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ESP32-audioI2S
Unverified Commit 81171154 authored by Wolle's avatar Wolle Committed by GitHub
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Arduino V3.0 ready

parent 2ef8cf7f
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src/Audio.cpp
View file @ 81171154
......@@ -3,8 +3,8 @@
*
* Created on: Oct 26.2018
*
* Version 3.0.7
* Updated on: Oct 08.2023
* Version 3.0.7a
* Updated on: Oct 09.2023
* Author: Wolle (schreibfaul1)
*
*/
......@@ -184,8 +184,8 @@ Audio::Audio(bool internalDAC /* = false */, uint8_t channelEnabled /* = I2S_DAC
m_i2s_config.bits_per_sample = I2S_BITS_PER_SAMPLE_16BIT;
m_i2s_config.channel_format = I2S_CHANNEL_FMT_RIGHT_LEFT;
m_i2s_config.intr_alloc_flags = ESP_INTR_FLAG_LEVEL1; // interrupt priority
m_i2s_config.dma_buf_count = 16;
m_i2s_config.dma_buf_len = 512;
m_i2s_config.dma_desc_num = 16;
m_i2s_config.dma_frame_num = 512;
m_i2s_config.use_apll = APLL_DISABLE; // must be disabled in V2.0.1-RC1
m_i2s_config.tx_desc_auto_clear = true; // new in V1.0.1
m_i2s_config.fixed_mclk = I2S_PIN_NO_CHANGE;
......@@ -273,36 +273,6 @@ esp_err_t Audio::I2Sstop(uint8_t i2s_num) {
return i2s_stop((i2s_port_t) i2s_num);
}
//---------------------------------------------------------------------------------------------------------------------
esp_err_t Audio::i2s_mclk_pin_select(const uint8_t pin) {
// IDF >= 4.4 use setPinout(BCLK, LRC, DOUT, DIN, MCK) only, i2s_mclk_pin_select() is no longer needed
if(pin != 0 && pin != 1 && pin != 3) {
log_e("Only support GPIO0/GPIO1/GPIO3, gpio_num:%d", pin);
return ESP_ERR_INVALID_ARG;
}
#ifdef CONFIG_IDF_TARGET_ESP32
switch(pin){
case 0:
PIN_FUNC_SELECT(PERIPHS_IO_MUX_GPIO0_U, FUNC_GPIO0_CLK_OUT1);
WRITE_PERI_REG(PIN_CTRL, 0xFFF0);
break;
case 1:
PIN_FUNC_SELECT(PERIPHS_IO_MUX_U0TXD_U, FUNC_U0TXD_CLK_OUT3);
WRITE_PERI_REG(PIN_CTRL, 0xF0F0);
break;
case 3:
PIN_FUNC_SELECT(PERIPHS_IO_MUX_U0RXD_U, FUNC_U0RXD_CLK_OUT2);
WRITE_PERI_REG(PIN_CTRL, 0xFF00);
break;
default:
break;
}
#endif
return ESP_OK;
}
//---------------------------------------------------------------------------------------------------------------------
Audio::~Audio() {
//I2Sstop(m_i2s_num);
//InBuff.~AudioBuffer(); #215 the AudioBuffer is automatically destroyed by the destructor
......@@ -334,7 +304,7 @@ void Audio::setDefaults() {
_client = static_cast<WiFiClient*>(&client); /* default to *something* so that no NULL deref can happen */
ts_parsePacket(0, 0, 0); // reset ts routine
AUDIO_INFO("buffers freed, free Heap: %u bytes", ESP.getFreeHeap());
AUDIO_INFO("buffers freed, free Heap: %lu bytes", ESP.getFreeHeap());
m_f_chunked = false; // Assume not chunked
m_f_firstmetabyte = false;
......@@ -502,7 +472,7 @@ bool Audio::connecttohost(const char* host, const char* user, const char* pwd) {
if(res){
uint32_t dt = millis() - t;
strcpy(m_lastHost, l_host);
AUDIO_INFO("%s has been established in %u ms, free Heap: %u bytes",
AUDIO_INFO("%s has been established in %lu ms, free Heap: %lu bytes",
m_f_ssl?"SSL":"Connection", dt, ESP.getFreeHeap());
m_f_running = true;
}
......@@ -1184,7 +1154,7 @@ int Audio::read_WAV_Header(uint8_t* data, size_t len) {
AUDIO_INFO("FormatCode: %u", fc);
// AUDIO_INFO("Channel: %u", nic);
// AUDIO_INFO("SampleRate: %u", sr);
AUDIO_INFO("DataRate: %u", dr);
AUDIO_INFO("DataRate: %lu", dr);
AUDIO_INFO("DataBlockSize: %u", dbs);
AUDIO_INFO("BitsPerSample: %u", bps);
......@@ -1278,7 +1248,7 @@ int Audio::read_FLAC_Header(uint8_t *data, size_t len) {
m_controlCounter = FLAC_MAGIC;
if(getDatamode() == AUDIO_LOCALFILE){
m_contentlength = getFileSize();
AUDIO_INFO("Content-Length: %u", m_contentlength);
AUDIO_INFO("Content-Length: %lu", m_contentlength);
}
return 0;
}
......@@ -1347,7 +1317,7 @@ int Audio::read_FLAC_Header(uint8_t *data, size_t len) {
vTaskDelay(2);
uint32_t nextval = bigEndian(data + 13, 3);
m_flacSampleRate = nextval >> 4;
AUDIO_INFO("FLAC sampleRate: %u", m_flacSampleRate);
AUDIO_INFO("FLAC sampleRate: %lu", m_flacSampleRate);
vTaskDelay(2);
m_flacNumChannels = ((nextval & 0x06) >> 1) + 1;
AUDIO_INFO("FLAC numChannels: %u", m_flacNumChannels);
......@@ -1363,13 +1333,13 @@ int Audio::read_FLAC_Header(uint8_t *data, size_t len) {
AUDIO_INFO("FLAC bitsPerSample: %u", m_flacBitsPerSample);
m_flacTotalSamplesInStream = bigEndian(data + 17, 4);
if(m_flacTotalSamplesInStream){
AUDIO_INFO("total samples in stream: %u", m_flacTotalSamplesInStream);
AUDIO_INFO("total samples in stream: %lu", m_flacTotalSamplesInStream);
}
else{
AUDIO_INFO("total samples in stream: N/A");
}
if(bps != 0 && m_flacTotalSamplesInStream) {
AUDIO_INFO("audio file duration: %u seconds", m_flacTotalSamplesInStream / m_flacSampleRate);
AUDIO_INFO("audio file duration: %lu seconds", m_flacTotalSamplesInStream / m_flacSampleRate);
}
m_controlCounter = FLAC_MBH; // METADATA_BLOCK_HEADER
retvalue = l + 3;
......@@ -1463,7 +1433,7 @@ int Audio::read_ID3_Header(uint8_t *data, size_t len) {
if(getDatamode() == AUDIO_LOCALFILE){
ID3version = 0;
m_contentlength = getFileSize();
AUDIO_INFO("Content-Length: %u", m_contentlength);
AUDIO_INFO("Content-Length: %lu", m_contentlength);
}
m_controlCounter ++;
APIC_seen = false;
......@@ -1892,10 +1862,10 @@ int Audio::read_M4A_Header(uint8_t *data, size_t len) {
if(streamType!= 5) { log_e("Streamtype is not audio!"); }
uint32_t maxBr = bigEndian(pos + 26, 4); // max bitrate
AUDIO_INFO("max bitrate: %i", maxBr);
AUDIO_INFO("max bitrate: %lu", maxBr);
uint32_t avrBr = bigEndian(pos + 30, 4); // avg bitrate
AUDIO_INFO("avr bitrate: %i", avrBr);
AUDIO_INFO("avr bitrate: %lu", avrBr);
uint16_t ASC = bigEndian(pos + 39, 2);
......@@ -1913,7 +1883,7 @@ int Audio::read_M4A_Header(uint8_t *data, size_t len) {
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350
};
uint8_t sRate = (ASC & 0x0600) >> 7; // next 4 bits Sampling Frequencies
AUDIO_INFO("Sampling Frequency: %u",samplingFrequencies[sRate]);
AUDIO_INFO("Sampling Frequency: %lu",samplingFrequencies[sRate]);
uint8_t chConfig = (ASC & 0x78) >> 3; // next 4 bits
if(chConfig == 0) AUDIO_INFO("Channel Configurations: AOT Specifc Config");
......@@ -2002,7 +1972,7 @@ int Audio::read_M4A_Header(uint8_t *data, size_t len) {
m_audioDataStart = headerSize;
// m_contentlength = headerSize + m_audioDataSize; // after this mdat atom there may be other atoms
if(getDatamode() == AUDIO_LOCALFILE){
AUDIO_INFO("Content-Length: %u", m_contentlength);
AUDIO_INFO("Content-Length: %lu", m_contentlength);
}
m_controlCounter = M4A_OKAY; // that's all
return 0;
......@@ -2758,7 +2728,7 @@ void Audio::processLocalFile() {
}
if(m_f_loop && f_stream){ //eof
AUDIO_INFO("loop from: %u to: %u", getFilePos(), m_audioDataStart); // loop
AUDIO_INFO("loop from: %lu to: %lu", getFilePos(), m_audioDataStart); // loop
setFilePos(m_audioDataStart);
if(m_codec == CODEC_FLAC) FLACDecoderReset();
m_audioCurrentTime = 0;
......@@ -3188,7 +3158,7 @@ void Audio::processWebStreamHLS() {
f_stream = true; // ready to play the audio data
uint16_t filltime = millis() - m_t0;
if(m_f_Log) AUDIO_INFO("stream ready");
if(m_f_Log) AUDIO_INFO("buffer filled in %d ms", filltime);
if(m_f_Log) AUDIO_INFO("buffer filled in %u ms", filltime);
}
// play audio data - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
......@@ -3358,7 +3328,7 @@ bool Audio::parseHttpResponseHeader() { // this is the response to a GET / reque
int32_t br = atoi(c_bitRate); // Found bitrate tag, read the bitrate in Kbit
br = br * 1000;
setBitrate(br);
sprintf(m_chbuf, "%d", getBitRate());
sprintf(m_chbuf, "%lu", getBitRate());
if(audio_bitrate) audio_bitrate(m_chbuf);
}
......@@ -3383,14 +3353,14 @@ bool Audio::parseHttpResponseHeader() { // this is the response to a GET / reque
int32_t i_cl = atoi(c_cl);
m_contentlength = i_cl;
m_streamType = ST_WEBFILE; // Stream comes from a fileserver
if(m_f_Log) AUDIO_INFO("content-length: %i", m_contentlength);
if(m_f_Log) AUDIO_INFO("content-length: %lu", m_contentlength);
}
else if(startsWith(rhl, "icy-description:")) {
const char* c_idesc = (rhl + 16);
while(c_idesc[0] == ' ') c_idesc++;
latinToUTF8(rhl, sizeof(rhl)); // if already UTF-8 do nothing, otherwise convert to UTF-8
if(specialIndexOf((uint8_t*)c_idesc, "24bit", strlen(c_idesc)) >= 0){
if(strlen(c_idesc) > 0 && specialIndexOf((uint8_t*)c_idesc, "24bit", 0) > 0){
AUDIO_INFO("icy-description: %s has to be 8 or 16", c_idesc);
stopSong();
}
......@@ -3458,7 +3428,7 @@ bool Audio:: initializeDecoder(){
}
gfH = ESP.getFreeHeap();
hWM = uxTaskGetStackHighWaterMark(NULL);
AUDIO_INFO("MP3Decoder has been initialized, free Heap: %u bytes , free stack %u DWORDs", gfH, hWM);
AUDIO_INFO("MP3Decoder has been initialized, free Heap: %lu bytes , free stack %lu DWORDs", gfH, hWM);
InBuff.changeMaxBlockSize(m_frameSizeMP3);
break;
case CODEC_AAC:
......@@ -3469,7 +3439,7 @@ bool Audio:: initializeDecoder(){
}
gfH = ESP.getFreeHeap();
hWM = uxTaskGetStackHighWaterMark(NULL);
AUDIO_INFO("AACDecoder has been initialized, free Heap: %u bytes , free stack %u DWORDs", gfH, hWM);
AUDIO_INFO("AACDecoder has been initialized, free Heap: %lu bytes , free stack %lu DWORDs", gfH, hWM);
InBuff.changeMaxBlockSize(m_frameSizeAAC);
}
break;
......@@ -3481,7 +3451,7 @@ bool Audio:: initializeDecoder(){
}
gfH = ESP.getFreeHeap();
hWM = uxTaskGetStackHighWaterMark(NULL);
AUDIO_INFO("AACDecoder has been initialized, free Heap: %u bytes , free stack %u DWORDs", gfH, hWM);
AUDIO_INFO("AACDecoder has been initialized, free Heap: %lu bytes , free stack %lu DWORDs", gfH, hWM);
InBuff.changeMaxBlockSize(m_frameSizeAAC);
}
break;
......@@ -3497,7 +3467,7 @@ bool Audio:: initializeDecoder(){
gfH = ESP.getFreeHeap();
hWM = uxTaskGetStackHighWaterMark(NULL);
InBuff.changeMaxBlockSize(m_frameSizeFLAC);
AUDIO_INFO("FLACDecoder has been initialized, free Heap: %u bytes , free stack %u DWORDs", gfH, hWM);
AUDIO_INFO("FLACDecoder has been initialized, free Heap: %lu bytes , free stack %lu DWORDs", gfH, hWM);
break;
case CODEC_OPUS:
if(!OPUSDecoder_AllocateBuffers()){
......@@ -3506,7 +3476,7 @@ bool Audio:: initializeDecoder(){
}
gfH = ESP.getFreeHeap();
hWM = uxTaskGetStackHighWaterMark(NULL);
AUDIO_INFO("OPUSDecoder has been initialized, free Heap: %u bytes , free stack %u DWORDs", gfH, hWM);
AUDIO_INFO("OPUSDecoder has been initialized, free Heap: %lu bytes , free stack %lu DWORDs", gfH, hWM);
InBuff.changeMaxBlockSize(m_frameSizeOPUS);
break;
case CODEC_VORBIS:
......@@ -3520,7 +3490,7 @@ bool Audio:: initializeDecoder(){
}
gfH = ESP.getFreeHeap();
hWM = uxTaskGetStackHighWaterMark(NULL);
AUDIO_INFO("VORBISDecoder has been initialized, free Heap: %u bytes, free stack %u DWORDs", gfH, hWM);
AUDIO_INFO("VORBISDecoder has been initialized, free Heap: %lu bytes, free stack %lu DWORDs", gfH, hWM);
InBuff.changeMaxBlockSize(m_frameSizeVORBIS);
break;
case CODEC_WAV:
......@@ -3770,10 +3740,10 @@ void Audio::showstreamtitle(const char* ml) {
void Audio::showCodecParams(){
// print Codec Parameter (mp3, aac) in audio_info()
AUDIO_INFO("Channels: %i", getChannels());
AUDIO_INFO("SampleRate: %i", getSampleRate());
AUDIO_INFO("BitsPerSample: %i", getBitsPerSample());
if(getBitRate()) {AUDIO_INFO("BitRate: %i", getBitRate());}
AUDIO_INFO("Channels: %u", getChannels());
AUDIO_INFO("SampleRate: %lu", getSampleRate());
AUDIO_INFO("BitsPerSample: %u", getBitsPerSample());
if(getBitRate()) {AUDIO_INFO("BitRate: %lu", getBitRate());}
else {AUDIO_INFO("BitRate: N/A");}
if(m_codec == CODEC_AAC){
......@@ -3837,7 +3807,7 @@ int Audio::findNextSync(uint8_t* data, size_t len){
}
if (nextSync == 0){
if(audio_info && swnf>0){
sprintf(m_chbuf, "syncword not found %i times", swnf);
sprintf(m_chbuf, "syncword not found %lu times", swnf);
audio_info(m_chbuf);
swnf = 0;
}
......@@ -4606,7 +4576,7 @@ void Audio::IIR_calculateCoefficients(int8_t G0, int8_t G1, int8_t G2){ // Infi
FcHS = getSampleRate() /2 - 100;
// according to the sampling theorem, the sample rate must be at least 2 * 6000 >= 12000Hz for a filter
// frequency of 6000Hz. If this is not the case, the filter frequency (plus a reserve of 100Hz) is lowered
AUDIO_INFO("Highshelf frequency lowered, from 6000Hz to %dHz", (uint32_t)FcHS);
AUDIO_INFO("Highshelf frequency lowered, from 6000Hz to %luHz", (uint32_t)FcHS);
}
float K, norm, Q, Fc, V ;
......
src/Audio.h
View file @ 81171154
......@@ -3,8 +3,8 @@
*
* Created on: Oct 28,2018
*
* Version 3.0.7
* Updated on: Oct 08.2023
* Version 3.0.7a
* Updated on: Oct 09.2023
* Author: Wolle (schreibfaul1)
*/
......@@ -158,7 +158,6 @@ public:
uint32_t getTotalPlayingTime();
uint16_t getVUlevel();
esp_err_t i2s_mclk_pin_select(const uint8_t pin);
uint32_t inBufferFilled(); // returns the number of stored bytes in the inputbuffer
uint32_t inBufferFree(); // returns the number of free bytes in the inputbuffer
void setTone(int8_t gainLowPass, int8_t gainBandPass, int8_t gainHighPass);
......
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