
/*****************************************************************************************************************************************************
    audio.cpp

    Created on: 28.10.2018                                                                                                  */
char audioI2SVers[] = "\
    Version 3.4.5b                                                                                                                            ";
/*  Updated on: Mar 11, 2026

    Author: Wolle (schreibfaul1)
    Audio library for ESP32, ESP32-S3 or ESP32-P4
    Arduino Vers. V3 is mandatory
    PSRAM is mandatory
    external DAC is mandatory

*****************************************************************************************************************************************************/

#include "Audio.h"
#include "aac_decoder/aac_decoder.h"
#include "flac_decoder/flac_decoder.h"
#include "mp3_decoder/mp3_decoder.h"
#include "opus_decoder/opus_decoder.h"
#include "psram_unique_ptr.hpp"
#include "vorbis_decoder/vorbis_decoder.h"
#include "wav_decoder/wav_decoder.h"

// constants
constexpr size_t m_frameSizeWav = 4096;
constexpr size_t m_frameSizeMP3 = 1600 * 2;
constexpr size_t m_frameSizeAAC = 1600 * 2;
constexpr size_t m_frameSizeFLAC = UINT16_MAX;   // max ogg size
constexpr size_t m_frameSizeOPUS = UINT16_MAX;   // max ogg size
constexpr size_t m_frameSizeVORBIS = UINT16_MAX; // OGG length is normally 4080 bytes, but can be reach 64KB in the metadata block
constexpr size_t m_outbuffSize = 4608 * 2;
constexpr size_t m_samplesBuff48KSize = m_outbuffSize * 8; // SRmin: 6KHz -> SRmax: 48K

constexpr size_t AUDIO_STACK_SIZE = 3500;

// static allocations for Audio task
StaticTask_t __attribute__((unused)) xAudioTaskBuffer;
StackType_t __attribute__((unused))  xAudioStack[AUDIO_STACK_SIZE];

// weak default implementation - can be overridden by user
__attribute__((weak)) void audio_process_i2s(int32_t* outBuff, int16_t validSamples, bool* continueI2S) {
    // Default: do nothing. User can provide their own implementation to process audio data.
}
__attribute__((weak)) void audio_process_raw_samples(int32_t* outBuff, int16_t validSamples) {
    // Default: do nothing. User can provide their own implementation to process audio data.
}

// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
// 📌📌📌  A U D I O B U F F E R  📌📌📌
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————

// AudioBuffer will be allocated in PSRAM
//
//  m_startPtr       m_readPtr                  m_writePtr                                                    m_endPtr
//   |                  |<------maxAvailableBytes----->|<--------------------- writeSpace ----------------------->|
//   ▼                  ▼                              ▼                                                          ▼
//   ---------------------------------------------------------------------------------------------------------------
//   |                         <--m_mainBuffSize-->                               |      <--m_resBuffSize -->     |
//   ---------------------------------------------------------------------------------------------------------------
//   |<---freeSpace---->|<------------filled---------->|<-------freeSpace-------->|
//                                                                                ▲
//                                                                                |
//                                                                             m_buffEnd
//   if resBuff is full copy data from resBuff to the beginning
//   if m_readPtr >= m_buff<end set m_readptr to m_startPtr + (m_readPtr - m_buffEnd)
//
//  m_startPtr                      m_writePtr                 m_readPtr                                      m_endPtr
//   |                                 |<-------writeSpace------>|<---------------maxAvailableBytes-------------->|
//   ▼                                 ▼                         ▼                ▼                               ▼
//   ---------------------------------------------------------------------------------------------------------------
//   |                        <--m_mainBuffSize-->                                |      <--m_resBuffSize -->     |
//   ---------------------------------------------------------------------------------------------------------------
//   |<------------filled------------->|<-------freeSpace------->|<----filled---->▲
//                                                                                |
//                                                                             m_buffEnd
//

AudioBuffer::AudioBuffer() {
    m_mainBuffSize = UINT16_MAX * 10;
    m_resBuffSize = UINT16_MAX;
}

AudioBuffer::~AudioBuffer() {
    ;
}

size_t AudioBuffer::getBufsize() {
    return m_mainBuffSize;
}

size_t AudioBuffer::init() {
    m_buffer.alloc(m_mainBuffSize + m_resBuffSize, "AudioBuffer");
    m_mutex = xSemaphoreCreateBinary();
    xSemaphoreGive(m_mutex);
    m_init = true;
    m_startPtr = m_buffer.get();
    m_endPtr = m_buffer.get() + m_mainBuffSize;
    m_buffEnd = m_endPtr + m_resBuffSize;
    reset();
    return m_mainBuffSize;
}

void AudioBuffer::setMaxBlocksize(uint32_t mbs) {
    m_maxBlockSize = mbs;
}

size_t AudioBuffer::getMaxBlockSize() {
    return m_maxBlockSize;
}
//----------------------------------------------------------------------------------------------------------------------------------------------------
size_t AudioBuffer::freeSpace() {
    if (!m_init) return 0;
    if (m_readPtr == m_writePtr) {
        if (m_isEmpty) { return m_mainBuffSize; }
        if (m_isFull) { return 0; }
        log_e("writePtr == readPtr, writePtr %i, readPtr %i", m_writePtr - m_startPtr, m_readPtr - m_startPtr);
    }
    if (m_readPtr < m_writePtr) {
        if (m_writePtr > m_endPtr) {
            return (m_readPtr - m_startPtr);
        } else {
            return (m_endPtr - m_writePtr) + (m_readPtr - m_startPtr);
        }
    }
    return m_readPtr - m_writePtr;
}
//----------------------------------------------------------------------------------------------------------------------------------------------------
size_t AudioBuffer::bufferFilled() {
    if (!m_init) return 0;
    xSemaphoreTake(m_mutex, portMAX_DELAY);
    size_t bufferFilled = 0;
    if (m_readPtr == m_writePtr) {
        if (m_isEmpty) {
            bufferFilled = 0;
            goto end;
        }
        if (m_isFull) {
            bufferFilled = m_mainBuffSize;
            goto end;
        }
        log_e("%s %i writePtr == readPtr, writePtr %i, readPtr %i", __FILE__, __LINE__, m_writePtr - m_startPtr, m_readPtr - m_startPtr);
    }
    if (m_readPtr < m_writePtr) {
        bufferFilled = (size_t)(m_writePtr - m_readPtr);
        goto end;
    }
    bufferFilled = (m_endPtr - m_readPtr) + (m_writePtr - m_startPtr);
end:
    xSemaphoreGive(m_mutex);
    return bufferFilled;
}
//----------------------------------------------------------------------------------------------------------------------------------------------------
size_t AudioBuffer::writeSpace() {
    if (!m_init) return 0;
    xSemaphoreTake(m_mutex, portMAX_DELAY);
    m_writeSpace = 0;

    // Check whether a complete block still fits in at the end
    size_t spaceToEnd = m_buffEnd - m_writePtr;

    if (m_isFull) {
        m_writeSpace = 0;
        goto end;
    }

    if (m_isEmpty) {
        m_writeSpace = min(m_maxRet, spaceToEnd);
        goto end;
    }

    if (spaceToEnd == 0) { // be sure that the resBuff is full
        // Only copy if the read pointer is not in the way
        if (m_readPtr > m_startPtr + m_resBuffSize) {
            memcpy(m_startPtr, m_endPtr, m_resBuffSize);
            // log_w("wrap copy %u bytes", m_resBuffSize);
            m_writePtr = m_startPtr + m_resBuffSize;
        }
    }

    if (m_writePtr < m_readPtr) {
        if (m_readPtr >= m_endPtr) {                                       // readPtr is in resBuff?
            m_writeSpace = min(m_maxRet, (size_t)(m_endPtr - m_writePtr)); // writePtr does not enter resbuff, wait for copy im readspace
        } else {
            m_writeSpace = min(m_maxRet, (size_t)(m_readPtr - m_writePtr));
        }
        goto end;
    }
    if (m_writePtr > m_readPtr) {
        m_writeSpace = min(m_maxRet, spaceToEnd);
        goto end;
    }

    log_e("%s %i writePtr == readPtr, writePtr %i, readPtr %i", __FILE__, __LINE__, m_writePtr - m_startPtr, m_readPtr - m_startPtr);

end:
    xSemaphoreGive(m_mutex);
    return m_writeSpace;
}
//----------------------------------------------------------------------------------------------------------------------------------------------------
void AudioBuffer::bytesWritten(size_t bw) {
    if (!m_init) return;
    xSemaphoreTake(m_mutex, portMAX_DELAY);
    if (!bw) goto end;

    if (bw > m_writeSpace) log_e("writeSpace < bw, writeSpace %i, bw %i", m_writeSpace, bw); // bw must not be larger than the queried m_writeSpace

    if (m_writePtr < m_readPtr && m_writePtr + bw > m_readPtr) {
        log_e("writePtr overrruns readPtr, writePtr %i, readPtr %i, bw %i", m_writePtr - m_startPtr, m_readPtr - m_startPtr, bw);
        m_writePtr = m_readPtr;
        goto end;
    }
    if (m_writePtr + bw > m_buffEnd) {
        log_e("writePtr overrruns buffEnd, writePtr %i, buffEnd %i, bw %i", m_writePtr - m_startPtr, m_buffEnd - m_startPtr, bw);
        m_writePtr = m_buffEnd;
        goto end;
    }

    m_writePtr += bw;
    if (bw) {
        if (m_writePtr == m_readPtr) m_isFull = true;
        m_isEmpty = false;
    }

end:
    xSemaphoreGive(m_mutex);
    return;
}
//----------------------------------------------------------------------------------------------------------------------------------------------------
size_t AudioBuffer::readSpace() {
    if (!m_init) return 0;
    xSemaphoreTake(m_mutex, portMAX_DELAY);

    if (m_readPtr >= m_endPtr && m_writePtr <= m_endPtr) {
        size_t len = m_readPtr - m_endPtr;
        if (m_writePtr > m_startPtr + len) {
            log_d("set new readptr to %i", len);
            m_readPtr = m_startPtr + len;
        }
    }

    m_readSpace = 0;
    if (m_isEmpty) { goto end; }

    if (m_isFull) {
        m_readSpace = min(m_maxRet, (size_t)(m_buffEnd - m_readPtr));
        goto end;
    }

    if (m_readPtr < m_writePtr) {
        m_readSpace = min(m_maxRet, (size_t)(m_writePtr - m_readPtr));
        goto end;
    }

    if (m_readPtr > m_writePtr) {
        m_readSpace = min(m_maxRet, (size_t)(m_buffEnd - m_readPtr));
        goto end;
    }

    log_e("%s %i writePtr == readPtr, writePtr %i, readPtr %i", __FILE__, __LINE__, m_writePtr - m_startPtr, m_readPtr - m_startPtr);
end:
    xSemaphoreGive(m_mutex);
    return m_readSpace;
}
//----------------------------------------------------------------------------------------------------------------------------------------------------
void AudioBuffer::bytesWasRead(size_t br) {
    if (!m_init) return;
    xSemaphoreTake(m_mutex, portMAX_DELAY);

    if (!br) goto end;

    if (m_readSpace < br) {
        log_e("readSpace < br, rspc %i, br %i", m_readSpace, br); // br must not be larger than the queried m_readSpace
        vTaskDelay(100);
        goto end;
    }

    if (m_readPtr < m_writePtr && m_readPtr + br > m_writePtr) {
        log_e("readPtr overrruns writePtr, readPtr %i, writePtr %i, br %i", m_readPtr - m_startPtr, m_writePtr - m_startPtr, br);
        m_readPtr = m_writePtr;
        vTaskDelay(100);
        goto end;
    }
    if (m_readPtr + br > m_buffEnd) {
        log_e("readPtr overrruns buffEnd, readPtr %i, buffEnd %i, bw %i", m_readPtr - m_startPtr, m_buffEnd - m_startPtr, br);
        m_readPtr = m_buffEnd;
        vTaskDelay(100);
        goto end;
    }

    m_readPtr += br;

    if (br) {
        if (m_readPtr == m_writePtr) {
            m_isEmpty = true;
            // log_d(" readPtr %i, writePtr %i, br %i", m_readPtr - m_startPtr, m_writePtr - m_startPtr, br);
        }
        m_isFull = false;
    }

end:
    xSemaphoreGive(m_mutex);
    return;
}
//----------------------------------------------------------------------------------------------------------------------------------------------------
uint8_t* AudioBuffer::getWritePtr() {
    return m_writePtr;
}

uint8_t* AudioBuffer::getReadPtr() {
    return m_readPtr;
}

void AudioBuffer::reset() {
    m_writePtr = m_buffer.get();
    m_readPtr = m_buffer.get();
    m_isEmpty = true;
    m_isFull = false;
    m_readSpace = 0;
    m_writeSpace = min(m_maxRet, (size_t)(m_buffEnd - m_writePtr));
}

void AudioBuffer::showStatus() {
    printf("\nfilled %i, free %i\n", bufferFilled(), freeSpace());
    printf("writeSpace %i, readSpace %i\n", writeSpace(), readSpace());
    printf("writePtr %i, readPtr %i\n", m_writePtr - m_startPtr, m_readPtr - m_startPtr);
    printf("isEmpty %i, isFull %i\n\n", m_isEmpty, m_isFull);
}

// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
// 📌📌📌  A U D I O   📌📌📌
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
Audio::Audio(uint8_t i2sPort) {

    m_f_I2S_init = false;
    mutex_playAudioData = xSemaphoreCreateMutex();
    mutex_audioTask = xSemaphoreCreateMutex();
    mutex_audioTaskIsDecoding = xSemaphoreCreateMutex();

    if (!psramFound()) AUDIO_LOG_ERROR("audioI2S requires PSRAM!");

    clientsecure.setInsecure();
    m_i2s_items.i2s_num = i2sPort; // i2s port number

    // -------- I2S configuration -------------------------------------------------------------------------------------------
    memset(&m_i2s_chan_cfg, 0, sizeof(i2s_chan_config_t));
    m_i2s_chan_cfg.id = (i2s_port_t)m_i2s_items.i2s_num;  // I2S_NUM_AUTO, I2S_NUM_0, I2S_NUM_1
    m_i2s_chan_cfg.role = I2S_ROLE_MASTER;                // I2S controller master role, bclk and lrc signal will be set to output
    m_i2s_chan_cfg.dma_desc_num = m_i2s_items.DESC_NUM;   // number of DMA buffer
    m_i2s_chan_cfg.dma_frame_num = m_i2s_items.FRAME_NUM; // I2S frame number in one DMA buffer.
    m_i2s_chan_cfg.auto_clear = true;                     // i2s will always send zero automatically if no data to send
    m_i2s_chan_cfg.allow_pd = false;
    m_i2s_chan_cfg.intr_priority = 2;
    i2s_new_channel(&m_i2s_chan_cfg, &m_i2s_tx_handle, NULL);

    memset(&m_i2s_std_cfg, 0, sizeof(i2s_std_config_t));
    m_i2s_std_cfg.slot_cfg = I2S_STD_PHILIPS_SLOT_DEFAULT_CONFIG(I2S_DATA_BIT_WIDTH_32BIT, I2S_SLOT_MODE_STEREO); // Set to enable bit shift in Philips mode
    m_i2s_std_cfg.gpio_cfg.bclk = I2S_GPIO_UNUSED;                                                                // BCLK, Assignment in setPinout()
    m_i2s_std_cfg.gpio_cfg.din = I2S_GPIO_UNUSED;                                                                 // not used
    m_i2s_std_cfg.gpio_cfg.dout = I2S_GPIO_UNUSED;                                                                // DOUT, Assignment in setPinout()
    m_i2s_std_cfg.gpio_cfg.mclk = I2S_GPIO_UNUSED;                                                                // MCLK, Assignment in setPinout()
    m_i2s_std_cfg.gpio_cfg.ws = I2S_GPIO_UNUSED;                                                                  // LRC,  Assignment in setPinout()
    m_i2s_std_cfg.gpio_cfg.invert_flags.mclk_inv = false;
    m_i2s_std_cfg.gpio_cfg.invert_flags.bclk_inv = false;
    m_i2s_std_cfg.gpio_cfg.invert_flags.ws_inv = false;
    m_i2s_std_cfg.clk_cfg.sample_rate_hz = m_i2s_items.sampleRate;
    m_i2s_std_cfg.clk_cfg.clk_src = I2S_CLK_SRC_DEFAULT;
    m_i2s_std_cfg.clk_cfg.mclk_multiple = I2S_MCLK_MULTIPLE_256;
    i2s_channel_init_std_mode(m_i2s_tx_handle, &m_i2s_std_cfg);

    calculateVolumeLimits(); // first init, vol = 21, vol_steps = 21
    startAudioTask();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
Audio::~Audio() {
    stopSong();
    setDefaults();

    i2s_channel_disable(m_i2s_tx_handle);
    i2s_del_channel(m_i2s_tx_handle);
    stopAudioTask();
    vSemaphoreDelete(mutex_playAudioData);
    vSemaphoreDelete(mutex_audioTask);
    vSemaphoreDelete(mutex_audioTaskIsDecoding);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::destroy_decoder() {
    if (m_decoder && m_decoder->isValid()) {
        info(*this, evt_info, "%sDecoder has been destroyed", m_decoder->whoIsIt());
        m_decoder->reset();
        m_decoder.reset();
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
std::unique_ptr<Decoder> Audio::createDecoder(const std::string& type) {
    destroy_decoder();
    if (type == "MP3") return std::make_unique<MP3Decoder>(*this);
    if (type == "FLAC") return std::make_unique<FlacDecoder>(*this);
    if (type == "OPUS") return std::make_unique<OpusDecoder>(*this);
    if (type == "AAC") return std::make_unique<AACDecoder>(*this);
    if (type == "VORBIS") return std::make_unique<VorbisDecoder>(*this);
    if (type == "WAV") return std::make_unique<WavDecoder>(*this);
    return nullptr;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::initInBuff() {
    if (!InBuff.isInitialized()) {
        size_t size = InBuff.init();
        if (size > 0) { info(*this, evt_info, "inputBufferSize: %u bytes", size - 1); }
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
esp_err_t Audio::I2Sstart() {
    zeroI2Sbuff();
    esp_err_t err = ESP_FAIL;
    if (!m_f_i2s_channel_enabled) {
        err = i2s_channel_enable(m_i2s_tx_handle);
        if (err == ESP_OK) m_f_i2s_channel_enabled = true;
    }
    return err;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
esp_err_t Audio::I2Sstop() {
    m_outBuff.clear();                                                  // Clear OutputBuffer
    m_samplesBuff48K.clear();                                           // Clear samplesBuff48K
    std::fill(std::begin(m_inputHistory), std::end(m_inputHistory), 0); // Clear history in samplesBuff48K
    esp_err_t err = ESP_FAIL;
    if (m_f_i2s_channel_enabled) err = i2s_channel_disable(m_i2s_tx_handle);
    m_f_i2s_channel_enabled = false;
    return err;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::zeroI2Sbuff() {
    uint8_t buff[2] = {0, 0}; // From IDF V5 there is no longer the zero_dma_buff() function.
    size_t  bytes_loaded = 0; // As a replacement, we write a small amount of zeros in the buffer and thus reset the entire buffer.
    i2s_channel_preload_data(m_i2s_tx_handle, buff, 2, &bytes_loaded);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setDefaults() {
    stopSong();
    initInBuff(); // initialize InputBuffer if not already done
    InBuff.reset();
    m_outBuff.clear();        // Clear OutputBuffer
    m_samplesBuff48K.clear(); // Clear samplesBuff48K
    vector_clear_and_shrink(m_playlistURL);
    vector_clear_and_shrink(m_playlistContent);
    vector_clear_and_shrink(m_syltLines);
    m_syltTimeStamp.clear();
    m_hashQueue.clear();
    m_hashQueue.shrink_to_fit(); // uint32_t vector
    client.stop();
    clientsecure.stop();
    m_client = static_cast<NetworkClient*>(&client); /* default to *something* so that no NULL deref can happen */
    ts_parsePacket(0, 0, 0);                         // reset ts routine
    m_lastM3U8host.reset();

    AUDIO_LOG_DEBUG("buffers freed, free Heap: %lu bytes", (long unsigned int)ESP.getFreeHeap());

    m_f_timeout = false;
    m_f_chunked = false; // Assume not chunked
    m_f_firstmetabyte = false;
    m_f_playing = false;
    //    m_f_ssl = false;
    m_f_metadata = false;
    m_f_tts = false;
    m_f_firstCall = true;       // InitSequence for processWebstream and processLocalFile
    m_cat.firstCall = true;     // InitSequence for calculateAudioTime
    m_pplM3U8.firstCall = true; // InitSequence for parsePlaylist_M3U8
    m_f_firstPlayCall = true;   // InitSequence for playAudioData
    //    m_f_running = false;       // already done in stopSong
    m_f_firstLoop = true;
    m_f_unsync = false;   // set within ID3 tag but not used
    m_f_exthdr = false;   // ID3 extended header
    m_f_rtsp = false;     // RTSP (m3u8)stream
    m_f_m3u8data = false; // set again in processM3U8entries() if necessary
    m_f_continue = false;
    m_f_ts = false;
    m_f_ogg = false;
    m_f_m4aID3dataAreRead = false;
    m_f_stream = false;
    m_f_decode_ready = false;
    m_f_eof = false;
    m_f_ID3v1TagFound = false;
    m_f_lockInBuffer = false;
    m_f_acceptRanges = false;
    m_f_connectionClose = false;

    m_streamType = ST_NONE;
    m_codec = CODEC_NONE;
    m_playlistFormat = FORMAT_NONE;
    m_dataMode = AUDIO_NONE;
    m_streamTitle.assign("");
    m_resumeFilePos = -1;
    m_audioCurrentTime = 0; // Reset playtimer
    m_audioFileDuration = 0;
    m_audioDataStart = 0;
    m_audioDataSize = 0;
    m_audioFileSize = 0;
    m_avr_bitrate = 0;
    m_nominal_bitrate = 0;
    m_bytesNotConsumed = 0; // counts all not decodable bytes
    m_chunkcount = 0;       // for chunked streams
    m_curSample = 0;
    m_metaint = 0;        // No metaint yet
    m_LFcount = 0;        // For end of header detection
    m_controlCounter = 0; // Status within readID3data() and readWaveHeader()
    m_channels = 2;       // assume stereo #209
    m_ID3Size = 0;
    m_haveNewFilePos = 0;
    m_validSamples = 0;
    m_M4A_chConfig = 0;
    m_M4A_objectType = 0;
    m_M4A_sampleRate = 0;
    m_lastGranulePosition = 0;
    m_validSamples = 0;
    std::fill(std::begin(m_inputHistory), std::end(m_inputHistory), 0);
    if (m_f_reset_m3u8Codec) { m_m3u8Codec = CODEC_AAC; } // reset to default
    m_f_reset_m3u8Codec = true;
}

// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setConnectionTimeout(uint16_t timeout_ms, uint16_t timeout_ms_ssl) {
    if (timeout_ms) m_timeout_ms = timeout_ms;
    if (timeout_ms_ssl) m_timeout_ms_ssl = timeout_ms_ssl;
}

/*
    Text to speech API provides a speech endpoint based on our TTS (text-to-speech) model.
    More info: https://platform.openai.com/docs/guides/text-to-speech/text-to-speech

    Request body:
    model (string) [Required] - One of the available TTS models: tts-1 or tts-1-hd
    input (string) [Required] - The text to generate audio for. The maximum length is 4096 characters.
    instructions (string) [Optional] - A description of the desired characteristics of the generated audio.
    voice (string) [Required] - The voice to use when generating the audio. Supported voices are alloy, echo, fable, onyx, nova, and shimmer.
    response_format (string) [Optional] - Defaults to mp3. The format to audio in. Supported formats are mp3, opus, aac, and flac.
    speed (number) [Optional] - Defaults to 1. The speed of the generated audio. Select a value from 0.25 to 4.0. 1.0 is the default.

    Usage: audio.openai_speech(OPENAI_API_KEY, "tts-1", input, instructions, "shimmer", "mp3", "1");
*/
bool Audio::openai_speech(const String& api_key, const String& model, const String& input, const String& instructions, const String& voice, const String& response_format, const String& speed) {
    ps_ptr<char> host;
    host.assign("api.openai.com");
    char path[] = "/v1/audio/speech";

    if (input == "") {
        AUDIO_LOG_WARN("input text is empty");
        stopSong();
        return false;
    }
    xSemaphoreTakeRecursive(mutex_playAudioData, 0.3 * configTICK_RATE_HZ);

    setDefaults();
    m_f_ssl = true;

    m_speechtxt.assign(input.c_str());

    // Escape special characters in input
    String input_clean = "";
    for (int i = 0; i < input.length(); i++) {
        char c = input.charAt(i);
        if (c == '\"') {
            input_clean += "\\\"";
        } else if (c == '\n') {
            input_clean += "\\n";
        } else if (c == '\r') {
            input_clean += "\\r";
        } else if (c == '\t') {
            input_clean += "\\t";
        } else if (c == '\\') {
            input_clean += "\\\\";
        } else if (c == '\b') {
            input_clean += "\\b";
        } else if (c == '\f') {
            input_clean += "\\f";
        } else {
            input_clean += c;
        }
    }

    // Escape special characters in instructions
    String instructions_clean = "";
    for (int i = 0; i < instructions.length(); i++) {
        char c = instructions.charAt(i);
        if (c == '\"') {
            instructions_clean += "\\\"";
        } else if (c == '\n') {
            instructions_clean += "\\n";
        } else if (c == '\r') {
            instructions_clean += "\\r";
        } else if (c == '\t') {
            instructions_clean += "\\t";
        } else if (c == '\\') {
            instructions_clean += "\\\\";
        } else if (c == '\b') {
            instructions_clean += "\\b";
        } else if (c == '\f') {
            instructions_clean += "\\f";
        } else {
            instructions_clean += c;
        }
    }

    String post_body = "{"
                       "\"model\": \"" +
                       model + "\"," + "\"stream\": true," + // add
                       "\"input\": \"" + input_clean + "\"," + "\"instructions\": \"" + instructions_clean + "\"," + "\"voice\": \"" + voice + "\"," + "\"response_format\": \"" + response_format +
                       "\"," + "\"speed\": " + speed + "}";

    String http_request =
        //  "POST " + String(path) + " HTTP/1.0\r\n" // UNKNOWN ERROR CODE (0050) - crashing on HTTP/1.1 need to use HTTP/1.0
        "POST " + String(path) + " HTTP/1.1\r\n" + "Host: " + host.get() + "\r\n" + "Authorization: Bearer " + api_key + "\r\n" + "Accept-Encoding: identity;q=1,*;q=0\r\n" +
        "User-Agent: nArija/1.0\r\n" + "Content-Type: application/json; charset=utf-8\r\n" + "Content-Length: " + post_body.length() +
        "\r\n"
        //  + "Connection: close\r\n" + "\r\n"
        + "\r\n" + post_body + "\r\n";

    bool res = true;
    int  port = 443;
    m_client = static_cast<NetworkClientSecure*>(&clientsecure);

    uint32_t t = millis();
    info(*this, evt_info, "Connect to: \"%s\"", host.get());
    res = m_client->connect(host.get(), port, m_timeout_ms_ssl);
    if (res) {
        uint32_t dt = millis() - t;
        m_lastHost.assign(host.get());
        m_currentHost.clone_from(host);
        info(*this, evt_info, "%s has been established in %lu ms", m_f_ssl ? "SSL" : "Connection", (long unsigned int)dt);
        m_f_running = true;
    }

    m_expectedCodec = CODEC_NONE;
    m_expectedPlsFmt = FORMAT_NONE;

    if (res) {
        m_client->print(http_request);
        if (response_format == "mp3") m_expectedCodec = CODEC_MP3;
        if (response_format == "opus") m_expectedCodec = CODEC_OPUS;
        if (response_format == "aac") m_expectedCodec = CODEC_AAC;
        if (response_format == "flac") m_expectedCodec = CODEC_FLAC;

        m_dataMode = HTTP_RESPONSE_HEADER;
        m_f_tts = true;
    } else {
        AUDIO_LOG_WARN("Request %s failed!", host.get());
    }
    xSemaphoreGiveRecursive(mutex_playAudioData);
    return res;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
audiolib::hwoe_t Audio::dismantle_host(const char* host) {
    if (!host) return {};

    audiolib::hwoe_t result;

    const char* p = host;

    // 🔐 1. SSL check
    if (strncmp(p, "https://", 8) == 0) {
        result.ssl = true;
        p += 8;
    } else if (strncmp(p, "http://", 7) == 0) {
        result.ssl = false;
        p += 7;
    } else { // No valid scheme -> error or acceptance http
        result.ssl = false;
    }

    // ❓ 2. extract host (from p to ':' or '/' or '?')
    const char* host_start = p;
    const char* port_sep = strchr(p, ':');
    const char* path_sep = strchr(p, '/');
    const char* query_sep = strchr(p, '?');

    const char* host_end = p + strlen(p); // default: end of string
    if (port_sep && port_sep < host_end) host_end = port_sep;
    if (path_sep && path_sep < host_end) host_end = path_sep;
    if (query_sep && query_sep < host_end) host_end = query_sep;
    result.hwoe.copy_from(host_start, host_end - host_start);
    result.rqh_host.clone_from(result.hwoe);

    // ❓ 3. extract port
    result.port = result.ssl ? 443 : 80; // default
    if (port_sep && (!path_sep || port_sep < path_sep)) {
        result.port = atoi(port_sep + 1);
        result.rqh_host.appendf(":%u", result.port);
    }

    // ❓ 4. extract extension (path)
    if (path_sep) {
        const char* path_start = path_sep + 1;
        const char* path_end = query_sep ? query_sep : host + strlen(host);
        result.extension.copy_from(path_start, path_end - path_start);
    }

    // ❓ 5. extract query string
    if (query_sep) { result.query_string.assign(query_sep + 1); }

    return result;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::connecttohost(const char* host, const char* user, const char* pwd) { // user and pwd for authentification only, can be empty

    if (!host) {
        AUDIO_LOG_ERROR("Hostaddress is empty");
        stopSong();
        return false;
    }
    if (strlen(host) > 2048) {
        AUDIO_LOG_ERROR("Hostaddress is too long");
        stopSong();
        return false;
    } // max length in Chrome DevTools

    bool     res = false;   // return value
    uint16_t port = 0;      // port number
    uint16_t authLen = 0;   // length of authorization
    uint32_t timestamp = 0; // timeout surveillance

    ps_ptr<char> c_host;       // copy of host
    ps_ptr<char> hwoe;         // host without extension
    ps_ptr<char> rqh_host;     // host for request header
    ps_ptr<char> extension;    // extension
    ps_ptr<char> query_string; // parameter
    ps_ptr<char> path;         // extension + '?' + parameter
    ps_ptr<char> rqh;          // request header

    xSemaphoreTakeRecursive(mutex_playAudioData, 0.3 * configTICK_RATE_HZ);

    c_host.copy_from(host);
    c_host.trim();
    auto dismantledHost = dismantle_host(c_host.get());

    //  https://edge.live.mp3.mdn.newmedia.nacamar.net:8000/ps-charivariwb/livestream.mp3;?user=ps-charivariwb;&pwd=ps-charivariwb-------
    //      |   |                                     |    |                              |
    //      |   |                                     |    |                              |             (query string)
    //  ssl?|   |<-----host without extension-------->|port|<----- --extension----------->|<-first parameter->|<-second parameter->.......
    //                                                     |
    //                                                     |<-----------------------------path------------------------------------>......

    m_f_ssl = dismantledHost.ssl;
    port = dismantledHost.port;
    if (dismantledHost.hwoe.valid()) hwoe.clone_from(dismantledHost.hwoe);
    if (dismantledHost.rqh_host.valid()) rqh_host.clone_from(dismantledHost.rqh_host);
    if (dismantledHost.extension.valid()) extension.clone_from(dismantledHost.extension);
    if (dismantledHost.query_string.valid()) query_string.clone_from(dismantledHost.query_string);

    if (extension.valid()) path.assign(extension.get());
    if (query_string.valid()) {
        path.append("?");
        path.append(query_string.get());
    }
    if (!hwoe.valid()) hwoe.assign("");
    if (!extension.valid()) extension.assign("");
    if (!path.valid()) path.assign("");

    path = urlencode(path.get(), true);

    // optional basic authorization
    if (user && pwd) authLen = strlen(user) + strlen(pwd);
    ps_ptr<char> authorization;
    ps_ptr<char> toEncode;
    authorization.alloc(base64_encode_expected_len(authLen + 1) + 1);
    authorization.clear();
    if (authLen > 0) {
        toEncode.assign(user);
        toEncode.append(":");
        toEncode.append(pwd);
        b64encode((const char*)toEncode.get(), toEncode.strlen(), authorization.get());
    }

    setDefaults();

    rqh.assign("GET /");
    rqh.append(path.get());
    rqh.append(" HTTP/1.1\r\n");
    rqh.appendf("Host: %s\r\n", rqh_host.get());
    rqh.append("Icy-MetaData:1\r\n");
    rqh.append("Icy-MetaData:2\r\n");
    rqh.append("Pragma: no-cache\r\n");
    rqh.append("Cache-Control: no-cache\r\n");
    rqh.append("Range: bytes=0-\r\n");
    rqh.append("Accept: */*\r\n");
    rqh.append("User-Agent: VLC/3.0.21 LibVLC/3.0.21 AppleWebKit/537.36 (KHTML, like Gecko)\r\n");
    if (authLen > 0) {
        rqh.append("Authorization: Basic ");
        rqh.append(authorization.get());
        rqh.append("\r\n");
    }
    rqh.append("Accept-Encoding: identity;q=1,*;q=0\r\n");
    rqh.append("Connection: keep-alive\r\n\r\n");

    if (m_f_ssl) {
        m_client = static_cast<NetworkClientSecure*>(&clientsecure);
        if (port == 80) port = 443;
    } else {
        m_client = static_cast<NetworkClient*>(&client);
    }

    timestamp = millis();
    m_client->setTimeout(m_f_ssl ? m_timeout_ms_ssl : m_timeout_ms);

    info(*this, evt_info, "connect to: \"%s\" on port %d path \"/%s\"", hwoe.get(), port, path.get());
    res = m_client->connect(hwoe.get(), port);

    m_expectedCodec = CODEC_NONE;
    m_expectedPlsFmt = FORMAT_NONE;

    if (res) {
        uint32_t dt = millis() - timestamp;
        info(*this, evt_info, "%s has been established in %lu ms", m_f_ssl ? "SSL" : "Connection", (long unsigned int)dt);
        m_f_running = true;
        m_client->print(rqh.get());
        if (extension.ends_with_icase(".mp3")) m_expectedCodec = CODEC_MP3;
        if (extension.ends_with_icase(".aac")) m_expectedCodec = CODEC_AAC;
        if (extension.ends_with_icase(".wav")) m_expectedCodec = CODEC_WAV;
        if (extension.ends_with_icase(".m4a")) m_expectedCodec = CODEC_M4A;
        if (extension.ends_with_icase(".ogg")) m_expectedCodec = CODEC_OGG;
        if (extension.ends_with_icase(".flac")) m_expectedCodec = CODEC_FLAC;
        if (extension.ends_with_icase("-flac")) m_expectedCodec = CODEC_FLAC;
        if (extension.ends_with_icase(".opus")) m_expectedCodec = CODEC_OPUS;
        if (extension.ends_with_icase("/opus")) m_expectedCodec = CODEC_OPUS;
        if (extension.ends_with_icase(".asx")) m_expectedPlsFmt = FORMAT_ASX;
        if (extension.ends_with_icase(".m3u")) m_expectedPlsFmt = FORMAT_M3U;
        if (extension.ends_with_icase(".pls")) m_expectedPlsFmt = FORMAT_PLS;
        if (extension.contains(".m3u8")) m_expectedPlsFmt = FORMAT_M3U8;

        m_currentHost.clone_from(c_host);
        m_lastHost.clone_from(c_host);
        info(*this, evt_lasthost, "%s", m_lastHost.c_get());
        m_dataMode = HTTP_RESPONSE_HEADER; // Handle header
        m_streamType = ST_WEBSTREAM;
    } else {
        AUDIO_LOG_ERROR("Request %s failed!", c_host.get());
        m_f_running = false;
        info(*this, evt_name, "");
        info(*this, evt_streamtitle, "");
        info(*this, evt_icydescription, "");
        info(*this, evt_icyurl, "");
    }
    xSemaphoreGiveRecursive(mutex_playAudioData);
    return res;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::httpPrint(const char* host) {
    // user and pwd for authentification only, can be empty
    if (!m_f_running) return false;
    if (host == NULL) {
        AUDIO_LOG_ERROR("Hostaddress is empty");
        stopSong();
        return false;
    }

    uint16_t     port = 0;     // port number
    ps_ptr<char> c_host;       // copy of host
    ps_ptr<char> hwoe;         // host without extension
    ps_ptr<char> rqh_host;     // host in request header
    ps_ptr<char> extension;    // extension
    ps_ptr<char> query_string; // parameter
    ps_ptr<char> path;         // extension + '?' + parameter
    ps_ptr<char> rqh;          // request header
    ps_ptr<char> cur_hwoe;     // m_currenthost without extension

    c_host.copy_from(host);
    c_host.trim();
    auto dismantledHost = dismantle_host(c_host.get());

    //  https://edge.live.mp3.mdn.newmedia.nacamar.net:8000/ps-charivariwb/livestream.mp3;?user=ps-charivariwb;&pwd=ps-charivariwb-------
    //      |   |                                     |    |                              |
    //      |   |                                     |    |                              |             (query string)
    //  ssl?|   |<-----host without extension-------->|port|<----- --extension----------->|<-first parameter->|<-second parameter->.......
    //                                                     |
    //                                                     |<-----------------------------path------------------------------------------->

    m_f_ssl = dismantledHost.ssl;
    port = dismantledHost.port;
    if (dismantledHost.hwoe.valid()) hwoe.clone_from(dismantledHost.hwoe);
    if (dismantledHost.rqh_host.valid()) rqh_host.clone_from(dismantledHost.rqh_host);
    if (dismantledHost.extension.valid()) extension.clone_from(dismantledHost.extension);
    if (dismantledHost.query_string.valid()) query_string.clone_from(dismantledHost.query_string);

    if (extension.valid()) path.assign(extension.get());
    if (query_string.valid()) {
        path.append("?");
        path.append(query_string.get());
    }
    if (!hwoe.valid()) hwoe.assign("");
    if (!extension.valid()) extension.assign("");
    if (!path.valid()) path.assign("");

    path = urlencode(path.get(), true);

    if (!m_currentHost.valid()) m_currentHost.assign("");
    auto dismantledLastHost = dismantle_host(m_currentHost.get());
    cur_hwoe.clone_from(dismantledLastHost.hwoe);

    bool f_equal = true;
    if (hwoe.equals(cur_hwoe) && port == dismantledLastHost.port) {
        f_equal = true;
    } else {
        f_equal = false;
    }

    rqh.assign("GET /");
    rqh.append(path.get());
    rqh.append(" HTTP/1.1\r\n");
    rqh.appendf("Host: %s\r\n", rqh_host.get());
    rqh.append("Icy-MetaData:1\r\n");
    rqh.append("Icy-MetaData:2\r\n");
    rqh.append("Accept:*/*\r\n");
    rqh.append("User-Agent: VLC/3.0.21 LibVLC/3.0.21 AppleWebKit/537.36 (KHTML, like Gecko)\r\n");
    rqh.append("Accept-Encoding: identity;q=1,*;q=0\r\n");
    rqh.append("Connection: keep-alive\r\n\r\n");

    info(*this, evt_info, "next URL: \"%s\"", c_host.get());

    if (f_equal == false) {
        if (m_client->connected()) m_client->stop();
    }
    if (!m_client->connected()) {
        if (m_f_ssl) {
            m_client = static_cast<NetworkClientSecure*>(&clientsecure);
            if (m_f_ssl && port == 80) port = 443;
        } else {
            m_client = static_cast<NetworkClient*>(&client);
        }
        if (f_equal) info(*this, evt_info, "The host has disconnected, reconnecting");

        if (!m_client->connect(hwoe.get(), port)) {
            AUDIO_LOG_ERROR("connection lost %s", c_host.c_get());
            stopSong();
            return false;
        }
    }
    m_currentHost.clone_from(c_host);
    m_client->print(rqh.get());

    if (extension.ends_with_icase(".mp3"))
        m_expectedCodec = CODEC_MP3;
    else if (extension.ends_with_icase(".aac"))
        m_expectedCodec = CODEC_AAC;
    else if (extension.ends_with_icase(".wav"))
        m_expectedCodec = CODEC_WAV;
    else if (extension.ends_with_icase(".m4a"))
        m_expectedCodec = CODEC_M4A;
    else if (extension.ends_with_icase(".flac"))
        m_expectedCodec = CODEC_FLAC;
    else
        m_expectedCodec = CODEC_NONE;

    if (extension.ends_with_icase(".asx"))
        m_expectedPlsFmt = FORMAT_ASX;
    else if (extension.ends_with_icase(".m3u"))
        m_expectedPlsFmt = FORMAT_M3U;
    else if (extension.contains(".m3u8"))
        m_expectedPlsFmt = FORMAT_M3U8;
    else if (extension.ends_with_icase(".pls"))
        m_expectedPlsFmt = FORMAT_PLS;
    else
        m_expectedPlsFmt = FORMAT_NONE;

    m_audioFileSize = 0;
    m_dataMode = HTTP_RESPONSE_HEADER; // Handle header
    m_streamType = ST_WEBSTREAM;
    m_f_chunked = false;
    AUDIO_LOG_DEBUG("playlistFormat %s, dataMode %s, streamType: %s", plsFmtStr[m_playlistFormat], dataModeStr[m_dataMode], streamTypeStr[m_streamType]);
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::httpRange(uint32_t seek, uint32_t length) {

    if (!m_f_running) return false;

    uint16_t     port = 0;     // port number
    ps_ptr<char> c_host;       // copy of host
    ps_ptr<char> hwoe;         // host without extension
    ps_ptr<char> rqh_host;     // host in request header
    ps_ptr<char> extension;    // extension
    ps_ptr<char> query_string; // parameter
    ps_ptr<char> path;         // extension + '?' + parameter
    ps_ptr<char> rqh;          // request header
    ps_ptr<char> cur_hwoe;     // m_currenthost without extension
    ps_ptr<char> range;        // e.g. "Range: bytes=124-"

    c_host.clone_from(m_currentHost);
    c_host.trim();
    auto dismantledHost = dismantle_host(c_host.get());

    //  https://edge.live.mp3.mdn.newmedia.nacamar.net:8000/ps-charivariwb/livestream.mp3;?user=ps-charivariwb;&pwd=ps-charivariwb-------
    //      |   |                                     |    |                              |
    //      |   |                                     |    |                              |             (query string)
    //  ssl?|   |<-----host without extension-------->|port|<----- --extension----------->|<-first parameter->|<-second parameter->.......
    //                                                     |
    //                                                     |<-----------------------------path------------------------------------------->

    m_f_ssl = dismantledHost.ssl;
    port = dismantledHost.port;
    if (dismantledHost.hwoe.valid()) hwoe.clone_from(dismantledHost.hwoe);
    if (dismantledHost.rqh_host.valid()) rqh_host.clone_from(dismantledHost.rqh_host);
    if (dismantledHost.extension.valid()) extension.clone_from(dismantledHost.extension);
    if (dismantledHost.query_string.valid()) query_string.clone_from(dismantledHost.query_string);

    if (extension.valid()) path.assign(extension.get());
    if (query_string.valid()) {
        path.append("?");
        path.append(query_string.get());
    }
    if (!hwoe.valid()) hwoe.assign("");
    if (!extension.valid()) extension.assign("");
    if (!path.valid()) path.assign("");

    path = urlencode(path.get(), true);

    if (!m_currentHost.valid()) m_currentHost.assign("");
    auto dismantledLastHost = dismantle_host(m_currentHost.get());
    cur_hwoe.clone_from(dismantledLastHost.hwoe);

    if (length == UINT32_MAX)
        range.assignf("Range: bytes=%li-\r\n", seek);
    else
        range.assignf("Range: bytes=%li-%li\r\n", seek, seek + length);

    rqh.assignf("GET /%s HTTP/1.1\r\n", path.get());
    rqh.appendf("Host: %s\r\n", rqh_host.get());
    rqh.append("Accept: */*\r\n");
    rqh.append("Accept-Encoding: identity;q=1,*;q=0\r\n");
    rqh.append("Cache-Control: no-cache\r\n");
    rqh.append("Connection: keep-alive\r\n");
    rqh.appendf(range.c_get());
    rqh.appendf("Referer: %s\r\n", m_currentHost.c_get());
    rqh.append("Sec-GPC: 1\r\n");
    rqh.append("User-Agent: VLC/3.0.21 LibVLC/3.0.21 AppleWebKit/537.36 (KHTML, like Gecko)\r\n\r\n");

    if (m_client->connected()) { m_client->stop(); }
    if (m_f_ssl) {
        m_client = static_cast<NetworkClientSecure*>(&clientsecure);
        if (m_f_ssl && port == 80) port = 443;
    } else {
        m_client = static_cast<NetworkClient*>(&client);
    }

    if (!m_client->connect(hwoe.get(), port)) {
        AUDIO_LOG_ERROR("connection lost %s", c_host.c_get());
        stopSong();
        return false;
    }

    // AUDIO_LOG_INFO("rqh \n%s", rqh.get());

    m_client->print(rqh.c_get());
    m_resumeFilePos = seek; // used in processWebFile()
    m_dataMode = HTTP_RANGE_HEADER;
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::connecttoFS(fs::FS& fs, const char* path, int32_t fileStartTime) {

    xSemaphoreTakeRecursive(mutex_playAudioData, 0.3 * configTICK_RATE_HZ);
    ps_ptr<char> c_path;
    ps_ptr<char> audioPath;
    bool         res = false;
    m_fileStartTime = fileStartTime;
    m_codec = CODEC_NONE;

    if (!path) {
        AUDIO_LOG_ERROR("file path is not set");
        goto exit;
    } // guard
    c_path.copy_from(path); // copy from path
    c_path.trim();
    if (!c_path.contains(".")) {
        AUDIO_LOG_ERROR("No file extension found");
        goto exit;
    } // guard
    setDefaults(); // free buffers an set defaults

    if (c_path.ends_with_icase(".mp3")) m_codec = CODEC_MP3;
    if (c_path.ends_with_icase(".m4a")) m_codec = CODEC_M4A;
    if (c_path.ends_with_icase(".aac")) m_codec = CODEC_AAC;
    if (c_path.ends_with_icase(".wav")) m_codec = CODEC_WAV;
    if (c_path.ends_with_icase(".flac")) m_codec = CODEC_FLAC;
    if (c_path.ends_with_icase(".opus")) m_codec = CODEC_OGG;
    if (c_path.ends_with_icase(".ogg")) m_codec = CODEC_OGG;
    if (c_path.ends_with_icase(".oga")) m_codec = CODEC_OGG;

    if (m_codec == CODEC_OGG) m_f_ogg = true;
    if (m_codec == CODEC_NONE) { // guard
        int dotPos = c_path.last_index_of('.');
        AUDIO_LOG_WARN("The %s format is not supported", path + dotPos);
        goto exit;
    }

    if (!c_path.starts_with("/")) c_path.insert("/", 0);

    if (!fs.exists(c_path.get())) {
        AUDIO_LOG_WARN("file not found: %s", c_path.get());
        goto exit;
    }
    info(*this, evt_info, "Reading file: \"%s\"", c_path.get());
    m_audiofile = fs.open(c_path.get());
    m_dataMode = AUDIO_LOCALFILE;
    m_audioFileSize = m_audiofile.size();
    m_f_running = true;
    res = true;

exit:
    xSemaphoreGiveRecursive(mutex_playAudioData);
    return res;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::connecttospeech(const char* speech, const char* lang) {
    xSemaphoreTakeRecursive(mutex_playAudioData, 0.3 * configTICK_RATE_HZ);

    setDefaults();
    char host[] = "translate.google.com.vn";
    char path[] = "/translate_tts";

    m_speechtxt.assign(speech);             // unique pointer takes care of the memory management
    auto urlStr = urlencode(speech, false); // percent encoding

    ps_ptr<char> req; // request header
    req.assign("GET ");
    req.append(path);
    req.append("?ie=UTF-8&tl=");
    req.append(lang);
    req.append("&client=tw-ob&q=");
    req.append(urlStr.get());
    req.append(" HTTP/1.1\r\n");
    req.append("Host: ");
    req.append(host);
    req.append("\r\n");
    req.append("User-Agent: Mozilla/5.0 \r\n");
    req.append("Accept-Encoding: identity\r\n");
    req.append("Accept: text/html\r\n");
    req.append("Connection: close\r\n\r\n");

    m_client = static_cast<NetworkClient*>(&client);
    info(*this, evt_info, "connect to \"%s\"", host);
    if (!m_client->connect(host, 80)) {
        AUDIO_LOG_ERROR("Connection failed");
        xSemaphoreGiveRecursive(mutex_playAudioData);
        return false;
    }
    m_client->print(req.get());

    m_f_running = true;
    m_f_ssl = false;
    m_f_tts = true;
    m_dataMode = HTTP_RESPONSE_HEADER;
    m_lastHost.assign(host);
    m_currentHost.copy_from(host);
    xSemaphoreGiveRecursive(mutex_playAudioData);
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::showID3Tag(const char* tag, const char* value) {
    ps_ptr<char> id3tag;
    // V2.2
    if (!strcmp(tag, "CNT")) id3tag.appendf("Play counter: %s", value, "id3tag");
    if (!strcmp(tag, "COM")) id3tag.appendf("Comments: %s", value, "id3tag");
    if (!strcmp(tag, "CRA")) id3tag.appendf("Audio encryption: %s", value, "id3tag");
    if (!strcmp(tag, "CRM")) id3tag.appendf("Encrypted meta frame: %s", value, "id3tag");
    if (!strcmp(tag, "ETC")) id3tag.appendf("Event timing codes: %s", value, "id3tag");
    if (!strcmp(tag, "EQU")) id3tag.appendf("Equalization: %s", value, "id3tag");
    if (!strcmp(tag, "IPL")) id3tag.appendf("Involved people list: %s", value, "id3tag");
    if (!strcmp(tag, "PIC")) id3tag.appendf("Attached picture: %s", value, "id3tag");
    if (!strcmp(tag, "SLT")) id3tag.appendf("Synchronized lyric/text: %s", value, "id3tag");
    if (!strcmp(tag, "TAL")) id3tag.appendf("Album/Movie/Show title: %s", value, "id3tag");
    if (!strcmp(tag, "TBP")) id3tag.appendf("BPM (Beats Per Minute): %s", value, "id3tag");
    if (!strcmp(tag, "TCM")) id3tag.appendf("Composer: %s", value, "id3tag");
    if (!strcmp(tag, "TCO")) id3tag.appendf("Content type: %s", value, "id3tag");
    if (!strcmp(tag, "TCR")) id3tag.appendf("Copyright message: %s", value, "id3tag");
    if (!strcmp(tag, "TDA")) id3tag.appendf("Date: %s", value, "id3tag");
    if (!strcmp(tag, "TDY")) id3tag.appendf("Playlist delay: %s", value, "id3tag");
    if (!strcmp(tag, "TEN")) id3tag.appendf("Encoded by: %s", value, "id3tag");
    if (!strcmp(tag, "TFT")) id3tag.appendf("File type: %s", value, "id3tag");
    if (!strcmp(tag, "TIM")) id3tag.appendf("Time: %s", value, "id3tag");
    if (!strcmp(tag, "TKE")) id3tag.appendf("Initial key: %s", value, "id3tag");
    if (!strcmp(tag, "TLA")) id3tag.appendf("Language(s): %s", value, "id3tag");
    if (!strcmp(tag, "TLE")) id3tag.appendf("Length: %s", value, "id3tag");
    if (!strcmp(tag, "TMT")) id3tag.appendf("Media type: %s", value, "id3tag");
    if (!strcmp(tag, "TOA")) id3tag.appendf("Original artist(s)/performer(s): %s", value, "id3tag");
    if (!strcmp(tag, "TOF")) id3tag.appendf("Original filename: %s", value, "id3tag");
    if (!strcmp(tag, "TOL")) id3tag.appendf("Original Lyricist(s)/text writer(s): %s", value, "id3tag");
    if (!strcmp(tag, "TOR")) id3tag.appendf("Original release year: %s", value, "id3tag");
    if (!strcmp(tag, "TOT")) id3tag.appendf("Original album/Movie/Show title: %s", value, "id3tag");
    if (!strcmp(tag, "TP1")) id3tag.appendf("Lead artist(s)/Lead performer(s)/Soloist(s)/Performing group: %s", value, "id3tag");
    if (!strcmp(tag, "TP2")) id3tag.appendf("Band/Orchestra/Accompaniment: %s", value, "id3tag");
    if (!strcmp(tag, "TP3")) id3tag.appendf("Conductor/Performer refinement: %s", value, "id3tag");
    if (!strcmp(tag, "TP4")) id3tag.appendf("Interpreted, remixed, or otherwise modified by: %s", value, "id3tag");
    if (!strcmp(tag, "TPA")) id3tag.appendf("Part of a set: %s", value, "id3tag");
    if (!strcmp(tag, "TPB")) id3tag.appendf("Publisher: %s", value, "id3tag");
    if (!strcmp(tag, "TRC")) id3tag.appendf("ISRC (International Standard Recording Code): %s", value, "id3tag");
    if (!strcmp(tag, "TRD")) id3tag.appendf("Recording dates: %s", value, "id3tag");
    if (!strcmp(tag, "TRK")) id3tag.appendf("Track number/Position in set: %s", value, "id3tag");
    if (!strcmp(tag, "TSI")) id3tag.appendf("Size: %s", value, "id3tag");
    if (!strcmp(tag, "TSS")) id3tag.appendf("Software/hardware and settings used for encoding: %s", value, "id3tag");
    if (!strcmp(tag, "TT1")) id3tag.appendf("Content group description: %s", value, "id3tag");
    if (!strcmp(tag, "TT2")) id3tag.appendf("Title/Songname/Content description: %s", value, "id3tag");
    if (!strcmp(tag, "TT3")) id3tag.appendf("Subtitle/Description refinement: %s", value, "id3tag");
    if (!strcmp(tag, "TXT")) id3tag.appendf("Lyricist/text writer: %s", value, "id3tag");
    if (!strcmp(tag, "TXX")) id3tag.appendf("User defined text information frame: %s", value, "id3tag");
    if (!strcmp(tag, "TYE")) id3tag.appendf("Year: %s", value, "id3tag");
    if (!strcmp(tag, "UFI")) id3tag.appendf("Unique file identifier: %s", value, "id3tag");
    if (!strcmp(tag, "ULT")) id3tag.appendf("Unsychronized lyric/text transcription: %s", value, "id3tag");
    if (!strcmp(tag, "WAF")) id3tag.appendf("Official audio file webpage: %s", value, "id3tag");
    if (!strcmp(tag, "WAR")) id3tag.appendf("Official artist/performer webpage: %s", value, "id3tag");
    if (!strcmp(tag, "WAS")) id3tag.appendf("Official audio source webpage: %s", value, "id3tag");
    if (!strcmp(tag, "WCM")) id3tag.appendf("Commercial information: %s", value, "id3tag");
    if (!strcmp(tag, "WCP")) id3tag.appendf("Copyright/Legal information: %s", value, "id3tag");
    if (!strcmp(tag, "WPB")) id3tag.appendf("Publishers official webpage: %s", value, "id3tag");
    if (!strcmp(tag, "WXX")) id3tag.appendf("User defined URL link frame: %s", value, "id3tag");

    // V2.3 V2.4 tags
    if (!strcmp(tag, "COMM")) id3tag.appendf("Comment: %s", value, "id3tag");
    if (!strcmp(tag, "OWNE")) id3tag.appendf("Ownership: %s", value, "id3tag");
    if (!strcmp(tag, "PRIV")) id3tag.appendf("Private: %s", value, "id3tag");
    if (!strcmp(tag, "SYLT")) id3tag.appendf("SynLyrics: %s", value, "id3tag");
    if (!strcmp(tag, "TALB")) id3tag.appendf("Album: %s", value, "id3tag");
    if (!strcmp(tag, "TBPM")) id3tag.appendf("BeatsPerMinute: %s", value, "id3tag");
    if (!strcmp(tag, "TCAT")) id3tag.appendf("PodcastCategory: %s", value, "id3tag");
    if (!strcmp(tag, "TCMP")) id3tag.appendf("Compilation: %s", value, "id3tag");
    if (!strcmp(tag, "TCOM")) id3tag.appendf("Composer: %s", value, "id3tag");
    if (!strcmp(tag, "TCON")) id3tag.appendf("ContentType: %s", value, "id3tag");
    if (!strcmp(tag, "TCOP")) id3tag.appendf("Copyright: %s", value, "id3tag");
    if (!strcmp(tag, "TDAT")) id3tag.appendf("Date: %s", value, "id3tag");
    if (!strcmp(tag, "TDES")) id3tag.appendf("PodcastDescription: %s", value, "id3tag");
    if (!strcmp(tag, "TDOR")) id3tag.appendf("Original Release Year: %s", value, "id3tag");
    if (!strcmp(tag, "TDRC")) id3tag.appendf("Publication date: %s", value, "id3tag");
    if (!strcmp(tag, "TDRL")) id3tag.appendf("ReleaseTime: %s", value, "id3tag");
    if (!strcmp(tag, "TENC")) id3tag.appendf("Encoded by: %s", value, "id3tag");
    if (!strcmp(tag, "TEXT")) {} // id3tag.appendf("Lyricist: %s", value, "id3tag");
    if (!strcmp(tag, "TGID")) id3tag.appendf("PodcastID: %s", value, "id3tag");
    if (!strcmp(tag, "TIME")) id3tag.appendf("Time: %s", value, "id3tag");
    if (!strcmp(tag, "TIT1")) id3tag.appendf("Grouping: %s", value, "id3tag");
    if (!strcmp(tag, "TIT2")) id3tag.appendf("Title: %s", value, "id3tag");
    if (!strcmp(tag, "TIT3")) id3tag.appendf("Subtitle: %s", value, "id3tag");
    if (!strcmp(tag, "TLAN")) id3tag.appendf("Language: %s", value, "id3tag");
    if (!strcmp(tag, "TLEN")) id3tag.appendf("Length (ms): %s", value, "id3tag");
    if (!strcmp(tag, "TMED")) id3tag.appendf("Media: %s", value, "id3tag");
    if (!strcmp(tag, "TOAL")) id3tag.appendf("OriginalAlbum: %s", value, "id3tag");
    if (!strcmp(tag, "TOPE")) id3tag.appendf("OriginalArtist: %s", value, "id3tag");
    if (!strcmp(tag, "TOLY")) id3tag.appendf("OriginalLyricist: %s", value, "id3tag");
    if (!strcmp(tag, "TORY")) id3tag.appendf("OriginalReleaseYear: %s", value, "id3tag");
    if (!strcmp(tag, "TPE1")) id3tag.appendf("Artist: %s", value, "id3tag");
    if (!strcmp(tag, "TPE2")) id3tag.appendf("Band: %s", value, "id3tag");
    if (!strcmp(tag, "TPE3")) id3tag.appendf("Conductor: %s", value, "id3tag");
    if (!strcmp(tag, "TPE4")) id3tag.appendf("InterpretedBy: %s", value, "id3tag");
    if (!strcmp(tag, "TPOS")) id3tag.appendf("PartOfSet: %s", value, "id3tag");
    if (!strcmp(tag, "TPUB")) id3tag.appendf("Publisher: %s", value, "id3tag");
    if (!strcmp(tag, "TRCK")) id3tag.appendf("Track: %s", value, "id3tag");
    if (!strcmp(tag, "TRSN")) id3tag.appendf("InternetRadioStationName: %s", value, "id3tag");
    if (!strcmp(tag, "TSSE")) id3tag.appendf("SettingsForEncoding: %s", value, "id3tag");
    if (!strcmp(tag, "TRDA")) id3tag.appendf("RecordingDates: %s", value, "id3tag");
    if (!strcmp(tag, "TSO2")) id3tag.appendf("AlbumArtistSortOrder: %s", value, "id3tag");
    if (!strcmp(tag, "TSOA")) id3tag.appendf("AlbumSortOrder: %s", value, "id3tag");
    if (!strcmp(tag, "TSOP")) id3tag.appendf("PerformerSortOrder: %s", value, "id3tag");
    if (!strcmp(tag, "TSOC")) id3tag.appendf("ComposerSortOrder: %s", value, "id3tag");
    if (!strcmp(tag, "TSOT")) id3tag.appendf("TitleSortOrder: %s", value, "id3tag");
    if (!strcmp(tag, "TXXX")) id3tag.appendf("UserDefinedText: %s", value, "id3tag");
    if (!strcmp(tag, "TYER")) id3tag.appendf("Year: %s", value, "id3tag");
    if (!strcmp(tag, "USER")) id3tag.appendf("TermsOfUse: %s", value, "id3tag");
    if (!strcmp(tag, "USLT")) id3tag.appendf("Lyrics: %s", value, "id3tag");
    if (!strcmp(tag, "WCOM")) id3tag.appendf("Commercial URL: %s", value, "id3tag");
    if (!strcmp(tag, "WFED")) id3tag.appendf("Podcast URL: %s", value, "id3tag");
    if (!strcmp(tag, "WOAF")) id3tag.appendf("File URL: %s", value, "id3tag");
    if (!strcmp(tag, "WOAR")) id3tag.appendf("OfficialArtistWebpage: %s", value, "id3tag");
    if (!strcmp(tag, "WOAS")) id3tag.appendf("Source URL: %s", value, "id3tag");
    if (!strcmp(tag, "WORS")) id3tag.appendf("InternetRadioStationURL: %s", value, "id3tag");
    if (!strcmp(tag, "WXXX")) id3tag.appendf("User defined URL link frame: %s", value, "id3tag");
    if (!strcmp(tag, "XDOR")) id3tag.appendf("OriginalReleaseTime: %s", value, "id3tag");

    if (!id3tag.valid()) {
        AUDIO_LOG_DEBUG("unknown tag: %s", tag);
        return;
    }
    latinToUTF8(id3tag);
    if (id3tag.contains("?xml")) {
        showstreamtitle(id3tag.get());
        return;
    }
    latinToUTF8(id3tag);
    if (id3tag.starts_with("Grouping")) {
        showstreamtitle(id3tag.get());
        return;
    }
    if (id3tag.strlen()) { info(*this, evt_id3data, "%s", id3tag.get()); }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::latinToUTF8(ps_ptr<char>& buff, bool UTF8check) {
    // most stations send  strings in UTF-8 but a few sends in latin. To standardize this, all latin strings are
    // converted to UTF-8. If UTF-8 is already present, nothing is done and true is returned.
    // A conversion to UTF-8 extends the string. Therefore it is necessary to know the buffer size. If the converted
    // string does not fit into the buffer, false is returned

    uint16_t pos = 0;
    uint16_t in = 0;
    uint16_t out = 0;
    bool     isUTF8 = true;

    // We cannot detect if a given string (or byte sequence) is a UTF-8 encoded text as for example each and every series
    // of UTF-8 octets is also a valid (if nonsensical) series of Latin-1 (or some other encoding) octets.
    // However not every series of valid Latin-1 octets are valid UTF-8 series. So you can rule out strings that do not conform
    // to the UTF-8 encoding schema:

    if (UTF8check) {
        while (buff[pos] != '\0') {
            if (buff[pos] <= 0x7F) { // 0xxxxxxx: ASCII
                pos++;
            } else if ((buff[pos] & 0xE0) == 0xC0) {
                if (pos + 1 >= buff.strlen()) isUTF8 = false;
                // 110xxxxx 10xxxxxx: 2-byte
                if ((buff[pos + 1] & 0xC0) != 0x80) isUTF8 = false;
                if (buff[pos] < 0xC2) isUTF8 = false; // Overlong encoding
                pos += 2;
            } else if ((buff[pos] & 0xF0) == 0xE0) {
                // 1110xxxx 10xxxxxx 10xxxxxx: 3-byte
                if ((buff[pos + 1] & 0xC0) != 0x80 || (buff[pos + 2] & 0xC0) != 0x80) isUTF8 = false;
                if (buff[pos] == 0xE0 && buff[pos + 1] < 0xA0) isUTF8 = false;  // Overlong
                if (buff[pos] == 0xED && buff[pos + 1] >= 0xA0) isUTF8 = false; // UTF-16 surrogate
                pos += 3;
            } else if ((buff[pos] & 0xF8) == 0xF0) {
                // 11110xxx 10xxxxxx 10xxxxxx 10xxxxxx: 4-byte
                if ((buff[pos + 1] & 0xC0) != 0x80 || (buff[pos + 2] & 0xC0) != 0x80 || (buff[pos + 3] & 0xC0) != 0x80) isUTF8 = false;
                if (buff[pos] == 0xF0 && buff[pos + 1] < 0x90) isUTF8 = false;                       // Overlong
                if (buff[pos] > 0xF4 || (buff[pos] == 0xF4 && buff[pos + 1] > 0x8F)) isUTF8 = false; // > U+10FFFF
                pos += 4;
            } else {
                isUTF8 = false; // Invalid first byte
            }
            if (!isUTF8) break;
        }
        if (isUTF8) return; // is UTF-8, do nothing
    }
    ps_ptr<char> iso8859_1;
    iso8859_1.assign(buff.get());

    // Worst-case: all chars are latin1 > 0x7F became 2 Bytes → max length is twice +1
    std::size_t requiredSize = strlen(iso8859_1.get()) * 2 + 1;

    if (buff.size() < requiredSize) { buff.realloc(requiredSize); }

    // coding into UTF-8
    while (iso8859_1[in] != '\0') {
        if (iso8859_1[in] < 0x80) {
            buff[out++] = iso8859_1[in++];
        } else {
            buff[out++] = (0xC0 | (iso8859_1[in] >> 6));
            buff[out++] = (0x80 | (iso8859_1[in] & 0x3F));
            in++;
        }
    }
    buff[out] = '\0';
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::htmlToUTF8(char* str) { // convert HTML to UTF-8

    typedef struct { // --- EntityMap Definition ---
        const char* name;
        uint32_t    codepoint;
    } EntityMap;

    const EntityMap entities[] = {
        {"amp", 0x0026},    // &
        {"lt", 0x003C},     // <
        {"gt", 0x003E},     // >
        {"quot", 0x0022},   // "
        {"apos", 0x0027},   // '
        {"nbsp", 0x00A0},   // non-breaking space
        {"euro", 0x20AC},   // €
        {"copy", 0x00A9},   // ©
        {"reg", 0x00AE},    // ®
        {"trade", 0x2122},  // ™
        {"hellip", 0x2026}, // …
        {"ndash", 0x2013},  // –
        {"mdash", 0x2014},  // —
        {"sect", 0x00A7},   // §
        {"para", 0x00B6}    // ¶
    };

    // --- EntityMap Lookup ---
    auto find_entity = [&](const char* p, uint32_t* codepoint, int* entity_len) {
        for (size_t i = 0; i < sizeof(entities) / sizeof(entities[0]); i++) {
            const char* name = entities[i].name;
            size_t      len = strlen(name);
            if (strncmp(p + 1, name, len) == 0 && p[len + 1] == ';') {
                *codepoint = entities[i].codepoint;
                *entity_len = (int)(len + 2); // &name;
                return 1;
            }
        }
        return 0;
    };

    auto codepoint_to_utf8 = [&](uint32_t cp, char* dst) { // Convert a Codepoint (Unicode) to UTF-8, writes in DST, there is number of bytes back
        if (cp <= 0x7F) {
            dst[0] = cp;
            return 1;
        } else if (cp <= 0x7FF) {
            dst[0] = 0xC0 | (cp >> 6);
            dst[1] = 0x80 | (cp & 0x3F);
            return 2;
        } else if (cp <= 0xFFFF) {
            dst[0] = 0xE0 | (cp >> 12);
            dst[1] = 0x80 | ((cp >> 6) & 0x3F);
            dst[2] = 0x80 | (cp & 0x3F);
            return 3;
        } else if (cp <= 0x10FFFF) {
            dst[0] = 0xF0 | (cp >> 18);
            dst[1] = 0x80 | ((cp >> 12) & 0x3F);
            dst[2] = 0x80 | ((cp >> 6) & 0x3F);
            dst[3] = 0x80 | (cp & 0x3F);
            return 4;
        }
        return -1; // invalid Codepoint
    };

    char* p = str;
    while (*p != '\0') {
        if (p[0] == '&') {
            uint32_t cp;
            int      consumed;
            if (find_entity(p, &cp, &consumed)) { // looking for entity, such as &copy;
                char utf8[5] = {0};
                int  len = codepoint_to_utf8(cp, utf8);
                if (len > 0) {
                    size_t tail_len = strlen(p + consumed);
                    memmove(p + len, p + consumed, tail_len + 1);
                    memcpy(p, utf8, len);
                    p += len;
                    continue;
                }
            }
        }
        if (p[0] == '&' && p[1] == '#') {
            char*    endptr;
            uint32_t codepoint = strtol(p + 2, &endptr, 10);

            if (*endptr == ';' && codepoint <= 0x10FFFF) {
                char utf8[5] = {0};
                int  utf8_len = codepoint_to_utf8(codepoint, utf8);
                if (utf8_len > 0) {
                    //    size_t entity_len = endptr - p + 1;
                    size_t tail_len = strlen(endptr + 1);

                    // Show residual ring to the left
                    memmove(p + utf8_len, endptr + 1, tail_len + 1); // +1 because of '\0'

                    // Copy UTF-8 characters
                    memcpy(p, utf8, utf8_len);

                    // weiter bei neuem Zeichen
                    p += utf8_len;
                    continue;
                }
            }
        }
        p++;
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
size_t Audio::readAudioHeader(uint32_t bytes) {
    size_t bytesReaded = 0;
    if (m_codec == CODEC_WAV) {
        int res = read_WAV_Header(InBuff.getReadPtr(), bytes);
        if (res >= 0)
            bytesReaded = res;
        else { // error, skip header
            m_controlCounter = 100;
        }
    }
    if (m_codec == CODEC_MP3) {
        int res = read_ID3_Header(InBuff.getReadPtr(), bytes);
        if (res > 0 || m_controlCounter < 100)
            bytesReaded = res;
        else { // error, skip header
            m_controlCounter = 100;
        }
    }
    if (m_codec == CODEC_M4A) {
        int res = read_M4A_Header(InBuff.getReadPtr(), bytes);
        if (res >= 0)
            bytesReaded = res;
        else { // error, skip header
            m_controlCounter = 100;
        }
    }
    if (m_codec == CODEC_AAC) {
        // stream only, no header
        m_audioDataSize = m_audioFileSize;
        m_controlCounter = 100;
    }
    if (m_codec == CODEC_FLAC) {
        int res = read_FLAC_Header(InBuff.getReadPtr(), bytes);
        if (res >= 0)
            bytesReaded = res;
        else { // error
            stopSong();
        }
    }
    if (m_codec == CODEC_OPUS) { m_controlCounter = 100; }
    if (m_codec == CODEC_VORBIS) { m_controlCounter = 100; }
    if (m_codec == CODEC_OGG) { m_controlCounter = 100; }
    if (!isRunning()) {
        AUDIO_LOG_ERROR("Processing stopped due to invalid audio header");
        return 0;
    }
    return bytesReaded;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::read_WAV_Header(uint8_t* data, size_t len) {

    if (m_controlCounter == 0) {
        m_rwh.cs = 0;
        m_rwh.bts = 0;
        m_controlCounter++;
        if ((*data != 'R') || (*(data + 1) != 'I') || (*(data + 2) != 'F') || (*(data + 3) != 'F')) {
            AUDIO_LOG_ERROR("file has no RIFF tag");
            m_rwh.headerSize = 0;
            return -1; // false;
        } else {
            m_rwh.headerSize = 4;
            return 4; // ok
        }
    }

    if (m_controlCounter == 1) {
        m_controlCounter++;
        m_rwh.cs = (uint32_t)(*data + (*(data + 1) << 8) + (*(data + 2) << 16) + (*(data + 3) << 24) - 8);
        m_rwh.headerSize += 4;
        return 4; // ok
    }

    if (m_controlCounter == 2) {
        m_controlCounter++;
        if ((*data != 'W') || (*(data + 1) != 'A') || (*(data + 2) != 'V') || (*(data + 3) != 'E')) {
            AUDIO_LOG_ERROR("format tag is not WAVE");
            return -1; // false;
        } else {
            m_rwh.headerSize += 4;
            return 4;
        }
    }

    if (m_controlCounter == 3) {
        if ((*data == 'f') && (*(data + 1) == 'm') && (*(data + 2) == 't')) {
            m_controlCounter++;
            m_rwh.headerSize += 4;
            return 4;
        } else {
            m_rwh.headerSize += 4;
            return 4;
        }
    }

    if (m_controlCounter == 4) {
        m_controlCounter++;
        m_rwh.cs = (uint32_t)(*data + (*(data + 1) << 8));
        if (m_rwh.cs > 40) return -1; // false, something going wrong
        m_rwh.bts = m_rwh.cs - 16;    // bytes to skip if fmt chunk is >16
        m_rwh.headerSize += 4;
        return 4;
    }

    if (m_controlCounter == 5) {
        m_controlCounter++;
        uint16_t fc = (uint16_t)(*(data + 0) + (*(data + 1) << 8));                                               // Format code
        uint16_t ch = (uint16_t)(*(data + 2) + (*(data + 3) << 8));                                               // Number of interleaved channels
        uint32_t sr = (uint32_t)(*(data + 4) + (*(data + 5) << 8) + (*(data + 6) << 16) + (*(data + 7) << 24));   // Samplerate
        uint32_t dr = (uint32_t)(*(data + 8) + (*(data + 9) << 8) + (*(data + 10) << 16) + (*(data + 11) << 24)); // Datarate
        uint16_t dbs = (uint16_t)(*(data + 12) + (*(data + 13) << 8));                                            // Data block size
        uint16_t bps = (uint16_t)(*(data + 14) + (*(data + 15) << 8));                                            // Bits per sample

        info(*this, evt_info, "FormatCode: %u", fc);
        // info(*this, evt_info, "Channel: %u", nic);
        // info(*this, evt_info, "SampleRate (Hz): %u", sr);
        info(*this, evt_info, "DataRate: %lu", (long unsigned int)dr);
        info(*this, evt_info, "DataBlockSize: %u", dbs);
        info(*this, evt_info, "BitsPerSample: %u", bps);

        if ((bps != 8) && (bps != 16) && (bps != 24) && (bps != 32)) {
            info(*this, evt_info, "BitsPerSample is %u,  must be 8, 16, 24 or 32", bps);
            stopSong();
            return -1;
        }
        if ((ch != 1) && (ch != 2)) {
            info(*this, evt_info, "num channels is %u,  must be 1 or 2", ch);
            stopSong();
            return -1;
        }
        if (fc != 1) {
            AUDIO_LOG_ERROR("format code is not 1 (PCM)");
            stopSong();
            return -1; // false;
        }
        m_decoder->setRawBlockParams(ch, sr, bps, 0, 0);
        m_nominal_bitrate = ch * sr * bps;
        m_rwh.headerSize += 16;
        return 16; // ok
    }

    if (m_controlCounter == 6) {
        m_controlCounter++;
        m_rwh.headerSize += m_rwh.bts;
        return m_rwh.bts; // skip to data
    }

    if (m_controlCounter == 7) {
        if ((*(data + 0) == 'd') && (*(data + 1) == 'a') && (*(data + 2) == 't') && (*(data + 3) == 'a')) {
            m_controlCounter++;
            //    vTaskDelay(30);
            m_rwh.headerSize += 4;
            return 4;
        } else {
            m_rwh.headerSize++;
            return 1;
        }
    }

    if (m_controlCounter == 8) {
        m_controlCounter++;
        size_t cs = *(data + 0) + (*(data + 1) << 8) + (*(data + 2) << 16) + (*(data + 3) << 24); // read chunkSize
        m_rwh.headerSize += 4;
        if (cs) {
            m_audioDataSize = cs - 44;
        } else { // sometimes there is nothing here
            m_audioDataSize = m_audioFileSize - m_rwh.headerSize;
        }

        m_audioFileDuration = m_audioDataSize / (getSampleRate() * getChannels());
        if (getBitsPerSample() == 16) m_audioFileDuration /= 2;
        info(*this, evt_info, "Duration (s): %u", m_audioFileDuration);
        info(*this, evt_bitrate, "%i", m_nominal_bitrate);
        info(*this, evt_info, "Bitrate (b/s): %lu", m_nominal_bitrate);
        return 4;
    }
    m_controlCounter = 100; // header succesfully read
    m_audioDataStart = m_rwh.headerSize;
    info(*this, evt_info, "Audio-Data-Start: %u", m_audioDataStart);
    info(*this, evt_info, "Audio-Length: %u", m_audioDataSize);
    return 0;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::read_FLAC_Header(uint8_t* data, size_t len) {

    if (m_rflh.retvalue && m_controlCounter != 0) {
        if (m_rflh.retvalue > len) { // if returnvalue > bufferfillsize
            if (len > InBuff.getMaxBlockSize()) len = InBuff.getMaxBlockSize();
            m_rflh.retvalue -= len; // and wait for more bufferdata
            return len;
        } else {
            size_t tmp = m_rflh.retvalue;
            m_rflh.retvalue = 0;
            return tmp;
        }
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_BEGIN) { // init
        m_rflh.reset();
        m_controlCounter = FLAC_MAGIC;
        m_rflh.picVec.clear();
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_MAGIC) {            /* check MAGIC STRING */
        if (specialIndexOf(data, "OggS", 10) == 0) { // is ogg
            m_rflh.headerSize = 0;
            m_rflh.retvalue = 0;
            m_controlCounter = FLAC_OKAY;
            return 0;
        }
        if (specialIndexOf(data, "fLaC", 10) != 0) {
            if (specialIndexOf(data, "ID3", 10) == 0) { // has ID3 before fLaC
                // Synchsafe-Size
                uint32_t size = (data[6] << 21) | (data[7] << 14) | (data[8] << 7) | data[9];
                size += 10;             // header + body
                m_rflh.retvalue = size; // skip ID3 header + body
                return 0;
            }
            AUDIO_LOG_ERROR("Magic String 'fLaC' not found in header");
            stopSong();
            return -1;
        }

        m_controlCounter = FLAC_MBH; // METADATA_BLOCK_HEADER
        m_rflh.headerSize = 4;
        m_rflh.retvalue = 4;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_MBH) { /* METADATA_BLOCK_HEADER */
        uint8_t blockType = *data;

        if (!m_rflh.f_lastMetaBlock) {
            if ((blockType & 127) == 0) m_controlCounter = FLAC_SINFO;
            if ((blockType & 127) == 1) m_controlCounter = FLAC_PADDING;
            if ((blockType & 127) == 2) m_controlCounter = FLAC_APP;
            if ((blockType & 127) == 3) m_controlCounter = FLAC_SEEK;
            if ((blockType & 127) == 4) m_controlCounter = FLAC_VORBIS;
            if ((blockType & 127) == 5) m_controlCounter = FLAC_CUESHEET;
            if ((blockType & 127) == 6) m_controlCounter = FLAC_PICTURE;
            if ((blockType & 128)) { m_rflh.f_lastMetaBlock = true; }
            m_rflh.headerSize += 1;
            m_rflh.retvalue = 1;
            return 0;
        }

        m_controlCounter = FLAC_OKAY;

        m_audioDataStart = m_rflh.headerSize;
        m_audioDataSize = m_audioFileSize - m_audioDataStart;
        m_decoder->setRawBlockParams(m_rflh.numChannels, m_rflh.sampleRate, m_rflh.bitsPerSample, m_rflh.totalSamplesInStream, (uint32_t)m_audioDataSize);
        if (m_rflh.picLen) {
            size_t pos = m_audioDataStart;
            info(*this, evt_image, m_rflh.picVec);
        }

        info(*this, evt_info, "Audio-Data-Start: %u", m_audioDataStart);
        info(*this, evt_info, "Audio-Length: %u", m_audioDataSize);
        if (m_rflh.duration) {
            m_rflh.nominalBitrate = (m_audioDataSize * 8) / m_rflh.duration;
            m_nominal_bitrate = m_rflh.nominalBitrate;
            m_audioFileDuration = m_rflh.duration;
            info(*this, evt_info, "Duration (s): %u", m_rflh.duration);
            info(*this, evt_bitrate, "%i", m_nominal_bitrate);
            info(*this, evt_info, "Bitrate (b/s): %lu", m_nominal_bitrate);
        }
        m_rflh.retvalue = 0;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_SINFO) { /* Stream info block */
        size_t l = bigEndian(data, 3);
        vTaskDelay(2);
        m_rflh.maxBlockSize = bigEndian(data + 5, 2);
        info(*this, evt_info, "FLAC maxBlockSize: %u", m_rflh.maxBlockSize);
        vTaskDelay(2);
        m_rflh.maxFrameSize = bigEndian(data + 10, 3);
        if (m_rflh.maxFrameSize) {
            info(*this, evt_info, "FLAC maxFrameSize: %u", m_rflh.maxFrameSize);
        } else {
            info(*this, evt_info, "FLAC maxFrameSize: N/A");
        }
        if (m_rflh.maxFrameSize > InBuff.getMaxBlockSize()) {
            AUDIO_LOG_ERROR("FLAC maxFrameSize too large!");
            stopSong();
            return -1;
        }
        vTaskDelay(2);
        uint32_t nextval = bigEndian(data + 13, 3);
        m_rflh.sampleRate = nextval >> 4;
        info(*this, evt_info, "FLAC sampleRate (Hz): %lu", (long unsigned int)m_rflh.sampleRate);
        vTaskDelay(2);
        m_rflh.numChannels = ((nextval & 0x06) >> 1) + 1;
        info(*this, evt_info, "FLAC numChannels: %u", m_rflh.numChannels);
        vTaskDelay(2);
        uint8_t bps = (nextval & 0x01) << 4;
        bps += (*(data + 16) >> 4) + 1;
        m_rflh.bitsPerSample = bps;
        if ((bps != 8) && (bps != 16) && (bps != 24)) {
            AUDIO_LOG_ERROR("bits per sample must be 8, 16 or 24, is %i", bps);
            stopSong();
            return -1;
        }
        info(*this, evt_info, "FLAC bitsPerSample: %u", m_rflh.bitsPerSample);
        m_rflh.totalSamplesInStream = bigEndian(data + 17, 4);
        if (m_rflh.totalSamplesInStream) {
            info(*this, evt_info, "total samples in stream: %lu", (long unsigned int)m_rflh.totalSamplesInStream);
        } else {
            info(*this, evt_info, "total samples in stream: N/A");
        }
        if (bps != 0 && m_rflh.totalSamplesInStream && m_rflh.sampleRate) { m_rflh.duration = (long unsigned int)m_rflh.totalSamplesInStream / (long unsigned int)m_rflh.sampleRate; }
        m_controlCounter = FLAC_MBH; // METADATA_BLOCK_HEADER
        m_rflh.retvalue = l + 3;
        m_rflh.headerSize += m_rflh.retvalue;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_PADDING) { /* PADDING */
        size_t l = bigEndian(data, 3);
        m_controlCounter = FLAC_MBH;
        m_rflh.retvalue = l + 3;
        m_rflh.headerSize += m_rflh.retvalue;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_APP) { /* APPLICATION */
        size_t l = bigEndian(data, 3);
        m_controlCounter = FLAC_MBH;
        m_rflh.retvalue = l + 3;
        m_rflh.headerSize += m_rflh.retvalue;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_SEEK) { /* SEEKTABLE */
        size_t l = bigEndian(data, 3);
        m_controlCounter = FLAC_MBH;
        m_rflh.retvalue = l + 3;
        m_rflh.headerSize += m_rflh.retvalue;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_VORBIS) { /* VORBIS COMMENT */ // field names

        auto parse_flac_timestamp = [&](const char* s) {
            if (!s || *s != '[') return -1;
            int mm = 0, ss = 0, hh = 0;
            if (sscanf(s, "[%d:%d.%d]", &mm, &ss, &hh) == 3) { return mm * 60000 + ss * 1000 + hh * 10; }
            return -1;
        };

        ps_ptr<char> vendorString;
        ps_ptr<char> commentString;
        ps_ptr<char> lyricsBuffer;
        size_t       vendorLength = bigEndian(data, 3);
        size_t       idx = 0;
        data += 3;
        idx += 3;
        size_t vendorStringLength = data[0] + (data[1] << 8) + (data[2] << 16) + (data[3] << 24);
        if (vendorStringLength) {
            data += 4;
            idx += 4;
        }
        if (vendorStringLength > 495) vendorStringLength = 495; // guard
        vendorString.assign((const char*)data, vendorStringLength);
        vendorString.insert("VENDOR_STRING: ", 0);
        info(*this, evt_id3data, "%s", vendorString.get());
        data += vendorStringLength;
        idx += vendorStringLength;
        size_t commentListLength = data[0] + (data[1] << 8) + (data[2] << 16) + (data[3] << 24);
        data += 4;
        idx += 4;

        for (int i = 0; i < commentListLength; i++) {
            (void)i;
            size_t commentLength = data[0] + (data[1] << 8) + (data[2] << 16) + (data[3] << 24);
            data += 4;
            idx += 4;
            if (commentLength) { // guard
                commentString.assign((const char*)data, commentLength);
                if (commentString.starts_with("LYRICS="))
                    lyricsBuffer.clone_from(commentString);
                else
                    info(*this, evt_id3data, "%s", commentString.get());
            }
            data += commentLength;
            idx += commentLength;
            if (idx > vendorLength + 3) { AUDIO_LOG_ERROR("VORBIS COMMENT section is too long"); }
        }
        ps_ptr<char> tmp;
        ps_ptr<char> timestamp;
        timestamp.alloc(12);
        if (lyricsBuffer.valid()) {
            lyricsBuffer.remove_prefix("LYRICS=");
            idx = 0;
            while (idx < lyricsBuffer.size()) {
                int pos = lyricsBuffer.index_of('\n', idx);
                if (pos == -1) break;
                int len = pos - idx + 1;
                tmp.copy_from(lyricsBuffer.get() + idx, len);
                idx += len;
                if (tmp.ends_with("\n")) tmp.truncate_at('\n');
                // tmp content e.g.: [00:27.07]Jetzt kommt die Zeit
                //                   [00:28.84]Auf die ihr euch freut
                timestamp.copy_from(tmp.get(), 10);

                int ms = parse_flac_timestamp(timestamp.c_get());

                // timestamp.remove_chars("[]:.");
                // timestamp.append("0"); // to ms 0028840
                tmp.remove_before(10, true);
                if (tmp.strlen() > 0) {
                    m_syltLines.push_back(std::move(tmp));
                    m_syltTimeStamp.push_back(ms);
                }
            }
            info(*this, evt_info, "audiofile contains synchronized lyrics");
            // for(int i = 0; i < m_syltLines.size(); i++){
            //     info(*this, evt_lyrics, "%07i ms,   %s", m_syltTimeStamp[i], m_syltLines[i].c_get());
            // }
        }
        m_controlCounter = FLAC_MBH;
        m_rflh.retvalue = vendorLength + 3;
        m_rflh.headerSize += m_rflh.retvalue;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_CUESHEET) { /* CUESHEET */
        size_t l = bigEndian(data, 3);
        m_controlCounter = FLAC_MBH;
        m_rflh.retvalue = l + 3;
        m_rflh.headerSize += m_rflh.retvalue;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == FLAC_PICTURE) { /* PICTURE */
        m_rflh.picLen = bigEndian(data, 3);
        m_rflh.picPos = m_rflh.headerSize + 3;
        m_rflh.picVec.push_back(m_rflh.picPos);
        m_rflh.picVec.push_back(m_rflh.picLen);
        AUDIO_LOG_DEBUG("FLAC PICTURE, pos %i, size %i", m_rflh.picPos, m_rflh.picLen);
        m_controlCounter = FLAC_MBH;
        m_rflh.retvalue = m_rflh.picLen + 3;
        m_rflh.headerSize += m_rflh.retvalue;
        return 0;
    }
    return 0;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————

int Audio::read_ID3_Header(uint8_t* data, size_t len) {
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == MP3_BEGIN) { /* read ID3 tag and ID3 header size */
        m_controlCounter = MP3_ID3HEADER;
        m_ID3Hdr.reset(); // reset all
        m_ID3Hdr.iBuffSize = 4096;
        m_ID3Hdr.iBuff.alloc(m_ID3Hdr.iBuffSize + 10, "m_ID3Hdr.iBuff");
        memset(m_ID3Hdr.tag, 0, sizeof(m_ID3Hdr.tag));
        return 0;
    }
    if (m_controlCounter == MP3_NEXTID3) {
        m_controlCounter = MP3_ID3HEADER;
        // reset specific
        m_ID3Hdr.id3Size = 0;
        //    m_ID3Hdr.totalId3Size = 0; // if we have more header, id3_1_size + id3_2_size + ....
        m_ID3Hdr.remainingHeaderBytes = 0;
        m_ID3Hdr.v22_tag_length = 0;
        m_ID3Hdr.ID3version = 0;
        m_ID3Hdr.ehsz = 0;
        m_ID3Hdr.framesize = 0;
        m_ID3Hdr.compressed = false;
        m_ID3Hdr.SYLT.size = 0;
        m_ID3Hdr.SYLT.pos = 0;
        m_ID3Hdr.iBuffSize = 4096;
        m_ID3Hdr.iBuff.alloc(m_ID3Hdr.iBuffSize + 10, "m_ID3Hdr.iBuff");
        memset(m_ID3Hdr.tag, 0, sizeof(m_ID3Hdr.tag));
        return 0;
    }

    if (m_controlCounter == MP3_ID3HEADER) {
        m_controlCounter = MP3_EXTHEADER;
        int retval = 0;
        if (!m_f_m3u8data) info(*this, evt_info, "File-Size: %lu", m_audioFileSize);

        m_ID3Hdr.remainingHeaderBytes = 0;
        m_ID3Hdr.ehsz = 0;
        if (specialIndexOf(data, "ID3", 4) != 0) { // ID3 not found
            if (!m_f_m3u8data) { info(*this, evt_info, "file has no ID3 tag, skip metadata"); }
            m_audioDataSize = m_audioFileSize;
            // if(!m_f_m3u8data) info(*this, evt_info, "Audio-Length: %u", m_audioDataSize);
            m_controlCounter = MP3_XING; // have xing?
            return 0;                    // error, no ID3 signature found
        }
        m_ID3Hdr.ID3version = *(data + 3);
        switch (m_ID3Hdr.ID3version) {
            case 2:
                m_f_unsync = (*(data + 5) & 0x80);
                m_f_exthdr = false;
                break;
            case 3:
            case 4:
                m_f_unsync = (*(data + 5) & 0x80); // bit7
                m_f_exthdr = (*(data + 5) & 0x40); // bit6 extended header
                break;
        };
        m_ID3Hdr.id3Size = bigEndian(data + 6, 4, 7); //  ID3v2 size  4 * %0xxxxxxx (shift left seven times!!)
        m_ID3Hdr.id3Size += 10;

        retval = 10;

        // Every read from now may be unsync'd
        if (!m_f_m3u8data) info(*this, evt_info, "ID3 framesSize: %i", m_ID3Hdr.id3Size);
        if (!m_f_m3u8data) info(*this, evt_info, "ID3 version: 2.%i", m_ID3Hdr.ID3version);

        if (m_ID3Hdr.ID3version == 2) { m_controlCounter = MP3_ID3V22; }
        m_ID3Hdr.remainingHeaderBytes = m_ID3Hdr.id3Size;
        m_ID3Size = m_ID3Hdr.id3Size;
        m_ID3Hdr.remainingHeaderBytes -= 10;

        if (m_f_exthdr) {
            m_ID3Hdr.ehsz = bigEndian(data + 10, 4, 7); // syncSave
            info(*this, evt_info, "ID3 extended header, size: %i", m_ID3Hdr.ehsz);
            m_ID3Hdr.ehsz -= 4;
            m_ID3Hdr.remainingHeaderBytes -= 4;
            m_ID3Hdr.id3Size += 4;
            retval += 4;
        }
        return retval;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == MP3_EXTHEADER) { // skip extended header if exists
        if (m_ID3Hdr.ehsz > len) {
            m_ID3Hdr.ehsz -= len;
            m_ID3Hdr.remainingHeaderBytes -= len;
            return len;
        } // Throw it away
        else {
            m_controlCounter = MP3_ID3FRAME;
            m_ID3Hdr.remainingHeaderBytes -= m_ID3Hdr.ehsz;
            return m_ID3Hdr.ehsz;
        } // Throw it away
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == MP3_ID3FRAME) { // read a ID3 frame, get the tag
        if (m_ID3Hdr.remainingHeaderBytes == 0) {
            m_controlCounter = MP3_XING;
            return 0;
        }
        m_controlCounter = MP3_FRAMESIZE;
        m_ID3Hdr.frameid[0] = *(data + 0);
        m_ID3Hdr.frameid[1] = *(data + 1);
        m_ID3Hdr.frameid[2] = *(data + 2);
        m_ID3Hdr.frameid[3] = *(data + 3);
        m_ID3Hdr.frameid[4] = 0;
        for (uint8_t i = 0; i < 4; i++) m_ID3Hdr.tag[i] = m_ID3Hdr.frameid[i]; // tag = frameid
        if (m_ID3Hdr.frameid[0] == 0 && m_ID3Hdr.frameid[1] == 0 && m_ID3Hdr.frameid[2] == 0 && m_ID3Hdr.frameid[3] == 0) {
            // We're in padding
            m_controlCounter = MP3_LASTFRAMES; // all ID3 metadata processed
        }
        m_ID3Hdr.remainingHeaderBytes -= 4;
        return 4;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == MP3_FRAMESIZE) { // get the frame size
        m_controlCounter = MP3_TAG;

        if (m_ID3Hdr.ID3version == 4) {
            m_ID3Hdr.framesize = bigEndian(data, 4, 7); // << 7
        } else {
            m_ID3Hdr.framesize = bigEndian(data, 4); // << 8
        }

        uint8_t frameFlag_0 = *(data + 4);
        uint8_t frameFlag_1 = *(data + 5);
        (void)frameFlag_0;
        m_ID3Hdr.compressed = frameFlag_1 & 0x80; // Frame is compressed using [#ZLIB zlib] with 4 bytes for 'decompressed
        uint32_t decompsize = 0;
        if (m_ID3Hdr.compressed) {
            // AUDIO_LOG_INFO("iscompressed");
            decompsize = bigEndian(data + 6, 4);
            (void)decompsize;
            // AUDIO_LOG_INFO("decompsize=%u", decompsize);
            m_ID3Hdr.remainingHeaderBytes -= 6 + 4;
            return 6 + 4;
        }
        m_ID3Hdr.remainingHeaderBytes -= 6;
        return 6;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == MP3_SKIP) { // If the frame is larger than m_ID3Hdr.framesize, skip the rest
        if (m_ID3Hdr.framesize > len) {
            m_ID3Hdr.framesize -= len;
            m_ID3Hdr.remainingHeaderBytes -= len;
            return len;
        } else {
            m_controlCounter = MP3_ID3FRAME; // check next frame
            m_ID3Hdr.remainingHeaderBytes -= m_ID3Hdr.framesize;
            return m_ID3Hdr.framesize;
        }
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == MP3_TAG) { // Read the value
        m_controlCounter = MP3_SKIP;   // only read 256 bytes

        uint8_t textEncodingByte = *(data + 0); // ID3v2 Text-Encoding-Byte
        // $00 – ISO-8859-1 (LATIN-1, Identical to ASCII for values smaller than 0x80).
        // $01 – UCS-2 encoded Unicode with BOM (Byte Order Mark), in ID3v2.2 and ID3v2.3.
        // $02 – UTF-16BE encoded Unicode without BOM (Byte Order Mark) , in ID3v2.4.
        // $03 – UTF-8 encoded Unicode, in ID3v2.4.

        if (startsWith(m_ID3Hdr.tag, "APIC")) { // a image embedded in file, passing it to external function
            m_ID3Hdr.APIC_vec.push_back(m_ID3Hdr.totalId3Size + m_ID3Hdr.id3Size - m_ID3Hdr.remainingHeaderBytes);
            m_ID3Hdr.APIC_vec.push_back(m_ID3Hdr.framesize);
            AUDIO_LOG_DEBUG("APIC_pos %i, APIC_size %i", m_ID3Hdr.APIC_vec[2 * m_ID3Hdr.numID3Header], m_ID3Hdr.APIC_vec[2 * m_ID3Hdr.numID3Header + 1]);
            return 0;
        }

        if (startsWith(m_ID3Hdr.tag, "SYLT") || startsWith(m_ID3Hdr.tag, "USLT")) { // any lyrics embedded in file, passing it to external function
            m_controlCounter = MP3_SYLT;
            return 0;
        }

        if ( // proprietary not standard information
            startsWith(m_ID3Hdr.tag, "PRIV")) {
            ; // AUDIO_LOG_ERROR("PRIV");
            return 0;
        }
        if (startsWith(m_ID3Hdr.tag, "TSIZ") || startsWith(m_ID3Hdr.tag, "XTRA")) { // tag with invalid length, skip everything and find the first sync word
            m_ID3Hdr.framesize = 0;
            size_t tmp = m_ID3Hdr.remainingHeaderBytes;
            m_ID3Hdr.remainingHeaderBytes = 0;
            return tmp;
        }

        if (m_ID3Hdr.framesize == 0) return 0;

        ps_ptr<char> tmp;
        ps_ptr<char> content_descriptor;
        size_t       fs = m_ID3Hdr.framesize; // fs = size of the frame data field as read from header
        size_t       bytesToCopy = fs;
        size_t       textDataLength = 0;
        uint16_t     idx = 0;

        if (bytesToCopy >= m_ID3Hdr.iBuffSize) { bytesToCopy = m_ID3Hdr.iBuffSize - 1; } // make sure a zero terminator fits
        if (bytesToCopy > 0) { textDataLength = bytesToCopy - 1; }                       // Only if there are data that we can shorten
        for (int i = 0; i < textDataLength; i++) {
            m_ID3Hdr.iBuff[i] = *(data + i + 1); // Skipped the first byte (Encoding)
        }

        if (textEncodingByte == 1 || textEncodingByte == 2) { // is UTF-16LE or UTF-16BE
            m_ID3Hdr.iBuff[textDataLength] = 0;               // UTF-16: set double zero terminator
            m_ID3Hdr.iBuff[textDataLength + 1] = 0;           // second '\0' for UTF-16
        } else {
            m_ID3Hdr.iBuff[textDataLength] = 0; // only one '\0' for ISO-8859-1 or UTF-8
        }

        m_ID3Hdr.framesize -= fs;

        uint16_t dataLength = fs - 1;
        bool     isBigEndian = (textEncodingByte == 2);

        char encodingTab[4][12] = {"ISO-8859-1", "UTF-16", "UTF-16BE", "UTF-8"};

        if (startsWith(m_ID3Hdr.tag, "COMM")) { // language code
            m_ID3Hdr.lang[0] = m_ID3Hdr.iBuff[0];
            m_ID3Hdr.lang[1] = m_ID3Hdr.iBuff[1];
            m_ID3Hdr.lang[2] = m_ID3Hdr.iBuff[2];
            m_ID3Hdr.lang[3] = '\0';

            idx = 4;
            if (textEncodingByte == 1 || textEncodingByte == 2) idx++;

            uint16_t cd_len = 0;
            if (textEncodingByte == 0)
                cd_len = 1 + content_descriptor.copy_from_iso8859_1((const uint8_t*)(m_ID3Hdr.iBuff.get() + idx)); // iso8859_1
            else if (textEncodingByte == 3)
                cd_len = 1 + content_descriptor.copy_from((const char*)(m_ID3Hdr.iBuff.get() + idx)); // utf-8
            else
                cd_len = 2 + content_descriptor.copy_from_utf16((const uint8_t*)(m_ID3Hdr.iBuff.get() + idx), isBigEndian); // utf-16
            if (cd_len > 2) info(*this, evt_info, "Comment: text encoding; %s, language %s, content_descriptor: %s", encodingTab[textEncodingByte], m_ID3Hdr.lang, content_descriptor.c_get());

            idx += cd_len;
            // AUDIO_LOG_INFO("Tag: %s, Length: %i, Format: %s", m_ID3Hdr.tag, textDataLength, encodingTab[textEncodingByte]);
        } else {
            if (textEncodingByte == 0) {
                tmp.copy_from_iso8859_1((const uint8_t*)m_ID3Hdr.iBuff.get() + idx);
            } // ISO-8859-1
            else if (textEncodingByte == 1 || textEncodingByte == 2) {
                tmp.copy_from_utf16((const uint8_t*)m_ID3Hdr.iBuff.get() + idx, isBigEndian);
            } // UTF-16LE oder UTF-16BE
            else if (textEncodingByte == 3) {
                tmp.copy_from(m_ID3Hdr.iBuff.get() + idx);
            } // UTF-8 copy directly because no conversion is necessary
        }
        showID3Tag(m_ID3Hdr.tag, tmp.c_get());
        m_ID3Hdr.remainingHeaderBytes -= fs;
        return fs;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == MP3_SYLT) { // SYLT
        m_controlCounter = MP3_SKIP;
        if (m_dataMode == AUDIO_LOCALFILE || (m_streamType == ST_WEBFILE)) {
            ps_ptr<char> tmp;
            ps_ptr<char> content_descriptor;
            ps_ptr<char> syltBuff;
            bool         isBigEndian = true;
            size_t       len = 0;
            int          idx = 0;
            m_ID3Hdr.SYLT.pos = m_ID3Hdr.id3Size - m_ID3Hdr.remainingHeaderBytes;
            m_ID3Hdr.SYLT.size = m_ID3Hdr.framesize;
            if (m_ID3Hdr.SYLT.size < len) return 0;
            syltBuff.copy_from((const char*)data, m_ID3Hdr.SYLT.size);
            m_ID3Hdr.SYLT.text_encoding = syltBuff[0]; // 0=ISO-8859-1, 1=UTF-16, 2=UTF-16BE, 3=UTF-8
            if (m_ID3Hdr.SYLT.text_encoding == 1) isBigEndian = false;
            if (m_ID3Hdr.SYLT.text_encoding > 3) {
                AUDIO_LOG_ERROR("unknown text encoding: %i", m_ID3Hdr.SYLT.text_encoding);
                m_ID3Hdr.SYLT.text_encoding = 0;
            }
            char encodingTab[4][12] = {"ISO-8859-1", "UTF-16", "UTF-16BE", "UTF-8"};
            memcpy(m_ID3Hdr.SYLT.lang, syltBuff.get() + 1, 3);
            m_ID3Hdr.SYLT.lang[3] = '\0';
            info(*this, evt_info, "Lyrics: text_encoding: %s, language: %s, size %i", encodingTab[m_ID3Hdr.SYLT.text_encoding], m_ID3Hdr.SYLT.lang, m_ID3Hdr.SYLT.size);
            m_ID3Hdr.SYLT.time_stamp_format = syltBuff[4];
            m_ID3Hdr.SYLT.content_type = syltBuff[5];
            idx = 6;
            uint16_t cd_len = 0;
            if (m_ID3Hdr.SYLT.text_encoding == 0)
                cd_len = 1 + content_descriptor.copy_from_iso8859_1((const uint8_t*)(syltBuff.get() + idx)); // iso8859_1
            else if (m_ID3Hdr.SYLT.text_encoding == 3)
                cd_len = 1 + content_descriptor.copy_from((const char*)(syltBuff.get() + idx)); // utf-8
            else
                cd_len = 2 + content_descriptor.copy_from_utf16((const uint8_t*)(syltBuff.get() + idx), isBigEndian); // utf-16
            if (cd_len > 2) info(*this, evt_info, "Lyrics: content_descriptor: %s", content_descriptor.c_get());

            idx += cd_len;
            while (idx < m_ID3Hdr.SYLT.size) {
                if (m_ID3Hdr.SYLT.text_encoding == 0)
                    idx += 1 + tmp.copy_from_iso8859_1((const uint8_t*)syltBuff.get() + idx); // ISO8859_1
                else if (m_ID3Hdr.SYLT.text_encoding == 3)
                    idx += 1 + tmp.copy_from((const char*)syltBuff.get() + idx); // UTF-8
                else
                    idx += 2 + tmp.copy_from_utf16((const uint8_t*)(syltBuff.get() + idx), isBigEndian); // UTF-16LE, UTF-16BE

                if (tmp.starts_with("\n")) tmp.remove_before(1);
                m_syltLines.push_back(std::move(tmp));
                if (idx + 4 > m_ID3Hdr.SYLT.size) break; // no more 4 bytes?
                uint32_t timestamp = bigEndian((uint8_t*)syltBuff.get() + idx, 4);
                m_syltTimeStamp.push_back(timestamp);
                idx += 4;
            }
            info(*this, evt_info, "audiofile contains synchronized lyrics");
            // for(int i = 0; i < m_syltLines.size(); i++){
            //     info(*this, evt_lyrics, "%07i ms,   %s", m_syltTimeStamp[i], m_syltLines[i].c_get());
            // }
        }
        return 0;
    }

    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -

    // --- section V2.2 only , higher Vers above ----
    // see https://mutagen-specs.readthedocs.io/en/latest/id3/id3v2.2.html
    if (m_controlCounter == MP3_ID3V22) { // frames in V2.2, 3bytes identifier, 3bytes size descriptor

        if (m_ID3Hdr.v22_tag_length > 0) {
            if (m_ID3Hdr.v22_tag_length > len) {
                m_ID3Hdr.v22_tag_length -= len;
                return len;
            } // Throw it away
            else {
                uint32_t t = m_ID3Hdr.v22_tag_length;
                m_ID3Hdr.v22_tag_length = 0;
                return t;
            } // Throw it away
        }

        if (m_ID3Hdr.remainingHeaderBytes == 0) {
            m_controlCounter = MP3_LASTFRAMES;
            return 0;
        }

        ps_ptr<char> tag;
        ps_ptr<char> value;
        if (data[0] == 0) { // we are in padding
            m_controlCounter = MP3_LASTFRAMES;
            uint16_t padding = m_ID3Hdr.remainingHeaderBytes;
            m_ID3Hdr.remainingHeaderBytes = 0;
            return padding;
        }
        tag.copy_from((const char*)data, 3);
        m_ID3Hdr.v22_tag_length = bigEndian(data + 3, 3) + 6;
        value.copy_from((const char*)data + 7, min(m_ID3Hdr.v22_tag_length - 7, (size_t)1024));

        if (tag.starts_with("PIC")) { // image embedded in header
            size_t   pic_len = bigEndian(data + 3, 3);
            uint32_t pic_start = m_ID3Hdr.totalId3Size + m_ID3Hdr.id3Size - m_ID3Hdr.remainingHeaderBytes;

            m_ID3Hdr.APIC_vec.push_back(pic_start + 6);
            m_ID3Hdr.APIC_vec.push_back(pic_len);
        } else if (startsWith(m_ID3Hdr.tag, "SLT")) { // lyrics embedded in header
            if (m_dataMode == AUDIO_LOCALFILE) {

                ps_ptr<char> tmp;
                ps_ptr<char> content_descriptor;
                ps_ptr<char> syltBuff;
                bool         isBigEndian = true;
                size_t       len = 0;
                int          idx = 0;

                m_ID3Hdr.SYLT.pos = m_ID3Hdr.id3Size - m_ID3Hdr.remainingHeaderBytes;
                m_ID3Hdr.SYLT.size = m_ID3Hdr.v22_tag_length;
                if (m_ID3Hdr.SYLT.size < len) return 0;
                syltBuff.copy_from((const char*)data, m_ID3Hdr.SYLT.size);
                m_ID3Hdr.SYLT.text_encoding = syltBuff[0];
                memcpy(m_ID3Hdr.SYLT.lang, syltBuff.get() + 1, 3);
                m_ID3Hdr.SYLT.lang[3] = '\0';
                info(*this, evt_info, "Lyrics: text_encoding: %s, language: %s, size %i",
                     m_ID3Hdr.SYLT.text_encoding == 0   ? "ASCII"
                     : m_ID3Hdr.SYLT.text_encoding == 3 ? "UTF-8"
                                                        : "?",
                     m_ID3Hdr.SYLT.lang, m_ID3Hdr.SYLT.size);
                m_ID3Hdr.SYLT.time_stamp_format = syltBuff[4];
                m_ID3Hdr.SYLT.content_type = syltBuff[5];

                idx = 6;

                if (m_ID3Hdr.SYLT.text_encoding == 0 || m_ID3Hdr.SYLT.text_encoding == 3) { // utf-8
                    len = content_descriptor.copy_from((const char*)(syltBuff.get() + idx));
                } else { // utf-16
                    len = content_descriptor.copy_from_utf16((const uint8_t*)(syltBuff.get() + idx), isBigEndian);
                }
                if (len > 2) info(*this, evt_info, "Lyrics: content_descriptor: %s", content_descriptor.c_get());
                idx += len;

                while (idx < m_ID3Hdr.SYLT.size) {
                    // UTF-16LE, UTF-16BE
                    if (m_ID3Hdr.SYLT.text_encoding == 1 || m_ID3Hdr.SYLT.text_encoding == 2) {
                        idx += tmp.copy_from_utf16((const uint8_t*)(syltBuff.get() + idx), isBigEndian);
                    } else {
                        // ISO-8859-1 / UTF-8
                        idx += tmp.copy_from((const char*)syltBuff.get() + idx);
                    }
                    if (tmp.starts_with("\n")) tmp.remove_before(1);
                    m_syltLines.push_back(std::move(tmp));

                    if (idx + 4 > m_ID3Hdr.SYLT.size) break; // no more 4 bytes?

                    uint32_t timestamp = bigEndian((uint8_t*)syltBuff.get() + idx, 4);
                    m_syltTimeStamp.push_back(timestamp);

                    idx += 4;
                }
                info(*this, evt_info, "audiofile contains synchronized lyrics");
                // for(int i = 0; i < m_syltLines.size(); i++){
                //     info(*this, evt_lyrics, "%07i ms,   %s", m_syltTimeStamp[i], m_syltLines[i].c_get());
                // }
            }
        } else {
            showID3Tag(tag.c_get(), value.c_get());
        }

        m_ID3Hdr.remainingHeaderBytes -= m_ID3Hdr.v22_tag_length;

        return 0;
    }
    // -- end section V2.2 -----------

    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == MP3_LASTFRAMES) { // skip all ID3 metadata (mostly spaces)
        if (m_ID3Hdr.remainingHeaderBytes > len) {
            m_ID3Hdr.remainingHeaderBytes -= len;
            return len;
        } // Throw it away
        else {
            m_controlCounter = MP3_XING;
            return m_ID3Hdr.remainingHeaderBytes;
        } // Throw it away
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == MP3_XING) { //  exist another ID3tag?

        static const int samplerate_table[4][3] = {
            {11025, 12000, 8000},  // MPEG 2.5
            {0, 0, 0},             // reserved
            {22050, 24000, 16000}, // MPEG 2
            {44100, 48000, 32000}  // MPEG 1
        };

        static const int samples_per_frame[4][4] = {// layer:   0    1      2      3
                                                    /* V2.5 */ {0, 576, 1152, 384},
                                                    /* res. */ {0, 0, 0, 0},
                                                    /* V2   */ {0, 576, 1152, 384},
                                                    /* V1   */ {0, 1152, 1152, 384}};

        m_audioDataStart += m_ID3Hdr.id3Size;
        m_ID3Hdr.totalId3Size += m_ID3Hdr.id3Size;
        m_ID3Hdr.id3Size = 0;

        if ((*(data + 0) == 'I') && (*(data + 1) == 'D') && (*(data + 2) == '3')) {
            m_controlCounter = MP3_NEXTID3;
            m_ID3Hdr.numID3Header++;
            return 0;
        } else {
            // padding after ID3 body?
            uint32_t padding_counter = 0;
            while (data[padding_counter] == 0) {
                padding_counter++;
                m_audioDataStart++;
                m_audioDataSize--;
            }
            if (padding_counter) return padding_counter;

            m_controlCounter = MP3_OKAY; // 100 -> ok
            m_audioDataSize = m_audioFileSize - m_audioDataStart;
            if (!m_f_m3u8data) info(*this, evt_info, "Audio-Data-Start: %u", m_audioDataStart);
            if (!m_f_m3u8data) info(*this, evt_info, "Audio-Length: %u", m_audioDataSize);

            uint32_t hdr = bigEndian(data, 4);
            if ((hdr & 0xFFE00000) != 0xFFE00000) AUDIO_LOG_ERROR("Syncword not found"); // check sync
            int versionID = (hdr >> 19) & 0x3;                                           // 0=MPEG2.5, 2=MPEG2, 3=MPEG1
            int layerIndex = (hdr >> 17) & 0x3;                                          // 1=Layer3
            int bitrateIdx = (hdr >> 12) & 0xF;
            int samplerateIdx = (hdr >> 10) & 0x3;
            int mode = (hdr >> 6) & 0x3; // 0=stereo,3=mono
            int samplerate = samplerate_table[versionID][samplerateIdx];
            int spf = samples_per_frame[versionID][layerIndex];

            // Xing or Info Header present?
            int8_t mp3_xing = specialIndexOf(data, "Xing", 50);
            int8_t mp3_info = specialIndexOf(data, "Info", 50);

            uint8_t xingPos = 0;
            if (mp3_xing > 0) xingPos = mp3_xing;
            if (mp3_info > 0) xingPos = mp3_info;

            if (xingPos > 0 && layerIndex == 1) { // layer III only
                uint32_t frames = bigEndian(data + xingPos + 8, 4);
                AUDIO_LOG_DEBUG("frames %i", frames);
                uint32_t bytes = bigEndian(data + xingPos + 12, 4);
                AUDIO_LOG_DEBUG("bytes %i", bytes);
                uint32_t duration = frames * spf / samplerate;
                info(*this, evt_info, "Duration (s): %u", duration);
                m_audioFileDuration = duration;
                uint32_t bitrate = bytes * 8 / duration;
                info(*this, evt_info, "Bitrate (b/s): %u", bitrate);
                m_nominal_bitrate = bitrate;
                info(*this, evt_bitrate, "%i", m_nominal_bitrate);
            }

            if (m_ID3Hdr.APIC_vec.size()) { // if we have a APIC
                info(*this, evt_image, m_ID3Hdr.APIC_vec);
                m_ID3Hdr.APIC_vec.clear();
                AUDIO_LOG_DEBUG("m_ID3Hdr.totalId3Size %i, m_audioDataStart %i", m_ID3Hdr.totalId3Size, m_audioDataStart);
                if (m_ID3Hdr.totalId3Size != m_audioDataStart) audioFileSeek(m_audioDataStart);
            }
            m_ID3Hdr.numID3Header = 0;
            m_ID3Hdr.totalId3Size = 0;
            m_ID3Hdr.iBuff.reset();
            return 0;
        }
    }
    return 0;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::read_M4A_Header(uint8_t* data, size_t len) {
    bool         atom_struct = false;
    ps_ptr<char> atom_name;
    ps_ptr<char> atom_size;
    uint32_t     idx = 0;

    /*
         ftyp
           | - moov  -> trak -> ... -> mp4a contains raw block parameters
           |    L... -> ilst  contains artist, composer ....
         free (optional) // jump to another atoms at the end of mdat
           |
         mdat contains the audio data                                                      */

    if (m_m4aHdr.retvalue) {
        if (m_m4aHdr.retvalue > UINT16_MAX) {
            int res = audioFileSeek(m_m4aHdr.headerSize);
            if (res >= 0) {
                AUDIO_LOG_INFO("skip %li bytes", m_m4aHdr.retvalue);
                InBuff.reset();
                m_m4aHdr.retvalue = 0;
                return 0;
            }
        }
        if (m_m4aHdr.retvalue > len) {
            m_m4aHdr.retvalue -= len;
            m_m4aHdr.cnt += len;
            return len;
        } else {
            size_t tmp = m_m4aHdr.retvalue;
            m_m4aHdr.retvalue = 0;
            m_m4aHdr.cnt += tmp;
            return tmp;
        }
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_BEGIN) { // init
        memset(&m_m4aHdr, 0, sizeof(audiolib::m4aHdr_t));

        atom_size.big_endian(data, 4);
        atom_name.copy_from((const char*)data + 4, 4);

        if (atom_name.equals("ftyp")) {
            if (atom_struct) {
                AUDIO_LOG_WARN("atom %s @ %zu, size: %u, ends @ %zu", atom_name.c_get(), m_m4aHdr.headerSize, (uint32_t)atom_size.to_uint32(16), m_m4aHdr.headerSize + (size_t)atom_size.to_uint32(16));
                m_m4aHdr.sizeof_ftyp = atom_size.to_uint32(16);
            }
            m_controlCounter = M4A_FTYP;
        } else {
            AUDIO_LOG_ERROR("begin: ftyp not found");
            stopSong();
            return -1;
        }
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_FTYP) { /* check_m4a_file */

        int m4a = specialIndexOf(data, "M4A ", 20);
        int isom = specialIndexOf(data, "isom", 20);
        int mp42 = specialIndexOf(data, "mp42", 20);

        if ((m4a != 8) && (isom != 8) && (mp42 != 8)) {
            AUDIO_LOG_ERROR("subtype 'MA4 ', 'isom' or 'mp42' expected, but found '%s '", (data + 8));
            stopSong();
            return -1;
        }

        m_m4aHdr.retvalue += m_m4aHdr.sizeof_ftyp;
        m_m4aHdr.headerSize += m_m4aHdr.sizeof_ftyp;
        m_controlCounter = M4A_CHK;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_CHK) { /* check  Tag */
        atom_size.big_endian(data, 4);
        atom_name.copy_from((const char*)data + 4, 4);

        if (atom_name.equals("moov")) {
            if (!m_m4aHdr.mdat_seen) m_m4aHdr.progressive = true; // moov before mdat
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_moov = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_MOOV;
            return 0;
        }

        if (atom_name.equals("mdat")) {
            m_m4aHdr.mdat_seen = true;
            if (!m_m4aHdr.progressive) {
                m_m4aHdr.mdat_startPos = m_m4aHdr.headerSize + 8;
                m_m4aHdr.sizeof_mdat = atom_size.to_uint32(16);
                if (atom_struct) {
                    AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.mdat_startPos, m_m4aHdr.sizeof_mdat, m_m4aHdr.mdat_startPos + m_m4aHdr.sizeof_mdat);
                }
                info(*this, evt_info, "Audiofile is non progressive");
                m_m4aHdr.retvalue += m_m4aHdr.sizeof_mdat;
                m_m4aHdr.headerSize += m_m4aHdr.sizeof_mdat;
                return 0;
                stopSong();
                return -1;
            }
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_mdat = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_MDAT;
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        m_m4aHdr.sizeof_moov -= atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_MOOV) { // moov
        if (atom_struct) { AUDIO_LOG_WARN("moov size remain %i", m_m4aHdr.sizeof_moov); }
        if (m_m4aHdr.sizeof_moov == 0) {
            if (m_m4aHdr.progressive) {
                m_controlCounter = M4A_CHK;
                AUDIO_LOG_DEBUG("goto CHK");
            } else {
                audioFileSeek(0); // non progressive, back to mdat
                InBuff.reset();
                m_m4aHdr.headerSize = m_m4aHdr.mdat_startPos;
                m_m4aHdr.retvalue = m_m4aHdr.mdat_startPos; // set InBuff.getReadPtr to audiodatastart
                if (atom_struct) AUDIO_LOG_WARN("goto MDAT at %i", m_m4aHdr.mdat_startPos + 8);
                m_controlCounter = M4A_MDAT;
            }
            return 0;
        } // go back

        atom_name.copy_from((const char*)data + 4, 4);
        atom_size.big_endian(data, 4);
        m_m4aHdr.sizeof_moov -= atom_size.to_uint32(16);

        if (atom_name.equals("trak")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_trak = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_TRAK;
            return 0;
        }

        if (atom_name.equals("udta")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_udta = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_UDTA;
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_TRAK) { // trak
        if (atom_struct) { AUDIO_LOG_WARN("trak size remain %i", m_m4aHdr.sizeof_trak); }
        if (m_m4aHdr.sizeof_trak == 0) {
            m_controlCounter = M4A_MOOV;
            return 0;
        } // go back

        atom_name.copy_from((const char*)data + 4, 4);
        atom_size.big_endian(data, 4);
        m_m4aHdr.sizeof_trak -= atom_size.to_uint32(16);

        if (atom_name.equals("mdia")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_mdia = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_MDIA;
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_MDIA) { // mdia
        if (atom_struct) { AUDIO_LOG_WARN("mdia size remain %i", m_m4aHdr.sizeof_mdia); }
        if (m_m4aHdr.sizeof_mdia == 0) {
            m_controlCounter = M4A_TRAK;
            return 0;
        } // go back

        atom_name.copy_from((const char*)data + 4, 4);
        atom_size.big_endian(data, 4);
        m_m4aHdr.sizeof_mdia -= atom_size.to_uint32(16);

        if (atom_name.equals("mdhd")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_mdhd = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_MDHD;
            return 0;
        }

        if (atom_name.equals("minf")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_minf = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_MINF;
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_MINF) { // minf
        if (atom_struct) { AUDIO_LOG_WARN("minf size remain %i", m_m4aHdr.sizeof_minf); }
        if (m_m4aHdr.sizeof_minf == 0) {
            m_controlCounter = M4A_MDIA;
            return 0;
        } // go back

        atom_name.copy_from((const char*)data + 4, 4);
        atom_size.big_endian(data, 4);
        m_m4aHdr.sizeof_minf -= atom_size.to_uint32(16);

        if (atom_name.equals("stbl")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_stbl = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_STBL;
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_MDHD) { // mdhd
        if (atom_struct) { AUDIO_LOG_WARN("mdhd size remain %i", m_m4aHdr.sizeof_mdhd); }
        if (m_m4aHdr.sizeof_mdhd == 0) {
            m_controlCounter = M4A_MDIA;
            return 0;
        } // go back

        ps_ptr<char> mdhd_buffer;                                 // read ESDS content in a buffer
        mdhd_buffer.copy_from((char*)data, m_m4aHdr.sizeof_mdhd); // read MDHD content (without header)
                                                                  //    mdhd_buffer.hex_dump(m_m4aHdr.sizeof_mdhd);
                                                                  /* version;           1 Byte  offset 0   0 -> 32 bit, 1 -> 64 bit
                                                                     flags[3];          3 Bytes offset 1
                                                                     creation_time;     4 Bytes offset 4
                                                                     modification_time; 4 Bytes offset 8
                                                                     timescale;         4 Bytes offset 12
                                                                     duration;          4 Bytes offset 16
                                                                     language;          2 Bytes offset 20
                                                                     pre_defined;       2 Bytes offset 22 */
        m_m4aHdr.timescale = bigEndian((uint8_t*)mdhd_buffer.get() + 12, 4);
        m_m4aHdr.duration = bigEndian((uint8_t*)mdhd_buffer.get() + 16, 4);
        if (m_m4aHdr.timescale) {
            m_audioFileDuration = m_m4aHdr.duration / m_m4aHdr.timescale;
            info(*this, evt_info, "Duration (s): %u", m_audioFileDuration);
        }
        m_m4aHdr.retvalue += m_m4aHdr.sizeof_mdhd;
        m_m4aHdr.headerSize += m_m4aHdr.sizeof_mdhd;
        m_controlCounter = M4A_MDIA;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_STBL) { // minf
        if (atom_struct) { AUDIO_LOG_WARN("stbl size remain %i", m_m4aHdr.sizeof_stbl); }
        if (m_m4aHdr.sizeof_stbl == 0) {
            m_controlCounter = M4A_MINF;
            return 0;
        } // go back

        atom_size.big_endian(data, 4);
        atom_name.copy_from((const char*)data + 4, 4);
        m_m4aHdr.sizeof_stbl -= atom_size.to_uint32(16);

        if (atom_name.equals("stsd")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            uint8_t header_size = 8;
            m_m4aHdr.sizeof_stsd = atom_size.to_uint32(16) - 8 - header_size;
            m_m4aHdr.retvalue += 8 + header_size;
            m_m4aHdr.headerSize += 8 + header_size;
            m_controlCounter = M4A_STSD;
            return 0;
        }

        if (atom_name.equals("stsz")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            uint8_t header_size = 8;
            m_m4aHdr.sizeof_stsz = atom_size.to_uint32(16) - 8 - header_size;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_STSZ;
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_STSD) {
        if (atom_struct) { AUDIO_LOG_WARN("stsd size remain %i", m_m4aHdr.sizeof_stsd); }
        if (m_m4aHdr.sizeof_stsd == 0) {
            m_controlCounter = M4A_STBL;
            return 0;
        } // go back

        atom_size.big_endian(data, 4);
        atom_name.copy_from((const char*)data + 4, 4);
        m_m4aHdr.sizeof_stsd -= atom_size.to_uint32(16);

        if (atom_name.equals("mp4a")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_mp4a = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_MP4A;
            return 0;
        }

        if (atom_name.equals("alac")) {
            AUDIO_LOG_ERROR("Apple loseless audio codec (ALAC) not supported");
            stopSong();
            return 0;
        }

        if (atom_name.equals("ac-3")) {
            AUDIO_LOG_ERROR("Dolby Digital not supported");
            stopSong();
            return 0;
        }

        if (atom_name.equals("ec-3")) {
            AUDIO_LOG_ERROR("Dolby Digital Plud not supported");
            stopSong();
            return 0;
        }

        if (atom_name.equals("Opus")) {
            // todo
            stopSong();
            return 0;
        }

        if (atom_name.equals("mp3 ")) {
            // todo
            stopSong();
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_CHPL) {
        if (atom_struct) { AUDIO_LOG_WARN("chpl size remain %i", m_m4aHdr.sizeof_chpl); }
        if (m_m4aHdr.sizeof_chpl == 0) {
            m_controlCounter = M4A_UDTA;
            return 0;
        } // go back

        ps_ptr<char> title_name;
        // data[0] version
        // data[1] 4 flags
        uint32_t nr_of_chapters = bigEndian(data + 5, 4); // Number of chapters
        // data[9] uint64 timestamp
        uint8_t title_length = data[17];
        title_name.copy_from((char*)data + 18, title_length);

        info(*this, evt_info, "Number of chapters: %i", nr_of_chapters);
        info(*this, evt_info, "Chapter name: %s", title_name.c_get());

        m_m4aHdr.retvalue += m_m4aHdr.sizeof_chpl;
        m_m4aHdr.headerSize += m_m4aHdr.sizeof_chpl;
        m_m4aHdr.sizeof_chpl = 0; // there are no subatoms
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_MP4A) {
        if (atom_struct) { AUDIO_LOG_WARN("mp4a size remain %i", m_m4aHdr.sizeof_mp4a); }
        if (m_m4aHdr.sizeof_mp4a == 0) {
            m_controlCounter = M4A_STSD;
            return 0;
        } // go back

        if (!m_m4aHdr.sample_rate) { // read the header first
            // data[0]...[5] reserved
            // data[6] & [7] Data reference buffer_index
            // data[8]...[15] reserved
            m_m4aHdr.channel_count = data[16] * 256 + data[17];
            AUDIO_LOG_DEBUG("channels %i", m_m4aHdr.channel_count);
            m_m4aHdr.sample_size = data[18] * 256 + data[19];
            AUDIO_LOG_DEBUG("bit per sample %i", m_m4aHdr.sample_size);
            // data[20]...data[23] reserved
            m_m4aHdr.sample_rate = data[24] * 256 + data[25];
            AUDIO_LOG_DEBUG("sample rate %i", m_m4aHdr.sample_rate);
            // data[26]...data[27] sample rate fixed point is 0x00, 0x00
            m_m4aHdr.offset = 28;
            m_m4aHdr.retvalue += m_m4aHdr.offset;
            m_m4aHdr.headerSize += m_m4aHdr.offset;
            m_m4aHdr.sizeof_mp4a -= m_m4aHdr.offset;
            return 0;
        }
        atom_size.big_endian(data, 4);
        atom_name.copy_from((const char*)data + 4, 4);
        m_m4aHdr.sizeof_mp4a -= atom_size.to_uint32(16);

        if (atom_name.equals("esds")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_esds = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_ESDS;
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_UDTA) { // udta
        if (atom_struct) { AUDIO_LOG_WARN("udta size remain %i", m_m4aHdr.sizeof_udta); }
        if (m_m4aHdr.sizeof_udta == 0) {
            m_controlCounter = M4A_MOOV;
            return 0;
        } // go back

        atom_size.big_endian(data, 4);
        atom_name.copy_from((const char*)data + 4, 4);
        m_m4aHdr.sizeof_udta -= atom_size.to_uint32(16);

        if (atom_name.equals("meta")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_meta = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_META;
            return 0;
        }

        if (atom_name.equals("chpl")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_chpl = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_CHPL;
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_META) { // meta
        if (atom_struct) { AUDIO_LOG_WARN("meta size remain %i", m_m4aHdr.sizeof_meta); }
        if (m_m4aHdr.sizeof_meta == 0) {
            m_controlCounter = M4A_UDTA;
            return 0;
        } // go back,  4bytes typ + 4 bytes size

        if (!m_m4aHdr.version_flags) {
            // 0x00, 0x00, 0x00, 0x94, 0x6D, 0x65, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00,
            //  size                 |   m    e     t     a  |     flags
            m_m4aHdr.version_flags = true;
            m_m4aHdr.sizeof_meta -= 4;
            m_m4aHdr.retvalue += 4;
            m_m4aHdr.headerSize += 4;
            idx += 4;
        }
        atom_size.big_endian(data + idx, 4);
        atom_name.copy_from((const char*)data + 4 + idx, 4);
        m_m4aHdr.sizeof_meta -= atom_size.to_uint32(16);

        if (atom_name.equals("ilst")) {
            if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
            m_m4aHdr.sizeof_ilst = atom_size.to_uint32(16) - 8;
            m_m4aHdr.retvalue += 8;
            m_m4aHdr.headerSize += 8;
            m_controlCounter = M4A_ILST;
            return 0;
        }

        if (atom_struct) { AUDIO_LOG_WARN("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), m_m4aHdr.headerSize, atom_size.to_uint32(16), m_m4aHdr.headerSize + atom_size.to_uint32(16)); }
        m_m4aHdr.retvalue += atom_size.to_uint32(16);
        m_m4aHdr.headerSize += atom_size.to_uint32(16);
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_ESDS) {                           // Elementary Stream Descriptor
        ps_ptr<char> esds_buffer;                                 // read ESDS content in a buffer
        esds_buffer.copy_from((char*)data, m_m4aHdr.sizeof_esds); // read ESDS content (without header)
        // esds_buffer.hex_dump(m_m4aHdr.sizeof_esds);

        // search for decoderConfigDescriptor (tag 0x04)
        int32_t dec_config_descriptor_offset = esds_buffer.special_index_of("\x04\x80\x80\x80", 4, (uint32_t)m_m4aHdr.sizeof_esds);
        if (dec_config_descriptor_offset > 0) {                                                                     // decoderConfigDescriptor found
            uint8_t dec_config_descriptor_length = ((uint8_t*)esds_buffer.get())[dec_config_descriptor_offset + 4]; // Length after Tag + 3 Extended Length Bytes
            m_m4aHdr.objectTypeIndicator = ((uint8_t*)esds_buffer.get())[dec_config_descriptor_offset + 5];         // 0x40 (AAC)
            m_m4aHdr.streamType = ((uint8_t*)esds_buffer.get())[dec_config_descriptor_offset + 6];                  // 0x05 (Audio)
            m_m4aHdr.bufferSizeDB = bigEndian((uint8_t*)esds_buffer.get() + dec_config_descriptor_offset + 7, 3);   // 24 bit
            m_m4aHdr.maxBitrate = bigEndian((uint8_t*)esds_buffer.get() + dec_config_descriptor_offset + 10, 4);    // 32 bit
            m_m4aHdr.nomBitrate = bigEndian((uint8_t*)esds_buffer.get() + dec_config_descriptor_offset + 14, 4);    // 32 bit
            // info(*this, evt_info, "maxBitrate %u, nominalBitrate %u", m_m4aHdr.maxBitrate, m_m4aHdr.nomBitrate);
        }

        // search for decoderspecificinfo (tag 0x05)
        int32_t dec_specific_offset = esds_buffer.special_index_of("\x05\x80\x80\x80", 4, m_m4aHdr.sizeof_esds);
        if (dec_specific_offset >= 0) {                                                           // decoderSpecificInfo found
            uint8_t dec_specific_length = ((uint8_t*)esds_buffer.get())[dec_specific_offset + 4]; // Länge nach Tag + 3 Extended Length Bytes
            AUDIO_LOG_DEBUG("DecoderSpecificInfo found at offset %d, length: %u", dec_specific_offset, dec_specific_length);

            // extract decoderSpecificInfo-data
            ps_ptr<char> dec_specific_data;
            dec_specific_data.alloc(dec_specific_length);
            memcpy(dec_specific_data.get(), (uint8_t*)esds_buffer.get() + dec_specific_offset + 5, dec_specific_length);
            m_m4aHdr.aac_profile = (uint8_t)dec_specific_data.get()[0] >> 3;
            if (m_m4aHdr.aac_profile > 4) {
                AUDIO_LOG_ERROR("unsupported AAC Profile %i", m_m4aHdr.aac_profile);
                stopSong();
                return 0;
            }
            AUDIO_LOG_DEBUG("DecoderSpecificInfo data: 0x%02X 0x%02X", (uint8_t)dec_specific_data.get()[0], (uint8_t)dec_specific_data.get()[1]);
            char profile[5][33] = {"unknown", "AAC Main", "AAC LC (Low Complexity)", "AAC SSR (Scalable Sample Rate)", "AAC LTP (Long Term Prediction)"};
            info(*this, evt_info, "AAC Profile: %s", profile[m_m4aHdr.aac_profile]);
            info(*this, evt_info, "AAC Channels; %i", m_m4aHdr.channel_count);
            info(*this, evt_info, "AAC Sample Rate: %i", m_m4aHdr.sample_rate);
            info(*this, evt_info, "AAC Bits Per Sample: %i", m_m4aHdr.sample_size);
            m_M4A_chConfig = m_m4aHdr.channel_count;
            m_M4A_sampleRate = m_m4aHdr.sample_rate;
            m_M4A_objectType = m_m4aHdr.aac_profile;
        } else {
            AUDIO_LOG_WARN("No DecoderSpecificInfo found in esds");
        }
        m_m4aHdr.retvalue += m_m4aHdr.sizeof_esds;
        m_m4aHdr.headerSize += m_m4aHdr.sizeof_esds;
        m_controlCounter = M4A_MP4A;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_ILST) {

        auto parse_m4a_timestamp = [&](const char* s) {
            if (!s || *s != '[') return -1;
            int mm = 0, ss = 0, hh = 0;
            if (sscanf(s, "[%d:%d.%d]", &mm, &ss, &hh) == 3) { return mm * 60000 + ss * 1000 + hh * 10; }
            return -1;
        };

        struct TagInfo { // Definition of the Taginfo structure for iTunes-style metadata
            const uint8_t tag[4];
            const char*   name;
            const char*   descr;
        };
        const TagInfo tags[] = {
            // List of all usual tags
            {{0xA9, 0x6E, 0x61, 0x6D}, "©nam", "Title"},      {{0xA9, 0x41, 0x52, 0x54}, "©ART", "Artist"},           {{0xA9, 0x61, 0x72, 0x74}, "©art", "Artist"},
            {{0xA9, 0x61, 0x6C, 0x62}, "©alb", "Album"},      {{0xA9, 0x74, 0x6F, 0x6F}, "©too", "Encoder"},          {{0xA9, 0x63, 0x6D, 0x74}, "©cmt", "Comment"},
            {{0xA9, 0x77, 0x72, 0x74}, "©wrt", "Composer"},   {{0x74, 0x6D, 0x70, 0x6F}, "tmpo", "Tempo (BPM)"},      {{0x74, 0x72, 0x6B, 0x6E}, "trkn", "Track-Number"},
            {{0xA9, 0x64, 0x61, 0x79}, "©day", "Year"},       {{0x63, 0x70, 0x69, 0x6C}, "cpil", "Compilation-Flag"}, {{0x61, 0x41, 0x52, 0x54}, "aART", "Album Artist"},
            {{0xA9, 0x67, 0x65, 0x6E}, "©gen", "Genre"},      {{0x63, 0x6F, 0x76, 0x72}, "covr", "Cover Art"},        {{0x64, 0x69, 0x73, 0x6B}, "disk", "Disk-Nummer"},
            {{0xA9, 0x6C, 0x79, 0x72}, "©lyr", "Songtext"},   {{0xA9, 0x70, 0x72, 0x74}, "cprt", "Copyright"},        {{0x67, 0x6E, 0x72, 0x65}, "gnre", "Genre-ID"},
            {{0x72, 0x74, 0x6E, 0x67}, "rtng", "Evaluation"}, {{0x70, 0x67, 0x61, 0x70}, "pgap", "Gapless Playback"},
        };
        const size_t tags_count = sizeof(tags) / sizeof(tags[0]); // Number of tags

        ps_ptr<char> id3tag;
        ps_ptr<char> lyricsBuffer;
        uint32_t     pos = m_m4aHdr.headerSize;
        uint32_t     consumed = 0;
        while (true) {
            atom_size.big_endian(data + consumed, 4);
            atom_name.copy_from((const char*)data + 4 + consumed, 4);
            uint32_t as = atom_size.to_uint32(16);
            AUDIO_LOG_DEBUG("atom %s @ %i, size: %i, ends @ %i", atom_name.c_get(), pos, as, pos + as);
            m_m4aHdr.ilst_pos += as;
            pos += as;
            consumed += as;

            // 0x00, 0x00, 0x00, 0x1C, 0xA9, 0x64, 0x61, 0x79, 0x00, 0x00, 0x00, 0x14, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x32, 0x30, 0x32, 0x32,
            //     sub atom length   |   ©    d     a     y  |  sub sub atom length  |  d     a     t     a  | data type   1->UTF-8  |   reserved            |  2     0     2     2

            ps_ptr<char> sa; // sub atom
            sa.copy_from((const char*)data + consumed - as, min(as, (uint32_t)1024));

            char san[5] = {0};  // sub atom name
            char ssan[5] = {0}; // sub sub atom name (should be 'data')
            strncpy(san, &sa[4], 4);
            strncpy(ssan, &sa[12], 4);
            uint32_t ssal = bigEndian((uint8_t*)&sa[8], 4); // sub sub atom length
            uint32_t dty = bigEndian((uint8_t*)&sa[16], 4); // data type 1-UTF8
            if (strncmp(ssan, "data", 4) == 0) {
                for (int i = 0; i < tags_count; i++) {
                    if (memcmp(san, tags[i].tag, 4) == 0) {
                        id3tag.reset();
                        if (dty == 0 && strlen(&sa[24]) > 0) {
                            if (strcmp(san, "covr") == 0) {
                                m_m4aHdr.picLen = as - 24;
                                m_m4aHdr.picPos = pos + 24 - as;
                                AUDIO_LOG_DEBUG("cover jpeg start: %lu, len %lu", m_m4aHdr.picPos, m_m4aHdr.picLen);
                            }
                            id3tag.assignf("%s: %i", tags[i].descr, &sa[24]); // binary
                        } else if (dty == 1 && strlen(&sa[24]) > 0)
                            id3tag.assignf("%s: %s", tags[i].descr, &sa[24]); // UTF-8 Text
                        else if (dty == 0x0D) {
                            m_m4aHdr.picLen = as - 24;
                            m_m4aHdr.picPos = pos + 24 - as;
                            AUDIO_LOG_DEBUG("cover jpeg start: %lu, len %lu", m_m4aHdr.picPos, m_m4aHdr.picLen);
                        } // jpeg
                        else if (dty == 0x0E) {
                            m_m4aHdr.picLen = as - 24;
                            m_m4aHdr.picPos = pos + 24 - as;
                            AUDIO_LOG_WARN("cover png start: %lu, len %lu", m_m4aHdr.picPos, m_m4aHdr.picLen);
                        } // png
                        else if (dty == 21) {
                            if (memcmp(tags[i].tag, "cpil", 4) == 0) { break; } // Compilation Flag
                            if (memcmp(tags[i].tag, "pgap", 4) == 0) { break; } // Gapless Playback
                            if (memcmp(tags[i].tag, "rtng", 4) == 0) { break; } // Evaluation
                        }
                        if (id3tag.valid()) { info(*this, evt_id3data, "%s", id3tag.get()); }
                        if (strcmp(tags[i].descr, "Songtext") == 0) { lyricsBuffer.copy_from(&sa[24]); }
                        break;
                    }
                }
            } else
                AUDIO_LOG_DEBUG("%s tag not supported", san);

            if (consumed > len + m_m4aHdr.ilst_already_consumed) { // ILST > len can happen
                m_m4aHdr.ilst_already_consumed += len;
                m_m4aHdr.retvalue = consumed;
                m_m4aHdr.headerSize += consumed;
                m_controlCounter = M4A_ILST;
                return 0;
            }

            if (m_m4aHdr.sizeof_ilst <= m_m4aHdr.ilst_pos) {

                ps_ptr<char> tmp; // last todo if we have lyrics
                ps_ptr<char> timestamp;
                timestamp.alloc(12);
                if (lyricsBuffer.valid()) {
                    lyricsBuffer.remove_prefix("LYRICS=");
                    idx = 0;
                    while (idx < lyricsBuffer.size()) {
                        int pos = lyricsBuffer.index_of('\n', idx);
                        if (pos == -1) break;
                        int len = pos - idx + 1;
                        tmp.copy_from(lyricsBuffer.get() + idx, len);
                        idx += len;
                        if (tmp.ends_with("\n")) tmp.truncate_at('\n');
                        // tmp content e.g.: [00:27.07]Jetzt kommt die Zeit
                        //                   [00:28.84]Auf die ihr euch freut
                        timestamp.copy_from(tmp.get(), 10);

                        int ms = parse_m4a_timestamp(timestamp.c_get());

                        // timestamp.remove_chars("[]:.");
                        // timestamp.append("0"); // to ms 0028840
                        tmp.remove_before(10, true);
                        if (tmp.strlen() > 0) {
                            m_syltLines.push_back(std::move(tmp));
                            m_syltTimeStamp.push_back(ms);
                        }
                    }
                    info(*this, evt_info, "audiofile contains synchronized lyrics");
                    // for(int i = 0; i < m_syltLines.size(); i++){
                    //     info(*this, evt_lyrics, "%07i ms,   %s", m_syltTimeStamp[i], m_syltLines[i].c_get());
                    // }
                }
                m_m4aHdr.retvalue = consumed;
                m_m4aHdr.headerSize += consumed;
                m_controlCounter = M4A_META;
                break;
            }
        }
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_controlCounter == M4A_STSZ) {
        uint8_t version = data[0];
        (void)version;
        uint32_t flags = bigEndian(data + 1, 3);
        (void)flags;
        uint32_t sample_size = bigEndian(data + 4, 4);
        (void)sample_size;
        m_m4aHdr.stsz_num_entries = bigEndian(data + 8, 4) / 4; // is uint32_t
        m_m4aHdr.stsz_table_pos = m_m4aHdr.headerSize + 12;
        m_m4aHdr.retvalue += m_m4aHdr.sizeof_stsz + 8;
        m_m4aHdr.headerSize += m_m4aHdr.sizeof_stsz + 8;
        m_controlCounter = M4A_STBL;
        return 0;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    uint8_t extLen = 0;
    if (m_controlCounter == M4A_MDAT) { // mdat
        // ps_ptr<char> hd;
        // hd.copy_from((const char*)data, 30);
        // hd.hex_dump(30);
        m_audioDataSize = m_m4aHdr.sizeof_mdat; // length of this atom

        // Extended Size
        // 00 00 00 01 6D 64 61 74 00 00 00 00 00 00 16 64
        //        0001  m  d  a  t                    5732

        if (m_audioDataSize == 1) { // Extended Size
            m_audioDataSize = bigEndian(data + 8, 8);
            m_audioDataSize -= 16;
            extLen = 8;
        } else {
            m_audioDataSize -= 8;
        }
        m_m4aHdr.retvalue = extLen;
        m_m4aHdr.headerSize += extLen;
        m_controlCounter = M4A_AMRDY; // last step before starting the audio
        return 0;
    }

    if (m_controlCounter == M4A_AMRDY) { // almost ready
        m_audioDataStart = m_m4aHdr.headerSize;
        if (m_m4aHdr.picLen) {
            size_t                pos = m_audioFilePosition;
            std::vector<uint32_t> vec;
            vec.push_back(m_m4aHdr.picPos);
            vec.push_back(m_m4aHdr.picLen);
            info(*this, evt_image, vec);
            vec.clear();
            audioFileSeek(pos); // the filepointer could have been changed by the user, set it back
        }
        m_stsz_numEntries = m_m4aHdr.stsz_num_entries;
        m_stsz_position = m_m4aHdr.stsz_table_pos;
        info(*this, evt_info, "Audio-Data-Start: %u", m_audioDataStart);
        info(*this, evt_info, "Audio-Length: %i", m_audioDataSize);
        if (m_audioFileDuration) {
            m_nominal_bitrate = (m_audioDataSize * 8) / m_audioFileDuration;
            info(*this, evt_info, "Duration (s): %u", m_audioFileDuration);
            info(*this, evt_bitrate, "%i", m_nominal_bitrate);
            info(*this, evt_info, "Bitrate (b/s): %lu", m_nominal_bitrate);
        }

        m_controlCounter = M4A_OKAY; // that's all
        return 0;
    }
    // this section should never be reached
    AUDIO_LOG_ERROR("error");
    return 0;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
size_t Audio::process_m3u8_ID3_Header(uint8_t* packet) {
    uint8_t  ID3version;
    size_t   id3Size;
    bool     m_f_unsync = false, m_f_exthdr = false;
    uint64_t current_timestamp = 0;

    (void)m_f_unsync;        // suppress -Wunused-variable
    (void)current_timestamp; // suppress -Wunused-variable

    if (specialIndexOf(packet, "ID3", 4) != 0) { // ID3 not found
        // AUDIO_LOG_INFO("m3u8 file has no mp3 tag");
        return 0; // error, no ID3 signature found
    }
    ID3version = *(packet + 3);
    switch (ID3version) {
        case 2:
            m_f_unsync = (*(packet + 5) & 0x80);
            m_f_exthdr = false;
            break;
        case 3:
        case 4:
            m_f_unsync = (*(packet + 5) & 0x80); // bit7
            m_f_exthdr = (*(packet + 5) & 0x40); // bit6 extended header
            break;
    };
    id3Size = bigEndian(&packet[6], 4, 7); //  ID3v2 size  4 * %0xxxxxxx (shift left seven times!!)
    id3Size += 10;
    // AUDIO_LOG_INFO("ID3 framesSize: %i", id3Size);
    // AUDIO_LOG_INFO("ID3 version: 2.%i", ID3version);

    if (m_f_exthdr) {
        AUDIO_LOG_ERROR("ID3 extended header in m3u8 files not supported");
        return 0;
    }
    // AUDIO_LOG_INFO("ID3 normal frames");

    if (specialIndexOf(&packet[10], "PRIV", 5) != 0) { // tag PRIV not found
        AUDIO_LOG_ERROR("tag PRIV in m3u8 Id3 Header not found");
        return 0;
    }
    // if tag PRIV exists assume content is "com.apple.streaming.transportStreamTimestamp"
    // a time stamp is expected in the header.

    current_timestamp = (double)bigEndian(&packet[69], 4) / 90000; // seconds

    return id3Size;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::stopSong() {
    m_f_lockInBuffer = true; // wait for the decoding to finish
    uint8_t  maxWait = 0;
    uint32_t currTime = getAudioCurrentTime();

    xSemaphoreTake(mutex_audioTaskIsDecoding, 0.3 * configTICK_RATE_HZ); // wait for audioTask is ready
    {
        if (m_f_running) {
            m_f_running = false;
            if (m_client->connected()) {
                if (m_streamType == ST_WEBSTREAM) { info(*this, evt_info, "Closing web stream \"%s\"", m_lastHost.c_get()); }
                if (m_streamType == ST_WEBFILE) { info(*this, evt_info, "Closing web file \"%s\"", m_lastHost.c_get()); }
                m_client->stop();
            }
            if (m_audiofile) {
                info(*this, evt_info, "Closing audio file \"%s\"", m_audiofile.name());
                m_audiofile.close();
            }
        }
        destroy_decoder();
        m_validSamples = 0;
        m_audioCurrentTime = 0;
        m_audioFileDuration = 0;
        m_codec = CODEC_NONE;
        m_dataMode = AUDIO_NONE;
        m_streamType = ST_NONE;
        m_playlistFormat = FORMAT_NONE;
        m_f_lockInBuffer = false;
    }
    xSemaphoreGive(mutex_audioTaskIsDecoding);
    return currTime;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::pauseResume() {
    xSemaphoreTake(mutex_audioTask, 0.3 * configTICK_RATE_HZ);
    bool retVal = false;
    if (m_dataMode == AUDIO_LOCALFILE || m_streamType == ST_WEBSTREAM || m_streamType == ST_WEBFILE) {
        m_f_running = !m_f_running;
        retVal = true;
        if (!m_f_running) {
            m_outBuff.clear();
            memset(m_samplesBuff48K.get(), 0, m_samplesBuff48KSize * sizeof(int16_t)); // Clear SamplesBuffer
            m_validSamples = 0;
        }
    }
    xSemaphoreGive(mutex_audioTask);
    return retVal;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
audiolib::BiquadCoeffs Audio::makeButterworthLPF_Q31(float fs) { // Calculation of the biquad coefficients for the 48K resampler

    float fc = 0.45f * fs;
    if (fc > 20000.0f) fc = 20000.0f; // absolute upper limit (security)
    if (fc < 3000.0f) fc = 3000.0f;   // absolute lower limit (prevents “thin” sound)

    constexpr float Q = 0.70710678f;

    float w0 = 2.0f * M_PI * fc / fs;
    float cw = cosf(w0);
    float sw = sinf(w0);
    float alpha = sw / (2.0f * Q);

    float b0 = (1.0f - cw) * 0.5f;
    float b1 = 1.0f - cw;
    float b2 = (1.0f - cw) * 0.5f;
    float a0 = 1.0f + alpha;
    float a1 = -2.0f * cw;
    float a2 = 1.0f - alpha;

    // normalize
    b0 /= a0;
    b1 /= a0;
    b2 /= a0;
    a1 /= a0;
    a2 /= a0;

    // convert to Q31
    constexpr float Q31 = 2147483648.0f; // 2^31

    audiolib::BiquadCoeffs c;
    c.b0 = (int32_t)lrintf(b0 * Q31);
    c.b1 = (int32_t)lrintf(b1 * Q31);
    c.b2 = (int32_t)lrintf(b2 * Q31);
    c.a1 = (int32_t)lrintf(a1 * Q31);
    c.a2 = (int32_t)lrintf(a2 * Q31);

    return c;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::resampleTo48kStereo(audiolib::resampler_t& rs, int32_t* input, uint32_t inputSamples, int32_t* output) {

    auto lerp_q32 = [&](int32_t a, int32_t b, uint32_t frac) -> int32_t { return a + (int32_t)(((int64_t)(b - a) * frac) >> 32); };
    auto biquadProcess = [&](audiolib::Biquad& s, const audiolib::BiquadCoeffs& c, int32_t x) -> int32_t {
        // Q31 signal, Q31 coeffs → Q62 acc
        int64_t acc = (int64_t)c.b0 * x + s.z1;
        s.z1 = (int64_t)c.b1 * x - (int64_t)c.a1 * (acc >> 31) + s.z2;
        s.z2 = (int64_t)c.b2 * x - (int64_t)c.a2 * (acc >> 31);
        int64_t y = acc >> 31;
        if (y > INT32_MAX) return INT32_MAX;
        if (y < INT32_MIN) return INT32_MIN;
        return (int32_t)y;
    };

    uint32_t outFrames = 0;
    uint32_t i = 0;

    // If we have a "last" from the previous frame, start with that
    if (rs.hasLast && inputSamples > 0) {
        int32_t l0 = rs.lastL;
        int32_t r0 = rs.lastR;
        int32_t l1 = input[0];
        int32_t r1 = input[1];

        // Continue processing with the old phase value
        while ((rs.phase >> 32) == 0) {
            uint32_t frac = (uint32_t)rs.phase;
            int32_t  l = lerp_q32(l0, l1, frac);
            int32_t  r = lerp_q32(r0, r1, frac);

            l = biquadProcess(rs.lpLeft, rs.g_lpCoeffs, l);
            r = biquadProcess(rs.lpRight, rs.g_lpCoeffs, r);

            output[outFrames * 2] = l;
            output[outFrames * 2 + 1] = r;
            ++outFrames;

            rs.phase += rs.phaseStep;
        }
        rs.phase -= (1ULL << 32);
        // i remains 0, we haven't used input[0] as "l0" yet!
    }

    // Rest of the frame as usual
    for (; i + 1 < inputSamples; ++i) {
        int32_t l0 = input[i * 2];
        int32_t r0 = input[i * 2 + 1];
        int32_t l1 = input[(i + 1) * 2];
        int32_t r1 = input[(i + 1) * 2 + 1];

        while ((rs.phase >> 32) == 0) {
            uint32_t frac = (uint32_t)rs.phase;
            int32_t  l = lerp_q32(l0, l1, frac);
            int32_t  r = lerp_q32(r0, r1, frac);

            l = biquadProcess(rs.lpLeft, rs.g_lpCoeffs, l);
            r = biquadProcess(rs.lpRight, rs.g_lpCoeffs, r);

            output[outFrames * 2] = l;
            output[outFrames * 2 + 1] = r;
            ++outFrames;

            rs.phase += rs.phaseStep;
        }
        rs.phase -= (1ULL << 32);
    }

    // Save last sample for next frame
    if (inputSamples > 0) {
        rs.lastL = input[(inputSamples - 1) * 2];
        rs.lastR = input[(inputSamples - 1) * 2 + 1];
        rs.hasLast = true;
    }

    return outFrames;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void IRAM_ATTR Audio::playChunk() {
    if (m_validSamples == 0) return; // nothing to do

    bool continueI2S = true;
    m_plCh.i2s_bytesConsumed = 0;
    m_plCh.err = ESP_OK;

    constexpr int BYTES_PER_FRAME = 2 * sizeof(int32_t);

    if (m_plCh.count > 0) goto i2swrite; // Not all samples could be written to I2S during the last run
    audio_process_raw_samples(m_outBuff.get(), m_validSamples);
    //------------------------------------------------------------------------------------------
    for (int i = 0; i < m_validSamples; i++) {
        calculateVUlevel(&m_outBuff[i * 2]);
        IIR_filter(&m_outBuff[i * 2]);
        Gain(&m_outBuff[i * 2]);
    }
    processSpectrum();
    if (m_f_forceMono) stereo2mono(m_outBuff.get(), m_validSamples);
    //------------------------------------------------------------------------------------------
    if (m_f_output48KHz && m_i2s_items.sampleRate != 48000) {
        m_validSamples = resampleTo48kStereo(m_resampler, m_outBuff.get(), m_validSamples, m_samplesBuff48K.get()); // have new amount of samples
        audio_process_i2s(m_samplesBuff48K.get(), m_validSamples, &continueI2S);                                    // 48KHz stereo 32bps
    } else {
        audio_process_i2s(m_outBuff.get(), (int32_t)m_validSamples, &continueI2S);
    }
    //------------------------------------------------------------------------------------------------------
    if (!continueI2S) {
        m_validSamples = 0;
        m_plCh.count = 0;
        return;
    }
    //------------------------------------------------------------------------------------------------------

i2swrite:
    if (m_f_output48KHz && m_i2s_items.sampleRate != 48000) { // with resampler
        m_plCh.err = i2s_channel_write(m_i2s_tx_handle, m_samplesBuff48K.get() + m_plCh.count, m_validSamples * BYTES_PER_FRAME, &m_plCh.i2s_bytesConsumed, 50);
    } else { // without resampler
        m_plCh.err = i2s_channel_write(m_i2s_tx_handle, m_outBuff.get() + m_plCh.count, m_validSamples * BYTES_PER_FRAME, &m_plCh.i2s_bytesConsumed, 20);
    }

    if (!(m_plCh.err == ESP_OK || m_plCh.err == ESP_ERR_TIMEOUT)) goto exit;
    m_validSamples -= m_plCh.i2s_bytesConsumed / BYTES_PER_FRAME;
    m_plCh.count += m_plCh.i2s_bytesConsumed / 2;
    if (m_validSamples <= 0) {
        m_validSamples = 0;
        m_plCh.count = 0;
    }

    // ---- statistics, bytes written to I2S (every 10s)
    // static int cnt = 0;
    // static uint32_t t = millis();

    // if(t + 10000 < millis()){
    //     AUDIO_LOG_INFO("%i", cnt);
    //     cnt = 0;
    //     t = millis();
    // }
    // cnt+= i2s_bytesConsumed;
    //-------------------------------------------

    return;
exit:
    if (m_plCh.err == ESP_OK)
        return;
    else if (m_plCh.err == ESP_ERR_INVALID_ARG)
        AUDIO_LOG_ERROR("NULL pointer or this handle is not tx handle");
    else if (m_plCh.err == ESP_ERR_TIMEOUT)
        AUDIO_LOG_ERROR("Writing timeout, no writing event received from ISR within ticks_to_wait");
    else if (m_plCh.err == ESP_ERR_INVALID_STATE)
        AUDIO_LOG_ERROR("I2S is not ready to write");
    else
        AUDIO_LOG_ERROR("i2s err %i", m_plCh.err);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::loop() {
    if (!m_f_running) return;

    if (m_f_firstLoop) {
        m_f_firstLoop = false;
        memset(&m_lVar, 0, sizeof(m_lVar));
    }

    if (m_playlistFormat != FORMAT_M3U8) { // normal process
        switch (m_dataMode) {
            case AUDIO_LOCALFILE: processLocalFile(); break;
            case HTTP_RESPONSE_HEADER:
                if (!parseHttpResponseHeader()) {
                    if (m_f_timeout && m_lVar.count < 3) {
                        m_f_timeout = false;
                        m_lVar.count++;
                        connecttohost(m_lastHost.get());
                    }
                } else {
                    m_lVar.count = 0;
                }
                break;
            case AUDIO_PLAYLISTINIT: readPlayListData(); break;
            case AUDIO_PLAYLISTDATA:
                if (m_playlistFormat == FORMAT_M3U) httpPrint(parsePlaylist_M3U());
                if (m_playlistFormat == FORMAT_PLS) httpPrint(parsePlaylist_PLS());
                if (m_playlistFormat == FORMAT_ASX) httpPrint(parsePlaylist_ASX());
                break;
            case AUDIO_DATA:
                if (m_streamType == ST_WEBSTREAM) processWebStream();
                if (m_streamType == ST_WEBFILE) processWebFile();
                break;
        }
    } else { // m3u8 datastream only
        ps_ptr<char> host;
        if (m_lVar.no_host_timer > millis()) { return; }
        switch (m_dataMode) {
            case HTTP_RESPONSE_HEADER:
                if (!parseHttpResponseHeader()) {
                    if (m_f_timeout && m_lVar.count < 3) {
                        m_f_timeout = false;
                        m_lVar.count++;
                        m_f_reset_m3u8Codec = false;
                        connecttohost(m_lastHost.get());
                    }
                } else {
                    m_lVar.count = 0;
                    m_f_firstCall = true;
                }
                break;
            case AUDIO_PLAYLISTINIT:
                if (readPlayListData())
                    break;
                else { // readPlayListData == false means connect to m3u8 URL
                    if (m_lastM3U8host.valid()) {
                        m_f_reset_m3u8Codec = false;
                        httpPrint(m_lastM3U8host.get());
                    } else {
                        httpPrint(m_lastHost.get());
                    } // if url has no first redirection
                    m_dataMode = HTTP_RESPONSE_HEADER; // we have a new playlist now
                    break;
                }
            case AUDIO_PLAYLISTDATA:
                host = parsePlaylist_M3U8();
                if (!host.valid())
                    m_lVar.no_host_cnt++;
                else {
                    m_lVar.no_host_cnt = 0;
                    m_lVar.no_host_timer = millis();
                }

                if (m_lVar.no_host_cnt == 2) { m_lVar.no_host_timer = millis() + 2000; } // no new url? wait 2 seconds
                if (host.valid()) {                                                      // host contains the next playlist URL
                    httpPrint(host.get());
                    m_dataMode = HTTP_RESPONSE_HEADER;
                } else { // host == NULL means connect to m3u8 URL
                    if (m_lastM3U8host.valid()) {
                        m_f_reset_m3u8Codec = false;
                        httpPrint(m_lastM3U8host.get());
                    } else {
                        httpPrint(m_lastHost.get());
                    } // if url has no first redirection
                    m_dataMode = HTTP_RESPONSE_HEADER; // we have a new playlist now
                }
                break;
            case AUDIO_DATA:
                if (m_f_ts) {
                    processWebStreamTS();
                } // aac or aacp with ts packets
                else {
                    processWebStreamHLS();
                } // aac or aacp normal stream

                if (m_f_continue) { // at this point m_f_continue is true, means processWebStream() needs more data
                    m_dataMode = AUDIO_PLAYLISTDATA;
                    m_f_continue = false;
                }
                break;
        }
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::readPlayListData() {

    uint32_t     chunksize = 0;
    uint16_t     readedBytes = 0;
    ps_ptr<char> pl;
    uint32_t     ctl = 0;
    uint16_t     plSize = 0;

    auto detectTimeout = [&]() -> bool {
        uint32_t t = millis();
        while (!m_client->available()) {
            vTaskDelay(2);
            if (t + 1000 < millis()) {
                AUDIO_LOG_WARN("Playlist is incomplete, fetch again");
                if (m_f_chunked) getChunkSize(0, true);
                return true;
            }
        }
        return false;
    };

    if (m_dataMode != AUDIO_PLAYLISTINIT) {
        AUDIO_LOG_ERROR("wrong datamode %s", dataModeStr[m_dataMode]);
        goto exit;
    }

    getChunkSize(0, true);
    if (m_f_chunked) chunksize = getChunkSize(&readedBytes);
    plSize = max(m_audioFileSize, chunksize);

    if (!plSize) { // maybe playlist without contentLength or chunkSize
        if (detectTimeout()) goto exit;
        plSize = m_client->available();
    }

    // delete all memory in m_playlistContent
    if (m_playlistFormat == FORMAT_M3U8 && !psramFound()) { AUDIO_LOG_ERROR("m3u8 playlists requires PSRAM enabled!"); }
    vector_clear_and_shrink(m_playlistContent);

    while (true) { // outer while
        uint32_t ctime = millis();
        uint32_t timeout = 2000; // ms
        pl.alloc(1024);
        pl.clear(); // playlistLine

        while (true) { // inner while
            uint16_t pos = 0;
            while (true) { // super inner while :-))
                uint32_t t = millis();
                if (ctl == plSize) break;
                if (detectTimeout()) goto exit;
                pl[pos] = audioFileRead();
                ctl++;
                if (pl[pos] == '\n') {
                    pl[pos] = '\0';
                    pos++;
                    break;
                }
                if (pl[pos] == '\r') {
                    pl[pos] = '\0';
                    pos++;
                    continue;
                }
                pos++;
                if (pos == 1022) {
                    pos--;
                    continue;
                }
                if (ctl == plSize) {
                    pl[pos] = '\0';
                    break;
                }
            }
            if (pos) {
                pl[pos] = '\0';
                break;
            }
            if (ctl == plSize) break;
            if (detectTimeout()) goto exit;
        } // inner while

        if (pl.starts_with_icase("<!DOCTYPE")) {
            AUDIO_LOG_WARN("url is a webpage!");
            goto exit;
        }
        if (pl.starts_with_icase("<html")) {
            AUDIO_LOG_WARN("url is a webpage!");
            goto exit;
        }
        // AUDIO_LOG_INFO("current playlist line: %s", pl.get());
        if (pl.size() > 0) m_playlistContent.emplace_back(std::move(pl));

        // termination conditions
        // 1. The http response header returns a value for contentLength -> read chars until contentLength is reached
        // 2. no contentLength, but Transfer-Encoding:chunked -> compute chunksize and read until chunksize is reached
        // 3. no chunksize and no contentlengt, but Connection: close -> read all available chars
        if (ctl == plSize) { break; }
    } // outer while

    if (m_f_chunked) getChunkSize(&readedBytes); // expected: "\r\n\0\r\n\r\n"
    m_dataMode = AUDIO_PLAYLISTDATA;
    return true;

exit:
    vector_clear_and_shrink(m_playlistContent);
    getChunkSize(0, true);
    return false;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
const char* Audio::parsePlaylist_M3U() {

    uint8_t lines = m_playlistContent.size();
    int     pos = 0;
    char*   host = nullptr;

    for (int i = 0; i < lines; i++) {
        // m_playlistContent[i].println();
        if (m_playlistContent[i].contains("#EXTINF:")) { // Info?
            pos = m_playlistContent[i].index_of(",");    // Comma in this line?
            if (pos > 0) {
                // Show artist and title if present in metadata
                info(*this, evt_id3data, "%s", m_playlistContent[i].get() + pos + 1);
            }
            continue;
        }
        if (m_playlistContent[i].starts_with("#")) { // Commentline?
            continue;
        }

        pos = m_playlistContent[i].index_of("http://:@", 0); // ":@"??  remove that!
        if (pos >= 0) {
            AUDIO_LOG_INFO("Entry in playlist found: %s", (m_playlistContent[i].get() + pos + 9));
            host = m_playlistContent[i].get() + pos + 9;
            break;
        }
        // AUDIO_LOG_INFO("Entry in playlist found: %s", pl);
        pos = m_playlistContent[i].index_of("http", 0); // Search for "http"
        if (pos >= 0) {                                 // Does URL contain "http://"?
                                                        //    AUDIO_LOG_ERROR(""%s pos=%i", m_playlistContent[i], pos);
            host = m_playlistContent[i].get() + pos;    // Yes, set new host
            break;
        }
    }
    //    vector_clear_and_shrink(m_playlistContent);
    return host;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
const char* Audio::parsePlaylist_PLS() {
    uint8_t lines = m_playlistContent.size();
    int     pos = 0;
    char*   host = nullptr;

    for (int i = 0; i < lines; i++) {
        if (i == 0) {
            if (m_playlistContent[0].strlen() == 0) goto exit;     // empty line
            if (!m_playlistContent[0].starts_with("[playlist]")) { // first entry in valid pls
                m_dataMode = HTTP_RESPONSE_HEADER;                 // pls is not valid
                AUDIO_LOG_INFO("Playlist is not valid, switch to HTTP_RESPONSE_HEADER");
                goto exit;
            }
            continue;
        }
        if (m_playlistContent[i].starts_with("File1")) {
            if (host) continue;                             // we have already a url
            pos = m_playlistContent[i].index_of("http", 0); // File1=http://streamplus30.leonex.de:14840/;
            if (pos >= 0) {                                 // yes, URL contains "http"?
                host = m_playlistContent[i].get() + pos;    // Now we have an URL for a stream in host.
            }
            continue;
        }
        if (m_playlistContent[i].starts_with("Title1")) { // Title1=Antenne Tirol
            const char* plsStationName = (m_playlistContent[i].get() + 7);
            info(*this, evt_name, "%s", plsStationName);
            continue;
        }
        if (m_playlistContent[i].starts_with("Length1")) { continue; }
        if (m_playlistContent[i].contains("Invalid username")) { // Unable to access account:
            goto exit;                                           // Invalid username or password
        }
    }
    return host;

exit:
    m_f_running = false;
    stopSong();
    vector_clear_and_shrink(m_playlistContent);
    m_dataMode = AUDIO_NONE;
    return nullptr;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
const char* Audio::parsePlaylist_ASX() { // Advanced Stream Redirector
    uint8_t lines = m_playlistContent.size();
    bool    f_entry = false;
    int     pos = 0;
    char*   host = nullptr;

    for (int i = 0; i < lines; i++) {
        int p1 = m_playlistContent[i].index_of("<", 0);
        int p2 = m_playlistContent[i].index_of(">", 1);
        if (p1 >= 0 && p2 > p1) { // #196 set all between "< ...> to lowercase
            for (uint8_t j = p1; j < p2; j++) { m_playlistContent[i][j] = toLowerCase(m_playlistContent[i][j]); }
        }
        if (m_playlistContent[i].contains("<entry>")) f_entry = true; // found entry tag (returns -1 if not found)
        if (f_entry) {
            if (m_playlistContent[i].contains("ref href")) { //  <ref href="http://87.98.217.63:24112/stream" />
                pos = m_playlistContent[i].index_of("http", 0);
                if (pos > 0) {
                    host = (m_playlistContent[i].get() + pos); // http://87.98.217.63:24112/stream" />
                    int pos1 = indexOf(host, "\"", 0);         // http://87.98.217.63:24112/stream
                    if (pos1 > 0) host[pos1] = '\0';           // Now we have an URL for a stream in host.
                }
            }
        }
        pos = m_playlistContent[i].index_of("<title>", 0);
        if (pos >= 0) {
            char* plsStationName = (m_playlistContent[i].get() + pos + 7); // remove <Title>
            pos = indexOf(plsStationName, "</", 0);
            if (pos >= 0) {
                *(plsStationName + pos) = 0; // remove </Title>
            }
            info(*this, evt_name, "%s", plsStationName);
        }

        if (m_playlistContent[i].starts_with("http") && !f_entry) { // url only in asx
            host = m_playlistContent[i].get();
        }
    }
    return host;
}

// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
uint16_t Audio::accomplish_m3u8_url() {

    for (uint16_t i = 0; i < m_linesWithURL.size(); i++) {

        ps_ptr<char> tmp;

        if (!m_linesWithURL[i].starts_with("http")) { //  playlist:   http://station.com/aaa/bbb/xxx.m3u8
                                                      //  chunklist:  http://station.com/aaa/bbb/ddd.aac
                                                      //  result:     http://station.com/aaa/bbb/ddd.aac
            if (m_lastM3U8host.valid()) {
                tmp.clone_from(m_lastM3U8host);
            } else {
                tmp.clone_from(m_lastHost);
            }
            if (m_linesWithURL[i][0] != '/') { //  playlist:   http://station.com/aaa/bbb/xxx.m3u8  // tmp
                                               //  chunklist:  ddd.aac                              // m_linesWithURL[i]
                                               //  result:     http://station.com/aaa/bbb/ddd.aac   // m_linesWithURL[i]
                int idx = tmp.last_index_of('/');
                tmp[idx + 1] = '\0';
                tmp.append(m_linesWithURL[i].get());
                m_linesWithURL[i].clone_from(tmp);
                continue;
            } else { //  playlist:   http://station.com/aaa/bbb/xxx.m3u8
                     //  chunklist:  /aaa/bbb/ddd.aac
                     //  result:     http://station.com/aaa/bbb/ddd.aac
                int idx = tmp.index_of('/', 8);
                tmp[idx] = '\0';
                tmp.append(m_linesWithURL[i].get());
                m_linesWithURL[i].clone_from(tmp);
                continue;
            }
        } else { /* nothing todo*/
            continue;
        }
    }
    // for(uint16_t i = 0; i < m_linesWithURL.size(); i++) m_linesWithURL[i].println();
    return 0;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
int16_t Audio::prepare_first_m3u8_url(ps_ptr<char>& playlistBuff) {
    // in m_lineswithURL, searches for the first new URL through the recent URL.
    // URLs that have already been used are removed from the DEQUE
    // If all URLs are new, there is nothing to do and 0 is returned.
    // no new URL is found -1 returned
    int idx = -1;
    for (int i = 0; i < m_linesWithURL.size(); i++) {
        if (playlistBuff.equals(m_linesWithURL[i].get())) idx = i;
    }
    if (idx == -1) {
        AUDIO_LOG_DEBUG("nothing found, all entries are new");
        return 0;
    }
    if (idx == m_linesWithURL.size()) {
        AUDIO_LOG_DEBUG("only last entry found, nothing is new");
        return -1;
    }
    for (int j = 0; j < idx + 1; j++) { m_linesWithURL.pop_front(); }
    AUDIO_LOG_DEBUG("%i entries are known and removed", idx + 1);
    return idx + 1;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
ps_ptr<char> Audio::parsePlaylist_M3U8() {
    // example: audio chunks
    // #EXTM3U
    // #EXT-X-TARGETDURATION:10
    // #EXT-X-MEDIA-SEQUENCE:163374040
    // #EXT-X-DISCONTINUITY
    // #EXTINF:10,title="text=\"Spot Block End\" amgTrackId=\"9876543\"",artist=" ",url="length=\"00:00:00\""
    // http://n3fa-e2.revma.ihrhls.com/zc7729/63_sdtszizjcjbz02/main/163374038.aac
    // #EXTINF:10,title="text=\"Spot Block End\" amgTrackId=\"9876543\"",artist=" ",url="length=\"00:00:00\""
    // http://n3fa-e2.revma.ihrhls.com/zc7729/63_sdtszizjcjbz02/main/163374039.aac

    // #EXTM3U
    // #EXT-X-VERSION:3
    // #EXT-X-MEDIA-SEQUENCE:0
    // #EXT-X-TARGETDURATION:4
    // #EXTINF:3.997,90s90s - Rock
    // #EXT-X-PROGRAM-DATE-TIME:2025-08-17T14:08:21.088877044Z
    // https://hz71.streamabc.net/hls/d2gu4l4peavc72tgotd0/regc-90s90srock1436287-mp3-192-2191420/0.mp3
    // #EXTINF:3.997,90s90s - Rock
    // #EXT-X-PROGRAM-DATE-TIME:2025-08-17T14:08:24.088877044Z
    // https://hz71.streamabc.net/hls/d2gu4l4peavc72tgotd0/regc-90s90srock1436287-mp3-192-2191420/1.mp3
    // #EXTINF:3.997,90s90s - Rock
    // #EXT-X-PROGRAM-DATE-TIME:2025-08-17T14:08:28.088877044Z
    // https://hz71.streamabc.net/hls/d2gu4l4peavc72tgotd0/regc-90s90srock1436287-mp3-192-2191420/2.mp3
    // #EXTINF:3.997,90s90s - Rock
    // #EXT-X-PROGRAM-DATE-TIME:2025-08-17T14:08:32.088877044Z
    // https://hz71.streamabc.net/hls/d2gu4l4peavc72tgotd0/regc-90s90srock1436287-mp3-192-2191420/3.mp3

    if (!m_lastHost.valid()) {
        AUDIO_LOG_ERROR("m_lastHost is NULL");
        return {};
    } // guard

    uint8_t lines = m_playlistContent.size();
    bool    f_haveRedirection = false;
    char    llasc[21]; // uint64_t max = 18,446,744,073,709,551,615  thats 20 chars + \0

    if (lines) {
        bool addNextLine = false;
        deque_clear_and_shrink(m_linesWithURL);
        vector_clear_and_shrink(m_linesWithEXTINF);
        for (uint8_t i = 0; i < lines; i++) {
            // AUDIO_LOG_INFO("pl%i = %s", i, m_playlistContent[i].get());
            if (m_playlistContent[i].starts_with("#EXT-X-STREAM-INF:")) { f_haveRedirection = true; /*AUDIO_LOG_ERROR("we have a redirection");*/ }
            if (addNextLine) {
                if (startsWith(m_playlistContent[i].get(), "#EXT-X-PROGRAM-DATE-TIME:")) continue; // skip this line
                addNextLine = false;
                // size_t len = strlen(linesWithSeqNr[idx].get()) + strlen(m_playlistContent[i].get()) + 1;
                m_linesWithURL.emplace_back().clone_from(m_playlistContent[i]);
            }
            if (startsWith(m_playlistContent[i].get(), "#EXTINF:")) {
                m_linesWithEXTINF.emplace_back().clone_from(m_playlistContent[i]);
                addNextLine = true;
            }
        }
        if (!f_haveRedirection) {
            accomplish_m3u8_url();
            prepare_first_m3u8_url(m_playlistBuff);
            vector_clear_and_shrink(m_playlistContent);
        }
    }

    for (int i = 0; i < m_linesWithURL.size(); i++) { /*AUDIO_LOG_INFO("%s", m_linesWithURL[i].get())*/
        ;
    }
    for (int i = 0; i < m_linesWithEXTINF.size(); i++) { /*AUDIO_LOG_INFO("%s", m_linesWithEXTINF[i].get());*/
        showstreamtitle(m_linesWithEXTINF[i].get());
    }

    if (f_haveRedirection) {
        m_lastM3U8host.clone_from(m3u8redirection(&m_m3u8Codec));
        vector_clear_and_shrink(m_playlistContent);
        return {};
    }

    if (m_pplM3U8.firstCall) {
        m_pplM3U8.firstCall = false;
        m_pplM3U8.xMedSeq = 0;
        m_pplM3U8.f_mediaSeq_found = false;
    }

    if (m_codec == CODEC_NONE) {
        m_codec = CODEC_AAC;
        if (m_m3u8Codec == CODEC_MP3) m_codec = CODEC_MP3;
    } // if we have no redirection
      //----------------------------------------------------------------------------------------------------------------------------------------------------

    if (m_linesWithURL.size() > 0) {
        ps_ptr<char> playlistBuff;
        if (m_linesWithURL[0].valid()) {
            playlistBuff.assign(m_linesWithURL[0].get());
            m_linesWithURL.pop_front();
            m_linesWithURL.shrink_to_fit();
        }
        AUDIO_LOG_DEBUG("now playing %s", playlistBuff.get());
        if (endsWith(playlistBuff.get(), "ts")) m_f_ts = true;
        if (playlistBuff.contains(".ts?") > 0) m_f_ts = true;
        m_playlistBuff.clone_from(playlistBuff);
        return playlistBuff;
    }
    return {};
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————

ps_ptr<char> Audio::m3u8redirection(uint8_t* codec) {
    // example: redirection
    // #EXTM3U
    // #EXT-X-STREAM-INF:BANDWIDTH=117500,AVERAGE-BANDWIDTH=117000,CODECS="mp4a.40.2"
    // 112/playlist.m3u8?hlssid=7562d0e101b84aeea0fa35f8b963a174
    // #EXT-X-STREAM-INF:BANDWIDTH=69500,AVERAGE-BANDWIDTH=69000,CODECS="mp4a.40.5"
    // 64/playlist.m3u8?hlssid=7562d0e101b84aeea0fa35f8b963a174
    // #EXT-X-STREAM-INF:BANDWIDTH=37500,AVERAGE-BANDWIDTH=37000,CODECS="mp4a.40.29"
    // 32/playlist.m3u8?hlssid=7562d0e101b84aeea0fa35f8b963a174

    if (!m_lastHost.valid()) {
        AUDIO_LOG_ERROR("m_lastHost is empty");
        return {};
    } // guard
    const char codecString[9][11] = {
        "mp4a.40.34", // mp3 stream
        "mp4a.40.01", // AAC Main
        "mp4a.40.2",  // MPEG-4 AAC LC
        "mp4a.40.02", // MPEG-4 AAC LC, leading 0 for Aud-OTI compatibility
        "mp4a.40.29", // MPEG-4 HE-AAC v2 (AAC LC + SBR + PS)
        "mp4a.40.42", // xHE-AAC
        "mp4a.40.5",  // MPEG-4 HE-AAC v1 (AAC LC + SBR)
        "mp4a.40.05", // MPEG-4 HE-AAC v1 (AAC LC + SBR), leading 0 for Aud-OTI compatibility
        "mp4a.67",    // MPEG-2 AAC LC
    };

    uint16_t choosenLine = 0;
    uint16_t plcSize = m_playlistContent.size();
    int8_t   cS = 100;

    for (uint16_t i = 0; i < plcSize; i++) { // looking for lowest codeString
        if (m_playlistContent[i].contains("CODECS=\"mp4a")) {
            for (uint8_t j = 0; j < 9; j++) {
                if (m_playlistContent[i].contains(codecString[j])) {
                    if (j < cS) {
                        cS = j;
                        choosenLine = i;
                    }
                }
            }
        }
    }
    if (cS == 0)
        *codec = CODEC_MP3;
    else if (cS < 100)
        *codec = CODEC_AAC;

    if (cS == 100) {                             // "mp4a.xx.xx" not found
        *codec = CODEC_AAC;                      // assume AAC
        for (uint16_t i = 0; i < plcSize; i++) { // we have no codeString, looking for "http"
            if (m_playlistContent[i].contains("#EXT-X-STREAM-INF")) { choosenLine = i; }
        }
    }

    choosenLine++; // next line is the redirection url

    ps_ptr<char> result;
    ps_ptr<char> line;
    line.clone_from(m_playlistContent[choosenLine]);

    if (line.starts_with("../")) {
        // ../../2093120-b/RISMI/stream01/streamPlaylist.m3u8
        while (line.starts_with("../")) { line.remove_prefix("../"); }
        result.clone_from(m_currentHost);
        int idx = result.last_index_of('/');
        if (idx > 0 && (size_t)idx != result.strlen() - 1) result.truncate_at((size_t)idx + 1);
        result.append(line.get());
    } else if (!line.starts_with_icase("http")) {

        // http://arcast.com.ar:1935/radio/radionar.stream/playlist.m3u8               m_lastHost
        // chunklist_w789468022.m3u8                                                   line
        // http://arcast.com.ar:1935/radio/radionar.stream/chunklist_w789468022.m3u8   --> result

        result.clone_from(m_currentHost);
        int pos = m_currentHost.last_index_of('/');
        if (pos > 0) {
            result.truncate_at((size_t)pos + 1);
            result.append(line.get());
        }
    } else {
        result.clone_from(line);
    }

    return result; // it's a redirection, a new m3u8 playlist
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::processLocalFile() {
    if (!(m_audiofile && m_f_running && m_dataMode == AUDIO_LOCALFILE)) return; // guard

    m_prlf.availableBytes = 0;
    m_prlf.bytesAddedToBuffer = 0;

    if (m_f_firstCall) { // runs only one time per connection, prepare for start
        m_f_firstCall = false;
        m_f_stream = false;
        m_prlf.audioHeaderFound = false;
        m_prlf.newFilePos = 0;
        m_prlf.ctime = millis();
        m_audioFilePosition = 0;
        m_audioDataSize = m_audioFileSize;
        m_audioDataStart = 0;
        m_f_allDataReceived = false;
        m_prlf.timeout = 8000; // ms
    }

    if (m_resumeFilePos >= 0) { // we have a resume file position
        m_prlf.newFilePos = newInBuffStart(m_resumeFilePos);
        if (m_prlf.newFilePos < 0) AUDIO_LOG_WARN("skip to new position was not successful");
        m_haveNewFilePos = m_prlf.newFilePos;
        m_audioDataReadPtr = m_prlf.newFilePos;
        m_resumeFilePos = -1;
        m_f_allDataReceived = false;
        return;
    }

    m_prlf.availableBytes = min(InBuff.writeSpace(), (size_t)(m_audioFileSize - m_audioFilePosition));
    m_prlf.bytesAddedToBuffer = audioFileRead(InBuff.getWritePtr(), min(m_prlf.availableBytes, (uint32_t)UINT16_MAX));
    if (m_prlf.bytesAddedToBuffer > 0) { InBuff.bytesWritten(m_prlf.bytesAddedToBuffer); }
    if (m_audioDataSize && m_audioFilePosition >= m_audioDataSize) {
        if (!m_f_allDataReceived) m_f_allDataReceived = true;
    }
    if (!m_audioDataSize && m_audioFilePosition == m_audioFileSize) {
        if (!m_f_allDataReceived) m_f_allDataReceived = true;
    }
    // AUDIO_LOG_DEBUG("m_audioFilePosition %u >= m_audioDataSize %u, m_f_allDataReceived % i", m_audioFilePosition, m_audioDataSize, m_f_allDataReceived);

    if (!m_decoder && InBuff.bufferFilled() > 127) {
        if (!initializeDecoder()) return;
        m_prlf.maxFrameSize = InBuff.getMaxBlockSize();
    }

    if (!m_f_stream) {
        if (m_controlCounter != 100) {
            if (m_f_ogg) { m_controlCounter = 100; }
            if ((millis() - m_prlf.ctime) > m_prlf.timeout) {
                AUDIO_LOG_ERROR("audioHeader reading timeout");
                m_f_running = false;
                goto exit;
            }
            if (InBuff.bufferFilled() > m_prlf.maxFrameSize || (InBuff.bufferFilled() == m_audioFileSize) || m_f_allDataReceived) { // at least one complete frame or the file is smaller
                InBuff.bytesWasRead(readAudioHeader(InBuff.readSpace()));
            }
            if (m_controlCounter == 100) {
                if (m_audioDataStart > 0) { m_prlf.audioHeaderFound = true; }
                if (!m_audioDataSize) m_audioDataSize = m_audioFileSize;
            }
            return;
        } else {
            m_f_stream = true;
            info(*this, evt_info, "stream ready");
        }
    }

    if (m_fileStartTime > 0 && m_nominal_bitrate) {
        if (getBitRate() > 0)
            setAudioPlayTime(m_fileStartTime);
        else
            info(*this, evt_info, "can't set audio play time directly");
        m_fileStartTime = -1;
    }

    // end of file reached? - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_f_eof) { // m_f_eof and m_f_ID3v1TagFound will be set in playAudioData()
        if (m_f_ID3v1TagFound) readID3V1Tag();
    exit:
        ps_ptr<char> afn;                                // audio file name
        if (m_audiofile) afn.assign(m_audiofile.name()); // store temporary the name
        m_audioCurrentTime = 0;
        m_audioFileDuration = 0;
        m_resumeFilePos = -1;
        m_haveNewFilePos = 0;
        m_codec = CODEC_NONE;
        stopSong();

        if (afn.valid()) { info(*this, evt_eof, "%s", afn.c_get()); }
        return;
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
void Audio::processWebStream() {

    if (m_dataMode != AUDIO_DATA) return; // guard
    uint16_t readedBytes = 0;

    m_pwst.availableBytes = 0; // available from stream
    m_pwst.f_clientIsConnected = m_client->connected();

    // first call, set some values to default  - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_f_firstCall) { // runs only ont time per connection, prepare for start
        m_f_firstCall = false;
        m_f_stream = false;
        m_pwst.chunkSize = 0;
        m_metacount = m_metaint;
        m_f_allDataReceived = false;
        readMetadata(0, &readedBytes, true);
        getChunkSize(0, true);
        m_audioFilePosition = 0;
    }
    if (m_pwst.f_clientIsConnected) m_pwst.availableBytes = m_client->available(); // available from stream

    // chunked data tramsfer
    if (m_f_chunked && m_pwst.availableBytes) {
        if (m_pwst.chunkSize == 0) {
            vTaskDelay(1);
            int chunkLen = getChunkSize(&m_pwst.readedBytes);
            if (chunkLen < 0) return;
            if (chunkLen == 0) m_f_allDataReceived = true;
            m_pwst.chunkSize = chunkLen;
            m_pwst.readedBytes = 0; // readedBytes is not a part of chunkSize
        }
        m_pwst.availableBytes = min(m_pwst.availableBytes, m_pwst.chunkSize);
    }

    // we have metadata  - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_f_metadata && (m_metacount == 0)) {
        if (!m_pwst.availableBytes) return;
        readedBytes = 0;
        bool res = readMetadata(m_pwst.availableBytes, &readedBytes);
        m_pwst.readedBytes += readedBytes;
        if (m_f_chunked) m_pwst.chunkSize -= readedBytes; // reduce chunkSize by metadata length
        if (res == false) return;
        m_metacount = m_metaint;
        return;
    }
    if (m_f_metadata) m_pwst.availableBytes = min(m_pwst.availableBytes, m_metacount);

    // if the buffer is often almost empty issue a warning - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_f_stream) {
        if (!m_f_allDataReceived)
            if (streamDetection(m_pwst.availableBytes)) return;
        if (!m_pwst.f_clientIsConnected) {
            if (m_f_tts && !m_f_allDataReceived) m_f_allDataReceived = true;
        } // connection closed (OpenAi)
    }

    // buffer fill routine - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_pwst.availableBytes) {
        m_pwst.availableBytes = min(m_pwst.availableBytes, (uint32_t)InBuff.writeSpace());
        int32_t bytesAddedToBuffer = audioFileRead(InBuff.getWritePtr(), min(m_pwst.availableBytes, (uint32_t)UINT16_MAX));
        if (bytesAddedToBuffer > 0) {
            if (m_f_metadata) m_metacount -= bytesAddedToBuffer;
            if (m_f_chunked) m_pwst.chunkSize -= bytesAddedToBuffer;
            InBuff.bytesWritten(bytesAddedToBuffer);
        }
    }
    if (!m_decoder && InBuff.bufferFilled() > 127) {
        if (initializeDecoder())
            m_pwst.maxFrameSize = InBuff.getMaxBlockSize();
        else
            return;
    }

    // start audio decoding - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (InBuff.bufferFilled() * 2 > m_pwst.maxFrameSize && !m_f_stream) { // waiting for buffer filled
        info(*this, evt_info, "stream ready");
        m_f_stream = true; // ready to play the audio data
    }

    if (m_f_eof) {
        info(*this, evt_eof, "%s", m_lastHost.c_get());
        stopSong();
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::processWebFile() {

    if (!m_lastHost.valid()) {
        AUDIO_LOG_ERROR("m_lastHost is empty");
        return;
    } // guard
    uint32_t availableBytes = 0;
    int32_t  bytesAddedToBuffer = 0;
    // m_pwf.f_clientIsConnected = m_client->connected();     // we are not connected

    if (m_f_firstCall) { // runs only ont time per connection, prepare for start
        m_f_firstCall = false;
        m_f_stream = false;
        m_controlCounter = 0;
        m_pwf.audioHeaderFound = false;
        m_pwf.newFilePos = 0;
        m_pwf.ctime = millis();
        m_audioFilePosition = 0;
        m_audioDataSize = m_audioFileSize;
        m_audioDataStart = 0;
        m_f_allDataReceived = false;
        m_pwf.timeout = 8000; // ms
        if (!initializeDecoder()) return;
        m_pwf.maxFrameSize = InBuff.getMaxBlockSize(); // every frame is not bigger
    }

    if (m_resumeFilePos >= 0) { // we have a resume file position
        m_pwf.newFilePos = newInBuffStart(m_resumeFilePos);
        if (m_pwf.newFilePos < 0) AUDIO_LOG_WARN("skip to new position was not successful");
        m_haveNewFilePos = m_pwf.newFilePos;
        m_audioDataReadPtr = m_pwf.newFilePos;
        m_resumeFilePos = -1;
        m_f_allDataReceived = false;
        return;
    }

    m_pwf.availableBytes = min(m_client->available(), (int)InBuff.writeSpace());
    m_pwf.bytesAddedToBuffer = audioFileRead(InBuff.getWritePtr(), min(m_pwf.availableBytes, (uint32_t)UINT16_MAX));
    if (m_pwf.bytesAddedToBuffer > 0) { InBuff.bytesWritten(m_pwf.bytesAddedToBuffer); }
    if (m_audioDataSize && m_audioFilePosition >= m_audioDataSize) {
        if (!m_f_allDataReceived) m_f_allDataReceived = true;
    }
    if (!m_audioDataSize && m_audioFilePosition == m_audioFileSize) {
        if (!m_f_allDataReceived) m_f_allDataReceived = true;
    }
    // AUDIO_LOG_ERROR("m_audioFilePosition %u >= m_audioDataSize %u, m_f_allDataReceived % i", m_audioFilePosition, m_audioDataSize, m_f_allDataReceived);
    if (!m_decoder && InBuff.bufferFilled() > 127) {
        if (!initializeDecoder()) return;
    }

    if (!m_f_stream) {
        if (m_controlCounter != 100) {
            if (m_f_ogg) { m_controlCounter = 100; }
            if ((millis() - m_pwf.ctime) > m_pwf.timeout) {
                AUDIO_LOG_ERROR("audioHeader reading timeout");
                m_f_running = false;
                goto exit;
            }
            if (InBuff.bufferFilled() > m_pwf.maxFrameSize || (InBuff.bufferFilled() == m_audioFileSize) || m_f_allDataReceived) { // at least one complete frame or the file is smaller
                InBuff.bytesWasRead(readAudioHeader(InBuff.readSpace()));
            }
            if (m_controlCounter == 100) {
                if (m_audioDataStart > 0) { m_pwf.audioHeaderFound = true; }
                if (!m_audioDataSize) m_audioDataSize = m_audioFileSize;
            }
            return;
        } else {
            if (m_resumeFilePos == -1) {
                if (InBuff.bufferFilled() > 2 * InBuff.getMaxBlockSize()) {
                    m_f_stream = true;
                    info(*this, evt_info, "stream ready");
                }
            }
        }
    }

    if (m_fileStartTime > 0 && m_nominal_bitrate) {
        if (getBitRate() > 0)
            setAudioPlayTime(m_fileStartTime);
        else
            info(*this, evt_info, "can't set audio play time directly");
        m_fileStartTime = -1;
    }

    // end of file reached? - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_f_eof) { // m_f_eof and m_f_ID3v1TagFound will be set in playAudioData()
        if (m_f_ID3v1TagFound) readID3V1Tag();
    exit:
        stopSong();
        info(*this, evt_eof, "%s", m_lastHost.c_get());

        m_audioCurrentTime = 0;
        m_audioFileDuration = 0;
        m_resumeFilePos = -1;
        m_haveNewFilePos = 0;
        m_codec = CODEC_NONE;
        return;
    }
}
// ——————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::processWebStreamTS() {
    uint32_t availableBytes; // available bytes in stream
    uint8_t  ts_packetStart = 0;
    uint8_t  ts_packetLength = 0;

    // first call, set some values to default ———————————————————————————————————
    if (m_f_firstCall) { // runs only one time per connection, prepare for start
        m_f_firstCall = false;
        m_f_m3u8data = true;
        m_audioFilePosition = 0;
        m_pwsst.f_firstPacket = true;
        m_pwsst.f_chunkFinished = false;
        m_pwsst.f_nextRound = false;
        m_pwsst.byteCounter = 0;
        m_pwsst.chunkSize = 0;
        m_pwsst.ts_packetPtr = 0;
        m_t0 = millis();
        getChunkSize(0, true);
        ts_parsePacket(0, 0, 0);
        if (!m_pwsst.ts_packet.valid()) m_pwsst.ts_packet.alloc_array(m_pwsst.ts_packetsize, "m_pwsst.ts_packet"); // first init
        if (!m_decoder && !initializeDecoder()) return;
    } // —————————————————————————————————————————————————————————————————————————

    if (m_dataMode != AUDIO_DATA) return; // guard

nextRound:
    availableBytes = m_client->available();
    if (availableBytes) {
        /* If the m3u8 stream uses 'chunked data transfer' no content length is supplied. Then the chunk size determines the audio data to be processed.
           However, the chunk size in some streams is limited to 32768 bytes, although the chunk can be larger. Then the chunk size is
           calculated again. The data used to calculate (here readedBytes) the chunk size is not part of it.
        */
        uint16_t readedBytes = 0;
        uint32_t minAvBytes = 0;
        if (m_pwsst.f_chunkFinished) goto chunkFinished;
        if (m_f_chunked && m_pwsst.chunkSize == m_pwsst.byteCounter) {
            int cs = getChunkSize(&readedBytes);
            if (cs == -1) return;
            m_pwsst.chunkSize += cs;
            AUDIO_LOG_DEBUG("cs %i, rb %i", cs, readedBytes);
            if (cs == 0) {
                m_pwsst.f_chunkFinished = true;
                m_pwsst.chunkSize = 0;
                m_pwsst.byteCounter = 0;
                goto chunkFinished;
            }
        }
        if (m_pwsst.chunkSize)
            minAvBytes = min3(availableBytes, m_pwsst.ts_packetsize - m_pwsst.ts_packetPtr, m_pwsst.chunkSize - m_pwsst.byteCounter);
        else
            minAvBytes = min(availableBytes, (uint32_t)(m_pwsst.ts_packetsize - m_pwsst.ts_packetPtr));

        int res = audioFileRead(m_pwsst.ts_packet.get() + m_pwsst.ts_packetPtr, minAvBytes);
        if (res > 0) {
            m_pwsst.ts_packetPtr += res;
            m_pwsst.byteCounter += res;
            if (m_pwsst.ts_packetPtr < m_pwsst.ts_packetsize) return; // not enough data yet, the process must be repeated if the packet size (188 bytes) is not reached
            m_pwsst.ts_packetPtr = 0;
            if (m_pwsst.f_firstPacket) { // search for ID3 Header in the first packet
                m_pwsst.f_firstPacket = false;
                uint8_t ID3_HeaderSize = process_m3u8_ID3_Header(m_pwsst.ts_packet.get());
                if (ID3_HeaderSize > m_pwsst.ts_packetsize) {
                    AUDIO_LOG_ERROR("ID3 Header is too big");
                    stopSong();
                    return;
                }
                if (ID3_HeaderSize) {
                    memcpy(m_pwsst.ts_packet.get(), &m_pwsst.ts_packet.get()[ID3_HeaderSize], m_pwsst.ts_packetsize - ID3_HeaderSize);
                    m_pwsst.ts_packetPtr = m_pwsst.ts_packetsize - ID3_HeaderSize;
                    return;
                }
            }
            if (!ts_parsePacket(&m_pwsst.ts_packet.get()[0], &ts_packetStart, &ts_packetLength)) {
                stopSong();
                AUDIO_LOG_ERROR("song stopped");
                return;
            };

            if (ts_packetLength) {
                size_t ws = InBuff.writeSpace();
                if (ws >= ts_packetLength) {
                    memcpy(InBuff.getWritePtr(), m_pwsst.ts_packet.get() + ts_packetStart, ts_packetLength);
                    InBuff.bytesWritten(ts_packetLength);
                    m_pwsst.f_nextRound = true;
                } else {
                    int t = 0;
                    memcpy(InBuff.getWritePtr(), m_pwsst.ts_packet.get() + ts_packetStart, ws); // write everything that fits into the buffer
                    InBuff.bytesWritten(ws);
                    while (true) {
                        if (InBuff.writeSpace() >= ts_packetLength - ws) {
                            memcpy(InBuff.getWritePtr(), &m_pwsst.ts_packet.get()[ws + ts_packetStart], ts_packetLength - ws); // write the rest
                            InBuff.bytesWritten(ts_packetLength - ws);
                            break;
                        }
                        t++;
                        vTaskDelay(10); // wait until the buffer is free
                        if (t == 10) {
                            AUDIO_LOG_ERROR("InBuff is full");
                            break;
                        }
                    }
                }
            }
            if (m_audioFileSize && m_pwsst.byteCounter > m_audioFileSize) {
                AUDIO_LOG_ERROR("byteCounter overflow, byteCounter: %d, contentlength: %d", m_pwsst.byteCounter, m_audioFileSize);
                return;
            }
            if (m_pwsst.chunkSize && m_pwsst.byteCounter > m_pwsst.chunkSize) {
                AUDIO_LOG_ERROR("byteCounter overflow, byteCounter: %d, chunkSize: %d", m_pwsst.byteCounter, m_pwsst.chunkSize);
                return;
            }
        }
    }
    if (m_audioFileSize && m_pwsst.byteCounter == m_audioFileSize) {
        if (InBuff.bufferFilled() < 120000) {
            m_f_continue = true;
            m_pwsst.byteCounter = 0;
            m_pwsst.ts_packetPtr = 0;
            m_pwsst.f_nextRound = false;
        }
        goto exit;
    }

chunkFinished:
    if (m_pwsst.f_chunkFinished) {
        if (InBuff.bufferFilled() < 120000) {
            m_pwsst.f_chunkFinished = false;
            m_f_continue = true;
            m_pwsst.byteCounter = 0;
            m_pwsst.chunkSize = 0;
            m_pwsst.ts_packetPtr = 0;
        }
        goto exit;
    }

    // if the buffer is often almost empty issue a warning - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_f_stream) {
        if (streamDetection(availableBytes)) goto exit;
    }

    // buffer fill routine  - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (true) {                                             // statement has no effect
        if (InBuff.bufferFilled() > 60000 && !m_f_stream) { // waiting for buffer filled
            m_f_stream = true;                              // ready to play the audio data
            uint16_t filltime = millis() - m_t0;
            info(*this, evt_info, "stream ready");
            info(*this, evt_info, "buffer filled in %d ms", filltime);
        }
    }
    if (m_pwsst.f_nextRound) { goto nextRound; }
exit:
    return;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::processWebStreamHLS() {

    // first call, set some values to default - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (m_f_firstCall) { // runs only ont time per connection, prepare for start
        m_f_firstCall = false;
        m_f_m3u8data = true;
        m_t0 = millis();
        m_controlCounter = 0;
        m_audioFilePosition = 0;
        m_pwsHLS.ID3BuffSize = 4096;
        m_pwsHLS.availableBytes = 0;
        m_pwsHLS.firstBytes = true;
        m_pwsHLS.f_chunkFinished = false;
        m_pwsHLS.byteCounter = 0;
        m_pwsHLS.chunkSize = 0;
        m_pwsHLS.ID3WritePtr = 0;
        m_pwsHLS.ID3ReadPtr = 0;
        m_pwsHLS.ID3Buff.alloc(m_pwsHLS.ID3BuffSize, "m_pwsHLS.ID3Buff");
        getChunkSize(0, true);
        if (!m_decoder && !initializeDecoder()) return;
        m_pwsHLS.maxFrameSize = InBuff.getMaxBlockSize(); // every mp3/aac frame is not bigger
    }

    if (m_dataMode != AUDIO_DATA) return; // guard

    m_pwsHLS.availableBytes = m_client->available();
    if (m_pwsHLS.availableBytes) { // an ID3 header could come here
        uint16_t readedBytes = 0;

        if (m_f_chunked && !m_pwsHLS.chunkSize) {
            m_pwsHLS.chunkSize = getChunkSize(&readedBytes);
            m_pwsHLS.byteCounter += readedBytes;
        }

        if (m_pwsHLS.firstBytes) {
            if (m_pwsHLS.ID3WritePtr < m_pwsHLS.ID3BuffSize) {
                m_pwsHLS.ID3WritePtr += audioFileRead(&m_pwsHLS.ID3Buff[m_pwsHLS.ID3WritePtr], m_pwsHLS.ID3BuffSize - m_pwsHLS.ID3WritePtr);
                return;
            }
            if (m_controlCounter < 100) {
                int res = read_ID3_Header(&m_pwsHLS.ID3Buff[m_pwsHLS.ID3ReadPtr], m_pwsHLS.ID3BuffSize - m_pwsHLS.ID3ReadPtr);
                if (res >= 0) m_pwsHLS.ID3ReadPtr += res;
                if (m_pwsHLS.ID3ReadPtr > m_pwsHLS.ID3BuffSize) {
                    AUDIO_LOG_ERROR("buffer overflow");
                    stopSong();
                    return;
                }
                return;
            }
            if (m_controlCounter != 100) return;

            size_t ws = InBuff.writeSpace();
            if (ws >= m_pwsHLS.ID3BuffSize - m_pwsHLS.ID3ReadPtr) {
                memcpy(InBuff.getWritePtr(), &m_pwsHLS.ID3Buff[m_pwsHLS.ID3ReadPtr], m_pwsHLS.ID3BuffSize - m_pwsHLS.ID3ReadPtr);
                InBuff.bytesWritten(m_pwsHLS.ID3BuffSize - m_pwsHLS.ID3ReadPtr);
            } else {
                int t = 0;
                memcpy(InBuff.getWritePtr(), &m_pwsHLS.ID3Buff[m_pwsHLS.ID3ReadPtr], ws);
                InBuff.bytesWritten(ws);
                while (true) {
                    if (InBuff.writeSpace() >= m_pwsHLS.ID3BuffSize - (m_pwsHLS.ID3ReadPtr + ws)) {
                        memcpy(InBuff.getWritePtr(), &m_pwsHLS.ID3Buff[ws + m_pwsHLS.ID3ReadPtr], m_pwsHLS.ID3BuffSize - (m_pwsHLS.ID3ReadPtr + ws)); // write everything that fits into the buffer
                        InBuff.bytesWritten(m_pwsHLS.ID3BuffSize - (m_pwsHLS.ID3ReadPtr + ws));
                        break;
                    }
                    t++;
                    vTaskDelay(10); // wait until the buffer is free
                    if (t == 10) {
                        AUDIO_LOG_ERROR("InBuff is full");
                        break;
                    }
                }
            }
            m_pwsHLS.ID3Buff.reset();
            m_pwsHLS.byteCounter += m_pwsHLS.ID3BuffSize;
            m_pwsHLS.firstBytes = false;
        }

        size_t bytesWasWritten = 0;
        if (InBuff.writeSpace() >= m_pwsHLS.availableBytes) {
            //    if(availableBytes > 1024) availableBytes = 1024; // 1K throttle
            bytesWasWritten = audioFileRead(InBuff.getWritePtr(), m_pwsHLS.availableBytes);
        } else {
            bytesWasWritten = audioFileRead(InBuff.getWritePtr(), InBuff.writeSpace());
        }
        InBuff.bytesWritten(bytesWasWritten);

        m_pwsHLS.byteCounter += bytesWasWritten;

        if (m_pwsHLS.byteCounter == m_audioFileSize || m_pwsHLS.byteCounter == m_pwsHLS.chunkSize) {
            m_pwsHLS.f_chunkFinished = true;
            m_pwsHLS.byteCounter = 0;
        }
    }

    if (m_pwsHLS.f_chunkFinished) {
        if (InBuff.bufferFilled() < 50000) {
            m_pwsHLS.f_chunkFinished = false;
            m_f_continue = true;
        }
    }

    // if the buffer is often almost empty issue a warning or try a new connection - - - - - - - - - - - - - - - - - - -
    if (m_f_stream) {
        if (streamDetection(m_pwsHLS.availableBytes)) return;
    }

    if (InBuff.bufferFilled() > m_pwsHLS.maxFrameSize && !m_f_stream) { // waiting for buffer filled
        m_f_stream = true;                                              // ready to play the audio data
        // uint16_t filltime = millis() - m_t0;
        info(*this, evt_info, "stream ready");
        // info(*this, evt_info, "buffer filled in %u ms", filltime);
    }
    return;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::playAudioData() {

    if (!m_f_stream || m_f_eof || m_f_lockInBuffer || !m_f_running || m_codec == CODEC_NONE) {
        m_validSamples = 0;
        return;
    } // guard, stream not ready or eof reached or InBuff is locked or not running
    if (m_validSamples) {
        playChunk();
        return;
    } // guard, play samples first
    //--------------------------------------------------------------------------------
    m_pad.count = 0;
    m_pad.bytesToDecode = InBuff.readSpace();
    m_pad.bytesDecoded = 0;

    if (m_f_firstPlayCall) {
        m_audioDataReadPtr = 0;
        m_f_firstPlayCall = false;
        m_pad.count = 0;
        m_pad.oldAudioDataSize = 0;
        m_bytesNotConsumed = 0;
        m_pad.lastFrames = false;
        m_f_eof = false;
    }
    //--------------------------------------------------------------------------------

    bool isFile = false;
    bool isStream = false;

    if (m_dataMode == AUDIO_LOCALFILE) isFile = true;
    if (m_streamType == ST_WEBFILE && m_playlistFormat != FORMAT_M3U8) isFile = true; // local file or webfile but not m3u8 file
    if (m_streamType == ST_WEBSTREAM || m_playlistFormat == FORMAT_M3U8) isStream = true;
    if (!isFile && !isStream) return;

    xSemaphoreTake(mutex_audioTaskIsDecoding, 0.3 * configTICK_RATE_HZ);
    {
        m_pad.bytesDecoded = 0;
        if (isFile) {
            if (!m_audioDataSize) goto exit; // no data to decode if filesize is 0
            if (m_audioDataStart + m_audioDataSize - m_audioDataReadPtr == 128) {
                m_f_ID3v1TagFound = true;
                m_f_eof = true;
                goto exit;
            }
            if (m_audioDataSize <= m_audioDataReadPtr) {
                m_f_eof = true;
                goto exit;
            }

            if (m_audioDataStart + m_audioDataSize >= m_audioFilePosition) m_f_allDataReceived = true;
            if (m_audioDataSize - m_audioDataReadPtr <= InBuff.getMaxBlockSize()) m_pad.lastFrames = true;

            if (m_pad.lastFrames) {
                m_pad.bytesToDecode = min(InBuff.readSpace(), (size_t)(m_audioDataStart + m_audioDataSize - m_audioDataReadPtr));
                m_pad.bytesDecoded = sendBytes(InBuff.getReadPtr(), m_pad.bytesToDecode);
            } else {
                m_pad.bytesToDecode = InBuff.readSpace();
                if (m_pad.bytesToDecode >= InBuff.getMaxBlockSize()) { m_pad.bytesDecoded = sendBytes(InBuff.getReadPtr(), m_pad.bytesToDecode); }
            }
        }

        if (isStream) {
            if (m_f_allDataReceived) { // Google TTS, OpenAI
                m_pad.lastFrames = true;
                if (m_f_tts && !InBuff.bufferFilled()) {
                    m_f_eof = true;
                    goto exit;
                }
            }
            if (m_pad.lastFrames) {
                m_pad.bytesToDecode = min(InBuff.readSpace(), (size_t)(m_audioDataStart + m_audioDataSize - m_audioDataReadPtr));
                m_pad.bytesDecoded = sendBytes(InBuff.getReadPtr(), m_pad.bytesToDecode);
            } else {
                m_pad.bytesToDecode = InBuff.readSpace();
                if (m_pad.bytesToDecode >= InBuff.getMaxBlockSize()) { m_pad.bytesDecoded = sendBytes(InBuff.getReadPtr(), m_pad.bytesToDecode); }
            }
        }

        if (m_pad.bytesDecoded > 0) {
            InBuff.bytesWasRead(m_pad.bytesDecoded);
            m_audioDataReadPtr += m_pad.bytesDecoded;
        }
    }
exit:
    xSemaphoreGive(mutex_audioTaskIsDecoding);

    AUDIO_LOG_DEBUG("m_audioDataReadPtr %i, m_audioDataSize %i", m_audioDataReadPtr, m_audioDataSize);
    return;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::parseHttpResponseHeader() { // this is the response to a GET / request

    if (m_dataMode != HTTP_RESPONSE_HEADER) return false;
    if (!m_currentHost.valid()) {
        AUDIO_LOG_ERROR("m_currentHost is empty");
        return false;
    }

    m_phreh.reset();
    m_phreh.ctime = millis();
    m_phreh.timeout = 4500; // ms

    if (m_client->available() == 0) {
        if (!m_phreh.f_time) {
            m_phreh.stime = millis();
            m_phreh.f_time = true;
        }
        if ((millis() - m_phreh.stime) > m_phreh.timeout) {
            AUDIO_LOG_ERROR("timeout");
            m_phreh.f_time = false;
            return false;
        }
    }
    m_phreh.f_time = false;

    ps_ptr<char> rhl;
    rhl.alloc(1024, "rhl"); // responseHeaderline
    rhl.clear();
    bool ct_seen = false;

    while (true) { // outer while
        uint16_t pos = 0;
        if ((millis() - m_phreh.ctime) > m_phreh.timeout) {
            AUDIO_LOG_ERROR("timeout");
            m_f_timeout = true;
            goto exit;
        }
        while (m_client->available()) {
            uint8_t b = audioFileRead();
            if (b == '\n') {
                if (!pos) { // empty line received, is the last line of this responseHeader
                    if (ct_seen)
                        goto lastToDo;
                    else {
                        if (!m_client->available()) goto exit;
                    }
                }
                break;
            }
            if (b == '\r') rhl[pos] = 0;
            if (b < 0x20) continue;
            rhl[pos] = b;
            pos++;
            if (pos == 1023) {
                pos = 1022;
                continue;
            }
            if (pos == 1022) {
                rhl[pos] = '\0';
                AUDIO_LOG_WARN("responseHeaderline overflow");
            }
        } // inner while
        if (!pos) {
            vTaskDelay(5);
            continue;
        }

        // rhl.println();
        if (rhl.starts_with_icase("icy-")) {
            m_phreh.f_icy_data = true; // is webstrean
        }

        if (rhl.starts_with_icase("HTTP/")) { // HTTP status error code
            char statusCode[5];
            statusCode[0] = rhl[9];
            statusCode[1] = rhl[10];
            statusCode[2] = rhl[11];
            statusCode[3] = '\0';
            int sc = atoi(statusCode);
            if (sc > 310) { // e.g. HTTP/1.1 301 Moved Permanently
                info(*this, evt_streamtitle, "%s", rhl.get());
                goto exit;
            }
        } else if (rhl.starts_with_icase("content-type:")) { // content-type: text/html; charset=UTF-8
            int idx = rhl.index_of(';', 13);
            if (idx > 0) rhl[idx] = '\0';
            if (parseContentType(rhl.get() + 13))
                ct_seen = true;
            else {
                AUDIO_LOG_WARN("unknown contentType %s", rhl.get() + 13);
                goto exit;
            }
        }

        else if (rhl.starts_with_icase("location:")) {
            int pos = rhl.index_of_icase("http", 0);
            if (pos == -1) {
                int doubleSlash = rhl.index_of("//");                   // e.g. location: //frontend.streamonkey.net/...  host: http://webstream.radiof.de/
                if (doubleSlash >= 9) rhl.insert("http:", doubleSlash); // ==> http://frontend.streamonkey.net/fhn-radiof945/stream/mp3?aggregator=fh-tinyurl
                pos = rhl.index_of_icase("http", 0);
            }
            if (pos >= 0) {
                const char* c_host = (rhl.get() + pos);
                if (!m_currentHost.equals(c_host)) { // prevent a loop
                    int pos_slash = indexOf(c_host, "/", 9);
                    if (pos_slash > 9) {
                        if (!strncmp(c_host, m_currentHost.get(), pos_slash)) {
                            info(*this, evt_info, "redirect to new extension at existing host \"%s\"", c_host);
                            if (m_playlistFormat == FORMAT_M3U8) {
                                //    m_lastHost.assign(c_host);
                                m_f_m3u8data = true;
                            }
                            httpPrint(c_host);
                            while (m_client->available()) audioFileRead(); // empty client buffer
                            return true;
                        }
                    }
                    info(*this, evt_info, "redirect to new host \"%s\"", c_host);
                    m_f_reset_m3u8Codec = false;
                    httpPrint(c_host);
                    return true;
                }
            } else {
                AUDIO_LOG_WARN("unknown redirection: %s", rhl.c_get());
            }
        }

        else if (rhl.starts_with_icase("content-encoding:")) {
            info(*this, evt_info, "%s", rhl.get());
            if (rhl.contains("gzip")) {
                AUDIO_LOG_ERROR("can't extract gzip");
                goto exit;
            }
        }

        else if (rhl.starts_with_icase("content-disposition:")) { // e.g we have this headerline:  content-disposition: attachment; filename=stream.asx
            int idx = rhl.index_of_icase("filename=");
            if (idx >= 0) {
                ps_ptr<char> fn;
                fn.assign(rhl.get() + idx + 9); // Position directly after "filename="
                fn.replace("\"", "");           // remove '\"' around filename if present
                info(*this, evt_info, "Filename is %s", fn.get());
            }
        } else if (rhl.starts_with_icase("connection:")) {
            if (rhl.contains_with_icase("close")) { m_f_connectionClose = true; /* AUDIO_LOG_ERROR("connection will be closed"); */ } // ends after ogg last Page is set
        }

        else if (rhl.starts_with_icase("icy-genre:")) {
            info(*this, evt_info, "icy-genre: %s", rhl.get() + 10); // Ambient, Rock, etc
        }

        else if (rhl.starts_with_icase("icy-logo:")) {
            ps_ptr<char> icyLogo;
            icyLogo.assign(rhl.get() + 9); // Get logo URL
            icyLogo.trim();
            if (icyLogo.strlen() > 0) {
                info(*this, evt_info, "icy-logo: %s", icyLogo.get());
                info(*this, evt_icylogo, "%s", icyLogo.get());
            }
        }

        else if (rhl.starts_with_icase("icy-br:")) {
            ps_ptr<char> c_bitRate;
            c_bitRate.assign(rhl.get() + 7);
            c_bitRate.trim();
            c_bitRate.append("000"); // Found bitrate tag, read the bitrate in Kbit
            if (m_phreh.bitrate == c_bitRate.to_uint32(10))
                continue; // avoid doubles
            else
                m_phreh.bitrate = c_bitRate.to_uint32(10);
            m_nominal_bitrate = c_bitRate.to_uint32(10);
            info(*this, evt_bitrate, "%i", m_nominal_bitrate);
            info(*this, evt_info, "Bitrate (b/s): %lu", m_nominal_bitrate);
        }

        else if (rhl.starts_with_icase("icy-metaint:")) {
            const char* c_metaint = (rhl.get() + 12);
            int32_t     i_metaint = atoi(c_metaint);
            m_metaint = i_metaint;
            if (m_metaint) m_f_metadata = true; // Multimediastream
        }

        else if (rhl.starts_with_icase("icy-name:")) {
            //  AUDIO_LOG_INFO("%s", rhl.get());
            ps_ptr<char> icyName;
            icyName.assign(rhl.get() + 9); // Get station name
            icyName.trim();
            if (icyName.strlen() > 0) { info(*this, evt_name, "%s", icyName.get()); }
        }

        else if (rhl.starts_with_icase("content-length:")) {
            const char* c_cl = (rhl.get() + 15);
            int32_t     i_cl = atoi(c_cl);
            m_audioFileSize = i_cl;
            // info(*this, evt_info, "content-length: %lu", (long unsigned int)m_audioFileSize);
        }

        else if (rhl.starts_with_icase("icy-description:")) {
            const char* c_idesc = (rhl.get() + 16);
            while (c_idesc[0] == ' ') c_idesc++;
            latinToUTF8(rhl); // if already UTF-8 do nothing, otherwise convert to UTF-8
            if (strlen(c_idesc) > 0 && specialIndexOf((uint8_t*)c_idesc, "24bit", 0) > 0) {
                AUDIO_LOG_INFO("icy-description: %s has to be 8 or 16", c_idesc);
                stopSong();
            }
            info(*this, evt_icydescription, "%s", c_idesc);
        }

        else if (rhl.starts_with_icase("transfer-encoding:")) {
            if (rhl.ends_with_icase("chunked")) { // Station provides chunked transfer
                m_f_chunked = true;
                info(*this, evt_info, "chunked data transfer");
                m_chunkcount = 0; // Expect chunkcount in DATA
            }
        }

        else if (rhl.starts_with_icase("accept-ranges:")) {
            if (rhl.ends_with_icase("bytes")) m_f_acceptRanges = true;
            //    AUDIO_LOG_INFO("%s", rhl.c_get());
        }

        else if (rhl.starts_with_icase("content-range:")) {
            AUDIO_LOG_INFO("%s", rhl.c_get());
        }

        else if (rhl.starts_with_icase("icy-url:")) {
            char* icyurl = (rhl.get() + 8);
            trim(icyurl);
            info(*this, evt_icyurl, "%s", icyurl);
        }

        else if (rhl.starts_with_icase("www-authenticate:")) {
            AUDIO_LOG_WARN("authentification failed, wrong credentials?");
            goto exit;
        } else {
            ;
        }
    } // outer while

exit: // termination condition
    info(*this, evt_name, "");
    info(*this, evt_icydescription, "");
    info(*this, evt_icyurl, "");
    m_dataMode = AUDIO_NONE;
    stopSong();
    return false;

lastToDo:
    m_streamType = ST_WEBSTREAM;
    if (m_audioFileSize > 0) m_streamType = ST_WEBFILE; // content length found
    if (m_phreh.f_icy_data) m_streamType = ST_WEBSTREAM;
    if (m_f_tts) m_streamType = ST_WEBSTREAM; // this is from AI or GoogleTTS(AI response)

    if (m_codec != CODEC_NONE) {
        m_dataMode = AUDIO_DATA; // Expecting data now

    } else if (m_playlistFormat != FORMAT_NONE) {
        m_dataMode = AUDIO_PLAYLISTINIT; // playlist expected
        // AUDIO_LOG_INFO("now parse playlist");
    } else {
        AUDIO_LOG_INFO("unknown content found at: %s", m_currentHost.c_get());
        goto exit;
    }

    AUDIO_LOG_DEBUG("playlistFormat %s, dataMode %s, streamType: %s", plsFmtStr[m_playlistFormat], dataModeStr[m_dataMode], streamTypeStr[m_streamType]);
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::parseHttpRangeHeader() { // this is the response to a Range request

    ps_ptr<char> rhl;
    rhl.alloc(1024); // responseHeaderline
    bool ct_seen = false;

    if (m_dataMode != HTTP_RANGE_HEADER) {
        AUDIO_LOG_ERROR("wrong datamode %i", m_dataMode);
        goto exit;
    }

    m_phrah.ctime = millis();
    m_phrah.timeout = 4500; // ms

    if (m_client->available() == 0) {
        if (!m_phrah.f_time) {
            m_phrah.stime = millis();
            m_phrah.f_time = true;
        }
        if ((millis() - m_phrah.stime) > m_phrah.timeout) {
            AUDIO_LOG_ERROR("timeout");
            m_phrah.f_time = false;
            return false;
        }
    }
    m_phrah.f_time = false;

    rhl.clear();

    while (true) { // outer while
        uint16_t pos = 0;
        if ((millis() - m_phrah.ctime) > m_phrah.timeout) {
            AUDIO_LOG_ERROR("timeout");
            m_f_timeout = true;
            goto exit;
        }
        while (m_client->available()) {
            uint8_t b = audioFileRead();
            if (b == '\n') {
                if (!pos) { // empty line received, is the last line of this responseHeader
                    goto lastToDo;
                }
                break;
            }
            if (b == '\r') rhl[pos] = 0;
            if (b < 0x20) continue;
            rhl[pos] = b;
            pos++;
            if (pos == 1023) {
                pos = 1022;
                continue;
            }
            if (pos == 1022) {
                rhl[pos] = '\0';
                AUDIO_LOG_WARN("responseHeaderline overflow");
            }
        } // inner while
        if (!pos) {
            vTaskDelay(5);
            continue;
        }
        // AUDIO_LOG_WARN("rh %s", rhl.c_get());
        if (rhl.starts_with_icase("HTTP/")) { // HTTP status error code
            char statusCode[5];
            statusCode[0] = rhl[9];
            statusCode[1] = rhl[10];
            statusCode[2] = rhl[11];
            statusCode[3] = '\0';
            int sc = atoi(statusCode);
            if (sc > 310) { // e.g. HTTP/1.1 301 Moved Permanently
                info(*this, evt_name, "%s", rhl.get());
                goto exit;
            }
        }
        if (rhl.starts_with_icase("Server:")) { info(*this, evt_info, "%s", rhl.c_get()); }
        if (rhl.starts_with_icase("Content-Length:")) {
            //    info(*this, evt_info, "%s", rhl.c_get());
        }
        if (rhl.starts_with_icase("Content-Range:")) { info(*this, evt_info, "%s", rhl.c_get()); }
        if (rhl.starts_with_icase("Content-Type:")) {
            //    info(*this, evt_info, "%s", rhl.c_get());
        }
    }
exit:
    return false;
lastToDo:
    m_dataMode = AUDIO_DATA;
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::initializeDecoder() {
    if (m_decoder) {
        AUDIO_LOG_WARN("stopSong first");
        return true;
    }
    if (m_codec == CODEC_OGG) {
        m_codec = determineOggCodec();
        m_lastGranulePosition = getLastGranulePosition();
        audioFileSeek(0);
        AUDIO_LOG_DEBUG("lastGranulePosition %llu", m_lastGranulePosition);
    }
    const char* type = nullptr;
    switch (m_codec) {
        case CODEC_MP3:
            type = "MP3";
            InBuff.setMaxBlocksize(m_frameSizeMP3);
            break;
        case CODEC_AAC:
            type = "AAC";
            InBuff.setMaxBlocksize(m_frameSizeAAC);
            break;
        case CODEC_M4A:
            type = "AAC";
            InBuff.setMaxBlocksize(m_frameSizeAAC);
            break;
        case CODEC_FLAC:
            type = "FLAC";
            InBuff.setMaxBlocksize(m_frameSizeFLAC);
            break;
        case CODEC_OPUS:
            type = "OPUS";
            InBuff.setMaxBlocksize(m_frameSizeOPUS);
            break;
        case CODEC_VORBIS:
            type = "VORBIS";
            InBuff.setMaxBlocksize(m_frameSizeVORBIS);
            break;
        case CODEC_WAV:
            type = "WAV";
            InBuff.setMaxBlocksize(m_frameSizeWav);
            break;
        default:
            AUDIO_LOG_ERROR("unknown decoder");
            stopSong();
            return false;
            break;
    }

    if (type) {
        m_decoder = createDecoder(type);
        if (!m_decoder || !m_decoder->init()) {
            AUDIO_LOG_ERROR("The %sDecoder could not be initialized", type);
            stopSong();
            return false;
        }
    }
    info(*this, evt_info, "%sDecoder has been initialized", m_decoder->whoIsIt());
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::parseContentType(char* ct) {
    enum : int { CT_NONE, CT_MP3, CT_AAC, CT_M4A, CT_WAV, CT_FLAC, CT_PLS, CT_M3U, CT_ASX, CT_M3U8, CT_TXT, CT_AACP, CT_OPUS, CT_OGG, CT_VORBIS };

    // MIME types and their CT_Val values
    static const struct {
        const char* mime_type;
        int         ct_val;
    } mime_map[] = {
        {"audio/mpeg", CT_MP3},
        {"audio/mpeg3", CT_MP3},
        {"audio/x-mpeg", CT_MP3},
        {"audio/x-mpeg-3", CT_MP3},
        {"audio/mp3", CT_MP3},
        {"audio/aac", CT_AAC},
        {"audio/x-aac", CT_AAC},
        {"audio/aacp", CT_AAC},
        {"video/mp2t", CT_AAC},
        {"audio/mp2t", CT_AAC},
        {"audio/mp4", CT_M4A},
        {"audio/m4a", CT_M4A},
        {"audio/x-m4a", CT_M4A},
        {"audio/wav", CT_WAV},
        {"audio/x-wav", CT_WAV},
        {"audio/flac", CT_FLAC},
        {"audio/x-flac", CT_FLAC},
        {"audio/scpls", CT_PLS},
        {"audio/x-scpls", CT_PLS},
        {"application/pls+xml", CT_PLS},
        {"audio/mpegurl", CT_M3U},
        {"audio/x-mpegurl", CT_M3U},
        {"audio/ms-asf", CT_ASX},
        {"video/x-ms-asf", CT_ASX},
        {"audio/x-ms-asx", CT_ASX},
        {"application/ogg", CT_OGG},
        {"audio/ogg", CT_OGG},
        {"application/vnd.apple.mpegurl", CT_M3U8},
        {"application/x-mpegurl", CT_M3U8},
        {"application/octet-stream", CT_TXT},
        {"text/html", CT_TXT},
        {"text/plain", CT_TXT},
        {"application/json", CT_TXT},
        {NULL, CT_NONE} // Final marker
    };

    strlower(ct);
    trim(ct);

    m_codec = CODEC_NONE;
    int ct_val = CT_NONE;

    // Search in the Lookup table
    for (int i = 0; mime_map[i].mime_type != NULL; i++) {
        if (!strcmp(ct, mime_map[i].mime_type)) {
            ct_val = mime_map[i].ct_val;
            break;
        }
    }

    // Special cases for video/mp2t and audio/mp2t
    if (ct_val == CT_AAC && (strcmp(ct, "video/mp2t") == 0 || strcmp(ct, "audio/mp2t") == 0) && m_m3u8Codec == CODEC_MP3) { ct_val = CT_MP3; }
    // Special case for audio/mpegurl
    if (ct_val == CT_M3U && strcmp(ct, "audio/mpegurl") == 0 && m_expectedPlsFmt == FORMAT_M3U8) { ct_val = CT_M3U8; }
    // Special case for application/json
    if (ct_val == CT_TXT && strcmp(ct, "application/json") == 0 && m_expectedPlsFmt == FORMAT_M3U8) { ct_val = CT_M3U8; }

    // exam for invalid content types
    if (ct_val == CT_NONE) {
        AUDIO_LOG_WARN("ContentType %s not supported", ct);
        return false;
    }

    // allocation of m_codec and m_playlist format
    switch (ct_val) {
        case CT_MP3: m_codec = CODEC_MP3; break;
        case CT_AAC: m_codec = CODEC_AAC; break;
        case CT_M4A: m_codec = CODEC_M4A; break;
        case CT_FLAC: m_codec = CODEC_FLAC; break;
        case CT_OPUS:
            m_codec = CODEC_OPUS;
            m_f_ogg = true;
            break;
        case CT_VORBIS:
            m_codec = CODEC_VORBIS;
            m_f_ogg = true;
            break;
        case CT_WAV: m_codec = CODEC_WAV; break;
        case CT_OGG:
            m_codec = CODEC_OGG;
            m_f_ogg = true;
            break;
        case CT_PLS: m_playlistFormat = FORMAT_PLS; break;
        case CT_M3U: m_playlistFormat = FORMAT_M3U; break;
        case CT_ASX: m_playlistFormat = FORMAT_ASX; break;
        case CT_M3U8: m_playlistFormat = FORMAT_M3U8; break;
        case CT_TXT:
            if (m_expectedCodec == CODEC_AAC) m_codec = CODEC_AAC;
            if (m_expectedCodec == CODEC_MP3) m_codec = CODEC_MP3;
            if (m_expectedPlsFmt == FORMAT_ASX) m_playlistFormat = FORMAT_ASX;
            if (m_expectedPlsFmt == FORMAT_M3U) m_playlistFormat = FORMAT_M3U;
            if (m_expectedPlsFmt == FORMAT_M3U8) m_playlistFormat = FORMAT_M3U8;
            if (m_expectedPlsFmt == FORMAT_PLS) m_playlistFormat = FORMAT_PLS;
            break;
        default: AUDIO_LOG_ERROR("%s, unsupported audio format", ct); return false;
    }

    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::showstreamtitle(char* st) {
    if (!st) return;
    // example for ml:
    // StreamTitle='Oliver Frank - Mega Hitmix';StreamUrl='www.radio-welle-woerthersee.at';
    // or adw_ad='true';durationMilliseconds='10135';adId='34254';insertionType='preroll';
    // html: 'Bielszy odcie&#324; bluesa 682 cz.1' --> 'Bielszy odcień bluesa 682 cz.1'

    ps_ptr<char> ml;
    ps_ptr<char> title;
    ps_ptr<char> artist;
    ps_ptr<char> streamTitle;
    ps_ptr<char> sUrl;
    ml.htmlToUTF8(st); // convert to UTF-8

    int16_t idx1 = 0, idx2, idx4, idx5, idx6, idx7, titleLen = 0, artistLen = 0, titleStart = 0, artistStart = 0;

    // if(idx1 < 0) idx1 = indexOf(ml, "Title:", 0); // Title found (e.g. https://stream-hls.bauermedia.pt/comercial.aac/playlist.m3u8)
    // if(idx1 < 0) idx1 = indexOf(ml, "title:", 0); // Title found (e.g. #EXTINF:10,title="The Dan Patrick Show (M-F 9a-12p ET)",artist="zc1401"

    if (ml.index_of("StreamTitle='<?xml version=", 0) == 0) {
        /* e.g. xmlStreamTitle
              StreamTitle='<?xml version="1.0" encoding="utf-8"?><RadioInfo><Table><DB_ALBUM_ID>37364</DB_ALBUM_ID>
              <DB_ALBUM_IMAGE>00000037364.jpg</DB_ALBUM_IMAGE><DB_ALBUM_NAME>Boyfriend</DB_ALBUM_NAME>
              <DB_ALBUM_TYPE>Single</DB_ALBUM_TYPE><DB_DALET_ARTIST_NAME>DOVE CAMERON</DB_DALET_ARTIST_NAME>
              <DB_DALET_ITEM_CODE>CD4161</DB_DALET_ITEM_CODE><DB_DALET_TITLE_NAME>BOYFRIEND</DB_DALET_TITLE_NAME>
              <DB_FK_SITE_ID>2</DB_FK_SITE_ID><DB_IS_MUSIC>1</DB_IS_MUSIC><DB_LEAD_ARTIST_ID>26303</DB_LEAD_ARTIST_ID>
              <DB_LEAD_ARTIST_NAME>Dove Cameron</DB_LEAD_ARTIST_NAME><DB_RADIO_IMAGE>cidadefm.jpg</DB_RADIO_IMAGE>
              <DB_RADIO_NAME>Cidade</DB_RADIO_NAME><DB_SONG_ID>120126</DB_SONG_ID><DB_SONG_LYRIC>60981</DB_SONG_LYRIC>
              <DB_SONG_NAME>Boyfriend</DB_SONG_NAME></Table><AnimadorInfo><TITLE>Cidade</TITLE>
              <START_TIME_UTC>2022-11-15T22:00:00+00:00</START_TIME_UTC><END_TIME_UTC>2022-11-16T06:59:59+00:00
              </END_TIME_UTC><SHOW_NAME>Cidade</SHOW_NAME><SHOW_HOURS>22h às 07h</SHOW_HOURS><SHOW_PANEL>0</SHOW_PANEL>
              </AnimadorInfo></RadioInfo>';StreamUrl='';
        */
        idx4 = ml.index_of("<DB_DALET_TITLE_NAME>");
        idx5 = ml.index_of("</DB_DALET_TITLE_NAME>");
        idx6 = ml.index_of("<DB_LEAD_ARTIST_NAME>");
        idx7 = ml.index_of("</DB_LEAD_ARTIST_NAME>");
        if (idx4 == -1 || idx5 == -1) return;
        titleStart = idx4 + 21; // <DB_DALET_TITLE_NAME>
        titleLen = idx5 - titleStart;

        if (idx6 != -1 && idx7 != -1) {
            artistStart = idx6 + 21; // <DB_LEAD_ARTIST_NAME>
            artistLen = idx7 - artistStart;
        }
        if (titleLen) title.assign(ml.get() + titleStart, titleLen);
        if (artistLen) artist.assign(ml.get() + artistStart, artistLen);

        if (title.valid() && artist.valid()) {
            streamTitle.assign(title.get());
            streamTitle.append(" - ");
            streamTitle.append(artist.get());
        } else if (title.valid()) {
            streamTitle.assign(title.get());
        } else if (artist.valid()) {
            streamTitle.assign(artist.get());
        }
    }

    else if (ml.index_of("StreamTitle='") == 0) { // #1202
        int start = 13;
        int end = ml.index_of("';", start);
        if (end < 0) end = ml.index_of("'", start); // fallback — when the station is transmitting incorrectly
        if (end > start) {
            int len = end - start;
            streamTitle.assign(ml.get() + start, len);
        }
    }

    else if (ml.starts_with("Grouping:")) {
        streamTitle.assign(ml.get() + 9);
    }

    else if (ml.starts_with("#EXTINF")) {
        // extraxt StreamTitle from m3u #EXTINF line to icy-format
        // orig: #EXTINF:10,title="text="TitleName",artist="ArtistName"
        // conv: StreamTitle=TitleName - ArtistName
        // orig: #EXTINF:10,title="text=\"Spot Block End\" amgTrackId=\"9876543\"",artist=" ",url="length=\"00:00:00\""
        // conv: StreamTitle=text=\"Spot Block End\" amgTrackId=\"9876543\" -

        idx1 = ml.index_of("title=\"");
        idx2 = ml.index_of("artist=\"");
        if (idx1 > 0) {
            int titleStart = idx1 + 7;
            int idx3 = ml.index_of("\"", titleStart);
            if (idx3 > titleStart) {
                int titleLength = idx3 - titleStart;
                title.assign(ml.get() + titleStart, titleLength);
                if (title.starts_with("text=\\")) { // #EXTINF:10,title="text=\"Spot Block End\"",artist=" ",
                    titleStart += 7;
                    idx3 = ml.index_of("\\", titleStart);
                    if (idx3 > titleStart) {
                        int titleLength = idx3 - titleStart;
                        title.assign(ml.get() + titleStart, titleLength);
                    }
                }
            }
        }
        if (idx2 > 0) {
            int artistStart = idx2 + 8;
            int idx3 = ml.index_of("\"", artistStart);
            if (idx3 > artistStart) {
                int artistLength = idx3 - artistStart;
                artist.assign(ml.get() + artistStart, artistLength);
                if (strcmp(artist.get(), " ") == 0) artist.reset();
            }
        }
        if (title.valid() && artist.valid()) {
            streamTitle.assign(title.get());
            streamTitle.append(" - ");
            streamTitle.append(artist.get());
        } else if (title.valid()) {
            streamTitle.assign(title.get());
        } else if (artist.valid()) {
            streamTitle.assign(artist.get());
        }
    } else {
        ;
    }

    if (ml.index_of("StreamUrl=", 0) > 0) {
        idx1 = ml.index_of("StreamUrl=", 0);
        idx2 = ml.index_of(";", idx1);
        if (idx1 >= 0 && idx2 > idx1) { // StreamURL found
            uint16_t len = idx2 - idx1;
            sUrl.assign(ml.get() + idx1, len);
            if (sUrl.valid()) { info(*this, evt_info, "Stream URL; %s", sUrl.get()); }
        }
    }

    if (ml.index_of("adw_ad=", 0) > 0) {
        idx1 = ml.index_of("adw_ad=", 0);
        if (idx1 >= 0) { // Advertisement found
            idx1 = ml.index_of("durationMilliseconds=", 0);
            idx2 = ml.index_of(";", idx1);
            if (idx1 >= 0 && idx2 > idx1) {
                uint16_t     len = idx2 - idx1;
                ps_ptr<char> sAdv;
                sAdv.assign(ml.get() + idx1, len + 1);
                // info(*this, evt_info, "Adveritsement: %s", sAdv.get());
                uint8_t pos = 21;                                                              // remove "StreamTitle="
                if (sAdv[pos] == '\'') pos++;                                                  // remove leading  \'
                if (sAdv[strlen(sAdv.get()) - 1] == '\'') sAdv[strlen(sAdv.get()) - 1] = '\0'; // remove trailing \'
                info(*this, evt_info, "Advertisement: %s", sAdv.get() + pos);
            }
        }
        return;
    }
    if (!streamTitle.valid()) return;

    if (!m_streamTitle.equals(streamTitle)) {
        m_streamTitle.clone_from(streamTitle);
    } else {
        return; // is equal
    }

    if (m_streamTitle.starts_with("{\"")) { // maybe a json string
        // "{\"status\":0,\"error\":{\"info\":\"\\u042d\\u0442\\u043e \\u0440\\u0435\\u043a\\u043b\\u0430\\u043c\\u0430 \\u0438\\u043b\\u0438
        // \\u0434\\u0436\\u0438\\u043d\\u0433\\u043b\",\"code\":201},\"result\":\"\\u042d\\u0442\\u043e \\u0440\\u0435\\u043a\\u043b\\u0430\\u043c\\u0430 \\u0438\\u043b\\u0438
        // \\u0434\\u0436\\u0438\\u043d\\u0433\\u043b\"}\n0";
        ps_ptr<char> jsonIn;
        jsonIn.clone_from(m_streamTitle);
        jsonIn.truncate_at('\n'); // can be '{"status":1,"message":"Ok","result":"Ok","errorCode":0}\n0'
        if (!jsonIn.isJson()) return;
        int idx = jsonIn.index_of_icase("\"result\"");
        if (idx < 0) return;
        jsonIn.remove_before(idx + 10); // remove "result":
        jsonIn.truncate_at('"');        // remove after '"'    Ok","result":"Ok","errorCode":0} --> OK
        // AUDIO_LOG_INFO("jsonIn: %s", jsonIn.c_get());
        m_streamTitle.unicodeToUTF8(jsonIn.c_get());
    }

    info(*this, evt_streamtitle, "%s", m_streamTitle.c_get());
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::showCodecParams() {

    info(*this, evt_info, "Channels: %u", getChannels());
    info(*this, evt_info, "SampleRate (Hz): %lu", getSampleRate());
    info(*this, evt_info, "BitsPerSample: %u", getBitsPerSample());
    // if(getBitRate()) { info(*this, evt_info, "BitRate (b/s): %lu", getBitRate()); }
    // else { info(*this, evt_info, "BitRate (b/s): N/A"); }

    if (m_codec == CODEC_AAC) {
        info(*this, evt_info, "%s", m_decoder->arg2()); // AAC Format
        uint8_t answ = m_decoder->val2();               // SBR
        if (answ > 0 && answ < 4) {
            const char sbr[4][50] = {"without SBR", "upsampled SBR", "downsampled SBR", "no SBR used, but file is upsampled by a factor 2"};
            info(*this, evt_info, "Spectral band replication: %s", sbr[answ]);
        }
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::findNextSync(uint8_t* data, size_t len) {
    // Mp3 and aac audio data are divided into frames. At the beginning of each frame there is a sync word.
    // The sync word is 0xFFF. This is followed by information about the structure of the frame.
    // Wav files have no frames
    // Return: 0 the synchronous word was found at position 0
    //         > 0 is the offset to the next sync word
    //         -1 the sync word was not found within the block with the length len

    m_fnsy.nextSync = 0;

    if (m_codec == CODEC_WAV) {
        m_f_playing = true;
        m_fnsy.nextSync = 0;
    }
    if (m_codec == CODEC_MP3) {
        m_fnsy.nextSync = m_decoder->findSyncWord(data, (int32_t)len);
        if (m_fnsy.nextSync == -1) return len; // syncword not found, search next block
        m_decoder->clear();
    }
    if (m_codec == CODEC_AAC) { m_fnsy.nextSync = m_decoder->findSyncWord(data, (int32_t)len); }
    if (m_codec == CODEC_M4A) {
        if (!m_M4A_chConfig) m_M4A_chConfig = 2; // guard
        if (!m_M4A_sampleRate) m_M4A_sampleRate = 44100;
        if (!m_M4A_objectType) m_M4A_objectType = 2;
        m_decoder->setRawBlockParams(m_M4A_chConfig, m_M4A_sampleRate, 0, m_M4A_objectType, 0);
        m_f_playing = true;
        m_fnsy.nextSync = 0;
    }
    if (m_codec == CODEC_FLAC) {
        m_fnsy.nextSync = m_decoder->findSyncWord(data, (int32_t)len);
        if (m_fnsy.nextSync == -1) return len; // OggS not found, search next block
    }
    if (m_codec == CODEC_OPUS) {
        m_fnsy.nextSync = m_decoder->findSyncWord(data, (int32_t)len);
        if (m_fnsy.nextSync == -1) return len; // OggS not found, search next block
    }
    if (m_codec == CODEC_VORBIS) {
        m_fnsy.nextSync = m_decoder->findSyncWord(data, len);
        if (m_fnsy.nextSync == -1) return len; // OggS not found, search next block
    }
    if (m_fnsy.nextSync == -1) {
        if (m_fnsy.swnf == 0)
            info(*this, evt_info, "syncword not found");
        else {
            m_fnsy.swnf++; // syncword not found counter, can be multimediadata
        }
    }
    if (m_fnsy.nextSync == 0) {
        if (m_fnsy.swnf) {
            info(*this, evt_info, "syncword not found %lu times", (long unsigned int)m_fnsy.swnf);
            m_fnsy.swnf = 0;
        } else {
            info(*this, evt_info, "syncword found at pos 0");
            m_f_decode_ready = true;
        }
    }
    if (m_fnsy.nextSync > 0) { info(*this, evt_info, "syncword found at pos %i", m_fnsy.nextSync); }
    return m_fnsy.nextSync;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setDecoderItems() {
    setChannels(m_decoder->getChannels());
    setSampleRate(m_decoder->getSampleRate());
    setBitsPerSample(m_decoder->getBitsPerSample());
    if (m_decoder->arg1()) info(*this, evt_info, "%s", m_decoder->arg1());
    if (m_decoder->getAudioDataStart() > 0) { // only flac-ogg, native flac sets audioDataStart in readFlacHeader()
        m_audioDataStart = m_decoder->getAudioDataStart();
        info(*this, evt_info, "AudioDataStart: %i", m_audioDataStart);
        if (m_audioFileSize && !m_audioDataSize) m_audioDataSize = m_audioFileSize - m_audioDataStart;
    }
    if (m_lastGranulePosition && m_audioFileSize && m_i2s_items.sampleRate) {
        m_audioFileDuration = (uint32_t)(m_lastGranulePosition / m_i2s_items.sampleRate);
        m_nominal_bitrate = (m_audioFileSize - m_audioDataStart) * 8 / m_audioFileDuration;
        AUDIO_LOG_DEBUG("m_nominal_bitrate %i, m_lastGranulePosition %i", m_nominal_bitrate, m_lastGranulePosition);
        info(*this, evt_bitrate, "%i", m_nominal_bitrate);
        info(*this, evt_info, "Duration (s): %lu", m_audioFileDuration);
        info(*this, evt_info, "Bitrate (b/s): %lu", m_nominal_bitrate);
    }

    if (getBitsPerSample() != 8 && getBitsPerSample() != 16 && getBitsPerSample() != 24 && getBitsPerSample() != 32) {
        AUDIO_LOG_ERROR("Bits per sample must be 8, 16, 24 or 32 found %i", getBitsPerSample());
        stopSong();
    }

    if (getChannels() != 1 && getChannels() != 2) {
        AUDIO_LOG_ERROR("Num of channels must be 1 or 2, found %i", getChannels());
        stopSong();
    }
    showCodecParams();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::decodeError(int8_t res, uint8_t* data, int32_t bytesDecoded) {
    // for(int i = 0; i < 10; i++){printf("0x%02X ", data[i]);} printf("\n");
    if (res == -100) {
        stopSong();
        return bytesDecoded;
    } // serious error, e.g. decoder could not be initialized
    if (m_codec == CODEC_AAC && res == -21) { // mono <-> stereo change
        //  According to the specification, the channel configuration is transferred in the first ADTS header and no longer changes in the entire
        //  stream. Some streams send short mono blocks in a stereo stream. e.g. http://mp3.ffh.de/ffhchannels/soundtrack.aac
        //  This triggers error -21 because the faad2 decoder cannot switch automatically.
        m_sbyt.channels = 0;
        if ((data[0] == 0xFF) || ((data[1] & 0xF0) == 0xF0)) {
            int channel_config = ((data[2] & 0x01) << 2) | ((data[3] & 0xC0) >> 6);
            if (channel_config != m_sbyt.channels) {
                m_sbyt.channels = channel_config;
                info(*this, evt_info, "AAC channel config changed to %d", m_sbyt.channels);
            }
        }
        m_decoder->reset();
        m_decoder->init();
        return 0;
    }
    if (m_codec == CODEC_MP3) {
        if (res == MP3Decoder::MP3_NEED_RESTART) {
            info(*this, evt_info, "" ANSI_ESC_RED "Network error" ANSI_ESC_RESET "");
            connecttohost(m_lastHost.get());
            return 0;
        }
    }
    m_f_playing = false;             // seek for new syncword
    if (bytesDecoded == 0) return 1; // skip one byte and seek for the next sync word
    return bytesDecoded;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::decodeContinue(int8_t res, uint8_t* data, int32_t bytesDecoded, int32_t* bytesLeft) {
    // if(m_codec == CODEC_MP3){   if(res == MAD_ERROR_CONTINUE)    return bytesDecoded;} // nothing to play, mybe eof
    if (m_codec == CODEC_AAC) {
        if (res == AACDecoder::AAC_ID3_HDR) {
            uint32_t size = ((data[6 + bytesDecoded] & 0x7F) << 21) | ((data[7 + bytesDecoded] & 0x7F) << 14) | ((data[8 + bytesDecoded] & 0x7F) << 7) | (data[9 + bytesDecoded] & 0x7F);
            size += 10; // skip payload
            return bytesDecoded + size;
            if (bytesDecoded > *bytesLeft) AUDIO_LOG_ERROR("AAC input data too small bytesDecoded %i,> bytesLeft %i", bytesDecoded, *bytesLeft);
        }
    }

    if (m_codec == CODEC_FLAC) {
        if (res == FlacDecoder::FLAC_PARSE_OGG_DONE) return bytesDecoded;
        if (res == FlacDecoder::FLAC_DECODE_FRAMES_LOOP) return bytesDecoded;
    } // nothing to play
    if (m_codec == CODEC_OPUS) {
        if (res == OpusDecoder::OPUS_PARSE_OGG_DONE) return bytesDecoded;
        if (res == OpusDecoder::OPUS_END) return bytesDecoded;
    } // nothing to play
    if (m_codec == CODEC_VORBIS) {
        if (res == VorbisDecoder::VORBIS_PARSE_OGG_DONE) return bytesDecoded;
        if (res == VorbisDecoder::VORBIS_COMMENT_DONE) return bytesDecoded;
        if (res == VorbisDecoder::VORBIS_COMMENT_NEED_MORE) return bytesDecoded;
    } // nothing to play
    if (m_codec == CODEC_MP3) {
        if (res == MP3Decoder::MP3_NEXT_FRAME) return bytesDecoded;
    }
    return 0;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::sendBytes(uint8_t* data, size_t len) {
    if (!m_f_running) return 0; // guard
    if (!m_decoder) return 0;   // guard

    int                   res = 0;
    int                   bytesDecoded = 0;
    const char*           st = NULL;
    std::vector<uint32_t> vec;
    uint16_t              samples_out = 0;
    if (m_validSamples) { goto exit; } // nothing to decode, next round

    m_sbyt.bytesLeft = 0;
    m_sbyt.nextSync = 0;

    if (!m_f_playing) {
        m_sbyt.channels = 2; // assume aac stereo
        m_sbyt.isPS = 0;
        m_sbyt.f_setDecodeParamsOnce = true;
        m_sbyt.nextSync = findNextSync(data, len);
        if (m_sbyt.nextSync < 0) return len; // no syncword found
        if (m_sbyt.nextSync == 0) { m_f_playing = true; }
        if (m_sbyt.nextSync > 0) return m_sbyt.nextSync;
    }
    // m_f_playing is true at this pos

    m_sbyt.bytesLeft = len;

    if (m_codec == CODEC_NONE && m_playlistFormat == FORMAT_M3U8) return 0; // can happen when the m3u8 playlist is loaded
    if (!m_f_decode_ready) return 0;                                        // find sync first

    //-----------------------------------------------------------------
    res = m_decoder->decode(data, &m_sbyt.bytesLeft, m_outBuff.get());
    bytesDecoded = len - m_sbyt.bytesLeft;
    //-----------------------------------------------------------------

    // res - possible values are:
    //          0: okay, no error
    //        >= 100: the decoder needs more data
    //        < 0: there has been an error
    //       -100: serious error, stop song

    if (res < 0) { return decodeError(res, data, bytesDecoded); }                        // Error, skip the frame...
    if (res > 99) { return decodeContinue(res, data, bytesDecoded, &m_sbyt.bytesLeft); } // decoder needs more data...

    if ((bytesDecoded == 0) && (m_codec != CODEC_VORBIS && m_codec != CODEC_FLAC)) { // unlikely framesize, exept VORBIS decodes lastSegmentTable
        info(*this, evt_info, "framesize is 0, start decoding again");
        m_f_playing = false; // seek for new syncword
        // we're here because there was a wrong sync word so skip one byte and seek for the next
        return 1;
    }
    // status: bytesDecoded > 0 and res >= 0

    switch (m_codec) {
        case CODEC_WAV: m_validSamples = m_decoder->getOutputSamples(); break;
        case CODEC_MP3: m_validSamples = m_decoder->getOutputSamples(); break;
        case CODEC_AAC:
            m_validSamples = m_decoder->getOutputSamples() / getChannels();
            if (!m_sbyt.isPS && m_decoder->val1()) { // only change 0 -> 1
                m_sbyt.isPS = 1;
                info(*this, evt_info, "Parametric Stereo");
            } else
                m_sbyt.isPS = m_decoder->val1();
            break;
        case CODEC_M4A: m_validSamples = m_decoder->getOutputSamples() / getChannels(); break;
        case CODEC_FLAC:
            m_validSamples = m_decoder->getOutputSamples();
            st = m_decoder->getStreamTitle();
            if (st) { info(*this, evt_streamtitle, "%s", st); }
            vec = m_decoder->getMetadataBlockPicture();
            if (vec.size() > 0) { // get blockpic data
                // AUDIO_LOG_INFO("---------------------------------------------------------------------------");
                // AUDIO_LOG_INFO("ogg metadata blockpicture found:");
                // for(int i = 0; i < vec.size(); i += 2) { AUDIO_LOG_INFO("segment %02i, pos %07i, len %05i", i / 2, vec[i], vec[i + 1]); }
                // AUDIO_LOG_INFO("---------------------------------------------------------------------------");
                info(*this, evt_image, vec);
            }
            break;
        case CODEC_OPUS:
            m_validSamples = m_decoder->getOutputSamples();
            st = m_decoder->getStreamTitle();
            if (st) { info(*this, evt_streamtitle, st); }
            vec = m_decoder->getMetadataBlockPicture();
            if (vec.size() > 0) { // get blockpic data
                // AUDIO_LOG_INFO("---------------------------------------------------------------------------");
                // AUDIO_LOG_INFO("ogg metadata blockpicture found:");
                // for(int i = 0; i < vec.size(); i += 2) { AUDIO_LOG_INFO("segment %02i, pos %07i, len %05i", i / 2, vec[i], vec[i + 1]); }
                // AUDIO_LOG_INFO("---------------------------------------------------------------------------");
                info(*this, evt_image, vec);
            }
            if (m_decoder->arg1()) {
                if (m_sbyt.opus_mode != m_decoder->arg1()) {
                    info(*this, evt_info, m_decoder->arg1());
                    m_sbyt.opus_mode = m_decoder->arg1();
                }
            }
            break;

        case CODEC_VORBIS:
            m_validSamples = m_decoder->getOutputSamples();
            st = m_decoder->getStreamTitle();
            if (st) { info(*this, evt_streamtitle, st); }
            vec = m_decoder->getMetadataBlockPicture();
            if (vec.size() > 0) { // get blockpic data
                // AUDIO_LOG_INFO("---------------------------------------------------------------------------");
                // AUDIO_LOG_INFO("ogg metadata blockpicture found:");
                // for(int i = 0; i < vec.size(); i += 2) { AUDIO_LOG_INFO("segment %02i, pos %07i, len %05i", i / 2, vec[i], vec[i + 1]); }
                // AUDIO_LOG_INFO("---------------------------------------------------------------------------");
                info(*this, evt_image, vec);
            }
            break;
    }
    if (m_sbyt.f_setDecodeParamsOnce && m_validSamples) {
        m_sbyt.f_setDecodeParamsOnce = false;
        setDecoderItems();
    }
    samples_out = m_validSamples;

exit:
    m_curSample = 0;
    if (m_validSamples) {
        calculateAudioTime(bytesDecoded, samples_out);
        playChunk();
    }
    return bytesDecoded;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::calculateAudioTime(uint16_t bytesDecoderIn, uint16_t samples_decoder_out) {

    // if(m_dataMode != AUDIO_LOCALFILE && m_streamType != ST_WEBFILE) return; //guard

    float audioCurrentTime = 0.0;

    if (m_cat.firstCall) { // first call
        m_cat.firstCall = false;
        m_cat.reset();

        if (m_codec == CODEC_FLAC && m_decoder->getAudioFileDuration()) { // BITSTREAMINFO FLAC/OGG
            m_audioFileDuration = m_decoder->getAudioFileDuration();
            m_cat.nominalBitRate = (m_audioDataSize / m_decoder->getAudioFileDuration()) * 8;
        }

        if (m_nominal_bitrate) {
            m_cat.nominalBitRate = m_nominal_bitrate;
            m_audioFileDuration = round(((float)m_audioDataSize * 8 / m_cat.nominalBitRate));
            if (m_lastGranulePosition)
                m_cat.tota_samples = m_lastGranulePosition;
            else
                m_cat.tota_samples = m_audioFileDuration * m_i2s_items.sampleRate;
        }
    }

    m_cat.sumBytesIn += bytesDecoderIn;
    m_cat.deltaBytesIn += bytesDecoderIn;
    m_cat.sum_samples += samples_decoder_out;

    if (m_cat.timeStamp + 50 < millis()) {
        uint32_t t = millis();                  // time tracking
        uint32_t delta_t = t - m_cat.timeStamp; //    ---"---
        m_cat.timeStamp = t;                    //    ---"---

        if (m_cat.nominalBitRate) {
            audioCurrentTime = (uint32_t)(m_cat.sum_samples / m_i2s_items.sampleRate);
        } else {
            double instBitRate = (m_cat.deltaBytesIn * 8000.0) / delta_t;
            m_cat.counter++;
            m_cat.avrBitRate += (instBitRate - m_cat.avrBitRate) / m_cat.counter;
            if ((abs(m_cat.avrBitRate - m_cat.oldAvrBitrate < 50)) && !m_cat.avrBitrateStable && m_cat.avrBitRate > 1000) {
                m_cat.brCounter++;
                if (m_cat.brCounter > 6) {
                    m_cat.avrBitrateStable = m_cat.avrBitRate;
                    info(*this, evt_bitrate, "%i", m_cat.avrBitrateStable);
                    info(*this, evt_info, "estimated bitrate (b/s): %lu", m_cat.avrBitrateStable);
                }
            }
            m_avr_bitrate = m_cat.avrBitRate;
            audioCurrentTime = (float)m_cat.sumBytesIn * 8 / m_cat.avrBitRate;
            m_audioFileDuration = round(((float)m_audioDataSize * 8 / m_cat.avrBitRate));
            m_cat.oldAvrBitrate = m_avr_bitrate;
        }
        m_cat.deltaBytesIn = 0;
        m_audioCurrentTime = round(audioCurrentTime);
    }

    if (m_haveNewFilePos && (m_cat.avrBitRate || m_cat.nominalBitRate)) {
        uint32_t posWhithinAudioBlock = m_haveNewFilePos - m_audioDataStart;
        float    newTime = 0;
        if (m_cat.nominalBitRate) {
            newTime = (float)posWhithinAudioBlock / (m_cat.nominalBitRate / 8);
        } else {
            newTime = (float)posWhithinAudioBlock / (m_cat.avrBitRate / 8);
            m_avr_bitrate = m_cat.avrBitRate;
        }
        m_audioCurrentTime = round(newTime);
        m_cat.sumBytesIn = posWhithinAudioBlock;
        m_haveNewFilePos = 0;
        m_cat.syltIdx = 0;
        if (m_syltLines.size()) {
            while (m_cat.syltIdx < m_syltLines.size()) {
                if (m_audioCurrentTime * 1000 < m_syltTimeStamp[m_cat.syltIdx]) break;
                m_cat.syltIdx++;
            }
            if (m_cat.syltIdx) m_cat.syltIdx--;
        }
    }

    if (m_syltLines.size()) {
        //  AUDIO_LOG_INFO("%f", audioCurrentTime * 1000); // ms
        if (m_cat.syltIdx >= m_syltLines.size()) return;
        if (m_audioCurrentTime * 1000 > m_syltTimeStamp[m_cat.syltIdx]) {
            info(*this, evt_lyrics, "%s", m_syltLines[m_cat.syltIdx].c_get());
            m_cat.syltIdx++;
        }
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::setPinout(uint8_t BCLK, uint8_t LRC, uint8_t DOUT, int8_t MCLK) {

    bool result = true;

    i2s_std_gpio_config_t gpio_cfg = {};
    gpio_cfg.bclk = (gpio_num_t)BCLK;
    gpio_cfg.din = (gpio_num_t)I2S_GPIO_UNUSED;
    gpio_cfg.dout = (gpio_num_t)DOUT;
    gpio_cfg.mclk = (gpio_num_t)MCLK;
    gpio_cfg.ws = (gpio_num_t)LRC;

#if (ESP_ARDUINO_VERSION_MAJOR < 3)
    AUDIO_LOG_ERROR("Arduino Version must be 3.0.0 or higher!");
    result = false;
    goto exit;
#endif

    m_f_psramFound = psramInit();
    if (!m_f_psramFound) {
        AUDIO_LOG_ERROR("PSRAM not found");
        result = false;
        goto exit;
    }

    m_outBuff.alloc_array(m_outbuffSize, "m_outBuff");
    m_samplesBuff48K.alloc_array(m_samplesBuff48KSize, "m_samplesBuff48K");
    m_vu_items.delay_l.alloc_array(m_vu_items.DELAY_BUFFER_SIZE, "delay_l");
    m_vu_items.delay_r.alloc_array(m_vu_items.DELAY_BUFFER_SIZE, "delay_r");
    m_fft_items.buffer.alloc_array(m_fft_items.SIZE, "buffer");
    m_fft_items.window.alloc_array(m_fft_items.SIZE, "window");
    m_fft_items.work.alloc_array(m_fft_items.SIZE * 2, "work");

    if (!m_outBuff.valid() || !m_vu_items.delay_l.valid() || !m_vu_items.delay_r.valid() || !m_samplesBuff48K.valid() || !m_fft_items.buffer.valid() || !m_fft_items.buffer.valid() ||
        !m_fft_items.work.valid()) {
        result = false;
        goto exit;
    }

    //---------------------------- FFT INIT-----------------------------------------
    for (int i = 0; i < m_fft_items.SIZE; i++) { // Hann window
        m_fft_items.window[i] = 0.5f * (1.0f - cosf(2.0f * M_PI * i / (m_fft_items.SIZE - 1)));
    }

    if (dsps_fft2r_init_fc32(nullptr, m_fft_items.SIZE) == ESP_OK) {
        m_fft_items.initialized = true;
    } else {
        AUDIO_LOG_ERROR("FFT init failed (size=%d)", m_fft_items.SIZE);
        result = false;
        goto exit;
    }
    //-----------------------------------------------------------------------------

    trim(audioI2SVers);
    info(*this, evt_info, "audioI2S %s", audioI2SVers);

    I2Sstop();
    if (i2s_channel_reconfig_std_gpio(m_i2s_tx_handle, &gpio_cfg) != ESP_OK) {
        result = false;
        goto exit;
    }
    I2Sstart();

exit:
    m_f_I2S_init = result;
    return m_f_I2S_init;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::getFileSize() { // returns the size of webfile or local file
    if (!m_audiofile) {
        if (m_audioFileSize > 0) { return m_audioFileSize; }
        return 0;
    }
    return m_audiofile.size();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::getAudioFileDuration() {
    if (!getBitRate()) return 0;
    if (m_playlistFormat == FORMAT_M3U8) return 0;
    if (!m_audioDataSize) return 0;
    return m_audioFileDuration;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::getAudioCurrentTime() { // return current time in seconds
    return m_audioCurrentTime;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::getAudioFilePosition() {
    if (!m_f_stream) return 0;
    if ((m_dataMode != AUDIO_LOCALFILE) && (m_streamType != ST_WEBFILE)) {
        AUDIO_LOG_WARN("audio is not a file");
        return 0;
    }
    return m_audioFilePosition - inBufferFilled();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::setAudioFilePosition(uint32_t pos) {
    if (!m_f_stream) return false;
    if ((m_dataMode != AUDIO_LOCALFILE) && (m_streamType != ST_WEBFILE)) {
        AUDIO_LOG_WARN("audio is not a file");
        return false;
    }
    if ((m_streamType == ST_WEBFILE) && (!m_f_acceptRanges)) {
        AUDIO_LOG_WARN("server does not accept ranges");
        return false;
    }
    if (pos > m_audioDataStart + m_audioDataSize) {
        AUDIO_LOG_WARN("given position is too large");
        return false;
    }
    if (pos < m_audioDataStart) {
        AUDIO_LOG_WARN("set audiodatastart at %lu", m_audioDataStart);
        m_resumeFilePos = m_audioDataStart;
    }
    m_resumeFilePos = pos;

    /*   m_cat.tota_samples             m_cat.sum_samples
         ------------------  =  ----------------------------------
         m_audioDataSize        m_resumeFilePos - m_audioDataStart                                                  */

    m_cat.sum_samples = (float)m_cat.tota_samples * ((float)(m_resumeFilePos - m_audioDataStart) / m_audioDataSize);
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::setAudioPlayTime(uint16_t sec) {
    // e.g. setAudioPlayTime(300) sets the pointer at pos 5 min
    if ((m_dataMode != AUDIO_LOCALFILE) && (m_streamType != ST_WEBFILE)) return false; // guard
    if (!getBitRate()) return false;                                                   // guard
    if (!m_f_running) return false;                                                    // guard

    if (sec > getAudioFileDuration()) sec = getAudioFileDuration();
    uint32_t filepos = m_audioDataStart + (getBitRate() * sec / 8);
    m_resumeFilePos = filepos;
    m_cat.sum_samples = (float)m_cat.tota_samples * ((float)(m_resumeFilePos - m_audioDataStart) / m_audioDataSize);
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::setTimeOffset(int sec) { // fast forward or rewind the current position in seconds
    // info(*this, evt_info, "time offset %li sec", sec);
    if ((m_dataMode != AUDIO_LOCALFILE) && (m_streamType != ST_WEBFILE)) {
        AUDIO_LOG_WARN("%s", "not a file");
        return false;
    } // guard
    if (!getBitRate()) return false; // guard
    if (!m_f_running) return false;  // guard

    int32_t newTime = getAudioCurrentTime() + sec;
    if (newTime < 0) newTime = 0;
    if (newTime > getAudioFileDuration()) {
        stopSong();
        return true;
    }

    uint32_t oneSec = getBitRate() / 8; // bytes decoded in one sec
    int32_t  offset = oneSec * sec;     // bytes to be wind/rewind
    int32_t  pos = m_audioFilePosition - inBufferFilled();
    pos += offset;
    m_resumeFilePos = pos;
    m_cat.sum_samples = (float)m_cat.tota_samples * ((float)(m_resumeFilePos - m_audioDataStart) / m_audioDataSize);
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setVolumeSteps(uint8_t steps) {
    // clamp new steps
    uint8_t new_steps = max(steps, (uint8_t)21);
    uint8_t old_steps = m_audio_items.volume_steps;

    if (new_steps == old_steps) return;

    // ratio OLD → NEW
    float corr = (float)new_steps / (float)old_steps;

    // scale target volume
    float new_volume = (float)m_audio_items.volume * corr;

    // also scale the current volume (very important!)
    m_audio_items.cur_volume *= corr;

    // take over + clamp
    m_audio_items.volume_steps = new_steps;

    m_audio_items.volume = (uint8_t)lroundf(new_volume);
    if (m_audio_items.volume > new_steps) m_audio_items.volume = new_steps;

    if (m_audio_items.cur_volume > (float)new_steps) m_audio_items.cur_volume = (float)new_steps;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint8_t Audio::getVolumeSteps() {
    return m_audio_items.volume_steps;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setMute(bool mute) {
    m_audio_items.mute = mute;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::getMute() {
    return m_audio_items.mute;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int32_t Audio::audioFileRead(uint8_t* buff, size_t len) {
    if (buff && len == 0) return 0; // nothing to do
    int32_t  readed_bytes = 0;
    uint32_t offset = 0;
    // This method standardized reading files, regardless of the source (local or web) and the correct number of the bytes read must be determined.
    int32_t res = -1;

    if (!buff && !len) { // read one byte
        if (m_dataMode == AUDIO_LOCALFILE) {
            res = m_audiofile.read();
            if (res >= 0) m_audioFilePosition++;
        } else {
            res = m_client->read();
            if (res >= 0) m_audioFilePosition++;
        }
    } else { // read len
        uint32_t t = millis();
        while (len > 0) {
            if (m_dataMode == AUDIO_LOCALFILE) {
                readed_bytes = m_audiofile.read(buff + offset, len);

                if (readed_bytes >= 0) {
                    m_audioFilePosition += readed_bytes;
                    len -= readed_bytes;
                    offset += readed_bytes;
                    res = offset;
                    t = millis();
                }
                if (readed_bytes <= 0) break;
            } else {
                readed_bytes = m_client->read(buff + offset, len);
                if (readed_bytes > 0) {
                    m_audioFilePosition += readed_bytes;
                    len -= readed_bytes;
                    offset += readed_bytes;
                    res = offset;
                    t = millis();
                }
                if (readed_bytes <= 0) break;
            }
            if (t + 3000 < millis()) {
                AUDIO_LOG_ERROR("timeout");
                res = -1;
                break;
            }
        }
    }
    return res;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int32_t Audio::audioFileSeek(uint32_t position, size_t len) {
    int32_t res = -1;

    if (m_dataMode == AUDIO_LOCALFILE) {
        uint32_t actualPos = m_audiofile.position(); // starts with 1
        if (actualPos != m_audioFilePosition) {
            AUDIO_LOG_DEBUG("actualPos != m_audioFilePosition %lu != %lu", actualPos, m_audioFilePosition);
            m_audioFilePosition = actualPos;
        }
        if (!m_audiofile) return -1;
        if (position > m_audiofile.size()) {
            AUDIO_LOG_WARN("position larger than size %lu > %lu", position, m_audiofile.size());
            position = m_audiofile.size();
        }
        bool r = m_audiofile.seek(position);
        m_audioFilePosition = m_audiofile.position();
        if (r == false) {
            AUDIO_LOG_ERROR("something went wrong");
            return -1;
        } else {
            return position;
        }
    } else {
        if (m_f_acceptRanges) {
            bool r;
            if (len == 0) len = UINT32_MAX;
            r = httpRange(position, len);
            if (res == false) {
                AUDIO_LOG_ERROR("http range request was not successful");
                return 0;
            }
            r = parseHttpRangeHeader();
            if (r == false) {
                AUDIO_LOG_ERROR("http range response was not successful");
                return 0;
            }
            m_audioFilePosition = position;
            return position;
        }
    }
    return res;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::fsRange(uint32_t range) {

    if ((m_dataMode == AUDIO_LOCALFILE) && !m_audiofile) {
        AUDIO_LOG_WARN("%s", "local file not accessibble");
        return false;
    } // guard
    uint32_t startAB = m_audioDataStart;                 // audioblock begin
    uint32_t endAB = m_audioDataStart + m_audioDataSize; // audioblock end
    if (range < (int32_t)startAB) { range = startAB; }
    if (range >= (int32_t)endAB) { range = endAB; }

    m_validSamples = 0;
    m_resumeFilePos = range; // used in processLocalFile()
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::setSampleRate(uint32_t sampRate) {

    if (!sampRate) return false;
    if (sampRate < 8000) {
        AUDIO_LOG_WARN("Sample rate must not be smaller than 8kHz, found: %lu", sampRate);
        return false;
    }
    m_i2s_items.sampleRate = sampRate;
    reconfigI2S();
    IIR_calculateCoefficients();
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::getSampleRate() {
    return m_i2s_items.sampleRate;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::setBitsPerSample(int bits) {
    if ((bits != 32) && (bits != 24) && (bits != 16) && (bits != 8)) return false;
    m_bitsPerSample = bits;
    reconfigI2S();
    return true;
}
uint8_t Audio::getBitsPerSample() {
    return m_bitsPerSample;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::setChannels(int ch) {
    m_channels = ch;
    return true;
}
uint8_t Audio::getChannels() {
    if (m_channels == 0) { // this should not happen! #209
        m_channels = 2;
    }
    return m_channels;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::getBitRate() {
    if (m_nominal_bitrate) return m_nominal_bitrate;
    return m_avr_bitrate;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
uint64_t Audio::getLastGranulePosition() {
    if (m_codec != CODEC_OPUS && m_codec != CODEC_VORBIS) return 0;
    if (m_audioFileSize == 0) { return 0; } // only files

    uint64_t granulePos = 0;
    uint8_t* buff = (uint8_t*)ps_malloc(UINT16_MAX);
    if (!buff) {
        AUDIO_LOG_ERROR("oom");
        return 0;
    }
    int rangeStart = m_audioFileSize - UINT16_MAX - 1;
    AUDIO_LOG_INFO("rangeStart: %lu, audioFileSize: %li, len: %lu, %lu", __LINE__, rangeStart, m_audioFileSize, UINT16_MAX);
    audioFileSeek(rangeStart, UINT16_MAX);
    audioFileRead(buff, UINT16_MAX);
    int32_t pos = specialIndexOfLast(buff, "OggS", UINT16_MAX);
    if (buff[pos + 5] & 0x04) { // is last page;
        for (int j = 0; j < 8; j++) { granulePos |= ((uint64_t)buff[pos + 6 + j] << (j * 8)); }
    }
    AUDIO_LOG_DEBUG("granulePos %llu", granulePos);
    m_resumeFilePos = 0;
    if (buff) {
        free(buff);
        buff = NULL;
    }
    return granulePos;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setI2SCommFMT_LSB(bool commFMT) {
    // false: I2S communication format is by default I2S_COMM_FORMAT_I2S_MSB, right->left (AC101, PCM5102A)
    // true:  changed to I2S_COMM_FORMAT_I2S_LSB for some DACs (PT8211)
    //        Japanese or called LSBJ (Least Significant Bit Justified) format

    if (commFMT) {
        info(*this, evt_info, "commFMT = LSBJ (Least Significant Bit Justified)");
    } else {
        info(*this, evt_info, "commFMT = Philips");
    }
    m_i2s_items.commFMT = commFMT;
    reconfigI2S();
    return;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::reconfigI2S() {

    i2s_channel_disable(m_i2s_tx_handle);

    if (m_i2s_items.commFMT) {
        m_i2s_std_cfg.slot_cfg = I2S_STD_MSB_SLOT_DEFAULT_CONFIG(I2S_DATA_BIT_WIDTH_32BIT, I2S_SLOT_MODE_STEREO);
    } else {
        m_i2s_std_cfg.slot_cfg = I2S_STD_PHILIPS_SLOT_DEFAULT_CONFIG(I2S_DATA_BIT_WIDTH_32BIT, I2S_SLOT_MODE_STEREO);
    }
    i2s_channel_reconfig_std_slot(m_i2s_tx_handle, &m_i2s_std_cfg.slot_cfg);

    if (m_f_output48KHz) {
        m_resampler.phase = 0; // prepare resampler
        m_resampler.phaseStep = ((uint64_t)m_i2s_items.sampleRate << 32) / 48000;
        m_resampler.g_lpCoeffs = makeButterworthLPF_Q31(m_i2s_items.sampleRate);
        m_resampler.lpLeft = {};
        m_resampler.lpRight = {};
        m_i2s_std_cfg.clk_cfg.sample_rate_hz = 48000;
    } else {
        m_i2s_std_cfg.clk_cfg.sample_rate_hz = m_i2s_items.sampleRate;
    }
    i2s_channel_reconfig_std_clock(m_i2s_tx_handle, &m_i2s_std_cfg.clk_cfg);
    i2s_channel_enable(m_i2s_tx_handle);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::calculateVUlevel(int32_t* sample) { // Envelope-Follower

    // delay line
    m_vu_items.delay_l[m_vu_items.delay_line_index] = sample[LEFTCHANNEL];
    m_vu_items.delay_r[m_vu_items.delay_line_index] = sample[RIGHTCHANNEL];
    m_vu_items.delay_line_index++;
    if (m_vu_items.delay_line_index == m_vu_items.DELAY_BUFFER_SIZE) m_vu_items.delay_line_index = 0;

    int16_t pos = m_vu_items.delay_line_index - 1;
    if (pos == -1) pos = m_vu_items.DELAY_BUFFER_SIZE - 1;

    // FFT buffer
    float mono = 0.5f * (float)((m_vu_items.delay_l[pos] >> 20) + (m_vu_items.delay_r[pos] >> 20));

    // --- FFT analyzer AGC ---
    constexpr float TARGET = 0.1f; // gewünschte RMS-Amplitude
    constexpr float ATTACK = 0.05f;
    constexpr float RELEASE_FFT = 0.005f;

    float level = fabsf(mono);

    if (level > 1e-6f) {
        float desired = TARGET / level;
        if (desired < m_fft_items.gain)
            m_fft_items.gain += ATTACK * (desired - m_fft_items.gain);
        else
            m_fft_items.gain += RELEASE_FFT * (desired - m_fft_items.gain);
    }
    if (fabsf(mono) < 1e-4f) mono = 0.0f;
    mono *= m_fft_items.gain;

    m_fft_items.buffer[m_fft_items.buffer_index] = mono;
    m_fft_items.buffer_index++;
    if (m_fft_items.buffer_index == m_fft_items.SIZE) m_fft_items.buffer_index = 0;

    uint8_t l = 0, r = 0;
    l = abs(m_vu_items.delay_l[pos] >> 23);
    r = abs(m_vu_items.delay_r[pos] >> 23);

    // Attack immediately
    constexpr float RELEASE = 1.0f; // the bigger, the more sluggish
    if (l > m_vu_items.left) {
        m_vu_items.left = l;
    } else if (m_vu_items.left > RELEASE) {
        m_vu_items.left -= RELEASE;
    }
    if (r > m_vu_items.right) {
        m_vu_items.right = r;
    } else if (m_vu_items.right > RELEASE) {
        m_vu_items.right -= RELEASE;
    }

    constexpr uint16_t PEAK_HOLD_SAMPLES = 2000; // ca. 20 ms @ 48 kHz
    constexpr uint8_t  PEAK_RELEASE = 1;         // Fall rate

    // LEFT
    if (m_vu_items.left > m_vu_items.left_peak) {
        m_vu_items.left_peak = m_vu_items.left;
        m_vu_items.left_hold = PEAK_HOLD_SAMPLES;
    } else {
        if (m_vu_items.left_hold > 0) {
            m_vu_items.left_hold--;
        } else if (m_vu_items.left_peak > PEAK_RELEASE) {
            m_vu_items.left_peak -= PEAK_RELEASE;
        }
    }

    // RIGHT
    if (m_vu_items.right > m_vu_items.right_peak) {
        m_vu_items.right_peak = m_vu_items.right;
        m_vu_items.right_hold = PEAK_HOLD_SAMPLES;
    } else {
        if (m_vu_items.right_hold > 0) {
            m_vu_items.right_hold--;
        } else if (m_vu_items.right_peak > PEAK_RELEASE) {
            m_vu_items.right_peak -= PEAK_RELEASE;
        }
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint16_t Audio::getVUlevel() {
    if (!m_f_running) return 0;
    AUDIO_LOG_DEBUG("%i", (uint8_t)m_vu_items.left);
    // avg 0 ... 255                                                                                  MSB          LSB
    return ((uint8_t)m_vu_items.right_peak << 8) + (uint8_t)m_vu_items.left_peak; // returns  rrrrrrrrllllllll
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setTone(float gainLowPass, float gainBandPass, float gainHighPass) {

    // gainLowPass   set between -12 ... +12 dB
    // gainBandPass  set between -12 ... +12 dB
    // gainHighPass  set between -12 ... +12 dB

    m_audio_items.gain_ls_db = fminf(fmaxf(gainLowPass, -12.0f), 12.0f);
    m_audio_items.gain_peq_db = fminf(fmaxf(gainBandPass, -12.0f), 12.0f);
    m_audio_items.gain_hs_db = fminf(fmaxf(gainHighPass, -12.0f), 12.0f);

    IIR_calculateCoefficients();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::forceMono(bool m) { // #100 mono option
    m_f_forceMono = m;          // false stereo, true mono
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setOutput48KHz(bool f48) { //
    m_f_output48KHz = f48;             // false f_out is the origin sample rate, true f_out is always 48000 Hz
    reconfigI2S();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setBalance(float balance) { // left -16.0dB ... 0dB ... -16.0dB right
    m_audio_items.balance = fminf(fmaxf(balance, -16.0f), 16.0f);
    calculateVolumeLimits();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::setVolume(uint8_t volume, uint8_t curve) {

    m_audio_items.volume = min(volume, m_audio_items.volume_steps);

    // curve is obsolete
    calculateVolumeLimits();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint8_t Audio::getVolume() {
    return m_audio_items.volume;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint8_t Audio::getI2sPort() {
    return m_i2s_items.i2s_num;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::gain_ramp() {
    // determine goal
    float target = m_audio_items.mute ? 0.0f : (float)m_audio_items.volume;

    // maximum change per tick
    float step = (float)m_audio_items.volume_steps / 50.0f;

    float diff = target - m_audio_items.cur_volume;

    if (fabsf(diff) <= step) {
        // Goal achieved (no oscillation!)
        m_audio_items.cur_volume = target;
    } else {
        // Ramp
        m_audio_items.cur_volume += (diff > 0.0f ? step : -step);
    }

    // Security-Clamp
    if (m_audio_items.cur_volume < 0.0f)
        m_audio_items.cur_volume = 0.0f;
    else if (m_audio_items.cur_volume > m_audio_items.volume_steps)
        m_audio_items.cur_volume = m_audio_items.volume_steps;

    calculateVolumeLimits();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::calculateVolumeLimits() { // is calculated when the volume or balance changes

    constexpr float BALANCE_DB = -16.0f;
    constexpr float MIN_DB = -60.0f; // quiet
    constexpr float MAX_DB = 0.0f;   // full level

    auto volumeToLinear = [&](uint8_t volume, uint8_t steps) {
        if (volume == 0) {
            return 0.0f; // real silence
        }

        float t = (float)volume / (float)steps; // 0…1

        //    float dB = MIN_DB + t * (MAX_DB - MIN_DB);
        float dB = -112.0f * t * t * t + 172.0f * t * t + MIN_DB;

        return powf(10.0f, dB / 20.0f);
    };

    float vol = volumeToLinear(m_audio_items.cur_volume, m_audio_items.volume_steps);

    float l_db = 0.0f;
    float r_db = 0.0f;

    if (m_audio_items.balance > 0.0f) { // emphasize on the right → quieter on the left
        l_db = BALANCE_DB * ((float)m_audio_items.balance / 16.0f);
    } else if (m_audio_items.balance < 0.0f) { // emphasize on the left → quieter on the right
        r_db = BALANCE_DB * ((float)-m_audio_items.balance / 16.0f);
    }

    m_audio_items.limiter[LEFTCHANNEL] = vol * powf(10.0f, l_db / 20.0f);
    m_audio_items.limiter[RIGHTCHANNEL] = vol * powf(10.0f, r_db / 20.0f);
    AUDIO_LOG_DEBUG("m_limiter[LEFTCHANNEL] %f, m_limiter[RIGHTCHANNEL] %f", m_audio_items.limiter[LEFTCHANNEL], m_audio_items.limiter[RIGHTCHANNEL]);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::processSpectrum() {

    // --- 10 Hz update ---
    uint32_t now = millis();
    if (now - m_fft_items.last_ms < 100) return;
    m_fft_items.last_ms = now;

    // --- Window + real → complex ---
    for (int i = 0; i < m_fft_items.SIZE; i++) {
        m_fft_items.work[2 * i] = m_fft_items.buffer[i] * m_fft_items.window[i];
        m_fft_items.work[2 * i + 1] = 0.0f;
    }

    // --- FFT ---
    dsps_fft2r_fc32(m_fft_items.work.get(), m_fft_items.SIZE);
    dsps_bit_rev_fc32(m_fft_items.work.get(), m_fft_items.SIZE);
    dsps_cplx2reC_fc32(m_fft_items.work.get(), m_fft_items.SIZE);

    const float bin_hz = (float)m_i2s_items.sampleRate / m_fft_items.SIZE;
    const float norm = 2.0f / m_fft_items.SIZE;

    // --- 5 internal bands ---
    float band[m_fft_items.BANDS] = {0};
    int   bins[m_fft_items.BANDS] = {0};

    for (int i = 1; i < m_fft_items.SIZE / 2; i++) {

        float re = m_fft_items.work[2 * i];
        float im = m_fft_items.work[2 * i + 1];
        float mag = sqrtf(re * re + im * im) * norm;
        float f = i * bin_hz;

        int b = -1;
        if (f < 250)
            b = 0;
        else if (f < 400)
            b = -1;
        else if (f < 700)
            b = 1;
        else if (f < 1000)
            b = -1;
        else if (f < 1550)
            b = 2;
        else if (f < 2200)
            b = -1;
        else if (f < 4250)
            b = 3;
        else if (f < 6300)
            b = -1;
        else if (f < 11150)
            b = 4;
        else if (f < 16000)
            b = 5;
        else
            b = -1;

        if (b >= 0) {
            band[b] += mag * mag;
            bins[b]++;
        }
    }

    // --- RMS + weighting ---
    for (int i = 0; i < m_fft_items.BANDS; i++) {
        if (bins[i])
            band[i] = sqrtf(band[i] / bins[i]);
        else
            band[i] = 0.0f;
    }

    // band weighting (psychoacoustic / UI)
    band[0] *= 0.5f;
    band[1] *= 0.8f;
    band[2] *= 3.0f;
    band[3] *= 2.8f;
    band[4] *= 1.5f;
    band[5] *= 1.5f;

    // log scale
    for (int i = 0; i < m_fft_items.BANDS; i++) { band[i] = log10f(band[i] + 1e-6f); }

    // --- temporal smoothing (only displayed bands) ---
    auto smooth = [](float old, float in) {
        constexpr float ATTACK = 0.6f;
        constexpr float RELEASE = 0.6f;
        return (in > old) ? old + ATTACK * (in - old) : old + RELEASE * (in - old);
    };

    for (int i = 0; i < m_fft_items.BANDS; i++) { m_fft_items.spec_smooth[i] = smooth(m_fft_items.spec_smooth[i], band[i]); }

    // --- map to 0..255 using dB window ---
    constexpr float DB_MIN = -50.0f;
    constexpr float DB_MAX = 0.0f;

    for (int i = 0; i < m_fft_items.BANDS; i++) {

        float db = m_fft_items.spec_smooth[i] * 20.0f;

        if (db < DB_MIN) db = DB_MIN;
        if (db > DB_MAX) db = DB_MAX;

        float norm = (db - DB_MIN) / (DB_MAX - DB_MIN);

        m_fft_items.spectrum[i] = (uint8_t)(norm * 255.0f);
    }
    //    AUDIO_LOG_INFO("% 4i, % 4i, % 4i, % 4i, % 4i, % 4i ", m_fft_items.spectrum[0], m_fft_items.spectrum[1],  m_fft_items.spectrum[2],  m_fft_items.spectrum[3],  m_fft_items.spectrum[4],
    //    m_fft_items.spectrum[3]);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::Gain(int32_t* sample) {
    /* important: these multiplications must all be signed ints, or the result will be invalid */
    int32_t* s32 = (int32_t*)sample;
    s32[LEFTCHANNEL] *= m_audio_items.limiter[LEFTCHANNEL];
    s32[RIGHTCHANNEL] *= m_audio_items.limiter[RIGHTCHANNEL];
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::inBufferFilled() {
    // current audio input buffer fillsize in bytes
    return InBuff.bufferFilled();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::inBufferFree() {
    // current audio input buffer free space in bytes
    return InBuff.freeSpace();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::getInBufferSize() {
    // current audio input buffer size in bytes
    return InBuff.getBufsize();
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::stereo2mono(int32_t* buff, uint16_t validSamples) {

    for (uint16_t i = 0; i < validSamples * 2; i += 2) {
        int64_t l = buff[i];
        int64_t r = buff[i + 1];
        int32_t m = (int32_t)((l + r) >> 1); // average, without overflow
        buff[i] = m;                         // Left
        buff[i + 1] = m;                     // Right
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
//            ***     D i g i t a l   b i q u a d r a t i c     f i l t e r     ***
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::IIR_calculateCoefficients() { // Infinite Impulse Response (IIR) filters

    AUDIO_LOG_DEBUG("gain gain_ls_db %f, gain gain_peq_db %f, gain gain_hs_db %f", m_audio_items.gain_ls_db, m_audio_items.gain_peq_db, m_audio_items.gain_hs_db);

    const float FcLS = m_audio_items.freq_ls_Hz;    // Frequency LowShelf(Hz)
    const float FcPKEQ = m_audio_items.freq_peq_Hz; // Frequency PeakEQ(Hz)
    const float FcHS = m_audio_items.freq_hs_Hz;    // Frequency HighShelf(Hz)

    float       normFreqLS = FcLS / m_i2s_items.sampleRate;    // filter cut off frequency
    float       normFreqPEQ = FcPKEQ / m_i2s_items.sampleRate; // filter center frequency
    float       normFreqHS = FcHS / m_i2s_items.sampleRate;    // filter cut off frequency
    const float QS = 0.707;                                    // Quality Slope (Shelf)

    float total_boost_db = fmax(fmax(fmax(0, m_audio_items.gain_ls_db), m_audio_items.gain_peq_db), m_audio_items.gain_hs_db); // dynamic headroom
    m_audio_items.pre_gain = powf(10.0, -total_boost_db / 20);

    auto dsps_biquad_gen_peakingEQ_f32 = [&](float* c, float f, int8_t g, const float Q) -> void {
        float A = powf(10.0f, g / 40.0f);
        float w0 = 2.0f * M_PI * f;
        float cw = cosf(w0);
        float sw = sinf(w0);
        float alpha = sw / (2.0f * Q);
        float b0 = 1.0f + alpha * A;
        float b1 = -2.0f * cw;
        float b2 = 1.0f - alpha * A;
        float a0 = 1.0f + alpha / A;
        float a1 = -2.0f * cw;
        float a2 = 1.0f - alpha / A;
        c[0] = b0 / a0;
        c[1] = b1 / a0;
        c[2] = b2 / a0;
        c[3] = a1 / a0;
        c[4] = a2 / a0;
    };

    dsps_biquad_gen_lowShelf_f32(m_audio_items.coeffs[LOWSHELF], normFreqLS, m_audio_items.gain_ls_db, QS);
    dsps_biquad_gen_peakingEQ_f32(m_audio_items.coeffs[PEAKINGEQ], normFreqPEQ, m_audio_items.gain_peq_db, QS); // my own calc.
    dsps_biquad_gen_highShelf_f32(m_audio_items.coeffs[HIFGSHELF], normFreqHS, m_audio_items.gain_hs_db, QS);

    AUDIO_LOG_DEBUG("\n([%f, %f, %f], [1.0, %f, %f]), # LOWSHELF\n([%f,  %f,  %f ], [1.0, %f,  %f ]), # PEAKINGEQ\n([%f, %f, %f], [1.0, %f, %f]), # HIGHSHELF\n", m_audio_items.coeffs[0][0],
                    m_audio_items.coeffs[0][1], m_audio_items.coeffs[0][2], m_audio_items.coeffs[0][3], m_audio_items.coeffs[0][4], m_audio_items.coeffs[1][0], m_audio_items.coeffs[1][1],
                    m_audio_items.coeffs[1][2], m_audio_items.coeffs[1][3], m_audio_items.coeffs[1][4], m_audio_items.coeffs[2][0], m_audio_items.coeffs[2][1], m_audio_items.coeffs[2][2],
                    m_audio_items.coeffs[2][3], m_audio_items.coeffs[2][4]);
    AUDIO_LOG_DEBUG("m_audio_items.pre_gain %f", m_audio_items.pre_gain);
    memset(m_audio_items.state_biquad, 0, sizeof(m_audio_items.state_biquad));
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::IIR_filter(int32_t* sample) {

    int32_t* s32 = sample;
    float    s[2];
    s[LEFTCHANNEL] = (float)(s32[LEFTCHANNEL] * m_audio_items.pre_gain);
    s[RIGHTCHANNEL] = (float)(s32[RIGHTCHANNEL] * m_audio_items.pre_gain);
    dsps_biquad_sf32(s, s, 1, m_audio_items.coeffs[0], m_audio_items.state_biquad[0]);
    dsps_biquad_sf32(s, s, 1, m_audio_items.coeffs[1], m_audio_items.state_biquad[1]);
    dsps_biquad_sf32(s, s, 1, m_audio_items.coeffs[2], m_audio_items.state_biquad[2]);
    s32[LEFTCHANNEL] = (int32_t)lrintf(fminf(2147483647.0f, fmaxf(-2147483648.0f, s[LEFTCHANNEL])));
    s32[RIGHTCHANNEL] = (int32_t)lrintf(fminf(2147483647.0f, fmaxf(-2147483648.0f, s[RIGHTCHANNEL])));

    s32[LEFTCHANNEL] = (int32_t)s[LEFTCHANNEL];
    s32[RIGHTCHANNEL] = (int32_t)s[RIGHTCHANNEL];

    return;
}

// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
//    AAC - T R A N S P O R T S T R E A M
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
bool Audio::ts_parsePacket(uint8_t* packet, uint8_t* packetStart, uint8_t* packetLength) {

    bool log = false;

    const uint8_t TS_PACKET_SIZE = 188;
    const uint8_t PAYLOAD_SIZE = 184;
    const uint8_t PID_ARRAY_LEN = 4;

    (void)PAYLOAD_SIZE; // suppress [-Wunused-variable]

    if (packet == NULL) {
        if (log) AUDIO_LOG_WARN("parseTS reset");
        m_tspp.reset();
        return true;
    }

    // --------------------------------------------------------------------------------------------------------
    // 0. Byte SyncByte  | 0 | 1 | 0 | 0 | 0 | 1 | 1 | 1 | always bit pattern of 0x47
    //---------------------------------------------------------------------------------------------------------
    // 1. Byte           |PUSI|TP|   |PID|PID|PID|PID|PID|
    //---------------------------------------------------------------------------------------------------------
    // 2. Byte           |PID|PID|PID|PID|PID|PID|PID|PID|
    //---------------------------------------------------------------------------------------------------------
    // 3. Byte           |TSC|TSC|AFC|AFC|CC |CC |CC |CC |
    //---------------------------------------------------------------------------------------------------------
    // 4.-187. Byte      |Payload data if AFC==01 or 11  |
    //---------------------------------------------------------------------------------------------------------

    // PUSI Payload unit start indicator, set when this packet contains the first byte of a new payload unit.
    //      The first byte of the payload will indicate where this new payload unit starts.
    // TP   Transport priority, set when the current packet has a higher priority than other packets with the same PID.
    // PID  Packet Identifier, describing the payload data.
    // TSC  Transport scrambling control, '00' = Not scrambled.
    // AFC  Adaptation field control, 01 – no adaptation field, payload only, 10 – adaptation field only, no payload,
    //                                11 – adaptation field followed by payload, 00 – RESERVED for future use
    // CC   Continuity counter, Sequence number of payload packets (0x00 to 0x0F) within each stream (except PID 8191)

    // for(int i = 1; i < 188; i++) {printf("%02X ", packet[i - 1]); if(i && (i % 16 == 0)) printf("\n");}
    // printf("\n----------\n");

    if (packet[0] != 0x47) {
        AUDIO_LOG_ERROR("ts SyncByte not found, first bytes are 0x%02X 0x%02X 0x%02X 0x%02X", packet[0], packet[1], packet[2], packet[3]);
        stopSong();
        return false;
    }
    int PID = (packet[1] & 0x1F) << 8 | (packet[2] & 0xFF);
    if (log) AUDIO_LOG_DEBUG("PID: 0x%04X (%d)", PID, PID);
    int PUSI = (packet[1] & 0x40) >> 6;
    if (log) AUDIO_LOG_DEBUG("Payload Unit Start Indicator: %d", PUSI);
    int AFC = (packet[3] & 0x30) >> 4;
    if (log) AUDIO_LOG_DEBUG("Adaption Field Control: %d", AFC);

    int AFL = -1;
    if ((AFC & 0b10) == 0b10) { // AFC '11' Adaptation Field followed
        AFL = packet[4] & 0xFF; // Adaptation Field Length
        if (log) AUDIO_LOG_DEBUG("Adaptation Field Length: %d", AFL);
    }
    int PLS = PUSI ? 5 : 4;      // PayLoadStart, Payload Unit Start Indicator
    if (AFL > 0) PLS += AFL + 1; // skip adaption field

    if (AFC == 2) { // The TS package contains only an adaptation Field and no user data.
        *packetStart = AFL + 1;
        *packetLength = 0;
        return true;
    }

    if (PID == 0) {
        // Program Association Table (PAT) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
        if (log) AUDIO_LOG_DEBUG("PAT");
        m_tspp.pidNumber = 0;
        m_tspp.pidOfAAC = 0;

        int startOfProgramNums = 8;
        int lengthOfPATValue = 4;
        int sectionLength = ((packet[PLS + 1] & 0x0F) << 8) | (packet[PLS + 2] & 0xFF);
        if (log) AUDIO_LOG_DEBUG("Section Length: %d", sectionLength);
        int program_number, program_map_PID;
        int indexOfPids = 0;
        (void)program_number; // [-Wunused-but-set-variable]
        for (int i = startOfProgramNums; i <= sectionLength; i += lengthOfPATValue) {
            program_number = ((packet[PLS + i] & 0xFF) << 8) | (packet[PLS + i + 1] & 0xFF);
            program_map_PID = ((packet[PLS + i + 2] & 0x1F) << 8) | (packet[PLS + i + 3] & 0xFF);
            if (log) AUDIO_LOG_DEBUG("Program Num: 0x%04X(%d) PMT PID: 0x%04X(%d)", program_number, program_number, program_map_PID, program_map_PID);
            m_tspp.pids[indexOfPids++] = program_map_PID;
        }
        m_tspp.pidNumber = indexOfPids;
        *packetStart = 0;
        *packetLength = 0;
        return true;
    } else if (PID == m_tspp.pidOfAAC) {
        if (log) AUDIO_LOG_DEBUG("AAC");
        uint8_t posOfPacketStart = 4;
        if (AFL >= 0) {
            posOfPacketStart = 5 + AFL;
            if (log) AUDIO_LOG_DEBUG("posOfPacketStart: %d", posOfPacketStart);
        }
        // Packetized Elementary Stream (PES) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
        if (log) AUDIO_LOG_DEBUG("PES_DataLength %i", m_tspp.PES_DataLength);
        if (m_tspp.PES_DataLength > 0) {
            *packetStart = posOfPacketStart + m_tspp.fillData;
            *packetLength = TS_PACKET_SIZE - posOfPacketStart - m_tspp.fillData;
            if (log) AUDIO_LOG_DEBUG("packetlength %i", *packetLength);
            m_tspp.fillData = 0;
            m_tspp.PES_DataLength -= (*packetLength);
            return true;
        } else {
            int firstByte = packet[posOfPacketStart] & 0xFF;
            int secondByte = packet[posOfPacketStart + 1] & 0xFF;
            int thirdByte = packet[posOfPacketStart + 2] & 0xFF;
            if (log) AUDIO_LOG_DEBUG("First 3 bytes: 0x%02X, 0x%02X, 0x%02X", firstByte, secondByte, thirdByte);
            if (firstByte == 0x00 && secondByte == 0x00 && thirdByte == 0x01) { // Packet start code prefix
                // --------------------------------------------------------------------------------------------------------
                // posOfPacketStart + 0...2     0x00, 0x00, 0x01                                          PES-Startcode
                //---------------------------------------------------------------------------------------------------------
                // posOfPacketStart + 3         0xE0 (Video) od 0xC0 (Audio)                              StreamID
                //---------------------------------------------------------------------------------------------------------
                // posOfPacketStart + 4...5     0xLL, 0xLL                                                PES Packet length
                //---------------------------------------------------------------------------------------------------------
                // posOfPacketStart + 6...7                                                               PTS/DTS Flags
                //---------------------------------------------------------------------------------------------------------
                // posOfPacketStart + 8         0xXX                                                      header length
                //---------------------------------------------------------------------------------------------------------

                uint8_t StreamID = packet[posOfPacketStart + 3] & 0xFF;
                if (StreamID >= 0xC0 && StreamID <= 0xDF) { ; } // okay ist audio stream
                if (StreamID >= 0xE0 && StreamID <= 0xEF) {
                    AUDIO_LOG_ERROR("video stream!");
                    return false;
                }
                int PES_PacketLength = ((packet[posOfPacketStart + 4] & 0xFF) << 8) + (packet[posOfPacketStart + 5] & 0xFF);
                if (log) AUDIO_LOG_DEBUG("PES PacketLength: %d", PES_PacketLength);
                bool PTS_flag = false;
                bool DTS_flag = false;
                int  flag_byte1 = packet[posOfPacketStart + 6] & 0xFF;
                int  flag_byte2 = packet[posOfPacketStart + 7] & 0xFF;
                (void)flag_byte2; // unused yet
                if (flag_byte1 & 0b10000000) PTS_flag = true;
                if (flag_byte1 & 0b00000100) DTS_flag = true;
                if (log && PTS_flag) AUDIO_LOG_DEBUG("PTS_flag is set");
                if (log && DTS_flag) AUDIO_LOG_DEBUG("DTS_flag is set");
                uint8_t PES_HeaderDataLength = packet[posOfPacketStart + 8] & 0xFF;
                if (log) AUDIO_LOG_DEBUG("PES_headerDataLength %d", PES_HeaderDataLength);

                m_tspp.PES_DataLength = PES_PacketLength;
                int startOfData = PES_HeaderDataLength + 9;
                if (posOfPacketStart + startOfData >= 188) { // only fillers in packet
                    if (log) AUDIO_LOG_DEBUG("posOfPacketStart + startOfData %i", posOfPacketStart + startOfData);
                    *packetStart = 0;
                    *packetLength = 0;
                    m_tspp.PES_DataLength -= (PES_HeaderDataLength + 3);
                    m_tspp.fillData = (posOfPacketStart + startOfData) - 188;
                    if (log) AUDIO_LOG_DEBUG("fillData %i", m_tspp.fillData);
                    return true;
                }
                if (log) AUDIO_LOG_DEBUG("First AAC data byte: %02X", packet[posOfPacketStart + startOfData]);
                if (log) AUDIO_LOG_DEBUG("Second AAC data byte: %02X", packet[posOfPacketStart + startOfData + 1]);
                *packetStart = posOfPacketStart + startOfData;
                *packetLength = TS_PACKET_SIZE - posOfPacketStart - startOfData;
                m_tspp.PES_DataLength -= (*packetLength);
                m_tspp.PES_DataLength -= (PES_HeaderDataLength + 3);
                return true;
            }
            if (firstByte == 0 && secondByte == 0 && thirdByte == 0) {
                // PES packet startcode prefix is 0x000000
                // skip such packets
                return true;
            }
        }
        *packetStart = 0;
        *packetLength = 0;
        AUDIO_LOG_ERROR("PES not found");
        return false;
    } else if (m_tspp.pidNumber) {
        //  Program Map Table (PMT) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
        for (int i = 0; i < m_tspp.pidNumber; i++) {
            if (PID == m_tspp.pids[i]) {
                if (log) AUDIO_LOG_DEBUG("PMT");
                int staticLengthOfPMT = 12;
                int sectionLength = ((packet[PLS + 1] & 0x0F) << 8) | (packet[PLS + 2] & 0xFF);
                if (log) AUDIO_LOG_DEBUG("Section Length: %d", sectionLength);
                int programInfoLength = ((packet[PLS + 10] & 0x0F) << 8) | (packet[PLS + 11] & 0xFF);
                if (log) AUDIO_LOG_DEBUG("Program Info Length: %d", programInfoLength);
                int cursor = staticLengthOfPMT + programInfoLength;
                while (cursor < sectionLength - 1) {
                    int streamType = packet[PLS + cursor] & 0xFF;
                    int elementaryPID = ((packet[PLS + cursor + 1] & 0x1F) << 8) | (packet[PLS + cursor + 2] & 0xFF);
                    if (log) AUDIO_LOG_DEBUG("Stream Type: 0x%02X Elementary PID: 0x%04X", streamType, elementaryPID);

                    if (streamType == 0x0F || streamType == 0x11 || streamType == 0x04) {
                        if (log) AUDIO_LOG_DEBUG("AAC PID discover");
                        m_tspp.pidOfAAC = elementaryPID;
                    }
                    int esInfoLength = ((packet[PLS + cursor + 3] & 0x0F) << 8) | (packet[PLS + cursor + 4] & 0xFF);
                    if (log) AUDIO_LOG_DEBUG("ES Info Length: 0x%04X", esInfoLength);
                    cursor += 5 + esInfoLength;
                }
            }
        }
        *packetStart = 0;
        *packetLength = 0;
        return true;
    }
    // PES received before PAT and PMT seen
    *packetStart = 0;
    *packetLength = 0;
    if (PID > 0) { return true; }
    return false;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
//    W E B S T R E A M  -  H E L P   F U N C T I O N S
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
bool Audio::readMetadata(uint16_t maxBytes, uint16_t* readedBytes, bool first) {
    *readedBytes = 0;
    ps_ptr<char> buff;
    buff.alloc(4096 + 1); // max 4096 + 1 for null terminator, just to make library code 'safe'
    buff.clear();         // is max 256 *16
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (first) {
        m_rmet.pos_ml = 0; // determines the current position in metaline
        m_rmet.metaDataSize = 0;
        m_rmet.res = 0;
        return true;
    }
    // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    if (!maxBytes) return 0; // guard

    if (!m_rmet.metaDataSize) {
        int b = audioFileRead(); // First byte of metadata?
        if (b < 0) {
            AUDIO_LOG_WARN("client->read() failed (%d)", b);
            return false;
        }
        m_rmet.metaDataSize = b * 16; // New count for metadata including length byte, max 4096
        m_rmet.pos_ml = 0;
        buff[m_rmet.pos_ml] = 0; // Prepare for new line
        *readedBytes = 1;
        maxBytes -= 1;
    }
    if (!m_rmet.metaDataSize) { return *readedBytes; } // metalen is 0
    int32_t a = audioFileRead((uint8_t*)&buff[m_rmet.pos_ml], min((uint16_t)(m_rmet.metaDataSize - m_rmet.pos_ml), (uint16_t)(maxBytes)));

    if (a > 0) {
        m_rmet.res += a;
        *readedBytes += a;
        m_rmet.pos_ml += a;
    }
    if (m_rmet.pos_ml == m_rmet.metaDataSize) {
        buff[m_rmet.pos_ml] = '\0';
        // buff.hex_dump(m_rmet.metaDataSize);
        if (buff.strlen() > 0) { // Any info present?
            // metaline contains artist and song name.  For example:
            // "StreamTitle='Don McLean - American Pie';StreamUrl='';"
            // Sometimes it is just other info like:
            // "StreamTitle='60s 03 05 Magic60s';StreamUrl='';"
            // Isolate the StreamTitle, remove leading and trailing quotes if present.
            latinToUTF8(buff);                             // convert to UTF-8 if necessary
            int pos = buff.index_of_icase("song_spot", 0); // remove some irrelevant infos
            if (pos > 3) {                                 // e.g. song_spot="T" MediaBaseId="0" itunesTrackId="0"
                buff[pos] = 0;
            }
            showstreamtitle(buff.get()); // Show artist and title if present in metadata
        }
        m_metacount = m_metaint;
        m_rmet.metaDataSize = 0;
        m_rmet.pos_ml = 0;
        buff[m_rmet.pos_ml] = 0; // Prepare for new line
    } else {
        return false; // not enough data, next round
    }
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
int32_t Audio::getChunkSize(uint16_t* readedBytes, bool first) {
    uint32_t timeout = 2000; // ms
    uint32_t ctime;

    if (first) {
        m_gchs.oneByteOfTwo = false;
        m_gchs.f_skipCRLF = false;
        m_gchs.isHttpChunked = true; // default: We assume http chunked
        m_gchs.transportLimit = 0;   // 0 = not yet recognized
        return 0;
    }

    *readedBytes = 0;

    // if we are in transport chunk mode → return limit
    if (!m_gchs.isHttpChunked) { return m_gchs.transportLimit; }

    // skip CRLF from the previous chunk (only http-chunked)
    if (m_gchs.f_skipCRLF) {
        uint32_t t = millis();

        if (m_client->available() == 1 && !m_gchs.oneByteOfTwo) {
            int a = audioFileRead();
            m_gchs.oneByteOfTwo = true;
            *readedBytes = 1;
            if (a != 0x0D) AUDIO_LOG_WARN("chunk count error, expected: 0x0D, received: 0x%02X", a);

            return -1;
        }
        if (!m_gchs.oneByteOfTwo) {
            int a = audioFileRead();
            if (a != 0x0D) AUDIO_LOG_WARN("chunk count error, expected: 0x0D, received: 0x%02X", a);
            *readedBytes += 1;
            if (!m_client->available()) { return -1; }
        }
        m_gchs.oneByteOfTwo = false;
        int b = audioFileRead();
        if (b != 0x0A) AUDIO_LOG_WARN("chunk count error, expected: 0x0A, received: 0x%02X", b);
        *readedBytes += 1;
        m_gchs.f_skipCRLF = false;
        if (!m_client->available()) { return -1; }
    }

    // -------- HTTP-chunked-Read Logic --------
    std::string chunkLine;
    ctime = millis();

    while (true) {
        if ((millis() - ctime) > timeout) {
            AUDIO_LOG_ERROR("chunkedDataTransfer: timeout");
            stopSong();
            return 0;
        }
        if (!m_client->available()) continue;
        int b = audioFileRead();
        if (b < 0) continue;

        (*readedBytes)++;

        if (b == '\n') break; // End of the line
        if (b == '\r') continue;

        chunkLine += static_cast<char>(b);

        // Detection: if signs are not hexadecimal and not ';'→ No http chunk
        if (!isxdigit(b) && b != ';') {
            // We have no valid HTTP chunk line → assume transport chunking
            m_gchs.isHttpChunked = false;
            // determine limit from the current data volume + already read bytes
            m_gchs.transportLimit = m_client->available() + *readedBytes;
            AUDIO_LOG_DEBUG("No http chunked recognized-switch to transport chunking with limit %u", (unsigned)m_gchs.transportLimit);
            return m_gchs.transportLimit;
        }
    }

    // Extract the hex number (before possibly ';')
    size_t      semicolonPos = chunkLine.find(';');
    std::string hexSize = (semicolonPos != std::string::npos) ? chunkLine.substr(0, semicolonPos) : chunkLine;

    size_t chunksize = strtoul(hexSize.c_str(), nullptr, 16);

    if (chunksize > 0) {
        m_gchs.f_skipCRLF = true; // skip next CRLF after data
    } else {
        // last chunk: read the final CRLF
        uint8_t idx = 0;
        ctime = millis();
        while (idx < 2 && (millis() - ctime) < timeout) {
            int ch = audioFileRead();
            if (ch < 0) continue;
            idx++;
        }
    }
    return chunksize;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
bool Audio::readID3V1Tag() {
    if (m_codec != CODEC_MP3) return false;
    ps_ptr<char> chBuff;
    chBuff.alloc(256);

    const uint8_t* p = InBuff.getReadPtr();

    // Lambda for simplification
    auto readID3Field = [&](ps_ptr<char>& field, const uint8_t* src, size_t len, const char* label = nullptr) {
        field.alloc(len + 1, "field");
        memcpy(field.get(), src, len);
        field[len] = '\0';
        latinToUTF8(field);
        if (field.strlen() > 0 && label) {
            field.insert(label, 0);
            info(*this, evt_id3data, "%s", field.get());
        }
    };

    if (InBuff.bufferFilled() == 128 && startsWith((const char*)p, "TAG")) {
        ps_ptr<char> title, artist, album, year, comment;

        readID3Field(title, p + 3, 30, "Title: ");
        readID3Field(artist, p + 33, 30, "Artist: ");
        readID3Field(album, p + 63, 30, "Album: ");
        readID3Field(year, p + 93, 4, "Year: ");
        readID3Field(comment, p + 97, 30, "Comment: ");

        uint8_t zeroByte = p[125];
        uint8_t track = p[126];
        uint8_t genre8 = p[127];

        info(*this, evt_info, zeroByte ? "ID3 version: 1" : "ID3 Version 1.1");

        if (zeroByte == 0) {
            sprintf(chBuff.get(), "Track Number: %d", track);
            info(*this, evt_id3data, "%s", chBuff.get());
        }

        if (genre8 < 192) {
            sprintf(chBuff.get(), "Genre: %d", genre8);
            info(*this, evt_id3data, "%s", chBuff.get());
        }

        return true;
    }

    if (InBuff.bufferFilled() == 227 && startsWith((const char*)p, "TAG+")) { // "TAG+" "can exist as an extension, does not overwrite" TAG"
        info(*this, evt_info, "ID3 version: 1 - Enhanced TAG");

        ps_ptr<char> title, artist, album, genre;

        readID3Field(title, p + 4, 60, "Title: ");
        readID3Field(artist, p + 64, 60, "Artist: ");
        readID3Field(album, p + 124, 60, "Album: ");
        readID3Field(genre, p + 185, 30, "Genre: ");

        // optional expansion: speed, start-time, end-time
        return true;
    }

    return false;
    // [1] https://en.wikipedia.org/wiki/List_of_ID3v1_Genres
    // [2] https://en.wikipedia.org/wiki/ID3#ID3v1_and_ID3v1.1[5]
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
int32_t Audio::newInBuffStart(int32_t resumeFilePos) {

    if ((m_controlCounter != 100) || (m_resumeFilePos >= (int32_t)m_audioDataStart + m_audioDataSize) || ((m_codec == CODEC_M4A) && !m_stsz_position)) {
        AUDIO_LOG_WARN("timeOffset not possible");
        return 0;
    }

    // keep resumeFilePos within the audio data
    if (resumeFilePos < (int32_t)m_audioDataStart) resumeFilePos = m_audioDataStart;

    uint32_t buffFillValue = std::min<uint32_t>(m_audioDataSize - resumeFilePos, UINT16_MAX);

    AUDIO_LOG_DEBUG("new InBuff start at m_resumeFilePos %i, m_audioDataStart %i", m_resumeFilePos, m_audioDataStart);

    // --- enter critical area ------------------------------------------
    xSemaphoreTake(mutex_audioTaskIsDecoding, 1 * configTICK_RATE_HZ);
    m_f_lockInBuffer = true;

    // ------- prepare InBuff ------
    m_f_allDataReceived = false;
    audioFileSeek(resumeFilePos);
    InBuff.reset();
    audioFileRead(InBuff.getWritePtr(), buffFillValue);
    InBuff.bytesWritten(buffFillValue);

    int32_t offset = 0;
    int32_t newFilePos = resumeFilePos;

    // ---------------------------------------------------------------------------
    // codec specific handling
    // ---------------------------------------------------------------------------

    auto fail = [&]() {
        AUDIO_LOG_ERROR("can't set newFilePos");
        InBuff.bytesWasRead(0);
        m_f_lockInBuffer = false;
        xSemaphoreGive(mutex_audioTaskIsDecoding);
        stopSong();
        return -1;
    };

    switch (m_codec) {
        case CODEC_M4A:
            offset = m4a_correctResumeFilePos();
            if (offset < 0) return fail();
            break;

        case CODEC_WAV:
            // WAV muss auf 4-Byte-Grenze
            offset = wav_correctResumeFilePos();
            if (offset < 0) return fail();
            break;

        case CODEC_MP3:
            offset = mp3_correctResumeFilePos();
            if (offset < 0) return fail();
            break;

        case CODEC_FLAC:
            offset = flac_correctResumeFilePos();
            if (offset < 0) return fail();
            break;

        case CODEC_VORBIS:
            offset = ogg_correctResumeFilePos(); // next OggS
            if (offset < 0) return fail();
            break;

        case CODEC_OPUS:
            offset = ogg_correctResumeFilePos(); // next OggS
            if (offset < 0) return fail();
            break;

        default: return fail();
    }

    AUDIO_LOG_DEBUG("offset %i", offset);
    newFilePos += offset;
    m_decoder->clear();

    // --- leaf critical area ----------------------------------------------
    InBuff.bytesWasRead(offset);
    m_f_lockInBuffer = false;
    xSemaphoreGive(mutex_audioTaskIsDecoding);

    return newFilePos;
}

// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
boolean Audio::streamDetection(uint32_t bytesAvail) {
    if (!m_lastHost.valid()) {
        AUDIO_LOG_ERROR("m_lastHost is empty");
        return false;
    }

    // if within one second the content of the audio buffer falls below the size of an audio frame 100 times,
    // issue a message
    if (m_sdet.tmr_slow + 1000 < millis()) {
        m_sdet.tmr_slow = millis();
        if (m_sdet.cnt_slow > 100) info(*this, evt_info, "slow stream, dropouts are possible");
        m_sdet.cnt_slow = 0;
    }
    if (InBuff.bufferFilled() < InBuff.getMaxBlockSize()) m_sdet.cnt_slow++;
    if (bytesAvail) {
        m_sdet.tmr_lost = millis() + 1000;
        m_sdet.cnt_lost = 0;
    }
    if (InBuff.bufferFilled() > InBuff.getMaxBlockSize() * 2) return false; // enough data available to play

    // if no audio data is received within three seconds, a new connection attempt is started.
    if (m_sdet.tmr_lost < millis()) {
        m_sdet.cnt_lost++;
        m_sdet.tmr_lost = millis() + 1000;
        if (m_sdet.cnt_lost == 5) { // 5s no data?
            m_sdet.cnt_lost = 0;
            info(*this, evt_info, "Stream lost -> try new connection");
            m_f_reset_m3u8Codec = false;
            InBuff.reset();
            httpPrint(m_lastHost.get());
            return true;
        }
    }
    return false;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
uint32_t Audio::m4a_correctResumeFilePos() {
    // In order to jump within an m4a file, the exact beginning of an aac block must be found. Since m4a cannot be
    // streamed, i.e. there is no syncword, an imprecise jump can lead to a crash.

    if (!m_stsz_position) return m_audioDataStart; // guard

    typedef union {
        uint8_t  u8[4];
        uint32_t u32;
    } tu;
    tu uu = {};

    uint32_t i = 0, pos = m_audioDataStart;
    uint32_t filePtr = m_audioFilePosition;
    bool     found = false;
    audioFileSeek(m_stsz_position, m_stsz_numEntries * 4);

    auto read_next = [&]() -> int {
        int r = 0;
        int cnt = 0;
        while (true) {
            r = audioFileRead();
            if (r >= 0) break;
            vTaskDelay(10);
            cnt++;
            if (cnt > 300) {
                AUDIO_LOG_ERROR("timeout");
                break;
            }
        }
        return r;
    };

    while (i < m_stsz_numEntries * 4) { // is uint32_t
        i++;
        uu.u8[3] = read_next();
        uu.u8[2] = read_next();
        uu.u8[1] = read_next();
        uu.u8[0] = read_next();

        pos += uu.u32;
        if (pos >= m_resumeFilePos) {
            found = true;
            break;
        }
    }
    if (!found) return -1; // not found

    audioFileSeek(filePtr);       // restore file pointer
    return pos - m_resumeFilePos; // return the number of bytes to jump
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
uint32_t Audio::ogg_correctResumeFilePos() {
    // The starting point is the next OggS magic word
    if (InBuff.bufferFilled() < 0xFFFF) return 0;
    vTaskDelay(1);
    // // AUDIO_LOG_INFO("in_resumeFilePos %i", resumeFilePos);

    auto find_sync_word = [&](uint8_t* pos, uint32_t av) -> int {
        int steps = 0;
        while (av--) {
            if (pos[steps] == 'O') {
                if (pos[steps + 1] == 'g') {
                    if (pos[steps + 2] == 'g') {
                        if (pos[steps + 3] == 'S') { // Check for the second part of magic word
                            return steps;            // Magic word found, return the number of steps
                        }
                    }
                }
            }
            steps++;
        }
        return -1; // Return -1 if OggS magic word is not found
    };

    uint8_t* readPtr = InBuff.getReadPtr();
    size_t   av = InBuff.readSpace();
    int32_t  steps = 0;

    if (av < InBuff.getMaxBlockSize()) return -1; // guard

    steps = find_sync_word(readPtr, av);
    if (steps == -1) return -1;
    return steps; // Return the number of steps to the sync word
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
int32_t Audio::flac_correctResumeFilePos() {
    uint8_t* p = InBuff.getReadPtr();
    size_t   av = InBuff.readSpace();

    if (av < 32) return -1; // sicher gehen

    auto utf8_length = [&](uint8_t first) -> int {
        if ((first & 0x80) == 0x00) return 1; // 0xxxxxxx
        if ((first & 0xE0) == 0xC0) return 2; // 110xxxxx
        if ((first & 0xF0) == 0xE0) return 3; // 1110xxxx
        if ((first & 0xF8) == 0xF0) return 4; // 11110xxx
        if ((first & 0xFC) == 0xF8) return 5; // 111110xx
        if ((first & 0xFE) == 0xFC) return 6; // 1111110x
        if (first == 0xFE) return 7;          // 11111110
        return -1;                            // invalid UTF-8
    };

    // FLAC CRC-8
    auto crc8_update = [](uint8_t crc, uint8_t data) -> uint8_t {
        crc ^= data;
        for (int i = 0; i < 8; ++i) {
            if (crc & 0x80)
                crc = (crc << 1) ^ 0x07;
            else
                crc <<= 1;
        }
        return crc;
    };

    for (size_t i = 0; i + 14 < av; ++i) {
        // 14-Bit-Sync: 0xFFF8 oder 0xFFF9 (last 2 Bits variable!)
        uint8_t b0 = p[i + 0]; // 11111111  sync
        uint8_t b1 = p[i + 1]; // 111111rb  sync         r-reserved    b-blocking strategy
        if (b0 != 0xFF) continue;
        if ((b1 & 0xFE) != 0xF8) continue;

        uint8_t b2 = p[i + 2]; // bbbbssss   b-blocksize   s samplerate
        uint8_t blocksize_code = (b2 & 0xF0) >> 4;
        uint8_t samplerate_code = (b2 & 0x0F);

        uint8_t b3 = p[i + 3]; // ccccsssr   c-channel assignment, s sample size, r reserved
        uint8_t channel_assign = (b3 & 0xF0) >> 4;
        uint8_t bps_code = (b3 & 0x0E) >> 1; // bits per sample
        uint8_t reserved_bit = b3 & 0x01;    // has to be 0

        if (reserved_bit != 0) continue;
        if (channel_assign >= 11) continue; // 11-15 reserved
        if (bps_code == 0x07) continue;     // reserved

        size_t pos = i + 4; // first byte of UTF8
        int    len = utf8_length(p[pos]);
        if (len < 1 || pos + len >= av) continue; // invalid/incomplete header

        size_t crc_end = pos + len; // this is where the CRC byte lives

        AUDIO_LOG_DEBUG("samplerate_code %i, bps_code %i channel_assign %i, reserved_bit %i", samplerate_code, bps_code, channel_assign, reserved_bit);
        AUDIO_LOG_DEBUG("b0 %i, b1 %i, b2 %i, b3 %i", b0, b1, b2, b3);

        uint8_t crc = 0;
        for (size_t k = i; k < crc_end; ++k) crc = crc8_update(crc, p[k]);

        if (crc != p[crc_end]) continue;

        // === Wenn wir hier ankommen: 99,9999 % sicher ein echter Frame ===
        AUDIO_LOG_ERROR(">>> ECHTER FLAC-FRAME gefunden bei Offset %u", (uint32_t)i);
        return (int32_t)i;
    }

    return -1; // kein Frame in den ersten 64 KB gefunden
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
int32_t Audio::mp3_correctResumeFilePos() {

    int32_t  steps = 0;
    int32_t  sumSteps = 0;
    uint8_t* pos = InBuff.getReadPtr();
    size_t   av = InBuff.readSpace();

    if (av < InBuff.getMaxBlockSize()) return -1; // guard

    while (true) {
        steps = m_decoder->findSyncWord(pos, av);
        if (steps == 0) break;
        if (steps == -1) return -1;
        pos += steps;
        sumSteps += steps;
    }
    // AUDIO_LOG_INFO("found sync word at %i  sync1 = 0x%02X, sync2 = 0x%02X", readPtr - pos, *readPtr, *(readPtr + 1));
    return sumSteps; // return the position of the first byte of the frame
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
int32_t Audio::wav_correctResumeFilePos() {
    int32_t pos = m_resumeFilePos;
    uint8_t offset = 0;
    ;
    // WAV must be on 4 byte limit
    while (pos % 4 != 0) {
        offset++;
        pos++;
    }
    if (pos >= m_audioDataStart + m_audioDataSize) return 0;
    return offset;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————-
uint8_t Audio::determineOggCodec() {
    // if we have contentType == application/ogg; codec cn be OPUS, FLAC or VORBIS
    // let's have a look, what it is
    uint8_t res = CODEC_NONE;
    int     idx = -1;
    idx = specialIndexOf(InBuff.getReadPtr(), "OggS", 127);
    AUDIO_LOG_DEBUG("idx %i", idx);
    if (idx == 0) {
        idx = specialIndexOf(InBuff.getReadPtr(), "OpusHead", 127);
        if (idx >= 28) { res = CODEC_OPUS; }
        idx = specialIndexOf(InBuff.getReadPtr(), "fLaC", 127);
        if (idx >= 28) { res = CODEC_FLAC; }
        idx = specialIndexOf(InBuff.getReadPtr(), "vorbis", 127);
        if (idx >= 28) { res = CODEC_VORBIS; }
    }
    return res;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::strlower(char* str) {
    unsigned char* p = (unsigned char*)str;
    while (*p) {
        *p = tolower((unsigned char)*p);
        p++;
    }
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::trim(char* str) {
    char* start = str; // keep the original pointer
    char* end;
    while (isspace((unsigned char)*start)) start++; // find the first non-space character

    if (*start == 0) { // all characters were spaces
        str[0] = '\0'; // return a empty string
        return;
    }

    end = start + strlen(start) - 1; // find the end of the string

    while (end > start && isspace((unsigned char)*end)) end--;
    end[1] = '\0'; // Null-terminate the string after the last non-space character

    // Move the trimmed string to the beginning of the memory area
    memmove(str, start, strlen(start) + 1); // +1 for '\0'
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::startsWith(const char* base, const char* str) {
    // fb
    char c;
    while ((c = *str++) != '\0')
        if (c != *base++) return false;
    return true;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::endsWith(const char* base, const char* searchString) {
    int32_t slen = strlen(searchString);
    if (slen == 0) return false;
    const char* p = base + strlen(base);
    //  while(p > base && isspace(*p)) p--;  // rtrim
    p -= slen;
    if (p < base) return false;
    return (strncmp(p, searchString, slen) == 0);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::indexOf(const char* base, const char* str, int startIndex) {
    // fbi
    const char* p = base;
    for (; startIndex > 0; startIndex--)
        if (*p++ == '\0') return -1;
    char* pos = strstr(p, str);
    if (pos == nullptr) return -1;
    return pos - base;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::indexOf(const char* base, char ch, int startIndex) {
    // fb
    const char* p = base;
    for (; startIndex > 0; startIndex--)
        if (*p++ == '\0') return -1;
    char* pos = strchr(p, ch);
    if (pos == nullptr) return -1;
    return pos - base;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::lastIndexOf(const char* haystack, const char* needle) {
    // fb
    int nlen = strlen(needle);
    if (nlen == 0) return -1;
    const char* p = haystack - nlen + strlen(haystack);
    while (p >= haystack) {
        int i = 0;
        while (needle[i] == p[i])
            if (++i == nlen) return p - haystack;
        p--;
    }
    return -1;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::lastIndexOf(const char* haystack, const char needle) {
    // fb
    const char* p = strrchr(haystack, needle);
    return (p ? p - haystack : -1);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::specialIndexOf(uint8_t* base, const char* str, int baselen, bool exact) {
    int result = 0;                       // seek for str in buffer or in header up to baselen, not nullterninated
    if (strlen(str) > baselen) return -1; // if exact == true seekstr in buffer must have "\0" at the end
    for (int i = 0; i < baselen - strlen(str); i++) {
        result = i;
        for (int j = 0; j < strlen(str) + exact; j++) {
            if (*(base + i + j) != *(str + j)) {
                result = -1;
                break;
            }
        }
        if (result >= 0) break;
    }
    return result;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
// Find the last instance of Str in the buffer backwards and checks end-of-stream-bit
int Audio::specialIndexOfLast(uint8_t* base, const char* str, int baselen) {
    int result = -1;
    if (strlen(str) > baselen) return -1; // too short buffer

    for (int i = baselen - strlen(str); i >= 0; i--) { // search backwards, start at the end of the buffer
        int match = 1;
        for (int j = 0; j < strlen(str); j++) {
            if (base[i + j] != str[j]) {
                match = 0;
                break;
            }
        }
        if (match) return i;
    }
    return result;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int Audio::find_utf16_null_terminator(const uint8_t* buf, int start, int max) {
    for (int i = start; i + 1 < max; i += 2) {
        if (buf[i] == 0x00 && buf[i + 1] == 0x00) return i; // Index to the first zero-byte
    }
    return -1; // not found
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
int32_t Audio::min3(int32_t a, int32_t b, int32_t c) {
    uint32_t min_val = a;
    if (b < min_val) min_val = b;
    if (c < min_val) min_val = c;
    return min_val;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
//  some other functions
uint64_t Audio::bigEndian(uint8_t* base, uint8_t numBytes, uint8_t shiftLeft) {
    uint64_t result = 0; // Use uint64_t for greater caching
    if (numBytes < 1 || numBytes > 8) return 0;
    for (int i = 0; i < numBytes; i++) {
        result |= (uint64_t)(*(base + i)) << ((numBytes - i - 1) * shiftLeft); // Make sure the calculation is done correctly
    }
    if (result > SIZE_MAX) {
        log_e("range overflow");
        return 0;
    }
    return result;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
bool Audio::b64encode(const char* source, uint16_t sourceLength, char* dest) {
    size_t size = base64_encode_expected_len(sourceLength) + 1;
    char*  buffer = (char*)malloc(size);
    if (buffer) {
        base64_encodestate _state;
        base64_init_encodestate(&_state);
        int len = base64_encode_block(&source[0], sourceLength, &buffer[0], &_state);
        base64_encode_blockend((buffer + len), &_state);
        memcpy(dest, buffer, strlen(buffer));
        dest[strlen(buffer)] = '\0';
        free(buffer);
        return true;
    }
    return false;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::vector_clear_and_shrink(std::vector<ps_ptr<char>>& vec) {
    for (int i = 0; i < vec.size(); i++) vec[i].reset();
    vec.clear();         // unique_ptr takes care of free()
    vec.shrink_to_fit(); // put back memory
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
void Audio::deque_clear_and_shrink(std::deque<ps_ptr<char>>& deq) {
    for (int i = 0; i < deq.size(); i++) deq[i].reset();
    deq.clear();         // unique_ptr takes care of free()
    deq.shrink_to_fit(); // put back memory
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::simpleHash(const char* str) {
    if (str == NULL) return 0;
    uint32_t hash = 0;
    for (int i = 0; i < strlen(str); i++) {
        if (str[i] < 32) continue; // ignore control sign
        hash += (str[i] - 31) * i * 32;
    }
    return hash;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
ps_ptr<char> Audio::urlencode(const char* str, bool spacesOnly) {
    if (!str) { return {}; } // Enter is zero

    // Reserve memory for the result (3x the length of the input string, worst-case)
    size_t       inputLength = strlen(str);
    size_t       bufferSize = inputLength * 3 + 1; // Worst-case-Szenario
    ps_ptr<char> encoded;
    encoded.alloc(bufferSize);
    if (!encoded.valid()) { return {}; } // memory allocation failed

    const char* p_input = str;               // Copy of the input pointer
    char*       p_encoded = encoded.get();   // pointer of the output buffer
    size_t      remainingSpace = bufferSize; // remaining space in the output buffer

    while (*p_input) {
        if (isalnum((unsigned char)*p_input)) {
            // adopt alphanumeric characters directly
            if (remainingSpace > 1) {
                *p_encoded++ = *p_input;
                remainingSpace--;
            } else {
                return {}; // security check failed
            }
        } else if (spacesOnly && *p_input != 0x20) {
            // Nur Leerzeichen nicht kodieren
            if (remainingSpace > 1) {
                *p_encoded++ = *p_input;
                remainingSpace--;
            } else {
                return {}; // security check failed
            }
        } else {
            // encode unsafe characters as '%XX'
            if (remainingSpace > 3) {
                int written = snprintf(p_encoded, remainingSpace, "%%%02X", (unsigned char)*p_input);
                if (written < 0 || written >= (int)remainingSpace) {
                    return {}; // error writing to buffer
                }
                p_encoded += written;
                remainingSpace -= written;
            } else {
                return {}; // security check failed
            }
        }
        p_input++;
    }

    // Null-terminieren
    if (remainingSpace > 0) {
        *p_encoded = '\0';
    } else {
        return {}; // security check failed
    }
    encoded.shrink_to_fit();
    return encoded;
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
//  Function to reverse the byte order of a 32-bit value (big-endian to little-endian)
uint32_t Audio::bswap32(uint32_t x) {
    return ((x & 0xFF000000) >> 24) | ((x & 0x00FF0000) >> 8) | ((x & 0x0000FF00) << 8) | ((x & 0x000000FF) << 24);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
//  Function to reverse the byte order of a 64-bit value (big-endian to little-endian)
uint64_t Audio::bswap64(uint64_t x) {
    return ((x & 0xFF00000000000000ULL) >> 56) | ((x & 0x00FF000000000000ULL) >> 40) | ((x & 0x0000FF0000000000ULL) >> 24) | ((x & 0x000000FF00000000ULL) >> 8) | ((x & 0x00000000FF000000ULL) << 8) |
           ((x & 0x0000000000FF0000ULL) << 24) | ((x & 0x000000000000FF00ULL) << 40) | ((x & 0x00000000000000FFULL) << 56);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
// separate task for decoding and outputting the data. 'playAudioData()' is started periodically and fetches the data from the InBuffer. This ensures
// that the I2S-DMA is always sufficiently filled, even if the Arduino 'loop' is stuck.
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————

void Audio::setAudioTaskCore(uint8_t coreID) { // Recommendation:If the ARDUINO RUNNING CORE is 1, the audio task should be core 0 or vice versa
    if (coreID > 1) return;
    stopAudioTask();
    xSemaphoreTake(mutex_audioTask, 0.3 * configTICK_RATE_HZ);
    m_audioTaskCoreId = coreID;
    xSemaphoreGive(mutex_audioTask);
    startAudioTask();
}

void Audio::startAudioTask() {

    if (m_f_audioTaskIsRunning) {
        AUDIO_LOG_INFO("Task is already running.");
        return;
    }
    m_f_audioTaskIsRunning = true;

    m_audioTaskHandle = xTaskCreateStaticPinnedToCore(&Audio::audioTaskWrapper, /* Function to implement the task */
                                                      "PeriodicTask",           /* Name of the task */
                                                      AUDIO_STACK_SIZE,         /* Stack size in words */
                                                      this,                     /* Task input parameter */
                                                      2,                        /* Priority of the task */
                                                      xAudioStack,              /* Task stack */
                                                      &xAudioTaskBuffer,        /* Memory for the task's control block */
                                                      m_audioTaskCoreId         /* Core where the task should run */
    );
}

void Audio::stopAudioTask() {
    if (!m_f_audioTaskIsRunning) {
        AUDIO_LOG_INFO("audio task is not running.");
        return;
    }
    xSemaphoreTake(mutex_audioTask, 0.3 * configTICK_RATE_HZ);
    m_f_audioTaskIsRunning = false;
    if (m_audioTaskHandle != nullptr) {
        vTaskDelete(m_audioTaskHandle);
        m_audioTaskHandle = nullptr;
    }
    xSemaphoreGive(mutex_audioTask);
}

void Audio::audioTaskWrapper(void* param) {
    Audio* audioRunner = static_cast<Audio*>(param);
    audioRunner->audioTask();
}

void Audio::audioTask() {
    while (m_f_audioTaskIsRunning) {
        vTaskDelay(1 / portTICK_PERIOD_MS); // periodically every x ms
        if (m_f_I2S_init) performAudioTask();
    }
    vTaskDelete(nullptr); // Delete this task
}

void Audio::performAudioTask() {
    if (!m_f_running) {
        int32_t c[2] = {0};
        calculateVUlevel(c);
        gain_ramp();
        vTaskDelay(20);
        return;
    }
    if (!m_f_stream) return;
    if (m_codec == CODEC_NONE) return; // wait for codec is  set
    if (m_codec == CODEC_OGG) return;  // wait for FLAC, VORBIS or OPUS
    xSemaphoreTake(mutex_audioTask, 0.3 * configTICK_RATE_HZ);
    while (m_validSamples) {
        vTaskDelay(20 / portTICK_PERIOD_MS);
        playChunk();
    } // I2S buffer full
    playAudioData();
    gain_ramp();
    xSemaphoreGive(mutex_audioTask);
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::getHighWatermark() {
    UBaseType_t highWaterMark = uxTaskGetStackHighWaterMark(m_audioTaskHandle);
    return highWaterMark; // dwords
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
