639-S-pdif
+    Teensy Audio Library 
+      https://github.com/PaulStoffregen/Audio/blob/master/output_spdif2.cpp
+   
+  Adapted for ESP8266Audio
+
+  NOTE: This module operates I2S at 4x sampling rate, as it needs to 
+        send out each bit as two output symbols, packed into 
+        32-bit words. Even for mono sound, S/PDIF is specified minimum
+        for 2 channels, each as 32-bits sub-frame. This drains I2S 
+        buffers 4x more quickly so you may need 4x bigger output 
+        buffers than usual, configurable with 'dma_buf_count' 
+        constructor parameter.
+
+  Copyright (C) 2020 Ivan Kostoski
+
+  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/>.
+*/
+#if defined(ESP32) || defined(ESP8266)
+
+#include <Arduino.h>
+#if defined(ESP32)
+  #include "driver/i2s.h"
+  #include "soc/rtc.h"
+#elif defined(ESP8266)
+  #include "driver/SinglePinI2SDriver.h"
+#endif
+#include "AudioOutputSPDIF.h"
+
+// BMC (Biphase Mark Coded) values (bit order reversed, i.e. LSB first)
+static const uint16_t spdif_bmclookup[256] PROGMEM = { 
+	0xcccc, 0x4ccc, 0x2ccc, 0xaccc, 0x34cc, 0xb4cc, 0xd4cc, 0x54cc,
+	0x32cc, 0xb2cc, 0xd2cc, 0x52cc, 0xcacc, 0x4acc, 0x2acc, 0xaacc,
+	0x334c, 0xb34c, 0xd34c, 0x534c, 0xcb4c, 0x4b4c, 0x2b4c, 0xab4c,
+	0xcd4c, 0x4d4c, 0x2d4c, 0xad4c, 0x354c, 0xb54c, 0xd54c, 0x554c,
+	0x332c, 0xb32c, 0xd32c, 0x532c, 0xcb2c, 0x4b2c, 0x2b2c, 0xab2c,
+	0xcd2c, 0x4d2c, 0x2d2c, 0xad2c, 0x352c, 0xb52c, 0xd52c, 0x552c,
+	0xccac, 0x4cac, 0x2cac, 0xacac, 0x34ac, 0xb4ac, 0xd4ac, 0x54ac,
+	0x32ac, 0xb2ac, 0xd2ac, 0x52ac, 0xcaac, 0x4aac, 0x2aac, 0xaaac,
+	0x3334, 0xb334, 0xd334, 0x5334, 0xcb34, 0x4b34, 0x2b34, 0xab34,
+	0xcd34, 0x4d34, 0x2d34, 0xad34, 0x3534, 0xb534, 0xd534, 0x5534,
+	0xccb4, 0x4cb4, 0x2cb4, 0xacb4, 0x34b4, 0xb4b4, 0xd4b4, 0x54b4,
+	0x32b4, 0xb2b4, 0xd2b4, 0x52b4, 0xcab4, 0x4ab4, 0x2ab4, 0xaab4,
+	0xccd4, 0x4cd4, 0x2cd4, 0xacd4, 0x34d4, 0xb4d4, 0xd4d4, 0x54d4,
+	0x32d4, 0xb2d4, 0xd2d4, 0x52d4, 0xcad4, 0x4ad4, 0x2ad4, 0xaad4,
+	0x3354, 0xb354, 0xd354, 0x5354, 0xcb54, 0x4b54, 0x2b54, 0xab54,
+	0xcd54, 0x4d54, 0x2d54, 0xad54, 0x3554, 0xb554, 0xd554, 0x5554,
+	0x3332, 0xb332, 0xd332, 0x5332, 0xcb32, 0x4b32, 0x2b32, 0xab32,
+	0xcd32, 0x4d32, 0x2d32, 0xad32, 0x3532, 0xb532, 0xd532, 0x5532,
+	0xccb2, 0x4cb2, 0x2cb2, 0xacb2, 0x34b2, 0xb4b2, 0xd4b2, 0x54b2,
+	0x32b2, 0xb2b2, 0xd2b2, 0x52b2, 0xcab2, 0x4ab2, 0x2ab2, 0xaab2,
+	0xccd2, 0x4cd2, 0x2cd2, 0xacd2, 0x34d2, 0xb4d2, 0xd4d2, 0x54d2,
+	0x32d2, 0xb2d2, 0xd2d2, 0x52d2, 0xcad2, 0x4ad2, 0x2ad2, 0xaad2,
+	0x3352, 0xb352, 0xd352, 0x5352, 0xcb52, 0x4b52, 0x2b52, 0xab52,
+	0xcd52, 0x4d52, 0x2d52, 0xad52, 0x3552, 0xb552, 0xd552, 0x5552,
+	0xccca, 0x4cca, 0x2cca, 0xacca, 0x34ca, 0xb4ca, 0xd4ca, 0x54ca,
+	0x32ca, 0xb2ca, 0xd2ca, 0x52ca, 0xcaca, 0x4aca, 0x2aca, 0xaaca,
+	0x334a, 0xb34a, 0xd34a, 0x534a, 0xcb4a, 0x4b4a, 0x2b4a, 0xab4a,
+	0xcd4a, 0x4d4a, 0x2d4a, 0xad4a, 0x354a, 0xb54a, 0xd54a, 0x554a,
+	0x332a, 0xb32a, 0xd32a, 0x532a, 0xcb2a, 0x4b2a, 0x2b2a, 0xab2a,
+	0xcd2a, 0x4d2a, 0x2d2a, 0xad2a, 0x352a, 0xb52a, 0xd52a, 0x552a,
+	0xccaa, 0x4caa, 0x2caa, 0xacaa, 0x34aa, 0xb4aa, 0xd4aa, 0x54aa,
+	0x32aa, 0xb2aa, 0xd2aa, 0x52aa, 0xcaaa, 0x4aaa, 0x2aaa, 0xaaaa
+};
+
+AudioOutputSPDIF::AudioOutputSPDIF(int dout_pin, int port, int dma_buf_count)
+{
+  this->portNo = port;
+#if defined(ESP32)
+  // Configure ESP32 I2S to roughly compatible to ESP8266 peripheral
+  i2s_config_t i2s_config_spdif = {
+    .mode = (i2s_mode_t)(I2S_MODE_MASTER | I2S_MODE_TX),
+    .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),
+    .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
+  };
+  if (i2s_driver_install((i2s_port_t)portNo, &i2s_config_spdif, 0, NULL) != ESP_OK) {
+    audioLogger->println(F("ERROR: Unable to install I2S drivers"));
+    return;
+  }
+  i2s_zero_dma_buffer((i2s_port_t)portNo);
+  SetPinout(I2S_PIN_NO_CHANGE, I2S_PIN_NO_CHANGE, dout_pin);
+  rate_multiplier = 2; // 2x32bit words
+#elif defined(ESP8266)
+  (void) dout_pin;
+  if (!I2SDriver.begin(dma_buf_count, DMA_BUF_SIZE_DEFAULT)) {
+    audioLogger->println(F("ERROR: Unable to start I2S driver"));
+    return;
+  }
+  rate_multiplier = 4; // 4x16 bit words
+#endif
+  i2sOn = true;
+  mono = false;
+  bps = 16;
+  channels = 2;
+  frame_num = 0;
+  SetGain(1.0);
+  hertz = 0;
+  SetRate(44100);
+}
+
+AudioOutputSPDIF::~AudioOutputSPDIF()
+{
+#if defined(ESP32)
+  if (i2sOn) {
+    i2s_stop((i2s_port_t)this->portNo);
+    audioLogger->printf("UNINSTALL I2S\n");
+    i2s_driver_uninstall((i2s_port_t)this->portNo); //stop & destroy i2s driver
+  }
+#elif defined(ESP8266)
+  if (i2sOn) I2SDriver.stop();
+#endif
+  i2sOn = false;
+}
+
+bool AudioOutputSPDIF::SetPinout(int bclk, int wclk, int dout)
+{
+#if defined(ESP32)
+  i2s_pin_config_t pins = {
+    .bck_io_num = bclk,
+    .ws_io_num = wclk,
+    .data_out_num = dout,
+    .data_in_num = I2S_PIN_NO_CHANGE
+  };
+  if (i2s_set_pin((i2s_port_t)portNo, &pins) != ESP_OK) {
+    audioLogger->println("ERROR setting up S/PDIF I2S pins\n");
+    return false;
+  }
+  return true;
+#else
+  (void) bclk;
+  (void) wclk;
+  (void) dout;
+  return false;
+#endif
+}
+
+bool AudioOutputSPDIF::SetRate(int hz)
+{
+  if (!i2sOn) return false;
+  if (hz < 32000) return false;
+  if (hz == this->hertz) return true;
+  this->hertz = hz;
+  int adjustedHz = AdjustI2SRate(hz);
+#if defined(ESP32)
+  if (i2s_set_sample_rates((i2s_port_t)portNo, adjustedHz) == ESP_OK) {
+    if (adjustedHz == 88200) {
+      // Manually fix the APLL rate for 44100. 
+      // See: https://github.com/espressif/esp-idf/issues/2634
+      // sdm0 = 28, sdm1 = 8, sdm2 = 5, odir = 0 -> 88199.977
+      rtc_clk_apll_enable(1, 28, 8, 5, 0); 
+    }
+  } else {
+    audioLogger->println("ERROR changing S/PDIF sample rate");
+  } 
+#elif defined(ESP8266)
+  I2SDriver.setRate(adjustedHz);
+  audioLogger->printf_P(PSTR("S/PDIF rate set: %.3f\n"), I2SDriver.getActualRate()/4);
+#endif
+  return true;
+}
+
+bool AudioOutputSPDIF::SetBitsPerSample(int bits)
+{
+  if ( (bits != 16) && (bits != 8) ) return false;
+  this->bps = bits;
+  return true;
+}
+
+bool AudioOutputSPDIF::SetChannels(int channels)
+{
+  if ( (channels < 1) || (channels > 2) ) return false;
+  this->channels = channels;
+  return true;
+}
+
+bool AudioOutputSPDIF::SetOutputModeMono(bool mono)
+{
+  this->mono = mono;
+  // Just use the left channel for mono
+  if (mono) SetChannels(1);
+  return true;
+}
+
+bool AudioOutputSPDIF::begin()
+{
+  return true;
+}
+
+bool AudioOutputSPDIF::ConsumeSample(int16_t sample[2])
+{
+  if (!i2sOn) return true; // Sink the data
+  int16_t ms[2];
+  uint16_t hi, lo, aux;
+  uint32_t buf[4];
+
+  ms[0] = sample[0];
+  ms[1] = sample[1];
+  MakeSampleStereo16(ms);
+
+  // S/PDIF encoding: 
+  //   http://www.hardwarebook.info/S/PDIF
+  // Original sources: Teensy Audio Library 
+  //   https://github.com/PaulStoffregen/Audio/blob/master/output_spdif2.cpp
+  // 
+  // Order of bits, before BMC encoding, from the definition of SPDIF format
+  //   PPPP AAAA  SSSS SSSS  SSSS SSSS  SSSS VUCP
+  // are sent rearanged as
+  //   VUCP PPPP  AAAA 0000  SSSS SSSS  SSSS SSSS
+  // This requires a bit less shifting as 16 sample bits align and can be 
+  // BMC encoded with two table lookups (and at the same time flipped to LSB first).
+  // There is no separate word-clock, so hopefully the receiver won't notice.
+
+  uint16_t sample_left = Amplify(ms[LEFTCHANNEL]);
+  // BMC encode and flip left channel bits
+  hi = pgm_read_word(&spdif_bmclookup[(uint8_t)(sample_left >> 8)]);
+  lo = pgm_read_word(&spdif_bmclookup[(uint8_t)sample_left]);
+  // Low word is inverted depending on first bit of high word
+  lo ^= (~((int16_t)hi) >> 16);
+  buf[0] = ((uint32_t)lo << 16) | hi;
+  // Fixed 4 bits auxillary-audio-databits, the first used as parity
+  // Depending on first bit of low word, invert the bits
+  aux = 0xb333 ^ (((uint32_t)((int16_t)lo)) >> 17);
+  // Send 'B' preamble only for the first frame of data-block
+  if (frame_num == 0) {
+    buf[1] = VUCP_PREAMBLE_B | aux;
+  } else {
+    buf[1] = VUCP_PREAMBLE_M | aux;
+  }
+
+  uint16_t sample_right = Amplify(ms[RIGHTCHANNEL]); 
+  // BMC encode right channel, similar as above
+  hi = pgm_read_word(&spdif_bmclookup[(uint8_t)(sample_right >> 8)]);
+  lo = pgm_read_word(&spdif_bmclookup[(uint8_t)sample_right]);
+  lo ^= (~((int16_t)hi) >> 16);
+  buf[2] = ((uint32_t)lo << 16) | hi;
+  aux = 0xb333 ^ (((uint32_t)((int16_t)lo)) >> 17);
+  buf[3] = VUCP_PREAMBLE_W | aux;
+
+#if defined(ESP32)
+  // Assume DMA buffers are multiples of 16 bytes. Either we write all bytes or none.
+  uint32_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;  
+#elif defined(ESP8266)
+  if (!I2SDriver.writeInterleaved(buf)) return false;
+#endif
+  // Increment and rotate frame number
+  if (++frame_num > 191) frame_num = 0;
+  return true;
+}
+
+bool AudioOutputSPDIF::stop()
+{
+#if defined(ESP32)
+  i2s_zero_dma_buffer((i2s_port_t)portNo);
+#elif defined(ESP8266)
+  I2SDriver.stop();
+#endif
+  frame_num = 0;
+  return true;
+}
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputSPDIF.h

@@ -0,0 +1,76 @@
+/*
+  AudioOutputSPDIF
+  
+  S/PDIF output via I2S
+  
+  Needs transciever from CMOS level to either optical or coaxial interface
+  See: https://www.epanorama.net/documents/audio/spdif.html
+
+  Original idea and sources: 
+    Forum thread dicussing implementation
+      https://forum.pjrc.com/threads/28639-S-pdif
+    Teensy Audio Library 
+      https://github.com/PaulStoffregen/Audio/blob/master/output_spdif2.cpp
+   
+  Adapted for ESP8266Audio
+
+  Copyright (C) 2020 Ivan Kostoski
+
+  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/>.
+*/
+
+#if defined(ESP32) || defined(ESP8266)
+#pragma once
+
+#include "AudioOutput.h"
+
+#if defined(ESP32)
+#define SPDIF_OUT_PIN_DEFAULT  27
+#define DMA_BUF_COUNT_DEFAULT  8
+#define DMA_BUF_SIZE_DEFAULT   256
+#elif defined(ESP8266)
+#define SPDIF_OUT_PIN_DEFAULT  3
+#define DMA_BUF_COUNT_DEFAULT  32
+#define DMA_BUF_SIZE_DEFAULT   64
+#endif
+
+class AudioOutputSPDIF : public AudioOutput
+{
+  public:
+    AudioOutputSPDIF(int dout_pin=SPDIF_OUT_PIN_DEFAULT, int port=0, int dma_buf_count = DMA_BUF_COUNT_DEFAULT);
+    virtual ~AudioOutputSPDIF() override;
+    bool SetPinout(int bclkPin, int wclkPin, int doutPin);
+    virtual bool SetRate(int hz) override;
+    virtual bool SetBitsPerSample(int bits) override;
+    virtual bool SetChannels(int channels) override;
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+
+    bool SetOutputModeMono(bool mono);  // Force mono output no matter the input
+
+    const uint32_t VUCP_PREAMBLE_B = 0xCCE80000; // 11001100 11101000
+    const uint32_t VUCP_PREAMBLE_M = 0xCCE20000; // 11001100 11100010
+    const uint32_t VUCP_PREAMBLE_W = 0xCCE40000; // 11001100 11100100
+
+  protected:
+    virtual inline int AdjustI2SRate(int hz) { return rate_multiplier * hz; }
+    uint8_t portNo;
+    bool mono;
+    bool i2sOn;
+    uint8_t frame_num;
+    uint8_t rate_multiplier;
+};
+
+#endif // _AUDIOOUTPUTSPDIF_H

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputSPIFFSWAV.cpp

@@ -0,0 +1,113 @@
+/*
+  AudioOutputSPIFFSWAV
+  Writes a WAV file to the SPIFFS filesystem
+  
+  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>
+#include <FS.h>
+#ifdef ESP32
+#include "SPIFFS.h"
+#endif
+
+#include "AudioOutputSPIFFSWAV.h"
+
+static const uint8_t wavHeaderTemplate[] PROGMEM = { // Hardcoded simple WAV header with 0xffffffff lengths all around
+    0x52, 0x49, 0x46, 0x46, 0xff, 0xff, 0xff, 0xff, 0x57, 0x41, 0x56, 0x45,
+    0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0x02, 0x00, 0x22, 0x56, 0x00, 0x00, 0x88, 0x58, 0x01, 0x00, 0x04, 0x00, 0x10, 0x00,
+    0x64, 0x61, 0x74, 0x61, 0xff, 0xff, 0xff, 0xff };
+
+void AudioOutputSPIFFSWAV::SetFilename(const char *name)
+{
+  if (filename) free(filename);
+  filename = strdup(name);
+}
+
+bool AudioOutputSPIFFSWAV::begin()
+{
+  uint8_t wavHeader[sizeof(wavHeaderTemplate)];
+  memset(wavHeader, 0, sizeof(wavHeader));
+
+  if (f) return false; // Already open!
+  SPIFFS.remove(filename);
+  f = SPIFFS.open(filename, "w+");
+  if (!f) return false;
+  
+  // We'll fix the header up when we close the file
+  f.write(wavHeader, sizeof(wavHeader));
+  return true;
+}
+
+bool AudioOutputSPIFFSWAV::ConsumeSample(int16_t sample[2])
+{
+  for (int i=0; i<channels; i++) {
+    if (bps == 8) {
+      uint8_t l = sample[i] & 0xff;
+      f.write(&l, sizeof(l));
+    } else {
+      uint8_t l = sample[i] & 0xff;
+      uint8_t h = (sample[i] >> 8) & 0xff;
+      f.write(&l, sizeof(l));
+      f.write(&h, sizeof(h));
+    }
+  }
+  return true;
+}
+
+
+bool AudioOutputSPIFFSWAV::stop()
+{
+  uint8_t wavHeader[sizeof(wavHeaderTemplate)];
+
+  memcpy_P(wavHeader, wavHeaderTemplate, sizeof(wavHeaderTemplate));
+
+  int chunksize = f.size() - 8;
+  wavHeader[4] = chunksize & 0xff;
+  wavHeader[5] = (chunksize>>8)&0xff;
+  wavHeader[6] = (chunksize>>16)&0xff;
+  wavHeader[7] = (chunksize>>24)&0xff;
+
+  wavHeader[22] = channels & 0xff;
+  wavHeader[23] = 0;
+
+  wavHeader[24] = hertz & 0xff;
+  wavHeader[25] = (hertz >> 8) & 0xff;
+  wavHeader[26] = (hertz >> 16) & 0xff;
+  wavHeader[27] = (hertz >> 24) & 0xff;
+  int byteRate = hertz * bps * channels / 8;
+  wavHeader[28] = byteRate & 0xff;
+  wavHeader[29] = (byteRate >> 8) & 0xff;
+  wavHeader[30] = (byteRate >> 16) & 0xff;
+  wavHeader[31] = (byteRate >> 24) & 0xff;
+  wavHeader[32] = channels * bps / 8;
+  wavHeader[33] = 0;
+  wavHeader[34] = bps;
+  wavHeader[35] = 0;
+
+  int datasize = f.size() - sizeof(wavHeader);
+  wavHeader[40] = datasize & 0xff;
+  wavHeader[41] = (datasize>>8)&0xff;
+  wavHeader[42] = (datasize>>16)&0xff;
+  wavHeader[43] = (datasize>>24)&0xff;
+
+  // Write real header out
+  f.seek(0, SeekSet);
+  f.write(wavHeader, sizeof(wavHeader));
+  f.close();
+  return true;
+}
+ 

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputSPIFFSWAV.h

@@ -0,0 +1,45 @@
+/*
+  AudioOutputSPIFFSWAV
+  Writes a WAV file to the SPIFFS filesystem
+  
+  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/>.
+*/
+
+#ifndef _AUDIOOUTPUTSPIFFSWAV_H
+#define _AUDIOOUTPUTSPIFFSWAV_H
+
+#include <Arduino.h>
+#include <FS.h>
+
+#include "AudioOutput.h"
+
+class AudioOutputSPIFFSWAV : public AudioOutput
+{
+  public:
+    AudioOutputSPIFFSWAV() { filename = NULL; };
+    ~AudioOutputSPIFFSWAV() { free(filename); };
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+    void SetFilename(const char *name);
+
+  private:
+    File f;
+    char *filename;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputSTDIO.cpp

@@ -0,0 +1,115 @@
+/*
+  AudioOutputSTDIO
+  Writes a WAV file to the STDIO filesystem
+  Only for host-based testing
+
+  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>
+
+#ifndef ARDUINO
+
+#include "AudioOutputSTDIO.h"
+#include <unistd.h>
+
+static const uint8_t wavHeaderTemplate[] PROGMEM = { // Hardcoded simple WAV header with 0xffffffff lengths all around
+    0x52, 0x49, 0x46, 0x46, 0xff, 0xff, 0xff, 0xff, 0x57, 0x41, 0x56, 0x45,
+    0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0x02, 0x00, 0x22, 0x56, 0x00, 0x00, 0x88, 0x58, 0x01, 0x00, 0x04, 0x00, 0x10, 0x00,
+    0x64, 0x61, 0x74, 0x61, 0xff, 0xff, 0xff, 0xff };
+
+void AudioOutputSTDIO::SetFilename(const char *name)
+{
+  free(filename);
+  filename = strdup(name);
+}
+
+bool AudioOutputSTDIO::begin()
+{
+  uint8_t wavHeader[sizeof(wavHeaderTemplate)];
+  memset(wavHeader, 0, sizeof(wavHeader));
+
+  if (f) return false; // Already open!
+  unlink(filename);
+  f = fopen(filename, "wb+");
+  if (!f) return false;
+  
+  // We'll fix the header up when we close the file
+  fwrite(wavHeader, sizeof(wavHeader), 1, f);
+  return true;
+}
+
+bool AudioOutputSTDIO::ConsumeSample(int16_t sample[2])
+{
+  for (int i=0; i<channels; i++) {
+    if (bps == 8) {
+      uint8_t l = sample[i] & 0xff;
+      fwrite(&l, sizeof(l), 1, f);
+    } else {
+      uint8_t l = sample[i] & 0xff;
+      uint8_t h = (sample[i] >> 8) & 0xff;
+      fwrite(&l, sizeof(l), 1, f);
+      fwrite(&h, sizeof(h), 1, f);
+    }
+  }
+  return true;
+}
+
+
+bool AudioOutputSTDIO::stop()
+{
+  uint8_t wavHeader[sizeof(wavHeaderTemplate)];
+
+  memcpy_P(wavHeader, wavHeaderTemplate, sizeof(wavHeaderTemplate));
+
+  int chunksize = ftell(f) - 8;
+  wavHeader[4] = chunksize & 0xff;
+  wavHeader[5] = (chunksize>>8)&0xff;
+  wavHeader[6] = (chunksize>>16)&0xff;
+  wavHeader[7] = (chunksize>>24)&0xff;
+
+  wavHeader[22] = channels & 0xff;
+  wavHeader[23] = 0;
+
+  wavHeader[24] = hertz & 0xff;
+  wavHeader[25] = (hertz >> 8) & 0xff;
+  wavHeader[26] = (hertz >> 16) & 0xff;
+  wavHeader[27] = (hertz >> 24) & 0xff;
+  int byteRate = hertz * bps * channels / 8;
+  wavHeader[28] = byteRate & 0xff;
+  wavHeader[29] = (byteRate >> 8) & 0xff;
+  wavHeader[30] = (byteRate >> 16) & 0xff;
+  wavHeader[31] = (byteRate >> 24) & 0xff;
+  wavHeader[32] = channels * bps / 8;
+  wavHeader[33] = 0;
+  wavHeader[34] = bps;
+  wavHeader[35] = 0;
+
+  int datasize = ftell(f) - sizeof(wavHeader);
+  wavHeader[40] = datasize & 0xff;
+  wavHeader[41] = (datasize>>8)&0xff;
+  wavHeader[42] = (datasize>>16)&0xff;
+  wavHeader[43] = (datasize>>24)&0xff;
+
+  // Write real header out
+  fseek(f, 0, SEEK_SET);
+  fwrite(wavHeader, sizeof(wavHeader), 1, f);
+  fclose(f);
+  return true;
+}
+
+#endif
+ 

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputSTDIO.h

@@ -0,0 +1,47 @@
+/*
+  AudioOutputSTDIO
+  Writes a WAV file to the STDIO filesystem
+  Only for host-based testing
+
+  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/>.
+*/
+
+#ifndef _AUDIOOUTPUTSTDIO_H
+#define _AUDIOOUTPUTSTDIO_H
+
+#include <Arduino.h>
+#ifndef ARDUINO
+
+#include "AudioOutput.h"
+
+class AudioOutputSTDIO : public AudioOutput
+{
+  public:
+    AudioOutputSTDIO() { filename = NULL; f = NULL; };
+    ~AudioOutputSTDIO() { free(filename); };
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+    void SetFilename(const char *name);
+
+  private:
+    FILE *f;
+    char *filename;
+};
+
+#endif
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputSerialWAV.cpp

@@ -0,0 +1,78 @@
+/*
+  AudioOutputSerialWAV
+  Writes a mostly correct WAV file to the serial 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 "AudioOutputSerialWAV.h"
+
+static const uint8_t wavHeaderTemplate[] PROGMEM = { // Hardcoded simple WAV header with 0xffffffff lengths all around
+    0x52, 0x49, 0x46, 0x46, 0xff, 0xff, 0xff, 0xff, 0x57, 0x41, 0x56, 0x45,
+    0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0x02, 0x00, 0x22, 0x56, 0x00, 0x00, 0x88, 0x58, 0x01, 0x00, 0x04, 0x00, 0x10, 0x00,
+    0x64, 0x61, 0x74, 0x61, 0xff, 0xff, 0xff, 0xff };
+
+bool AudioOutputSerialWAV::begin()
+{
+  uint8_t wavHeader[sizeof(wavHeaderTemplate)];
+  memcpy_P(wavHeader, wavHeaderTemplate, sizeof(wavHeaderTemplate));
+  wavHeader[22] = channels & 0xff;
+  wavHeader[23] = 0;
+  wavHeader[24] = hertz & 0xff;
+  wavHeader[25] = (hertz >> 8) & 0xff;
+  wavHeader[26] = (hertz >> 16) & 0xff;
+  wavHeader[27] = (hertz >> 24) & 0xff;
+  int byteRate = hertz * bps * channels / 8;
+  wavHeader[28] = byteRate & 0xff;
+  wavHeader[29] = (byteRate >> 8) & 0xff;
+  wavHeader[30] = (byteRate >> 16) & 0xff;
+  wavHeader[31] = (byteRate >> 24) & 0xff;
+  wavHeader[32] = channels * bps / 8;
+  wavHeader[33] = 0;
+  wavHeader[34] = bps;
+  wavHeader[35] = 0;
+  Serial.write(wavHeader, sizeof(wavHeader));
+  count = 0;
+  return true;
+}
+
+bool AudioOutputSerialWAV::ConsumeSample(int16_t sample[2])
+{
+  if (++count == 200) {
+    count = 0;
+    return false;
+  }
+  for (int i=0; i<channels; i++) {
+    if (bps == 8) {
+      uint8_t l = sample[i] & 0xff;
+      Serial.write(l);
+    } else {
+      uint8_t l = sample[i] & 0xff;
+      uint8_t h = (sample[i] >> 8) & 0xff;
+      Serial.write(l);
+      Serial.write(h);
+    }
+  }
+  return true;
+}
+
+
+bool AudioOutputSerialWAV::stop()
+{
+  audioLogger->printf_P(PSTR("\n\n\nEOF\n\n\n"));
+  return true;
+}
+ 

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputSerialWAV.h

@@ -0,0 +1,39 @@
+/*
+  AudioOutputSerialWAV
+  Writes a mostly correct WAV file to the serial 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/>.
+*/
+
+#ifndef _AUDIOOUTPUTSERIALWAV_H
+#define _AUDIOOUTPUTSERIALWAV_H
+
+#include "AudioOutput.h"
+
+class AudioOutputSerialWAV : public AudioOutput
+{
+  public:
+    AudioOutputSerialWAV() {};
+    ~AudioOutputSerialWAV() {};
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+  private:
+    int count;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputULP.cpp

@@ -0,0 +1,262 @@
+/*
+  AudioOutputULP
+  Outputs to ESP32 DAC through the ULP, freeing I2S for other uses
+  
+  Copyright (C) 2020  Martin Laclaustra, based on bitluni's code
+
+  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/>.
+*/
+
+#ifdef ESP32
+
+#include "AudioOutputULP.h"
+#include <esp32/ulp.h>
+#include <driver/rtc_io.h>
+#include <driver/dac.h>
+#include <soc/rtc.h>
+#include <math.h>
+
+uint32_t create_I_WR_REG(uint32_t reg, uint32_t low_bit, uint32_t high_bit, uint32_t val){
+    typedef union {ulp_insn_t ulp_ins; uint32_t ulp_bin;} ulp_union;
+    const ulp_insn_t singleinstruction[] = {I_WR_REG(reg, low_bit, high_bit, val)};
+    ulp_union recover_ins;
+    recover_ins.ulp_ins=singleinstruction[0];
+    return (uint32_t)(recover_ins.ulp_bin);
+}
+
+uint32_t create_I_BXI(uint32_t imm_pc){
+    typedef union {ulp_insn_t ulp_ins; uint32_t ulp_bin;} ulp_union;
+    const ulp_insn_t singleinstruction[] = {I_BXI(imm_pc)};
+    ulp_union recover_ins;
+    recover_ins.ulp_ins=singleinstruction[0];
+    return (uint32_t)(recover_ins.ulp_bin);
+}
+
+bool AudioOutputULP::begin()
+{
+  if(!stereoOutput){
+    waitingOddSample = false;
+    //totalSampleWords += 512;
+    //dacTableStart2 = dacTableStart1;
+  }
+  
+  //calculate the actual ULP clock
+  unsigned long rtc_8md256_period = rtc_clk_cal(RTC_CAL_8MD256, 1000);
+  unsigned long rtc_fast_freq_hz = 1000000ULL * (1 << RTC_CLK_CAL_FRACT) * 256 / rtc_8md256_period;
+
+  //initialize DACs
+  if(activeDACs & 1){
+    dac_output_enable(DAC_CHANNEL_1);
+    dac_output_voltage(DAC_CHANNEL_1, 128);
+  }
+  if(activeDACs & 2){
+    dac_output_enable(DAC_CHANNEL_2);
+    dac_output_voltage(DAC_CHANNEL_2, 128);
+  }
+
+  int retAddress1 = 9;
+  int retAddress2 = 14;
+
+  int loopCycles = 134;
+  int loopHalfCycles1 = 90;
+  int loopHalfCycles2 = 44;
+  
+  Serial.print("Real RTC clock: ");
+  Serial.println(rtc_fast_freq_hz);
+
+  uint32_t dt = (rtc_fast_freq_hz / hertz) - loopCycles;
+  uint32_t dt2 = 0;
+  if(!stereoOutput){
+    dt = (rtc_fast_freq_hz / hertz) - loopHalfCycles1;
+    dt2 = (rtc_fast_freq_hz / hertz) - loopHalfCycles2;
+  }
+  
+  Serial.print("dt: ");
+  Serial.println(dt);
+  
+  Serial.print("dt2: ");
+  Serial.println(dt2);
+
+  const ulp_insn_t stereo[] = {
+    //reset offset register
+    I_MOVI(R3, 0),
+    //delay to get the right sampling rate
+    I_DELAY(dt), // 6 + dt
+    //reset sample index
+    I_MOVI(R0, 0), // 6
+    //write the index back to memory for the main cpu
+    I_ST(R0, R3, indexAddress), // 8
+    //load the samples
+    I_LD(R1, R0, bufferStart), // 8
+    //mask the lower 8 bits
+    I_ANDI(R2, R1, 0x00ff), // 6
+    //multiply by 2
+    I_LSHI(R2, R2, 1), // 6
+    //add start position
+    I_ADDI(R2, R2, dacTableStart1),// 6
+    //jump to the dac opcode
+    I_BXR(R2), // 4
+    //back from first dac
+    //delay between the two samples in mono rendering
+    I_DELAY(dt2), // 6 + dt2
+    //mask the upper 8 bits
+    I_ANDI(R2, R1, 0xff00), // 6
+    //shift the upper bits to right and multiply by 2
+    I_RSHI(R2, R2, 8 - 1), // 6
+    //add start position of second dac table
+    I_ADDI(R2, R2, dacTableStart2),// 6
+    //jump to the dac opcode
+    I_BXR(R2), // 4
+    //here we get back from writing the second sample
+    //load 0x8080 as sample
+    I_MOVI(R1, 0x8080), // 6
+    //write 0x8080 in the sample buffer
+    I_ST(R1, R0, indexAddress), // 8
+    //increment the sample index
+    I_ADDI(R0, R0, 1), // 6
+    //if reached end of the buffer, jump relative to index reset
+    I_BGE(-16, totalSampleWords), // 4
+    //wait to get the right sample rate (2 cycles more to compensate the index reset)
+    I_DELAY((unsigned int)dt + 2), // 8 + dt
+    //if not, jump absolute to where index is written to memory
+    I_BXI(3) // 4
+  };
+  // write io and jump back another 12 + 4 + 12 + 4
+
+  size_t load_addr = 0;
+  size_t size = sizeof(stereo)/sizeof(ulp_insn_t);
+  ulp_process_macros_and_load(load_addr, stereo, &size);
+  //  this is how to get the opcodes
+  //  for(int i = 0; i < size; i++)
+  //    Serial.println(RTC_SLOW_MEM[i], HEX);
+  
+  //create DAC opcode tables
+  switch(activeDACs){
+    case 1:
+      for(int i = 0; i < 256; i++)
+      {
+        RTC_SLOW_MEM[dacTableStart1 + i * 2] = create_I_WR_REG(RTC_IO_PAD_DAC1_REG,19,26,i); //dac1: 0x1D4C0121 | (i << 10)
+        RTC_SLOW_MEM[dacTableStart1 + 1 + i * 2] = create_I_BXI(retAddress1); // 0x80000000 + retAddress1 * 4
+        RTC_SLOW_MEM[dacTableStart2 + i * 2] = create_I_WR_REG(RTC_IO_PAD_DAC1_REG,19,26,i); //dac2: 0x1D4C0122 | (i << 10)
+        RTC_SLOW_MEM[dacTableStart2 + 1 + i * 2] = create_I_BXI(retAddress2); // 0x80000000 + retAddress2 * 4
+      }
+      break;
+    case 2:
+      for(int i = 0; i < 256; i++)
+      {
+        RTC_SLOW_MEM[dacTableStart1 + i * 2] = create_I_WR_REG(RTC_IO_PAD_DAC2_REG,19,26,i); //dac1: 0x1D4C0121 | (i << 10)
+        RTC_SLOW_MEM[dacTableStart1 + 1 + i * 2] = create_I_BXI(retAddress1); // 0x80000000 + retAddress1 * 4
+        RTC_SLOW_MEM[dacTableStart2 + i * 2] = create_I_WR_REG(RTC_IO_PAD_DAC2_REG,19,26,i); //dac2: 0x1D4C0122 | (i << 10)
+        RTC_SLOW_MEM[dacTableStart2 + 1 + i * 2] = create_I_BXI(retAddress2); // 0x80000000 + retAddress2 * 4
+      }
+      break;
+    case 3:
+      for(int i = 0; i < 256; i++)
+      {
+        RTC_SLOW_MEM[dacTableStart1 + i * 2] = create_I_WR_REG(RTC_IO_PAD_DAC1_REG,19,26,i); //dac1: 0x1D4C0121 | (i << 10)
+        RTC_SLOW_MEM[dacTableStart1 + 1 + i * 2] = create_I_BXI(retAddress1); // 0x80000000 + retAddress1 * 4
+        RTC_SLOW_MEM[dacTableStart2 + i * 2] = create_I_WR_REG(RTC_IO_PAD_DAC1_REG,19,26,i); //dac2: 0x1D4C0122 | (i << 10)
+        RTC_SLOW_MEM[dacTableStart2 + 1 + i * 2] = create_I_BXI(retAddress2); // 0x80000000 + retAddress2 * 4
+      }
+      break;
+  }
+
+  //set all samples to 128 (silence)
+  for(int i = 0; i < totalSampleWords; i++)
+    RTC_SLOW_MEM[bufferStart + i] = 0x8080;
+    
+  //start
+  RTC_SLOW_MEM[indexAddress] = 0;
+  ulp_run(0);
+
+  //wait until ULP starts using samples and the index of output sample advances
+  while(RTC_SLOW_MEM[indexAddress] == 0)
+    delay(1);
+
+  return true;
+}
+
+bool AudioOutputULP::ConsumeSample(int16_t sample[2])
+{
+  int16_t ms[2];
+  ms[0] = sample[0];
+  ms[1] = sample[1];
+  MakeSampleStereo16( ms );
+
+  // TODO: needs improvement (counting is different here with respect to ULP code)
+  int currentSample = RTC_SLOW_MEM[indexAddress] & 0xffff;
+  int currentWord = currentSample >> 1;
+
+  for (int i=0; i<2; i++) {
+    ms[i] = ((ms[i] >> 8) + 128) & 0xff;
+  }
+  if(!stereoOutput) // mix both channels
+    ms[0] = (uint16_t)(( (uint32_t)((int32_t)(ms[0]) + (int32_t)(ms[1])) >> 1 ) & 0xff);
+
+  if(waitingOddSample){ // always true for stereo because samples are consumed in pairs
+    if(lastFilledWord != currentWord) // accept sample if writing index lastFilledWord has not reached index of output sample
+    {
+      unsigned int w;
+      if(stereoOutput){
+        w = ms[0];
+        w |= ms[1] << 8;
+      } else {
+        w = bufferedOddSample;
+        w |= ms[0] << 8;
+        bufferedOddSample = 128;
+        waitingOddSample = false;
+      }
+      RTC_SLOW_MEM[bufferStart + lastFilledWord] = w;
+      lastFilledWord++;
+      if(lastFilledWord == totalSampleWords)
+        lastFilledWord = 0;
+      return true;
+    } else {
+      return false;
+    }
+  } else {
+    bufferedOddSample = ms[0];
+    waitingOddSample = true;
+    return true;
+  }
+}
+
+
+bool AudioOutputULP::stop()
+{
+  audioLogger->printf_P(PSTR("\n\n\nstop\n\n\n"));
+  const ulp_insn_t stopulp[] = {
+    //stop the timer
+    I_END(),
+    //end the program
+    I_HALT()};
+
+  size_t load_addr = 0;
+  size_t size = sizeof(stopulp)/sizeof(ulp_insn_t);
+  ulp_process_macros_and_load(load_addr, stopulp, &size);
+  
+  //start
+  ulp_run(0);
+
+  if(activeDACs & 1){
+    dac_output_voltage(DAC_CHANNEL_1, 128);
+  }
+  if(activeDACs & 2){
+    dac_output_voltage(DAC_CHANNEL_2, 128);
+  }
+
+  return true;
+}
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioOutputULP.h

@@ -0,0 +1,69 @@
+/*
+  AudioOutputULP
+  Outputs to ESP32 DAC through the ULP, freeing I2S for other uses
+  
+  Copyright (C) 2020  Martin Laclaustra, based on bitluni's code
+
+  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/>.
+*/
+
+// Instructions:
+// AudioOutputULP out = new AudioOutputULP(); // stereo
+// Connect left channel on pin 25
+// Connect right channel on pin 26
+// OR
+// Connect mono channel on either of them (stereo samples are downmixed)
+// AudioOutputULP out = new AudioOutputULP(1); // mono, only DAC 1
+// OR
+// AudioOutputULP out = new AudioOutputULP(2); // mono, only DAC 2
+
+
+#ifndef _AUDIOOUTPUTULP_H
+#define _AUDIOOUTPUTULP_H
+
+#include "AudioOutput.h"
+
+#ifdef ESP32
+
+class AudioOutputULP : public AudioOutput
+{
+  public:
+    AudioOutputULP(int argActiveDACs=3) {if(argActiveDACs<1||argActiveDACs>2)argActiveDACs=3;activeDACs=argActiveDACs;stereoOutput=activeDACs==3;};
+    ~AudioOutputULP() {};
+    virtual bool begin() override;
+    virtual bool ConsumeSample(int16_t sample[2]) override;
+    virtual bool stop() override;
+    enum : int { DAC1 = 1, DAC2 = 2 };
+  private:
+    int lastFilledWord = 0;
+    uint8_t bufferedOddSample = 128;
+    bool waitingOddSample = true; // must be set to false for mono output
+    int activeDACs = 3; // 1:DAC1; 2:DAC2; 3:both;
+    bool stereoOutput = true;
+    const int opcodeCount = 20;
+    const uint32_t dacTableStart1 = 2048 - 512;
+    const uint32_t dacTableStart2 = dacTableStart1 - 512;
+    uint32_t totalSampleWords = 2048 - 512 - 512 - (opcodeCount + 1); // add 512 for mono
+    const int totalSamples = totalSampleWords * 2;
+    const uint32_t indexAddress = opcodeCount;
+    const uint32_t bufferStart = indexAddress + 1;
+};
+
+#else
+
+#error Only the ESP32 supports ULP audio output
+
+#endif
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/AudioStatus.h

@@ -0,0 +1,57 @@
+/*
+  AudioStatus
+  Base class for Audio* status/metadata reporting
+  
+  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/>.
+*/
+
+#ifndef _AUDIOSTATUS_H
+#define _AUDIOSTATUS_H
+
+#include <Arduino.h>
+
+#include "AudioLogger.h"
+
+class AudioStatus
+{
+  public:
+    AudioStatus() { ClearCBs(); };
+    virtual ~AudioStatus() {};
+
+    void ClearCBs() { mdFn = NULL; stFn = NULL; };
+
+    typedef void (*metadataCBFn)(void *cbData, const char *type, bool isUnicode, const char *str);
+    bool RegisterMetadataCB(metadataCBFn f, void *cbData) { mdFn = f; mdData = cbData; return true; }
+
+    // Returns a unique warning/error code, varying by the object.  The string may be a PSTR, use _P functions!
+    typedef void (*statusCBFn)(void *cbData, int code, const char *string);
+    bool RegisterStatusCB(statusCBFn f, void *cbData) { stFn = f; stData = cbData; return true; }
+
+    // Safely call the md function, if defined
+    inline void md(const char *type, bool isUnicode, const char *string) { if (mdFn) mdFn(mdData, type, isUnicode, string); }
+
+    // Safely call the st function, if defined
+    inline void st(int code, const char *string) { if (stFn) stFn(stData, code, string); }
+
+  private:
+    metadataCBFn mdFn;
+    void *mdData;
+    statusCBFn stFn;
+    void *stData;
+};
+
+#endif
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/ESP8266Audio.h

@@ -0,0 +1,50 @@
+// Lazy "include all the things" header for simplicity.
+// In general a user should only include the specific headers they need
+// to miniimize build times.
+
+// Input stage
+#include "AudioFileSourceBuffer.h"
+#include "AudioFileSourceFATFS.h"
+#include "AudioFileSourceFS.h"
+#include "AudioFileSource.h"
+#include "AudioFileSourceHTTPStream.h"
+#include "AudioFileSourceICYStream.h"
+#include "AudioFileSourceID3.h"
+#include "AudioFileSourceLittleFS.h"
+#include "AudioFileSourcePROGMEM.h"
+#include "AudioFileSourceSD.h"
+#include "AudioFileSourceSPIFFS.h"
+#include "AudioFileSourceSPIRAMBuffer.h"
+#include "AudioFileSourceSTDIO.h"
+
+// Misc. plumbing
+#include "AudioFileStream.h"
+#include "AudioLogger.h"
+#include "AudioStatus.h"
+
+// Actual decode/audio generation logic
+#include "AudioGeneratorAAC.h"
+#include "AudioGeneratorFLAC.h"
+#include "AudioGenerator.h"
+#include "AudioGeneratorMIDI.h"
+#include "AudioGeneratorMOD.h"
+#include "AudioGeneratorMP3a.h"
+#include "AudioGeneratorMP3.h"
+#include "AudioGeneratorOpus.h"
+#include "AudioGeneratorRTTTL.h"
+#include "AudioGeneratorTalkie.h"
+#include "AudioGeneratorWAV.h"
+
+// Render(output) sounds
+#include "AudioOutputBuffer.h"
+#include "AudioOutputFilterDecimate.h"
+#include "AudioOutput.h"
+#include "AudioOutputI2S.h"
+#include "AudioOutputI2SNoDAC.h"
+#include "AudioOutputMixer.h"
+#include "AudioOutputNull.h"
+#include "AudioOutputSerialWAV.h"
+#include "AudioOutputSPDIF.h"
+#include "AudioOutputSPIFFSWAV.h"
+#include "AudioOutputSTDIO.h"
+#include "AudioOutputULP.h"

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/driver/SinglePinI2SDriver.cpp

@@ -0,0 +1,283 @@
+/*
+  SinglePinI2SDriver  
+  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.
+  Original code is licensed under LGPL 2.1 or above
+ 
+  Reasons for rewrite:
+  - Non-configurable size of DMA buffers
+  - We only need GPIO3 connected to I2SO_DATA
+  - No API to queue data from ISR. Callbacks are unusable 
+    as i2s_write_* functions are not in IRAM and may re-enable 
+    SLC interrupt before returning
+  - ISR overhead is not needed
+
+  Copyright (C) 2020 Ivan Kostoski
+
+  This library is free software; you can redistribute it and/or
+  modify it under the terms of the GNU Lesser General Public
+  License as published by the Free Software Foundation; either
+  version 2.1 of the License, or (at your option) any later version.
+
+  This library 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
+  Lesser General Public License for more details.
+
+  You should have received a copy of the GNU Lesser General Public
+  License along with this library; if not, write to the Free Software
+  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+*/
+
+#if defined(ESP8266)
+
+#include "SinglePinI2SDriver.h"
+
+SinglePinI2SDriver::~SinglePinI2SDriver()
+{
+  stopI2S();
+  freeBuffers(bufCount);
+}
+
+bool SinglePinI2SDriver::begin(uint8_t bufCount, uint16_t bufSize)
+{
+  if (bufCount < 4) return false;
+  if ((bufSize < 32) || (bufSize > 1020) || ((bufSize % 4) != 0)) return false;
+
+  stopSLC();
+
+  if ((bufCount != this->bufCount) || (bufSize != this->bufSize)) freeBuffers(this->bufCount);
+  this->bufCount = bufCount;
+  this->bufSize = bufSize;
+  if (!allocateBuffers()) return false;
+
+  zeroBuffers();
+  configureSLC();
+  startI2S();
+
+  // Send out at least one buffer, so we have valid address of 
+  // finished descriptor in SLCRXEDA
+  head = &descriptors[0];
+  head->owner = 1;
+  startSLC();
+  while (!lastSentDescriptor()) delayMicroseconds(1);
+  // By here, SLC should be stalled (next descriptor owner is 0)
+  head->owner = 0;
+  head = head->next_link_ptr;
+  headPos = 0;
+  underflowCount = -1;
+  return true;
+};
+
+void SinglePinI2SDriver::stop()
+{
+  stopI2S();
+}
+
+bool SinglePinI2SDriver::allocateBuffers() 
+{
+  // Allocate output (RXLINK) descriptors and bufferss
+  if (descriptors) return true;
+  descriptors = (SLCDecriptor*) calloc(bufCount, sizeof(SLCDecriptor));
+  if (!descriptors) return false;
+  int allocated;
+  for (allocated = 0; allocated < bufCount; allocated++) {
+    uint32_t* buffer = (uint32_t*)malloc(bufSize * sizeof(uint32_t));
+    if (!buffer) break;
+    auto descriptor = &descriptors[allocated];
+    descriptor->eof = 1; // Needed for SLC to update SLCRXEDA
+    descriptor->datalen = bufSize * sizeof(uint32_t);
+    descriptor->blocksize = bufSize * sizeof(uint32_t);
+    descriptor->buf_ptr = buffer;
+    descriptor->next_link_ptr = &descriptors[(allocated+1) % bufCount];
+  }
+  // Release memory if not all buffers were allocated
+  if (allocated < bufCount) {
+    freeBuffers(allocated);
+    return false;
+  }
+  zeroBuffers();
+  return true;
+}
+
+void SinglePinI2SDriver::freeBuffers(int allocated)
+{
+  if (!descriptors) return;
+  while (--allocated >= 0) {
+    if (descriptors[allocated].buf_ptr) {
+      free(descriptors[allocated].buf_ptr);
+      descriptors[allocated].buf_ptr = NULL;
+    }
+  }
+  free(descriptors);
+  descriptors = NULL;
+}
+
+void SinglePinI2SDriver::zeroBuffers()
+{
+  for(int i = 0; i < bufCount; i++) {    
+    auto descriptor = &descriptors[i];
+    ets_memset((void *)descriptor->buf_ptr, 0, bufSize * sizeof(uint32_t));
+  }
+}
+
+inline void SinglePinI2SDriver::advanceHead(const SLCDecriptor *lastSent) 
+{
+  if (headPos >= bufSize) {
+    head->owner = 1; // Owned by SLC
+    head = head->next_link_ptr;
+    head->owner = 0; // Owned by CPU
+    headPos = 0;
+    // If SLC stopped, fill-up the buffers before (re)starting
+    if (isSLCStopped() && (head->next_link_ptr == lastSent)) { 
+      underflowCount++;
+      startSLC();
+    }
+  }
+}
+
+bool SinglePinI2SDriver::write(uint32_t sample) 
+{
+  auto lastSent = lastSentDescriptor();
+  if (isSLCRunning() && (lastSent->next_link_ptr == head)) return false;
+  head->buf_ptr[headPos++] = sample;
+  advanceHead(lastSent);
+  return true;
+};
+
+bool SinglePinI2SDriver::writeInterleaved(uint32_t samples[4]) 
+{
+  auto lastSent = lastSentDescriptor();
+  if (isSLCRunning() && (lastSent->next_link_ptr == head)) return false;
+  uint32_t* buf = head->buf_ptr;
+  buf[headPos++] = samples[1];
+  buf[headPos++] = samples[0];
+  buf[headPos++] = samples[3];
+  buf[headPos++] = samples[2];
+  advanceHead(lastSent);
+  return true;
+};
+
+void SinglePinI2SDriver::configureSLC()
+{
+  ETS_SLC_INTR_DISABLE();
+  // Reset SLC, clear interrupts
+  SLCC0 |= SLCRXLR | SLCTXLR;
+  SLCC0 &= ~(SLCRXLR | SLCTXLR);
+  SLCIC = 0xFFFFFFFF;
+  // Configure SLC DMA in mode 1
+  SLCC0 &= ~(SLCMM << SLCM);
+  SLCC0 |= (1 << SLCM);
+  SLCRXDC |= SLCBINR | SLCBTNR;
+  SLCRXDC &= ~(SLCBRXFE | SLCBRXEM | SLCBRXFM);
+  // RXLINK is output DMA, but TXLINK must be configured with valid descriptr
+  SLCTXL &= ~(SLCTXLAM << SLCTXLA);
+  SLCRXL &= ~(SLCRXLAM << SLCRXLA);
+  SLCTXL |= (uint32)&descriptors[0] << SLCTXLA;
+  SLCRXL |= (uint32)&descriptors[0] << SLCRXLA;
+  // Not setting up interrupts. Not starting DMA here
+}
+
+void SinglePinI2SDriver::stopSLC()
+{
+  // Stop SLC, reset descriptors and clear interrupts
+  ETS_SLC_INTR_DISABLE();
+  SLCTXL |= SLCTXLE;
+  SLCRXL |= SLCRXLE;
+  SLCTXL &= ~(SLCTXLAM << SLCTXLA);
+  SLCRXL &= ~(SLCRXLAM << SLCRXLA);
+  SLCIC = 0xFFFFFFFF;
+}
+
+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
+  I2SC &= ~(I2STSM | I2SRSM | (I2SBMM << I2SBM) | (I2SBDM << I2SBD) | (I2SCDM << I2SCD));
+  // I2SRF = Send/recv right channel first
+  // I2SMR = MSB recv/xmit first
+  // I2SRMS, I2STMS = We don't really care about WS delay here
+  // div1, div2 = Set I2S WS clock frequency. BCLK seems to be generated from 32x this
+  I2SC |= I2SRF | I2SMR | I2SRMS | I2STMS | (div1 << I2SBD) | (div2 << I2SCD);
+}
+
+void SinglePinI2SDriver::setRate(const uint32_t rateHz)
+{
+  if (rateHz == currentRate) return;
+  currentRate = rateHz;
+
+  uint32_t scaledBaseFreq = I2SBASEFREQ / 32;
+  float bestDelta = scaledBaseFreq;
+  uint8_t sbdDivBest=1;
+  uint8_t scdDivBest=1;
+  for (uint8_t i = 1; i < 64; i++) {
+    for (uint8_t j = i; j < 64; j++) {
+      float delta = fabs(((float)scaledBaseFreq/i/j) - rateHz);
+      if (delta < bestDelta){
+      	bestDelta = delta;
+        sbdDivBest = i;
+        scdDivBest = j;
+      }
+    }
+  }
+
+  setDividers(sbdDivBest, scdDivBest);
+}
+
+float SinglePinI2SDriver::getActualRate()
+{
+  return (float)I2SBASEFREQ/32/((I2SC>>I2SBD) & I2SBDM)/((I2SC >> I2SCD) & I2SCDM);
+}
+
+void SinglePinI2SDriver::startI2S()
+{
+  // Setup only I2S SD pin for output
+  pinMode(I2SO_DATA, FUNCTION_1);
+  I2S_CLK_ENABLE();
+  // Reset I2S
+  I2SIC = 0x3F;
+  I2SIE = 0;
+  I2SC &= ~(I2SRST);
+  I2SC |= I2SRST;
+  I2SC &= ~(I2SRST);  
+  // I2STXFMM, I2SRXFMM=0 => 16-bit, dual channel data 
+  I2SFC &= ~(I2SDE | (I2STXFMM << I2STXFM) | (I2SRXFMM << I2SRXFM)); 
+  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
+  currentRate = 0;
+  setRate(44100);
+  // Start I2S peripheral
+  I2SC |= I2STXS; 
+}
+
+void SinglePinI2SDriver::stopI2S()
+{
+  // Disable any I2S send or receive
+  I2SC &= ~(I2STXS | I2SRXS);
+  // Reset I2S
+  I2SC &= ~(I2SRST);
+  I2SC |= I2SRST;
+  I2SC &= ~(I2SRST);
+
+  stopSLC();
+  // Reconnect RX0 ?
+  //pinMode(I2SO_DATA, SPECIAL);
+}
+
+int SinglePinI2SDriver::getUnderflowCount()
+{
+  int count = underflowCount;
+  underflowCount = 0;
+  return count;
+}
+
+// Global instance
+SinglePinI2SDriver I2SDriver;
+
+#endif // defined(ESP8266)

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/driver/SinglePinI2SDriver.h

@@ -0,0 +1,85 @@
+/*
+  SinglePinI2SDriver  
+  ESP8266Audio I2S Minimal driver
+
+  Copyright (C) 2020 Ivan Kostoski
+
+  This library is free software; you can redistribute it and/or
+  modify it under the terms of the GNU Lesser General Public
+  License as published by the Free Software Foundation; either
+  version 2.1 of the License, or (at your option) any later version.
+
+  This library 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
+  Lesser General Public License for more details.
+
+  You should have received a copy of the GNU Lesser General Public
+  License along with this library; if not, write to the Free Software
+  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+*/
+
+#pragma once
+#if defined(ESP8266)
+
+#include <Arduino.h>
+#include "osapi.h"
+#include "ets_sys.h"
+#include "i2s_reg.h"
+
+struct SLCDecriptor {
+  uint32_t          blocksize : 12;
+  uint32_t          datalen   : 12;
+  uint32_t          unused    : 5;
+  uint32_t          sub_sof   : 1;
+  uint32_t          eof       : 1;
+  volatile uint32_t owner     : 1;
+  uint32_t*         buf_ptr;
+  SLCDecriptor* next_link_ptr;
+};
+
+class SinglePinI2SDriver
+{
+  public:
+    enum : uint8_t { I2SO_DATA = 3 }; // GPIO3/RX0
+
+    SinglePinI2SDriver(): descriptors(NULL), bufCount(0), bufSize(0) {};
+    ~SinglePinI2SDriver();
+
+    bool begin(uint8_t bufCount, uint16_t bufSize);
+    void stop();
+    void setRate(const uint32_t rateHz);
+    float getActualRate();
+    bool write(uint32_t sample);
+    bool writeInterleaved(uint32_t samples[4]);
+    int getUnderflowCount();
+
+  protected:
+    bool allocateBuffers();
+    void freeBuffers(int allocated);
+    void zeroBuffers();
+    void advanceHead(const SLCDecriptor *finished);
+    void configureSLC();
+    void stopSLC();
+    void setDividers(uint8_t div1, uint8_t div2);
+    void startI2S();
+    void stopI2S();
+    SLCDecriptor *descriptors;
+    uint16_t bufCount;
+    uint16_t bufSize;
+    SLCDecriptor *head;
+    uint32_t headPos;
+    uint32_t currentRate;
+    int underflowCount;
+
+    // (Re)Start transmission ("TX" DMA always needed to be enabled)
+    static inline void startSLC() { SLCTXL |= SLCTXLS; SLCRXL |= SLCRXLS; }
+    static inline uint32_t isSLCRunning() { return (SLCRXS & SLCRXF); };  
+    static inline uint32_t isSLCStopped() { return (SLCRXS & SLCRXE); }; 
+    static inline SLCDecriptor *lastSentDescriptor() { return (SLCDecriptor*)SLCRXEDA; }
+};
+
+// Global instance
+extern SinglePinI2SDriver I2SDriver;
+
+#endif // defined(ESP8266)

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/AUTHORS

@@ -0,0 +1,58 @@
+/* FLAC - Free Lossless Audio Codec
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * This file is part the FLAC project.  FLAC is comprised of several
+ * components distributed under different licenses.  The codec libraries
+ * are distributed under Xiph.Org's BSD-like license (see the file
+ * COPYING.Xiph in this distribution).  All other programs, libraries, and
+ * plugins are distributed under the GPL (see COPYING.GPL).  The documentation
+ * is distributed under the Gnu FDL (see COPYING.FDL).  Each file in the
+ * FLAC distribution contains at the top the terms under which it may be
+ * distributed.
+ *
+ * Since this particular file is relevant to all components of FLAC,
+ * it may be distributed under the Xiph.Org license, which is the least
+ * restrictive of those mentioned above.  See the file COPYING.Xiph in this
+ * distribution.
+ */
+
+Current FLAC maintainer: Erik de Castro Lopo <erikd@mega-nerd.com>
+
+Original author: Josh Coalson <jcoalson@users.sourceforge.net>
+
+Website : https://www.xiph.org/flac/
+
+FLAC is an Open Source lossless audio codec originally developed by Josh Coalson
+between 2001 and 2009. From 2009 to 2012 FLAC was basically unmaintained. In
+2012 the Erik de Castro Lopo became the chief maintainer as part of the
+Xiph.Org Foundation.
+
+Other major contributors and their contributions:
+
+"lvqcl" <lvqcl@users.sourceforge.net>
+* Visual Studio build system.
+* Optimisations in the encoder and decoder.
+
+"Janne Hyvärinen" <cse@sci.fi>
+* Visual Studio build system.
+* Unicode handling on Windows.
+
+"Andrey Astafiev" <andrei@tvcell.ru>
+* Russian translation of the HTML documentation
+
+"Miroslav Lichvar" <lichvarm@phoenix.inf.upol.cz>
+* IA-32 assembly versions of several libFLAC routines
+
+"Brady Patterson" <bpat@users.sourceforge.net>
+* AIFF file support, PPC assembly versions of libFLAC routines
+
+"Daisuke Shimamura" <Daisuke_Shimamura@nifty.com>
+* i18n support in the XMMS plugin
+
+"X-Fixer" <x-fixer@narod.ru>
+* Configuration system, tag editing, and file info in the Winamp2 plugin
+
+"Matt Zimmerman" <mdz@debian.org>
+* Libtool/autoconf/automake make system, flac man page
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/COPYING.FDL

@@ -0,0 +1,397 @@
+		GNU Free Documentation License
+		  Version 1.2, November 2002
+
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FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/COPYING.GPL

@@ -0,0 +1,339 @@
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FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/COPYING.LGPL

@@ -0,0 +1,504 @@
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+Library), you indicate your acceptance of this License to do so, and
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+
+  10. Each time you redistribute the Library (or any work based on the
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+
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+
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+
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+
+           How to Apply These Terms to Your New Libraries
+
+  If you develop a new library, and you want it to be of the greatest
+possible use to the public, we recommend making it free software that
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+
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+    Copyright (C) <year>  <name of author>
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library 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
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
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+    Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
+
+Also add information on how to contact you by electronic and paper mail.
+
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+
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+
+  <signature of Ty Coon>, 1 April 1990
+  Ty Coon, President of Vice
+
+That's all there is to it!
+
+

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/COPYING.Xiph

@@ -0,0 +1,29 @@
+Copyright (C) 2000-2009  Josh Coalson
+Copyright (C) 2011-2016  Xiph.Org Foundation
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+
+- Redistributions of source code must retain the above copyright
+notice, this list of conditions and the following disclaimer.
+
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+
+- Neither the name of the Xiph.org Foundation nor the names of its
+contributors may be used to endorse or promote products derived from
+this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/FLAC/assert.h

@@ -0,0 +1,46 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__ASSERT_H
+#define FLAC__ASSERT_H
+
+/* we need this since some compilers (like MSVC) leave assert()s on release code (and we don't want to use their ASSERT) */
+#ifndef NDEBUG
+#include <assert.h>
+#define FLAC__ASSERT(x) assert(x)
+#define FLAC__ASSERT_DECLARATION(x) x
+#else
+#define FLAC__ASSERT(x)
+#define FLAC__ASSERT_DECLARATION(x)
+#endif
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/FLAC/callback.h

@@ -0,0 +1,185 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__CALLBACK_H
+#define FLAC__CALLBACK_H
+
+#include "ordinals.h"
+#include <stdlib.h> /* for size_t */
+
+/** \file include/FLAC/callback.h
+ *
+ *  \brief
+ *  This module defines the structures for describing I/O callbacks
+ *  to the other FLAC interfaces.
+ *
+ *  See the detailed documentation for callbacks in the
+ *  \link flac_callbacks callbacks \endlink module.
+ */
+
+/** \defgroup flac_callbacks FLAC/callback.h: I/O callback structures
+ *  \ingroup flac
+ *
+ *  \brief
+ *  This module defines the structures for describing I/O callbacks
+ *  to the other FLAC interfaces.
+ *
+ *  The purpose of the I/O callback functions is to create a common way
+ *  for the metadata interfaces to handle I/O.
+ *
+ *  Originally the metadata interfaces required filenames as the way of
+ *  specifying FLAC files to operate on.  This is problematic in some
+ *  environments so there is an additional option to specify a set of
+ *  callbacks for doing I/O on the FLAC file, instead of the filename.
+ *
+ *  In addition to the callbacks, a FLAC__IOHandle type is defined as an
+ *  opaque structure for a data source.
+ *
+ *  The callback function prototypes are similar (but not identical) to the
+ *  stdio functions fread, fwrite, fseek, ftell, feof, and fclose.  If you use
+ *  stdio streams to implement the callbacks, you can pass fread, fwrite, and
+ *  fclose anywhere a FLAC__IOCallback_Read, FLAC__IOCallback_Write, or
+ *  FLAC__IOCallback_Close is required, and a FILE* anywhere a FLAC__IOHandle
+ *  is required.  \warning You generally CANNOT directly use fseek or ftell
+ *  for FLAC__IOCallback_Seek or FLAC__IOCallback_Tell since on most systems
+ *  these use 32-bit offsets and FLAC requires 64-bit offsets to deal with
+ *  large files.  You will have to find an equivalent function (e.g. ftello),
+ *  or write a wrapper.  The same is true for feof() since this is usually
+ *  implemented as a macro, not as a function whose address can be taken.
+ *
+ * \{
+ */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** This is the opaque handle type used by the callbacks.  Typically
+ *  this is a \c FILE* or address of a file descriptor.
+ */
+typedef void* FLAC__IOHandle;
+
+/** Signature for the read callback.
+ *  The signature and semantics match POSIX fread() implementations
+ *  and can generally be used interchangeably.
+ *
+ * \param  ptr      The address of the read buffer.
+ * \param  size     The size of the records to be read.
+ * \param  nmemb    The number of records to be read.
+ * \param  handle   The handle to the data source.
+ * \retval size_t
+ *    The number of records read.
+ */
+typedef size_t (*FLAC__IOCallback_Read) (void *ptr, size_t size, size_t nmemb, FLAC__IOHandle handle);
+
+/** Signature for the write callback.
+ *  The signature and semantics match POSIX fwrite() implementations
+ *  and can generally be used interchangeably.
+ *
+ * \param  ptr      The address of the write buffer.
+ * \param  size     The size of the records to be written.
+ * \param  nmemb    The number of records to be written.
+ * \param  handle   The handle to the data source.
+ * \retval size_t
+ *    The number of records written.
+ */
+typedef size_t (*FLAC__IOCallback_Write) (const void *ptr, size_t size, size_t nmemb, FLAC__IOHandle handle);
+
+/** Signature for the seek callback.
+ *  The signature and semantics mostly match POSIX fseek() WITH ONE IMPORTANT
+ *  EXCEPTION: the offset is a 64-bit type whereas fseek() is generally 'long'
+ *  and 32-bits wide.
+ *
+ * \param  handle   The handle to the data source.
+ * \param  offset   The new position, relative to \a whence
+ * \param  whence   \c SEEK_SET, \c SEEK_CUR, or \c SEEK_END
+ * \retval int
+ *    \c 0 on success, \c -1 on error.
+ */
+typedef int (*FLAC__IOCallback_Seek) (FLAC__IOHandle handle, FLAC__int64 offset, int whence);
+
+/** Signature for the tell callback.
+ *  The signature and semantics mostly match POSIX ftell() WITH ONE IMPORTANT
+ *  EXCEPTION: the offset is a 64-bit type whereas ftell() is generally 'long'
+ *  and 32-bits wide.
+ *
+ * \param  handle   The handle to the data source.
+ * \retval FLAC__int64
+ *    The current position on success, \c -1 on error.
+ */
+typedef FLAC__int64 (*FLAC__IOCallback_Tell) (FLAC__IOHandle handle);
+
+/** Signature for the EOF callback.
+ *  The signature and semantics mostly match POSIX feof() but WATCHOUT:
+ *  on many systems, feof() is a macro, so in this case a wrapper function
+ *  must be provided instead.
+ *
+ * \param  handle   The handle to the data source.
+ * \retval int
+ *    \c 0 if not at end of file, nonzero if at end of file.
+ */
+typedef int (*FLAC__IOCallback_Eof) (FLAC__IOHandle handle);
+
+/** Signature for the close callback.
+ *  The signature and semantics match POSIX fclose() implementations
+ *  and can generally be used interchangeably.
+ *
+ * \param  handle   The handle to the data source.
+ * \retval int
+ *    \c 0 on success, \c EOF on error.
+ */
+typedef int (*FLAC__IOCallback_Close) (FLAC__IOHandle handle);
+
+/** A structure for holding a set of callbacks.
+ *  Each FLAC interface that requires a FLAC__IOCallbacks structure will
+ *  describe which of the callbacks are required.  The ones that are not
+ *  required may be set to NULL.
+ *
+ *  If the seek requirement for an interface is optional, you can signify that
+ *  a data source is not seekable by setting the \a seek field to \c NULL.
+ */
+typedef struct {
+	FLAC__IOCallback_Read read;
+	FLAC__IOCallback_Write write;
+	FLAC__IOCallback_Seek seek;
+	FLAC__IOCallback_Tell tell;
+	FLAC__IOCallback_Eof eof;
+	FLAC__IOCallback_Close close;
+} FLAC__IOCallbacks;
+
+/* \} */
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/FLAC/export.h

@@ -0,0 +1,97 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__EXPORT_H
+#define FLAC__EXPORT_H
+
+/** \file include/FLAC/export.h
+ *
+ *  \brief
+ *  This module contains \#defines and symbols for exporting function
+ *  calls, and providing version information and compiled-in features.
+ *
+ *  See the \link flac_export export \endlink module.
+ */
+
+/** \defgroup flac_export FLAC/export.h: export symbols
+ *  \ingroup flac
+ *
+ *  \brief
+ *  This module contains \#defines and symbols for exporting function
+ *  calls, and providing version information and compiled-in features.
+ *
+ *  If you are compiling with MSVC and will link to the static library
+ *  (libFLAC.lib) you should define FLAC__NO_DLL in your project to
+ *  make sure the symbols are exported properly.
+ *
+ * \{
+ */
+
+#if defined(FLAC__NO_DLL)
+#define FLAC_API
+
+#elif defined(_WIN32)
+#ifdef FLAC_API_EXPORTS
+#define	FLAC_API __declspec(dllexport)
+#else
+#define FLAC_API __declspec(dllimport)
+#endif
+
+#elif defined(FLAC__USE_VISIBILITY_ATTR)
+#define FLAC_API __attribute__ ((visibility ("default")))
+
+#else
+#define FLAC_API
+
+#endif
+
+/** These \#defines will mirror the libtool-based library version number, see
+ * http://www.gnu.org/software/libtool/manual/libtool.html#Libtool-versioning
+ */
+#define FLAC_API_VERSION_CURRENT 11
+#define FLAC_API_VERSION_REVISION 0 /**< see above */
+#define FLAC_API_VERSION_AGE 3 /**< see above */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** \c 1 if the library has been compiled with support for Ogg FLAC, else \c 0. */
+extern FLAC_API int FLAC_API_SUPPORTS_OGG_FLAC;
+
+#ifdef __cplusplus
+}
+#endif
+
+/* \} */
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/FLAC/ordinals.h

@@ -0,0 +1,85 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__ORDINALS_H
+#define FLAC__ORDINALS_H
+
+#if defined(_MSC_VER) && _MSC_VER < 1600
+
+/* Microsoft Visual Studio earlier than the 2010 version did not provide
+ * the 1999 ISO C Standard header file <stdint.h>.
+ */
+
+typedef signed __int8 FLAC__int8;
+typedef signed __int16 FLAC__int16;
+typedef signed __int32 FLAC__int32;
+typedef signed __int64 FLAC__int64;
+typedef unsigned __int8 FLAC__uint8;
+typedef unsigned __int16 FLAC__uint16;
+typedef unsigned __int32 FLAC__uint32;
+typedef unsigned __int64 FLAC__uint64;
+
+#else
+
+/* For MSVC 2010 and everything else which provides <stdint.h>. */
+
+#include <stdint.h>
+
+typedef int8_t FLAC__int8;
+typedef uint8_t FLAC__uint8;
+
+typedef int16_t FLAC__int16;
+typedef int32_t FLAC__int32;
+typedef int64_t FLAC__int64;
+typedef uint16_t FLAC__uint16;
+typedef uint32_t FLAC__uint32;
+typedef uint64_t FLAC__uint64;
+
+#endif
+
+typedef int FLAC__bool;
+
+typedef FLAC__uint8 FLAC__byte;
+
+
+#ifdef true
+#undef true
+#endif
+#ifdef false
+#undef false
+#endif
+#ifndef __cplusplus
+#define true 1
+#define false 0
+#endif
+
+#endif

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/README

@@ -0,0 +1,254 @@
+/* FLAC - Free Lossless Audio Codec
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * This file is part the FLAC project.  FLAC is comprised of several
+ * components distributed under different licenses.  The codec libraries
+ * are distributed under Xiph.Org's BSD-like license (see the file
+ * COPYING.Xiph in this distribution).  All other programs, libraries, and
+ * plugins are distributed under the LGPL or GPL (see COPYING.LGPL and
+ * COPYING.GPL).  The documentation is distributed under the Gnu FDL (see
+ * COPYING.FDL).  Each file in the FLAC distribution contains at the top the
+ * terms under which it may be distributed.
+ *
+ * Since this particular file is relevant to all components of FLAC,
+ * it may be distributed under the Xiph.Org license, which is the least
+ * restrictive of those mentioned above.  See the file COPYING.Xiph in this
+ * distribution.
+ */
+
+
+FLAC is an Open Source lossless audio codec developed by Josh Coalson from 2001
+to 2009.
+
+From January 2012 FLAC is being maintained by Erik de Castro Lopo under the
+auspices of the Xiph.org Foundation.
+
+FLAC is comprised of
+  * `libFLAC', a library which implements reference encoders and
+    decoders for native FLAC and Ogg FLAC, and a metadata interface
+  * `libFLAC++', a C++ object wrapper library around libFLAC
+  * `flac', a command-line program for encoding and decoding files
+  * `metaflac', a command-line program for viewing and editing FLAC
+    metadata
+  * player plugin for XMMS
+  * user and API documentation
+
+The libraries (libFLAC, libFLAC++) are
+licensed under Xiph.org's BSD-like license (see COPYING.Xiph).  All other
+programs and plugins are licensed under the GNU General Public License
+(see COPYING.GPL).  The documentation is licensed under the GNU Free
+Documentation License (see COPYING.FDL).
+
+
+===============================================================================
+FLAC - 1.3.2 - Contents
+===============================================================================
+
+- Introduction
+- Prerequisites
+- Note to embedded developers
+- Building in a GNU environment
+- Building with Makefile.lite
+- Building with MSVC
+- Building on Mac OS X
+
+
+===============================================================================
+Introduction
+===============================================================================
+
+This is the source release for the FLAC project.  See
+
+	doc/html/index.html
+
+for full documentation.
+
+A brief description of the directory tree:
+
+	doc/          the HTML documentation
+	examples/     example programs demonstrating the use of libFLAC and libFLAC++
+	include/      public include files for libFLAC and libFLAC++
+	man/          the man pages for `flac' and `metaflac'
+	src/          the source code and private headers
+	test/         the test scripts
+
+If you have questions about building FLAC that this document does not answer,
+please submit them at the following tracker so this document can be improved:
+
+	https://sourceforge.net/p/flac/support-requests/
+
+
+===============================================================================
+Prerequisites
+===============================================================================
+
+To build FLAC with support for Ogg FLAC you must have built and installed
+libogg according to the specific instructions below.  You must have
+libogg 1.1.2 or greater, or there will be seeking problems with Ogg FLAC.
+
+If you are building on x86 and want the assembly optimizations, you will
+need to have NASM >= 0.98.30 installed according to the specific instructions
+below.
+
+
+===============================================================================
+Note to embedded developers
+===============================================================================
+
+libFLAC has grown larger over time as more functionality has been
+included, but much of it may be unnecessary for a particular embedded
+implementation.  Unused parts may be pruned by some simple editing of
+configure.ac and src/libFLAC/Makefile.am; the following dependency
+graph shows which modules may be pruned without breaking things
+further down:
+
+metadata.h
+	stream_decoder.h
+	format.h
+
+stream_encoder.h
+	stream_decoder.h
+	format.h
+
+stream_decoder.h
+	format.h
+
+In other words, for pure decoding applications, both the stream encoder
+and metadata editing interfaces can be safely removed.
+
+There is a section dedicated to embedded use in the libFLAC API
+HTML documentation (see doc/html/api/index.html).
+
+Also, there are several places in the libFLAC code with comments marked
+with "OPT:" where a #define can be changed to enable code that might be
+faster on a specific platform.  Experimenting with these can yield faster
+binaries.
+
+
+===============================================================================
+Building in a GNU environment
+===============================================================================
+
+FLAC uses autoconf and libtool for configuring and building.
+Better documentation for these will be forthcoming, but in
+general, this should work:
+
+./configure && make && make check && make install
+
+The 'make check' step is optional; omit it to skip all the tests,
+which can take several hours and use around 70-80 megs of disk space.
+Even though it will stop with an explicit message on any failure, it
+does print out a lot of stuff so you might want to capture the output
+to a file if you're having a problem.  Also, don't run 'make check'
+as root because it confuses some of the tests.
+
+NOTE: Despite our best efforts it's entirely possible to have
+problems when using older versions of autoconf, automake, or
+libtool.  If you have the latest versions and still can't get it
+to work, see the next section on Makefile.lite.
+
+There are a few FLAC-specific arguments you can give to
+`configure':
+
+--enable-debug : Builds everything with debug symbols and some
+extra (and more verbose) error checking.
+
+--disable-asm-optimizations : Disables the compilation of the
+assembly routines.  Many routines have assembly versions for
+speed and `configure' is pretty good about knowing what is
+supported, but you can use this option to build only from the
+C sources.  May be necessary for building on OS X (Intel).
+
+--enable-sse : If you are building for an x86 CPU that supports
+SSE instructions, you can enable some of the faster routines
+if your operating system also supports SSE instructions.  flac
+can tell if the CPU supports the instructions but currently has
+no way to test if the OS does, so if it does, you must pass
+this argument to configure to use the SSE routines.  If flac
+crashes when built with this option you will have to go back and
+configure without --enable-sse.  Note that
+--disable-asm-optimizations implies --disable-sse.
+
+--enable-local-xmms-plugin : Installs the FLAC XMMS plugin in
+$HOME/.xmms/Plugins, instead of the global XMMS plugin area
+(usually /usr/lib/xmms/Input).
+
+--with-ogg=
+--with-xmms-prefix=
+--with-libiconv-prefix=
+Use these if you have these packages but configure can't find them.
+
+If you want to build completely from scratch (i.e. starting with just
+configure.ac and Makefile.am) you should be able to just run 'autogen.sh'
+but make sure and read the comments in that file first.
+
+
+===============================================================================
+Building with Makefile.lite
+===============================================================================
+
+There is a more lightweight build system for do-it-yourself-ers.
+It is also useful if configure isn't working, which may be the
+case since lately we've had some problems with different versions
+of automake and libtool.  The Makefile.lite system should work
+on GNU systems with few or no adjustments.
+
+From the top level just 'make -f Makefile.lite'.  You can
+specify zero or one optional target from 'release', 'debug',
+'test', or 'clean'.  The default is 'release'.  There is no
+'install' target but everything you need will end up in the
+obj/ directory.
+
+If you are not on an x86 system or you don't have nasm, you
+may have to change the DEFINES in src/libFLAC/Makefile.lite.  If
+you don't have nasm, remove -DFLAC__HAS_NASM.  If your target is
+not an x86, change -DFLAC__CPU_IA32 to -DFLAC__CPU_UNKNOWN.
+
+
+===============================================================================
+Building with MSVC
+===============================================================================
+
+There are .vcproj projects and a master FLAC.sln solution to build all
+the libraries and executables with MSVC 2005 or newer.
+
+Prerequisite: you must have the Ogg libraries installed as described
+later.
+
+Prerequisite: you must have nasm installed, and nasm.exe must be in
+your PATH, or the path to nasm.exe must be added to the list of
+directories for executable files in the MSVC global options.
+
+To build everything, run Visual Studio, do File|Open and open FLAC.sln.
+From the dropdown in the toolbar, select "Release" instead of "Debug",
+then do Build|Build Solution.
+
+This will build all libraries both statically (e.g.
+objs\release\lib\libFLAC_static.lib) and as DLLs (e.g.
+objs\release\lib\libFLAC.dll), and it will build all binaries, statically
+linked (e.g. objs\release\bin\flac.exe).
+
+Everything will end up in the "objs" directory.  DLLs and .exe files
+are all that are needed and can be copied to an installation area and
+added to the PATH.
+
+By default the code is configured with Ogg support. Before building FLAC
+you will need to get the Ogg source distribution
+(see http://xiph.org/downloads/), build libogg_static.lib (load
+win32\libogg_static.sln, change solution configuration to "Release" and
+code generation to "Multi-threaded (/MT)", then build), copy libogg_static.lib
+into FLAC's 'objs\release\lib' directory, and copy the entire include\ogg tree
+into FLAC's 'include' directory (so that there is an 'ogg' directory in FLAC's
+'include' directory with the files ogg.h, os_types.h and config_types.h).
+
+If you want to build without Ogg support, instead edit all .vcproj files
+and remove any "FLAC__HAS_OGG" definitions.
+
+
+===============================================================================
+Building on Mac OS X
+===============================================================================
+
+If you have Fink or a recent version of OS X with the proper autotools,
+the GNU flow above should work.

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/README.ESP8266

@@ -0,0 +1,5 @@
+This is LIBFLAC 1.3.2 ported to the ESP8266 by Earle F. Philhower, III
+<earlephilhower@yahoo.com>.  It's pretty much unchanged from base sourcew,
+with some CRC and string constants moved into PROGMEM.  Headers have been
+purged of references to STDIO/FILE*.  Files not needed for playback have
+been removed.

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/bitmath.c

@@ -0,0 +1,75 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+//#ifdef HAVE_CONFIG_H
+#  include "config.h"
+//#endif
+
+#pragma GCC optimize ("O3")
+
+#include "private/bitmath.h"
+
+/* An example of what FLAC__bitmath_silog2() computes:
+ *
+ * silog2(-10) = 5
+ * silog2(- 9) = 5
+ * silog2(- 8) = 4
+ * silog2(- 7) = 4
+ * silog2(- 6) = 4
+ * silog2(- 5) = 4
+ * silog2(- 4) = 3
+ * silog2(- 3) = 3
+ * silog2(- 2) = 2
+ * silog2(- 1) = 2
+ * silog2(  0) = 0
+ * silog2(  1) = 2
+ * silog2(  2) = 3
+ * silog2(  3) = 3
+ * silog2(  4) = 4
+ * silog2(  5) = 4
+ * silog2(  6) = 4
+ * silog2(  7) = 4
+ * silog2(  8) = 5
+ * silog2(  9) = 5
+ * silog2( 10) = 5
+ */
+uint32_t FLAC__bitmath_silog2(FLAC__int64 v)
+{
+    if(v == 0)
+		return 0;
+
+	if(v == -1)
+		return 2;
+
+	v = (v < 0) ? (-(v+1)) : v;
+	return FLAC__bitmath_ilog2_wide(v)+2;
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/config.h

@@ -0,0 +1,248 @@
+#define PGM_READ_UNALIGNED 0
+
+#ifdef DEBUG
+  #undef NDEBUG
+#else
+  #define NDEBUG
+#endif
+
+/* config.h.  Generated from config.h.in by configure.  */
+/* config.h.in.  Generated from configure.ac by autoheader.  */
+
+/* Define if building universal (internal helper macro) */
+/* #undef AC_APPLE_UNIVERSAL_BUILD */
+
+/* Target processor is big endian. */
+#define CPU_IS_BIG_ENDIAN 0
+
+/* Target processor is little endian. */
+#define CPU_IS_LITTLE_ENDIAN 1
+
+/* Set FLAC__BYTES_PER_WORD to 8 (4 is the default) */
+#define ENABLE_64_BIT_WORDS 0
+
+/* define to align allocated memory on 32-byte boundaries */
+#define FLAC__ALIGN_MALLOC_DATA 1
+
+/* define if building for ia32/i386 */
+/* #undef FLAC__CPU_IA32 */
+
+/* define if building for PowerPC */
+/* #undef FLAC__CPU_PPC */
+
+/* define if building for PowerPC with SPE ABI */
+/* #undef FLAC__CPU_PPC_SPE */
+
+/* define if building for SPARC */
+/* #undef FLAC__CPU_SPARC */
+
+/* define if building for x86_64 */
+#undef FLAC__CPU_X86_64
+
+/* define if you have docbook-to-man or docbook2man */
+#undef FLAC__HAS_DOCBOOK_TO_MAN
+
+/* define if you are compiling for x86 and have the NASM assembler */
+#undef FLAC__HAS_NASM
+
+/* define if you have the ogg library */
+#define FLAC__HAS_OGG 0
+
+/* Set to 1 if <x86intrin.h> is available. */
+#undef FLAC__HAS_X86INTRIN
+
+/* define to disable use of assembly code */
+#define FLAC__NO_ASM 1
+
+/* define if building for Darwin / MacOS X */
+/* #undef FLAC__SYS_DARWIN */
+
+/* define if building for Linux */
+#undef FLAC__SYS_LINUX
+
+/* define to enable use of Altivec instructions */
+#undef FLAC__USE_ALTIVEC
+
+/* define to enable use of AVX instructions */
+#undef FLAC__USE_AVX
+
+/* Compiler has the __builtin_bswap16 intrinsic */
+#undef HAVE_BSWAP16
+
+/* Compiler has the __builtin_bswap32 intrinsic */
+#undef HAVE_BSWAP32
+
+/* Define to 1 if you have the <byteswap.h> header file. */
+#undef HAVE_BYTESWAP_H
+
+/* define if you have clock_gettime */
+#undef HAVE_CLOCK_GETTIME
+
+/* Define to 1 if you have the <cpuid.h> header file. */
+#undef HAVE_CPUID_H
+
+/* Define to 1 if C++ supports variable-length arrays. */
+#define HAVE_CXX_VARARRAYS 1
+
+/* Define to 1 if C supports variable-length arrays. */
+#define HAVE_C_VARARRAYS 1
+
+/* Define to 1 if you have the <dlfcn.h> header file. */
+#undef HAVE_DLFCN_H
+
+/* Define to 1 if fseeko (and presumably ftello) exists and is declared. */
+#undef HAVE_FSEEKO
+
+/* Define to 1 if you have the `getopt_long' function. */
+#undef HAVE_GETOPT_LONG
+
+/* Define if you have the iconv() function and it works. */
+#undef HAVE_ICONV
+
+/* Define to 1 if you have the <inttypes.h> header file. */
+#undef HAVE_INTTYPES_H
+
+/* Define if you have <langinfo.h> and nl_langinfo(CODESET). */
+#define HAVE_LANGINFO_CODESET 1
+
+/* lround support */
+#define HAVE_LROUND 1
+
+/* Define to 1 if you have the <memory.h> header file. */
+#undef HAVE_MEMORY_H
+
+/* Define to 1 if the system has the type `socklen_t'. */
+#undef HAVE_SOCKLEN_T
+
+/* Define to 1 if you have the <stdint.h> header file. */
+#define HAVE_STDINT_H 1
+
+/* Define to 1 if you have the <stdlib.h> header file. */
+#undef HAVE_STDLIB_H
+
+/* Define to 1 if you have the <strings.h> header file. */
+#define HAVE_STRINGS_H 1
+
+/* Define to 1 if you have the <string.h> header file. */
+#define HAVE_STRING_H 1
+
+/* Define to 1 if you have the <sys/ioctl.h> header file. */
+#undef HAVE_SYS_IOCTL_H
+
+/* Define to 1 if you have the <sys/param.h> header file. */
+#undef HAVE_SYS_PARAM_H
+
+/* Define to 1 if you have the <sys/stat.h> header file. */
+#undef HAVE_SYS_STAT_H
+
+/* Define to 1 if you have the <sys/types.h> header file. */
+#undef HAVE_SYS_TYPES_H
+
+/* Define to 1 if you have the <termios.h> header file. */
+#undef HAVE_TERMIOS_H
+
+/* Define to 1 if typeof works with your compiler. */
+#define HAVE_TYPEOF 1
+
+/* Define to 1 if you have the <unistd.h> header file. */
+#undef HAVE_UNISTD_H
+
+/* Define to 1 if you have the <x86intrin.h> header file. */
+#undef HAVE_X86INTRIN_H
+
+/* Define as const if the declaration of iconv() needs const. */
+#define ICONV_CONST 
+
+/* Define to the sub-directory where libtool stores uninstalled libraries. */
+#define LT_OBJDIR ".libs/"
+
+/* Name of package */
+#define PACKAGE "flac"
+
+/* Define to the address where bug reports for this package should be sent. */
+#define PACKAGE_BUGREPORT "flac-dev@xiph.org"
+
+/* Define to the full name of this package. */
+#define PACKAGE_NAME "flac"
+
+/* Define to the full name and version of this package. */
+#define PACKAGE_STRING "flac 1.3.2"
+
+/* Define to the one symbol short name of this package. */
+#define PACKAGE_TARNAME "flac"
+
+/* Define to the home page for this package. */
+#define PACKAGE_URL "https://www.xiph.org/flac/"
+
+/* Define to the version of this package. */
+#define PACKAGE_VERSION "1.3.2"
+
+/* The size of `off_t', as computed by sizeof. */
+#define SIZEOF_OFF_T 4
+
+/* The size of `void*', as computed by sizeof. */
+#define SIZEOF_VOIDP 4
+
+/* Define to 1 if you have the ANSI C header files. */
+#define STDC_HEADERS 1
+
+/* Enable extensions on AIX 3, Interix.  */
+#ifndef _ALL_SOURCE
+# define _ALL_SOURCE 1
+#endif
+/* Enable GNU extensions on systems that have them.  */
+#ifndef _GNU_SOURCE
+# define _GNU_SOURCE 1
+#endif
+/* Enable threading extensions on Solaris.  */
+#ifndef _POSIX_PTHREAD_SEMANTICS
+# define _POSIX_PTHREAD_SEMANTICS 1
+#endif
+/* Enable extensions on HP NonStop.  */
+#ifndef _TANDEM_SOURCE
+# define _TANDEM_SOURCE 1
+#endif
+/* Enable general extensions on Solaris.  */
+#ifndef __EXTENSIONS__
+# define __EXTENSIONS__ 1
+#endif
+
+
+/* Version number of package */
+#define VERSION "1.3.2"
+
+/* Target processor is big endian. */
+#define WORDS_BIGENDIAN 0
+
+/* Enable large inode numbers on Mac OS X 10.5.  */
+#ifndef _DARWIN_USE_64_BIT_INODE
+# define _DARWIN_USE_64_BIT_INODE 1
+#endif
+
+/* Number of bits in a file offset, on hosts where this is settable. */
+/* #undef _FILE_OFFSET_BITS */
+
+/* Define to 1 to make fseeko visible on some hosts (e.g. glibc 2.2). */
+/* #undef _LARGEFILE_SOURCE */
+
+/* Define for large files, on AIX-style hosts. */
+/* #undef _LARGE_FILES */
+
+/* Define to 1 if on MINIX. */
+/* #undef _MINIX */
+
+/* Define to 2 if the system does not provide POSIX.1 features except with
+   this defined. */
+/* #undef _POSIX_1_SOURCE */
+
+/* Define to 1 if you need to in order for `stat' and other things to work. */
+/* #undef _POSIX_SOURCE */
+
+/* Define to `__inline__' or `__inline' if that's what the C compiler
+   calls it, or to nothing if 'inline' is not supported under any name.  */
+#ifndef __cplusplus
+/* #undef inline */
+#endif
+
+/* Define to __typeof__ if your compiler spells it that way. */
+/* #undef typeof */

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/cpu.c

@@ -0,0 +1,299 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+//#ifdef HAVE_CONFIG_H
+#  include "config.h"
+//#endif
+
+#include "private/cpu.h"
+#include "share/compat.h"
+#include <stdlib.h>
+#include <string.h>
+
+#if defined _MSC_VER
+#include <intrin.h> /* for __cpuid() and _xgetbv() */
+#elif defined __GNUC__ && defined HAVE_CPUID_H
+#include <cpuid.h> /* for __get_cpuid() and __get_cpuid_max() */
+#endif
+
+#ifndef NDEBUG
+//#include <stdio.h>
+#define dfprintf fprintf
+#else
+/* This is bad practice, it should be a static void empty function */
+#define dfprintf(file, format, ...)
+#endif
+
+#pragma GCC optimize ("O3")
+#if defined FLAC__CPU_PPC
+#include <sys/auxv.h>
+#endif
+
+#if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && (defined FLAC__HAS_NASM || FLAC__HAS_X86INTRIN) && !defined FLAC__NO_ASM
+
+/* these are flags in EDX of CPUID AX=00000001 */
+static const uint32_t FLAC__CPUINFO_X86_CPUID_CMOV    = 0x00008000;
+static const uint32_t FLAC__CPUINFO_X86_CPUID_MMX     = 0x00800000;
+static const uint32_t FLAC__CPUINFO_X86_CPUID_SSE     = 0x02000000;
+static const uint32_t FLAC__CPUINFO_X86_CPUID_SSE2    = 0x04000000;
+
+/* these are flags in ECX of CPUID AX=00000001 */
+static const uint32_t FLAC__CPUINFO_X86_CPUID_SSE3    = 0x00000001;
+static const uint32_t FLAC__CPUINFO_X86_CPUID_SSSE3   = 0x00000200;
+static const uint32_t FLAC__CPUINFO_X86_CPUID_SSE41   = 0x00080000;
+static const uint32_t FLAC__CPUINFO_X86_CPUID_SSE42   = 0x00100000;
+static const uint32_t FLAC__CPUINFO_X86_CPUID_OSXSAVE = 0x08000000;
+static const uint32_t FLAC__CPUINFO_X86_CPUID_AVX     = 0x10000000;
+static const uint32_t FLAC__CPUINFO_X86_CPUID_FMA     = 0x00001000;
+
+/* these are flags in EBX of CPUID AX=00000007 */
+static const uint32_t FLAC__CPUINFO_X86_CPUID_AVX2    = 0x00000020;
+
+static uint32_t
+cpu_xgetbv_x86(void)
+{
+#if (defined _MSC_VER || defined __INTEL_COMPILER) && FLAC__AVX_SUPPORTED
+	return (uint32_t)_xgetbv(0);
+#elif defined __GNUC__
+	uint32_t lo, hi;
+	__asm__ volatile (".byte 0x0f, 0x01, 0xd0" : "=a"(lo), "=d"(hi) : "c" (0));
+	return lo;
+#else
+	return 0;
+#endif
+}
+
+static uint32_t
+cpu_have_cpuid(void)
+{
+#if defined FLAC__CPU_X86_64 || defined __i686__ || defined __SSE__ || (defined _M_IX86_FP && _M_IX86_FP > 0)
+	/* target CPU does have CPUID instruction */
+	return 1;
+#elif defined FLAC__HAS_NASM
+	return FLAC__cpu_have_cpuid_asm_ia32();
+#elif defined __GNUC__ && defined HAVE_CPUID_H
+	if (__get_cpuid_max(0, 0) != 0)
+		return 1;
+	else
+		return 0;
+#elif defined _MSC_VER
+	FLAC__uint32 flags1, flags2;
+	__asm {
+		pushfd
+		pushfd
+		pop		eax
+		mov		flags1, eax
+		xor		eax, 0x200000
+		push	eax
+		popfd
+		pushfd
+		pop		eax
+		mov		flags2, eax
+		popfd
+	}
+	if (((flags1^flags2) & 0x200000) != 0)
+		return 1;
+	else
+		return 0;
+#else
+	return 0;
+#endif
+}
+
+static void
+cpuinfo_x86(FLAC__uint32 level, FLAC__uint32 *eax, FLAC__uint32 *ebx, FLAC__uint32 *ecx, FLAC__uint32 *edx)
+{
+#if defined _MSC_VER
+	int cpuinfo[4];
+	int ext = level & 0x80000000;
+	__cpuid(cpuinfo, ext);
+	if ((uint32_t)cpuinfo[0] >= level) {
+#if FLAC__AVX_SUPPORTED
+		__cpuidex(cpuinfo, level, 0); /* for AVX2 detection */
+#else
+		__cpuid(cpuinfo, level); /* some old compilers don't support __cpuidex */
+#endif
+		*eax = cpuinfo[0]; *ebx = cpuinfo[1]; *ecx = cpuinfo[2]; *edx = cpuinfo[3];
+		return;
+	}
+#elif defined __GNUC__ && defined HAVE_CPUID_H
+	FLAC__uint32 ext = level & 0x80000000;
+	__cpuid(ext, *eax, *ebx, *ecx, *edx);
+	if (*eax >= level) {
+		__cpuid_count(level, 0, *eax, *ebx, *ecx, *edx);
+		return;
+	}
+#elif defined FLAC__HAS_NASM && defined FLAC__CPU_IA32
+	FLAC__cpu_info_asm_ia32(level, eax, ebx, ecx, edx);
+	return;
+#endif
+	*eax = *ebx = *ecx = *edx = 0;
+}
+
+#endif
+
+static void
+x86_cpu_info (FLAC__CPUInfo *info)
+{
+#if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && (defined FLAC__HAS_NASM || FLAC__HAS_X86INTRIN) && !defined FLAC__NO_ASM
+	FLAC__bool x86_osxsave = false;
+	FLAC__bool os_avx = false;
+	FLAC__uint32 flags_eax, flags_ebx, flags_ecx, flags_edx;
+
+	info->use_asm = true; /* we assume a minimum of 80386 */
+	if (!cpu_have_cpuid())
+		return;
+
+	cpuinfo_x86(0, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+	info->x86.intel = (flags_ebx == 0x756E6547 && flags_edx == 0x49656E69 && flags_ecx == 0x6C65746E) ? true : false; /* GenuineIntel */
+	cpuinfo_x86(1, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+
+	info->x86.cmov  = (flags_edx & FLAC__CPUINFO_X86_CPUID_CMOV ) ? true : false;
+	info->x86.mmx   = (flags_edx & FLAC__CPUINFO_X86_CPUID_MMX  ) ? true : false;
+	info->x86.sse   = (flags_edx & FLAC__CPUINFO_X86_CPUID_SSE  ) ? true : false;
+	info->x86.sse2  = (flags_edx & FLAC__CPUINFO_X86_CPUID_SSE2 ) ? true : false;
+	info->x86.sse3  = (flags_ecx & FLAC__CPUINFO_X86_CPUID_SSE3 ) ? true : false;
+	info->x86.ssse3 = (flags_ecx & FLAC__CPUINFO_X86_CPUID_SSSE3) ? true : false;
+	info->x86.sse41 = (flags_ecx & FLAC__CPUINFO_X86_CPUID_SSE41) ? true : false;
+	info->x86.sse42 = (flags_ecx & FLAC__CPUINFO_X86_CPUID_SSE42) ? true : false;
+
+	if (FLAC__AVX_SUPPORTED) {
+		x86_osxsave     = (flags_ecx & FLAC__CPUINFO_X86_CPUID_OSXSAVE) ? true : false;
+		info->x86.avx   = (flags_ecx & FLAC__CPUINFO_X86_CPUID_AVX    ) ? true : false;
+		info->x86.fma   = (flags_ecx & FLAC__CPUINFO_X86_CPUID_FMA    ) ? true : false;
+		cpuinfo_x86(7, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+		info->x86.avx2  = (flags_ebx & FLAC__CPUINFO_X86_CPUID_AVX2   ) ? true : false;
+	}
+
+#if defined FLAC__CPU_IA32
+	dfprintf(stderr, "CPU info (IA-32):\n");
+#else
+	dfprintf(stderr, "CPU info (x86-64):\n");
+#endif
+	dfprintf(stderr, "  CMOV ....... %c\n", info->x86.cmov    ? 'Y' : 'n');
+	dfprintf(stderr, "  MMX ........ %c\n", info->x86.mmx     ? 'Y' : 'n');
+	dfprintf(stderr, "  SSE ........ %c\n", info->x86.sse     ? 'Y' : 'n');
+	dfprintf(stderr, "  SSE2 ....... %c\n", info->x86.sse2    ? 'Y' : 'n');
+	dfprintf(stderr, "  SSE3 ....... %c\n", info->x86.sse3    ? 'Y' : 'n');
+	dfprintf(stderr, "  SSSE3 ...... %c\n", info->x86.ssse3   ? 'Y' : 'n');
+	dfprintf(stderr, "  SSE41 ...... %c\n", info->x86.sse41   ? 'Y' : 'n');
+	dfprintf(stderr, "  SSE42 ...... %c\n", info->x86.sse42   ? 'Y' : 'n');
+
+	if (FLAC__AVX_SUPPORTED) {
+		dfprintf(stderr, "  AVX ........ %c\n", info->x86.avx     ? 'Y' : 'n');
+		dfprintf(stderr, "  FMA ........ %c\n", info->x86.fma     ? 'Y' : 'n');
+		dfprintf(stderr, "  AVX2 ....... %c\n", info->x86.avx2    ? 'Y' : 'n');
+	}
+
+	/*
+	 * now have to check for OS support of AVX instructions
+	 */
+	if (FLAC__AVX_SUPPORTED && info->x86.avx && x86_osxsave && (cpu_xgetbv_x86() & 0x6) == 0x6) {
+		os_avx = true;
+	}
+	if (os_avx) {
+		dfprintf(stderr, "  AVX OS sup . %c\n", info->x86.avx ? 'Y' : 'n');
+	}
+	if (!os_avx) {
+		/* no OS AVX support */
+		info->x86.avx     = false;
+		info->x86.avx2    = false;
+		info->x86.fma     = false;
+	}
+#else
+	info->use_asm = false;
+#endif
+}
+
+static void
+ppc_cpu_info (FLAC__CPUInfo *info)
+{
+#if defined FLAC__CPU_PPC
+#ifndef PPC_FEATURE2_ARCH_3_00
+#define PPC_FEATURE2_ARCH_3_00		0x00800000
+#endif
+
+#ifndef PPC_FEATURE2_ARCH_2_07
+#define PPC_FEATURE2_ARCH_2_07		0x80000000
+#endif
+
+#ifdef __linux__
+	if (getauxval(AT_HWCAP2) & PPC_FEATURE2_ARCH_3_00) {
+		info->ppc.arch_3_00 = true;
+	} else if (getauxval(AT_HWCAP2) & PPC_FEATURE2_ARCH_2_07) {
+		info->ppc.arch_2_07 = true;
+	}
+#elif defined(__FreeBSD__)
+	long hwcaps;
+	elf_aux_info(AT_HWCAP2, &hwcaps, sizeof(hwcaps));
+#else
+#error Unsupported platform! Please add support for reading ppc hwcaps.
+#endif
+
+	if (hwcaps & PPC_FEATURE2_ARCH_3_00) {
+		info->ppc.arch_3_00 = true;
+	} else if (hwcaps & PPC_FEATURE2_ARCH_2_07) {
+		info->ppc.arch_2_07 = true;
+	}
+#else
+	info->ppc.arch_2_07 = false;
+	info->ppc.arch_3_00 = false;
+#endif
+}
+
+void FLAC__cpu_info (FLAC__CPUInfo *info)
+{
+	memset(info, 0, sizeof(*info));
+
+#ifdef FLAC__CPU_IA32
+	info->type = FLAC__CPUINFO_TYPE_IA32;
+#elif defined FLAC__CPU_X86_64
+	info->type = FLAC__CPUINFO_TYPE_X86_64;
+#elif defined FLAC__CPU_PPC
+	info->type = FLAC__CPUINFO_TYPE_PPC;
+#else
+	info->type = FLAC__CPUINFO_TYPE_UNKNOWN;
+#endif
+
+	switch (info->type) {
+	case FLAC__CPUINFO_TYPE_IA32: /* fallthrough */
+	case FLAC__CPUINFO_TYPE_X86_64:
+		x86_cpu_info (info);
+		break;
+	case FLAC__CPUINFO_TYPE_PPC:
+		ppc_cpu_info (info);
+		break;
+	default:
+		info->use_asm = false;
+		break;
+	}
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/crc.c

@@ -0,0 +1,150 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+//#ifdef HAVE_CONFIG_H
+#  include "config.h"
+//#endif
+#define PGM_READ_UNALIGNED 0
+#include <Arduino.h>
+#include <pgmspace.h>
+
+#include "private/crc.h"
+
+#pragma GCC optimize ("O3")
+
+/* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */
+
+FLAC__byte const FLAC__crc8_table[256] PROGMEM = {
+	0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15,
+	0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D,
+	0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65,
+	0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D,
+	0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5,
+	0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD,
+	0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85,
+	0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD,
+	0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2,
+	0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA,
+	0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2,
+	0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A,
+	0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32,
+	0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A,
+	0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42,
+	0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A,
+	0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C,
+	0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4,
+	0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC,
+	0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4,
+	0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C,
+	0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44,
+	0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C,
+	0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34,
+	0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B,
+	0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63,
+	0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B,
+	0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13,
+	0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB,
+	0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83,
+	0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB,
+	0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3
+};
+
+/* CRC-16, poly = x^16 + x^15 + x^2 + x^0, init = 0 */
+
+unsigned const FLAC__crc16_table[256] PROGMEM = {
+	0x0000,  0x8005,  0x800f,  0x000a,  0x801b,  0x001e,  0x0014,  0x8011,
+	0x8033,  0x0036,  0x003c,  0x8039,  0x0028,  0x802d,  0x8027,  0x0022,
+	0x8063,  0x0066,  0x006c,  0x8069,  0x0078,  0x807d,  0x8077,  0x0072,
+	0x0050,  0x8055,  0x805f,  0x005a,  0x804b,  0x004e,  0x0044,  0x8041,
+	0x80c3,  0x00c6,  0x00cc,  0x80c9,  0x00d8,  0x80dd,  0x80d7,  0x00d2,
+	0x00f0,  0x80f5,  0x80ff,  0x00fa,  0x80eb,  0x00ee,  0x00e4,  0x80e1,
+	0x00a0,  0x80a5,  0x80af,  0x00aa,  0x80bb,  0x00be,  0x00b4,  0x80b1,
+	0x8093,  0x0096,  0x009c,  0x8099,  0x0088,  0x808d,  0x8087,  0x0082,
+	0x8183,  0x0186,  0x018c,  0x8189,  0x0198,  0x819d,  0x8197,  0x0192,
+	0x01b0,  0x81b5,  0x81bf,  0x01ba,  0x81ab,  0x01ae,  0x01a4,  0x81a1,
+	0x01e0,  0x81e5,  0x81ef,  0x01ea,  0x81fb,  0x01fe,  0x01f4,  0x81f1,
+	0x81d3,  0x01d6,  0x01dc,  0x81d9,  0x01c8,  0x81cd,  0x81c7,  0x01c2,
+	0x0140,  0x8145,  0x814f,  0x014a,  0x815b,  0x015e,  0x0154,  0x8151,
+	0x8173,  0x0176,  0x017c,  0x8179,  0x0168,  0x816d,  0x8167,  0x0162,
+	0x8123,  0x0126,  0x012c,  0x8129,  0x0138,  0x813d,  0x8137,  0x0132,
+	0x0110,  0x8115,  0x811f,  0x011a,  0x810b,  0x010e,  0x0104,  0x8101,
+	0x8303,  0x0306,  0x030c,  0x8309,  0x0318,  0x831d,  0x8317,  0x0312,
+	0x0330,  0x8335,  0x833f,  0x033a,  0x832b,  0x032e,  0x0324,  0x8321,
+	0x0360,  0x8365,  0x836f,  0x036a,  0x837b,  0x037e,  0x0374,  0x8371,
+	0x8353,  0x0356,  0x035c,  0x8359,  0x0348,  0x834d,  0x8347,  0x0342,
+	0x03c0,  0x83c5,  0x83cf,  0x03ca,  0x83db,  0x03de,  0x03d4,  0x83d1,
+	0x83f3,  0x03f6,  0x03fc,  0x83f9,  0x03e8,  0x83ed,  0x83e7,  0x03e2,
+	0x83a3,  0x03a6,  0x03ac,  0x83a9,  0x03b8,  0x83bd,  0x83b7,  0x03b2,
+	0x0390,  0x8395,  0x839f,  0x039a,  0x838b,  0x038e,  0x0384,  0x8381,
+	0x0280,  0x8285,  0x828f,  0x028a,  0x829b,  0x029e,  0x0294,  0x8291,
+	0x82b3,  0x02b6,  0x02bc,  0x82b9,  0x02a8,  0x82ad,  0x82a7,  0x02a2,
+	0x82e3,  0x02e6,  0x02ec,  0x82e9,  0x02f8,  0x82fd,  0x82f7,  0x02f2,
+	0x02d0,  0x82d5,  0x82df,  0x02da,  0x82cb,  0x02ce,  0x02c4,  0x82c1,
+	0x8243,  0x0246,  0x024c,  0x8249,  0x0258,  0x825d,  0x8257,  0x0252,
+	0x0270,  0x8275,  0x827f,  0x027a,  0x826b,  0x026e,  0x0264,  0x8261,
+	0x0220,  0x8225,  0x822f,  0x022a,  0x823b,  0x023e,  0x0234,  0x8231,
+	0x8213,  0x0216,  0x021c,  0x8219,  0x0208,  0x820d,  0x8207,  0x0202
+};
+
+
+void FLAC__crc8_update(const FLAC__byte data, FLAC__uint8 *crc)
+{
+	*crc = pgm_read_byte(&FLAC__crc8_table[*crc ^ data]);
+}
+
+void FLAC__crc8_update_block(const FLAC__byte *data, unsigned len, FLAC__uint8 *crc)
+{
+	while(len--)
+		*crc = pgm_read_byte(&FLAC__crc8_table[*crc ^ *data++]);
+}
+
+FLAC__uint8 FLAC__crc8(const FLAC__byte *data, unsigned len)
+{
+	FLAC__uint8 crc = 0;
+
+	while(len--)
+		crc = pgm_read_byte(&FLAC__crc8_table[crc ^ *data++]);
+
+	return crc;
+}
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wstrict-aliasing"
+unsigned FLAC__crc16(const FLAC__byte *data, unsigned len)
+{
+	unsigned crc = 0;
+
+	while(len--)
+		crc = ((crc<<8) ^ pgm_read_word(&FLAC__crc16_table[(crc>>8) ^ *data++])) & 0xffff;
+
+	return crc;
+}
+#pragma GCC diagnostic pop

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/fixed.c

@@ -0,0 +1,397 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+//#ifdef HAVE_CONFIG_H
+#  include "config.h"
+//#endif
+
+#include <math.h>
+#include <string.h>
+#include "share/compat.h"
+#include "private/bitmath.h"
+#include "private/fixed.h"
+#include "private/macros.h"
+#include "FLAC/assert.h"
+
+#ifdef local_abs
+#undef local_abs
+#endif
+#define local_abs(x) ((uint32_t)((x)<0? -(x) : (x)))
+
+#pragma GCC optimize ("O3")
+
+#ifdef FLAC__INTEGER_ONLY_LIBRARY
+/* rbps stands for residual bits per sample
+ *
+ *             (ln(2) * err)
+ * rbps = log  (-----------)
+ *           2 (     n     )
+ */
+static FLAC__fixedpoint local__compute_rbps_integerized(FLAC__uint32 err, FLAC__uint32 n)
+{
+	FLAC__uint32 rbps;
+	uint32_t bits; /* the number of bits required to represent a number */
+	int fracbits; /* the number of bits of rbps that comprise the fractional part */
+
+	FLAC__ASSERT(sizeof(rbps) == sizeof(FLAC__fixedpoint));
+	FLAC__ASSERT(err > 0);
+	FLAC__ASSERT(n > 0);
+
+	FLAC__ASSERT(n <= FLAC__MAX_BLOCK_SIZE);
+	if(err <= n)
+		return 0;
+	/*
+	 * The above two things tell us 1) n fits in 16 bits; 2) err/n > 1.
+	 * These allow us later to know we won't lose too much precision in the
+	 * fixed-point division (err<<fracbits)/n.
+	 */
+
+	fracbits = (8*sizeof(err)) - (FLAC__bitmath_ilog2(err)+1);
+
+	err <<= fracbits;
+	err /= n;
+	/* err now holds err/n with fracbits fractional bits */
+
+	/*
+	 * Whittle err down to 16 bits max.  16 significant bits is enough for
+	 * our purposes.
+	 */
+	FLAC__ASSERT(err > 0);
+	bits = FLAC__bitmath_ilog2(err)+1;
+	if(bits > 16) {
+		err >>= (bits-16);
+		fracbits -= (bits-16);
+	}
+	rbps = (FLAC__uint32)err;
+
+	/* Multiply by fixed-point version of ln(2), with 16 fractional bits */
+	rbps *= FLAC__FP_LN2;
+	fracbits += 16;
+	FLAC__ASSERT(fracbits >= 0);
+
+	/* FLAC__fixedpoint_log2 requires fracbits%4 to be 0 */
+	{
+		const int f = fracbits & 3;
+		if(f) {
+			rbps >>= f;
+			fracbits -= f;
+		}
+	}
+
+	rbps = FLAC__fixedpoint_log2(rbps, fracbits, (uint32_t)(-1));
+
+	if(rbps == 0)
+		return 0;
+
+	/*
+	 * The return value must have 16 fractional bits.  Since the whole part
+	 * of the base-2 log of a 32 bit number must fit in 5 bits, and fracbits
+	 * must be >= -3, these assertion allows us to be able to shift rbps
+	 * left if necessary to get 16 fracbits without losing any bits of the
+	 * whole part of rbps.
+	 *
+	 * There is a slight chance due to accumulated error that the whole part
+	 * will require 6 bits, so we use 6 in the assertion.  Really though as
+	 * long as it fits in 13 bits (32 - (16 - (-3))) we are fine.
+	 */
+	FLAC__ASSERT((int)FLAC__bitmath_ilog2(rbps)+1 <= fracbits + 6);
+	FLAC__ASSERT(fracbits >= -3);
+
+	/* now shift the decimal point into place */
+	if(fracbits < 16)
+		return rbps << (16-fracbits);
+	else if(fracbits > 16)
+		return rbps >> (fracbits-16);
+	else
+		return rbps;
+}
+
+static FLAC__fixedpoint local__compute_rbps_wide_integerized(FLAC__uint64 err, FLAC__uint32 n)
+{
+	FLAC__uint32 rbps;
+	uint32_t bits; /* the number of bits required to represent a number */
+	int fracbits; /* the number of bits of rbps that comprise the fractional part */
+
+	FLAC__ASSERT(sizeof(rbps) == sizeof(FLAC__fixedpoint));
+	FLAC__ASSERT(err > 0);
+	FLAC__ASSERT(n > 0);
+
+	FLAC__ASSERT(n <= FLAC__MAX_BLOCK_SIZE);
+	if(err <= n)
+		return 0;
+	/*
+	 * The above two things tell us 1) n fits in 16 bits; 2) err/n > 1.
+	 * These allow us later to know we won't lose too much precision in the
+	 * fixed-point division (err<<fracbits)/n.
+	 */
+
+	fracbits = (8*sizeof(err)) - (FLAC__bitmath_ilog2_wide(err)+1);
+
+	err <<= fracbits;
+	err /= n;
+	/* err now holds err/n with fracbits fractional bits */
+
+	/*
+	 * Whittle err down to 16 bits max.  16 significant bits is enough for
+	 * our purposes.
+	 */
+	FLAC__ASSERT(err > 0);
+	bits = FLAC__bitmath_ilog2_wide(err)+1;
+	if(bits > 16) {
+		err >>= (bits-16);
+		fracbits -= (bits-16);
+	}
+	rbps = (FLAC__uint32)err;
+
+	/* Multiply by fixed-point version of ln(2), with 16 fractional bits */
+	rbps *= FLAC__FP_LN2;
+	fracbits += 16;
+	FLAC__ASSERT(fracbits >= 0);
+
+	/* FLAC__fixedpoint_log2 requires fracbits%4 to be 0 */
+	{
+		const int f = fracbits & 3;
+		if(f) {
+			rbps >>= f;
+			fracbits -= f;
+		}
+	}
+
+	rbps = FLAC__fixedpoint_log2(rbps, fracbits, (uint32_t)(-1));
+
+	if(rbps == 0)
+		return 0;
+
+	/*
+	 * The return value must have 16 fractional bits.  Since the whole part
+	 * of the base-2 log of a 32 bit number must fit in 5 bits, and fracbits
+	 * must be >= -3, these assertion allows us to be able to shift rbps
+	 * left if necessary to get 16 fracbits without losing any bits of the
+	 * whole part of rbps.
+	 *
+	 * There is a slight chance due to accumulated error that the whole part
+	 * will require 6 bits, so we use 6 in the assertion.  Really though as
+	 * long as it fits in 13 bits (32 - (16 - (-3))) we are fine.
+	 */
+	FLAC__ASSERT((int)FLAC__bitmath_ilog2(rbps)+1 <= fracbits + 6);
+	FLAC__ASSERT(fracbits >= -3);
+
+	/* now shift the decimal point into place */
+	if(fracbits < 16)
+		return rbps << (16-fracbits);
+	else if(fracbits > 16)
+		return rbps >> (fracbits-16);
+	else
+		return rbps;
+}
+#endif
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+uint32_t FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], uint32_t data_len, float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#else
+uint32_t FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], uint32_t data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#endif
+{
+	FLAC__int32 last_error_0 = data[-1];
+	FLAC__int32 last_error_1 = data[-1] - data[-2];
+	FLAC__int32 last_error_2 = last_error_1 - (data[-2] - data[-3]);
+	FLAC__int32 last_error_3 = last_error_2 - (data[-2] - 2*data[-3] + data[-4]);
+	FLAC__int32 error, save;
+	FLAC__uint32 total_error_0 = 0, total_error_1 = 0, total_error_2 = 0, total_error_3 = 0, total_error_4 = 0;
+	uint32_t i, order;
+
+	for(i = 0; i < data_len; i++) {
+		error  = data[i]     ; total_error_0 += local_abs(error);                      save = error;
+		error -= last_error_0; total_error_1 += local_abs(error); last_error_0 = save; save = error;
+		error -= last_error_1; total_error_2 += local_abs(error); last_error_1 = save; save = error;
+		error -= last_error_2; total_error_3 += local_abs(error); last_error_2 = save; save = error;
+		error -= last_error_3; total_error_4 += local_abs(error); last_error_3 = save;
+	}
+
+	if(total_error_0 < flac_min(flac_min(flac_min(total_error_1, total_error_2), total_error_3), total_error_4))
+		order = 0;
+	else if(total_error_1 < flac_min(flac_min(total_error_2, total_error_3), total_error_4))
+		order = 1;
+	else if(total_error_2 < flac_min(total_error_3, total_error_4))
+		order = 2;
+	else if(total_error_3 < total_error_4)
+		order = 3;
+	else
+		order = 4;
+
+	/* Estimate the expected number of bits per residual signal sample. */
+	/* 'total_error*' is linearly related to the variance of the residual */
+	/* signal, so we use it directly to compute E(|x|) */
+	FLAC__ASSERT(data_len > 0 || total_error_0 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_1 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_2 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_3 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_4 == 0);
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+	residual_bits_per_sample[0] = (float)((total_error_0 > 0) ? log(M_LN2 * (double)total_error_0 / (double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[1] = (float)((total_error_1 > 0) ? log(M_LN2 * (double)total_error_1 / (double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[2] = (float)((total_error_2 > 0) ? log(M_LN2 * (double)total_error_2 / (double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[3] = (float)((total_error_3 > 0) ? log(M_LN2 * (double)total_error_3 / (double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[4] = (float)((total_error_4 > 0) ? log(M_LN2 * (double)total_error_4 / (double)data_len) / M_LN2 : 0.0);
+#else
+	residual_bits_per_sample[0] = (total_error_0 > 0) ? local__compute_rbps_integerized(total_error_0, data_len) : 0;
+	residual_bits_per_sample[1] = (total_error_1 > 0) ? local__compute_rbps_integerized(total_error_1, data_len) : 0;
+	residual_bits_per_sample[2] = (total_error_2 > 0) ? local__compute_rbps_integerized(total_error_2, data_len) : 0;
+	residual_bits_per_sample[3] = (total_error_3 > 0) ? local__compute_rbps_integerized(total_error_3, data_len) : 0;
+	residual_bits_per_sample[4] = (total_error_4 > 0) ? local__compute_rbps_integerized(total_error_4, data_len) : 0;
+#endif
+
+	return order;
+}
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+uint32_t FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], uint32_t data_len, float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#else
+uint32_t FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], uint32_t data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#endif
+{
+	FLAC__int32 last_error_0 = data[-1];
+	FLAC__int32 last_error_1 = data[-1] - data[-2];
+	FLAC__int32 last_error_2 = last_error_1 - (data[-2] - data[-3]);
+	FLAC__int32 last_error_3 = last_error_2 - (data[-2] - 2*data[-3] + data[-4]);
+	FLAC__int32 error, save;
+	/* total_error_* are 64-bits to avoid overflow when encoding
+	 * erratic signals when the bits-per-sample and blocksize are
+	 * large.
+	 */
+	FLAC__uint64 total_error_0 = 0, total_error_1 = 0, total_error_2 = 0, total_error_3 = 0, total_error_4 = 0;
+	uint32_t i, order;
+
+	for(i = 0; i < data_len; i++) {
+		error  = data[i]     ; total_error_0 += local_abs(error);                      save = error;
+		error -= last_error_0; total_error_1 += local_abs(error); last_error_0 = save; save = error;
+		error -= last_error_1; total_error_2 += local_abs(error); last_error_1 = save; save = error;
+		error -= last_error_2; total_error_3 += local_abs(error); last_error_2 = save; save = error;
+		error -= last_error_3; total_error_4 += local_abs(error); last_error_3 = save;
+	}
+
+	if(total_error_0 < flac_min(flac_min(flac_min(total_error_1, total_error_2), total_error_3), total_error_4))
+		order = 0;
+	else if(total_error_1 < flac_min(flac_min(total_error_2, total_error_3), total_error_4))
+		order = 1;
+	else if(total_error_2 < flac_min(total_error_3, total_error_4))
+		order = 2;
+	else if(total_error_3 < total_error_4)
+		order = 3;
+	else
+		order = 4;
+
+	/* Estimate the expected number of bits per residual signal sample. */
+	/* 'total_error*' is linearly related to the variance of the residual */
+	/* signal, so we use it directly to compute E(|x|) */
+	FLAC__ASSERT(data_len > 0 || total_error_0 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_1 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_2 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_3 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_4 == 0);
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+	residual_bits_per_sample[0] = (float)((total_error_0 > 0) ? log(M_LN2 * (double)total_error_0 / (double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[1] = (float)((total_error_1 > 0) ? log(M_LN2 * (double)total_error_1 / (double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[2] = (float)((total_error_2 > 0) ? log(M_LN2 * (double)total_error_2 / (double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[3] = (float)((total_error_3 > 0) ? log(M_LN2 * (double)total_error_3 / (double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[4] = (float)((total_error_4 > 0) ? log(M_LN2 * (double)total_error_4 / (double)data_len) / M_LN2 : 0.0);
+#else
+	residual_bits_per_sample[0] = (total_error_0 > 0) ? local__compute_rbps_wide_integerized(total_error_0, data_len) : 0;
+	residual_bits_per_sample[1] = (total_error_1 > 0) ? local__compute_rbps_wide_integerized(total_error_1, data_len) : 0;
+	residual_bits_per_sample[2] = (total_error_2 > 0) ? local__compute_rbps_wide_integerized(total_error_2, data_len) : 0;
+	residual_bits_per_sample[3] = (total_error_3 > 0) ? local__compute_rbps_wide_integerized(total_error_3, data_len) : 0;
+	residual_bits_per_sample[4] = (total_error_4 > 0) ? local__compute_rbps_wide_integerized(total_error_4, data_len) : 0;
+#endif
+
+	return order;
+}
+
+void FLAC__fixed_compute_residual(const FLAC__int32 data[], uint32_t data_len, uint32_t order, FLAC__int32 residual[])
+{
+	const int idata_len = (int)data_len;
+	int i;
+
+	switch(order) {
+		case 0:
+			FLAC__ASSERT(sizeof(residual[0]) == sizeof(data[0]));
+			memcpy(residual, data, sizeof(residual[0])*data_len);
+			break;
+		case 1:
+			for(i = 0; i < idata_len; i++)
+				residual[i] = data[i] - data[i-1];
+			break;
+		case 2:
+			for(i = 0; i < idata_len; i++)
+				residual[i] = data[i] - 2*data[i-1] + data[i-2];
+			break;
+		case 3:
+			for(i = 0; i < idata_len; i++)
+				residual[i] = data[i] - 3*data[i-1] + 3*data[i-2] - data[i-3];
+			break;
+		case 4:
+			for(i = 0; i < idata_len; i++)
+				residual[i] = data[i] - 4*data[i-1] + 6*data[i-2] - 4*data[i-3] + data[i-4];
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+}
+
+void FLAC__fixed_restore_signal(const FLAC__int32 residual[], uint32_t data_len, uint32_t order, FLAC__int32 data[])
+{
+	int i, idata_len = (int)data_len;
+
+	switch(order) {
+		case 0:
+			FLAC__ASSERT(sizeof(residual[0]) == sizeof(data[0]));
+			memcpy(data, residual, sizeof(residual[0])*data_len);
+			break;
+		case 1:
+			for(i = 0; i < idata_len; i++)
+				data[i] = residual[i] + data[i-1];
+			break;
+		case 2:
+			for(i = 0; i < idata_len; i++)
+				data[i] = residual[i] + 2*data[i-1] - data[i-2];
+			break;
+		case 3:
+			for(i = 0; i < idata_len; i++)
+				data[i] = residual[i] + 3*data[i-1] - 3*data[i-2] + data[i-3];
+			break;
+		case 4:
+			for(i = 0; i < idata_len; i++)
+				data[i] = residual[i] + 4*data[i-1] - 6*data[i-2] + 4*data[i-3] - data[i-4];
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+}

FW/leo_muziekdoos_sam51/lib/ESP8266Audio/src/libflac/float.c

@@ -0,0 +1,304 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004-2009  Josh Coalson
+ * Copyright (C) 2011-2016  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+//#ifdef HAVE_CONFIG_H
+#  include "config.h"
+//#endif
+
+#include "FLAC/assert.h"
+#include "share/compat.h"
+#include "private/float.h"
+
+#pragma GCC optimize ("O3")
+
+#ifdef FLAC__INTEGER_ONLY_LIBRARY
+
+const FLAC__fixedpoint FLAC__FP_ZERO = 0;
+const FLAC__fixedpoint FLAC__FP_ONE_HALF = 0x00008000;
+const FLAC__fixedpoint FLAC__FP_ONE = 0x00010000;
+const FLAC__fixedpoint FLAC__FP_LN2 = 45426;
+const FLAC__fixedpoint FLAC__FP_E = 178145;
+
+/* Lookup tables for Knuth's logarithm algorithm */
+#define LOG2_LOOKUP_PRECISION 16
+static const FLAC__uint32 log2_lookup[][LOG2_LOOKUP_PRECISION] PROGMEM = {
+	{
+		/*
+		 * 0 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00000001,
+		/* lg(4/3) = */ 0x00000000,
+		/* lg(8/7) = */ 0x00000000,
+		/* lg(16/15) = */ 0x00000000,
+		/* lg(32/31) = */ 0x00000000,
+		/* lg(64/63) = */ 0x00000000,
+		/* lg(128/127) = */ 0x00000000,
+		/* lg(256/255) = */ 0x00000000,
+		/* lg(512/511) = */ 0x00000000,
+		/* lg(1024/1023) = */ 0x00000000,
+		/* lg(2048/2047) = */ 0x00000000,
+		/* lg(4096/4095) = */ 0x00000000,
+		/* lg(8192/8191) = */ 0x00000000,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 4 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00000010,
+		/* lg(4/3) = */ 0x00000007,
+		/* lg(8/7) = */ 0x00000003,
+		/* lg(16/15) = */ 0x00000001,
+		/* lg(32/31) = */ 0x00000001,
+		/* lg(64/63) = */ 0x00000000,
+		/* lg(128/127) = */ 0x00000000,
+		/* lg(256/255) = */ 0x00000000,
+		/* lg(512/511) = */ 0x00000000,
+		/* lg(1024/1023) = */ 0x00000000,
+		/* lg(2048/2047) = */ 0x00000000,
+		/* lg(4096/4095) = */ 0x00000000,
+		/* lg(8192/8191) = */ 0x00000000,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 8 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00000100,
+		/* lg(4/3) = */ 0x0000006a,
+		/* lg(8/7) = */ 0x00000031,
+		/* lg(16/15) = */ 0x00000018,
+		/* lg(32/31) = */ 0x0000000c,
+		/* lg(64/63) = */ 0x00000006,
+		/* lg(128/127) = */ 0x00000003,
+		/* lg(256/255) = */ 0x00000001,
+		/* lg(512/511) = */ 0x00000001,
+		/* lg(1024/1023) = */ 0x00000000,
+		/* lg(2048/2047) = */ 0x00000000,
+		/* lg(4096/4095) = */ 0x00000000,
+		/* lg(8192/8191) = */ 0x00000000,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 12 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00001000,
+		/* lg(4/3) = */ 0x000006a4,
+		/* lg(8/7) = */ 0x00000315,
+		/* lg(16/15) = */ 0x0000017d,
+		/* lg(32/31) = */ 0x000000bc,
+		/* lg(64/63) = */ 0x0000005d,
+		/* lg(128/127) = */ 0x0000002e,
+		/* lg(256/255) = */ 0x00000017,
+		/* lg(512/511) = */ 0x0000000c,
+		/* lg(1024/1023) = */ 0x00000006,
+		/* lg(2048/2047) = */ 0x00000003,
+		/* lg(4096/4095) = */ 0x00000001,
+		/* lg(8192/8191) = */ 0x00000001,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 16 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00010000,
+		/* lg(4/3) = */ 0x00006a40,
+		/* lg(8/7) = */ 0x00003151,
+		/* lg(16/15) = */ 0x000017d6,
+		/* lg(32/31) = */ 0x00000bba,
+		/* lg(64/63) = */ 0x000005d1,
+		/* lg(128/127) = */ 0x000002e6,
+		/* lg(256/255) = */ 0x00000172,
+		/* lg(512/511) = */ 0x000000b9,
+		/* lg(1024/1023) = */ 0x0000005c,
+		/* lg(2048/2047) = */ 0x0000002e,
+		/* lg(4096/4095) = */ 0x00000017,
+		/* lg(8192/8191) = */ 0x0000000c,
+		/* lg(16384/16383) = */ 0x00000006,
+		/* lg(32768/32767) = */ 0x00000003
+	},
+	{
+		/*
+		 * 20 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00100000,
+		/* lg(4/3) = */ 0x0006a3fe,
+		/* lg(8/7) = */ 0x00031513,
+		/* lg(16/15) = */ 0x00017d60,
+		/* lg(32/31) = */ 0x0000bb9d,
+		/* lg(64/63) = */ 0x00005d10,
+		/* lg(128/127) = */ 0x00002e59,
+		/* lg(256/255) = */ 0x00001721,
+		/* lg(512/511) = */ 0x00000b8e,
+		/* lg(1024/1023) = */ 0x000005c6,
+		/* lg(2048/2047) = */ 0x000002e3,
+		/* lg(4096/4095) = */ 0x00000171,
+		/* lg(8192/8191) = */ 0x000000b9,
+		/* lg(16384/16383) = */ 0x0000005c,
+		/* lg(32768/32767) = */ 0x0000002e
+	},
+	{
+		/*
+		 * 24 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x01000000,
+		/* lg(4/3) = */ 0x006a3fe6,
+		/* lg(8/7) = */ 0x00315130,
+		/* lg(16/15) = */ 0x0017d605,
+		/* lg(32/31) = */ 0x000bb9ca,
+		/* lg(64/63) = */ 0x0005d0fc,
+		/* lg(128/127) = */ 0x0002e58f,
+		/* lg(256/255) = */ 0x0001720e,
+		/* lg(512/511) = */ 0x0000b8d8,
+		/* lg(1024/1023) = */ 0x00005c61,
+		/* lg(2048/2047) = */ 0x00002e2d,
+		/* lg(4096/4095) = */ 0x00001716,
+		/* lg(8192/8191) = */ 0x00000b8b,
+		/* lg(16384/16383) = */ 0x000005c5,
+		/* lg(32768/32767) = */ 0x000002e3
+	},
+	{
+		/*
+		 * 28 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x10000000,
+		/* lg(4/3) = */ 0x06a3fe5c,
+		/* lg(8/7) = */ 0x03151301,
+		/* lg(16/15) = */ 0x017d6049,
+		/* lg(32/31) = */ 0x00bb9ca6,
+		/* lg(64/63) = */ 0x005d0fba,
+		/* lg(128/127) = */ 0x002e58f7,
+		/* lg(256/255) = */ 0x001720da,
+		/* lg(512/511) = */ 0x000b8d87,
+		/* lg(1024/1023) = */ 0x0005c60b,
+		/* lg(2048/2047) = */ 0x0002e2d7,
+		/* lg(4096/4095) = */ 0x00017160,
+		/* lg(8192/8191) = */ 0x0000b8ad,
+		/* lg(16384/16383) = */ 0x00005c56,
+		/* lg(32768/32767) = */ 0x00002e2b
+	}
+};
+
+#if 0
+static const FLAC__uint64 log2_lookup_wide[] = {
+	{
+		/*
+		 * 32 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ FLAC__U64L(0x100000000),
+		/* lg(4/3) = */ FLAC__U64L(0x6a3fe5c6),
+		/* lg(8/7) = */ FLAC__U64L(0x31513015),
+		/* lg(16/15) = */ FLAC__U64L(0x17d60497),
+		/* lg(32/31) = */ FLAC__U64L(0x0bb9ca65),
+		/* lg(64/63) = */ FLAC__U64L(0x05d0fba2),
+		/* lg(128/127) = */ FLAC__U64L(0x02e58f74),
+		/* lg(256/255) = */ FLAC__U64L(0x01720d9c),
+		/* lg(512/511) = */ FLAC__U64L(0x00b8d875),
+		/* lg(1024/1023) = */ FLAC__U64L(0x005c60aa),
+		/* lg(2048/2047) = */ FLAC__U64L(0x002e2d72),
+		/* lg(4096/4095) = */ FLAC__U64L(0x00171600),
+		/* lg(8192/8191) = */ FLAC__U64L(0x000b8ad2),
+		/* lg(16384/16383) = */ FLAC__U64L(0x0005c55d),
+		/* lg(32768/32767) = */ FLAC__U64L(0x0002e2ac)
+	},
+	{
+		/*
+		 * 48 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ FLAC__U64L(0x1000000000000),
+		/* lg(4/3) = */ FLAC__U64L(0x6a3fe5c60429),
+		/* lg(8/7) = */ FLAC__U64L(0x315130157f7a),
+		/* lg(16/15) = */ FLAC__U64L(0x17d60496cfbb),
+		/* lg(32/31) = */ FLAC__U64L(0xbb9ca64ecac),
+		/* lg(64/63) = */ FLAC__U64L(0x5d0fba187cd),
+		/* lg(128/127) = */ FLAC__U64L(0x2e58f7441ee),
+		/* lg(256/255) = */ FLAC__U64L(0x1720d9c06a8),
+		/* lg(512/511) = */ FLAC__U64L(0xb8d8752173),
+		/* lg(1024/1023) = */ FLAC__U64L(0x5c60aa252e),
+		/* lg(2048/2047) = */ FLAC__U64L(0x2e2d71b0d8),
+		/* lg(4096/4095) = */ FLAC__U64L(0x1716001719),
+		/* lg(8192/8191) = */ FLAC__U64L(0xb8ad1de1b),
+		/* lg(16384/16383) = */ FLAC__U64L(0x5c55d640d),
+		/* lg(32768/32767) = */ FLAC__U64L(0x2e2abcf52)
+	}
+};
+#endif
+
+FLAC__uint32 FLAC__fixedpoint_log2(FLAC__uint32 x, uint32_t fracbits, uint32_t precision)
+{
+	const FLAC__uint32 ONE = (1u << fracbits);
+	const FLAC__uint32 *table = log2_lookup[fracbits >> 2];
+
+	FLAC__ASSERT(fracbits < 32);
+	FLAC__ASSERT((fracbits & 0x3) == 0);
+
+	if(x < ONE)
+		return 0;
+
+	if(precision > LOG2_LOOKUP_PRECISION)
+		precision = LOG2_LOOKUP_PRECISION;
+
+	/* Knuth's algorithm for computing logarithms, optimized for base-2 with lookup tables */
+	{
+		FLAC__uint32 y = 0;
+		FLAC__uint32 z = x >> 1, k = 1;
+		while (x > ONE && k < precision) {
+			if (x - z >= ONE) {
+				x -= z;
+				z = x >> k;
+				y += table[k];
+			}
+			else {
+				z >>= 1;
+				k++;
+			}
+		}
+		return y;
+	}
+}
+
+#endif /* defined FLAC__INTEGER_ONLY_LIBRARY */