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Audio.cpp
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Audio.cpp
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/*
* Audio.cpp
*
* Created on: Oct 27.2018
*
* Version 3.0.13
* Updated on: Oct 09.2024
* Author: Wolle (schreibfaul1)
*
*/
#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 "vorbis_decoder/vorbis_decoder.h"
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
AudioBuffer::AudioBuffer(size_t maxBlockSize) {
mutex_buffer = xSemaphoreCreateRecursiveMutex();
// if maxBlockSize isn't set use defaultspace (1600 bytes) is enough for aac and mp3 player
if(maxBlockSize) m_resBuffSizeRAM = maxBlockSize;
if(maxBlockSize) m_maxBlockSize = maxBlockSize;
}
AudioBuffer::~AudioBuffer() {
if(m_buffer) free(m_buffer);
m_buffer = NULL;
vSemaphoreDelete(mutex_buffer);
}
void AudioBuffer::setBufsize(int ram, int psram) {
if(ram > -1) // -1 == default / no change
m_buffSizeRAM = ram;
if(psram > -1) m_buffSizePSRAM = psram;
}
int32_t AudioBuffer::getBufsize() { return m_buffSize; }
size_t AudioBuffer::init() {
if(m_buffer) free(m_buffer);
m_buffer = NULL;
if(psramInit() && m_buffSizePSRAM > 0) {
// PSRAM found, AudioBuffer will be allocated in PSRAM
m_f_psram = true;
m_buffSize = m_buffSizePSRAM;
m_buffer = (uint8_t*)ps_calloc(m_buffSize, sizeof(uint8_t));
m_buffSize = m_buffSizePSRAM - m_resBuffSizePSRAM;
}
if(m_buffer == NULL) {
// PSRAM not found, not configured or not enough available
m_f_psram = false;
m_buffer = (uint8_t*)heap_caps_calloc(m_buffSizeRAM, sizeof(uint8_t), MALLOC_CAP_DEFAULT | MALLOC_CAP_INTERNAL);
m_buffSize = m_buffSizeRAM - m_resBuffSizeRAM;
}
if(!m_buffer) return 0;
m_f_init = true;
resetBuffer();
return m_buffSize;
}
void AudioBuffer::changeMaxBlockSize(uint16_t mbs) {
m_maxBlockSize = mbs;
return;
}
uint16_t AudioBuffer::getMaxBlockSize() { return m_maxBlockSize; }
size_t AudioBuffer::freeSpace() {
xSemaphoreTakeRecursive(mutex_buffer, 3 * configTICK_RATE_HZ);
if(m_readPtr == m_writePtr) {
if(m_f_isEmpty == true) m_freeSpace = m_buffSize;
else m_freeSpace = 0;
}
if(m_readPtr < m_writePtr) {
m_freeSpace = (m_endPtr - m_writePtr) + (m_readPtr - m_buffer);
}
if(m_readPtr > m_writePtr) {
m_freeSpace = m_readPtr - m_writePtr;
}
xSemaphoreGiveRecursive(mutex_buffer);
return m_freeSpace;
}
size_t AudioBuffer::writeSpace() {
xSemaphoreTakeRecursive(mutex_buffer, 3 * configTICK_RATE_HZ);
if(m_readPtr == m_writePtr) {
if(m_f_isEmpty == true) m_writeSpace = m_endPtr - m_writePtr;
else m_writeSpace = 0;
}
if(m_readPtr < m_writePtr) {
m_writeSpace = m_endPtr - m_writePtr;
}
if(m_readPtr > m_writePtr) {
m_writeSpace = m_readPtr - m_writePtr;
}
xSemaphoreGiveRecursive(mutex_buffer);
return m_writeSpace;
}
size_t AudioBuffer::bufferFilled() {
xSemaphoreTakeRecursive(mutex_buffer, 3 * configTICK_RATE_HZ);
if(m_readPtr == m_writePtr) {
if(m_f_isEmpty == true) m_dataLength = 0;
else m_dataLength = m_buffSize;
}
if(m_readPtr < m_writePtr) {
m_dataLength = m_writePtr - m_readPtr;
}
if(m_readPtr > m_writePtr) {
m_dataLength = (m_endPtr - m_readPtr) + (m_writePtr - m_buffer);
}
xSemaphoreGiveRecursive(mutex_buffer);
return m_dataLength;
}
size_t AudioBuffer::getMaxAvailableBytes() {
xSemaphoreTakeRecursive(mutex_buffer, 3 * configTICK_RATE_HZ);
if(m_readPtr == m_writePtr) {
// if(m_f_start)m_dataLength = 0;
if(m_f_isEmpty == true) m_dataLength = 0;
else m_dataLength = (m_endPtr - m_readPtr);
}
if(m_readPtr < m_writePtr) {
m_dataLength = m_writePtr - m_readPtr;
}
if(m_readPtr > m_writePtr) {
m_dataLength = (m_endPtr - m_readPtr);
}
xSemaphoreGiveRecursive(mutex_buffer);
return m_dataLength;
}
void AudioBuffer::bytesWritten(size_t bw) {
if(!bw) return;
xSemaphoreTakeRecursive(mutex_buffer, 3 * configTICK_RATE_HZ);
m_writePtr += bw;
if(m_writePtr == m_endPtr) { m_writePtr = m_buffer; }
if(m_writePtr > m_endPtr) log_e("m_writePtr %i, m_endPtr %i", m_writePtr, m_endPtr);
m_f_isEmpty = false;
xSemaphoreGiveRecursive(mutex_buffer);
}
void AudioBuffer::bytesWasRead(size_t br) {
if(!br) return;
xSemaphoreTakeRecursive(mutex_buffer, 3 * configTICK_RATE_HZ);
m_readPtr += br;
if(m_readPtr >= m_endPtr) {
size_t tmp = m_readPtr - m_endPtr;
m_readPtr = m_buffer + tmp;
}
if(m_readPtr == m_writePtr) m_f_isEmpty = true;
xSemaphoreGiveRecursive(mutex_buffer);
}
uint8_t* AudioBuffer::getWritePtr() { return m_writePtr; }
uint8_t* AudioBuffer::getReadPtr() {
xSemaphoreTakeRecursive(mutex_buffer, 3 * configTICK_RATE_HZ);
int32_t len = m_endPtr - m_readPtr;
if(len < m_maxBlockSize) { // be sure the last frame is completed
memcpy(m_endPtr, m_buffer, m_maxBlockSize - (len)); // cpy from m_buffer to m_endPtr with len
}
xSemaphoreGiveRecursive(mutex_buffer);
return m_readPtr;
}
void AudioBuffer::resetBuffer() {
m_writePtr = m_buffer;
m_readPtr = m_buffer;
m_endPtr = m_buffer + m_buffSize;
m_f_isEmpty = true;
vSemaphoreDelete(mutex_buffer);
mutex_buffer = xSemaphoreCreateRecursiveMutex(); // free semaphore is it set
}
uint32_t AudioBuffer::getWritePos() { return m_writePtr - m_buffer; }
uint32_t AudioBuffer::getReadPos() { return m_readPtr - m_buffer; }
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
// clang-format off
Audio::Audio(bool internalDAC /* = false */, uint8_t channelEnabled /* = I2S_SLOT_MODE_STEREO */, uint8_t i2sPort) {
mutex_playAudioData = xSemaphoreCreateMutex();
#ifdef AUDIO_LOG
m_f_Log = true;
#endif
#define AUDIO_INFO(...) { snprintf(m_ibuff, m_ibuffSize, __VA_ARGS__); if(audio_info) audio_info(m_ibuff); }
m_f_psramFound = psramInit();
if(m_f_psramFound) m_chbufSize = 4096; else m_chbufSize = 512 + 64;
if(m_f_psramFound) m_ibuffSize = 4096; else m_ibuffSize = 512 + 64;
m_lastHost = (char*) x_ps_malloc(2048);
m_outBuff = (int16_t*)x_ps_malloc(m_outbuffSize * sizeof(int16_t));
m_chbuf = (char*) x_ps_malloc(m_chbufSize);
m_ibuff = (char*) x_ps_malloc(m_ibuffSize);
if(!m_chbuf || !m_lastHost || !m_outBuff || !m_ibuff) log_e("oom");
clientsecure.setInsecure();
m_f_channelEnabled = channelEnabled;
m_f_internalDAC = internalDAC;
m_i2s_num = i2sPort; // i2s port number
// -------- I2S configuration -------------------------------------------------------------------------------------------
#if ESP_IDF_VERSION_MAJOR == 5
m_i2s_chan_cfg.id = (i2s_port_t)m_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 = 32; // number of DMA buffer
m_i2s_chan_cfg.dma_frame_num = 256; // 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
i2s_new_channel(&m_i2s_chan_cfg, &m_i2s_tx_handle, NULL);
m_i2s_std_cfg.slot_cfg = I2S_STD_PHILIPS_SLOT_DEFAULT_CONFIG(I2S_DATA_BIT_WIDTH_16BIT, 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 = 44100;
m_i2s_std_cfg.clk_cfg.clk_src = I2S_CLK_SRC_PLL_160M; // Select PLL_F160M as the default source clock
m_i2s_std_cfg.clk_cfg.mclk_multiple = I2S_MCLK_MULTIPLE_512; // mclk = sample_rate * 256
i2s_channel_init_std_mode(m_i2s_tx_handle, &m_i2s_std_cfg);
I2Sstart(m_i2s_num);
m_sampleRate = 44100;
if (internalDAC) {
#ifdef CONFIG_IDF_TARGET_ESP32 // ESP32S3 has no DAC
printf("internal DAC is not supported");
// no support in V5 ???
#endif
}
#else
m_i2s_config.sample_rate = 44100;
m_i2s_config.bits_per_sample = I2S_BITS_PER_SAMPLE_16BIT;
m_i2s_config.channel_format = I2S_CHANNEL_FMT_RIGHT_LEFT;
m_i2s_config.intr_alloc_flags = ESP_INTR_FLAG_LEVEL1; // interrupt priority
m_i2s_config.dma_buf_count = 16;
m_i2s_config.dma_buf_len = 512;
m_i2s_config.use_apll = APLL_DISABLE;
m_i2s_config.tx_desc_auto_clear = true;
m_i2s_config.fixed_mclk = true;
m_i2s_config.mclk_multiple = I2S_MCLK_MULTIPLE_128;
if (internalDAC) {
#ifdef CONFIG_IDF_TARGET_ESP32 // ESP32S3 has no DAC
printf("internal DAC");
m_i2s_config.mode = (i2s_mode_t)(I2S_MODE_MASTER | I2S_MODE_TX | I2S_MODE_DAC_BUILT_IN );
m_i2s_config.communication_format = (i2s_comm_format_t)(I2S_COMM_FORMAT_STAND_MSB); // vers >= 2.0.5
i2s_driver_install((i2s_port_t)m_i2s_num, &m_i2s_config, 0, NULL);
i2s_set_dac_mode((i2s_dac_mode_t)m_f_channelEnabled);
if(m_f_channelEnabled != I2S_DAC_CHANNEL_BOTH_EN) {
m_f_forceMono = true;
}
#endif
}
else {
m_i2s_config.mode = (i2s_mode_t)(I2S_MODE_MASTER | I2S_MODE_TX);
m_i2s_config.communication_format = (i2s_comm_format_t)(I2S_COMM_FORMAT_STAND_I2S); // Arduino vers. > 2.0.0
i2s_driver_install((i2s_port_t)m_i2s_num, &m_i2s_config, 0, NULL);
m_f_forceMono = false;
}
i2s_zero_dma_buffer((i2s_port_t) m_i2s_num);
#endif // ESP_IDF_VERSION_MAJOR == 5
for(int i = 0; i < 3; i++) {
m_filter[i].a0 = 1;
m_filter[i].a1 = 0;
m_filter[i].a2 = 0;
m_filter[i].b1 = 0;
m_filter[i].b2 = 0;
}
computeLimit(); // first init, vol = 21, vol_steps = 21
startAudioTask();
}
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
Audio::~Audio() {
// I2Sstop(m_i2s_num);
// InBuff.~AudioBuffer(); #215 the AudioBuffer is automatically destroyed by the destructor
setDefaults();
if(m_playlistBuff) {
free(m_playlistBuff);
m_playlistBuff = NULL;
}
#if ESP_IDF_VERSION_MAJOR == 5
i2s_channel_disable(m_i2s_tx_handle);
i2s_del_channel(m_i2s_tx_handle);
#else
i2s_driver_uninstall((i2s_port_t)m_i2s_num); // #215 free I2S buffer
#endif
if(m_chbuf) {free(m_chbuf); m_chbuf = NULL;}
if(m_lastHost) {free(m_lastHost); m_lastHost = NULL;}
if(m_outBuff) {free(m_outBuff); m_outBuff = NULL; }
if(m_ibuff) {free(m_ibuff); m_ibuff = NULL;}
if(m_lastM3U8host){free(m_lastM3U8host); m_lastM3U8host = NULL;}
stopAudioTask();
vSemaphoreDelete(mutex_playAudioData);
}
// clang-format on
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
void Audio::setBufsize(int rambuf_sz, int psrambuf_sz) {
if(InBuff.isInitialized()) {
log_e("Audio::setBufsize must not be called after audio is initialized");
return;
}
InBuff.setBufsize(rambuf_sz, psrambuf_sz);
};
void Audio::initInBuff() {
if(!InBuff.isInitialized()) {
size_t size = InBuff.init();
if(size > 0) { AUDIO_INFO("PSRAM %sfound, inputBufferSize: %u bytes", InBuff.havePSRAM() ? "" : "not ", size - 1); }
}
changeMaxBlockSize(1600); // default size mp3 or aac
}
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
esp_err_t Audio::I2Sstart(uint8_t i2s_num) {
#if ESP_IDF_VERSION_MAJOR == 5
return i2s_channel_enable(m_i2s_tx_handle);
#else
// It is not necessary to call this function after i2s_driver_install() (it is started automatically),
// however it is necessary to call it after i2s_stop()
return i2s_start((i2s_port_t)i2s_num);
#endif
}
esp_err_t Audio::I2Sstop(uint8_t i2s_num) {
#if ESP_IDF_VERSION_MAJOR == 5
return i2s_channel_disable(m_i2s_tx_handle);
#else
return i2s_stop((i2s_port_t)i2s_num);
#endif
}
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
void Audio::setDefaults() {
stopSong();
initInBuff(); // initialize InputBuffer if not already done
InBuff.resetBuffer();
MP3Decoder_FreeBuffers();
FLACDecoder_FreeBuffers();
AACDecoder_FreeBuffers();
OPUSDecoder_FreeBuffers();
VORBISDecoder_FreeBuffers();
memset(m_outBuff, 0, m_outbuffSize * sizeof(int16_t)); // Clear OutputBuffer
if(m_playlistBuff) {
free(m_playlistBuff);
m_playlistBuff = NULL;
} // free if stream is not m3u8
vector_clear_and_shrink(m_playlistURL);
vector_clear_and_shrink(m_playlistContent);
m_hashQueue.clear();
m_hashQueue.shrink_to_fit(); // uint32_t vector
client.stop();
// client.clear(); // delete all leftovers in the receive buffer
clientsecure.stop();
// clientsecure.clear(); // delete all leftovers in the receive buffer
_client = static_cast<WiFiClient*>(&client); /* default to *something* so that no NULL deref can happen */
ts_parsePacket(0, 0, 0); // reset ts routine
if(m_lastM3U8host) {
free(m_lastM3U8host);
m_lastM3U8host = NULL;
}
AUDIO_INFO("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_f_firstCurTimeCall = true; // InitSequence for computeAudioTime
m_f_firstM3U8call = true; // InitSequence for parsePlaylist_M3U8
m_f_firstPlayCall = true; // InitSequence for playAudioData
m_f_running = false;
m_f_loop = false; // Set if audio file should loop
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_m4aID3dataAreRead = false;
m_f_stream = false;
m_f_eof = false;
m_f_ID3v1TagFound = false;
m_f_lockInBuffer = false;
m_streamType = ST_NONE;
m_codec = CODEC_NONE;
m_playlistFormat = FORMAT_NONE;
m_dataMode = AUDIO_NONE;
m_audioCurrentTime = 0; // Reset playtimer
m_audioFileDuration = 0;
m_audioDataStart = 0;
m_audioDataSize = 0;
m_avr_bitrate = 0; // the same as m_bitrate if CBR, median if VBR
m_bitRate = 0; // Bitrate still unknown
m_bytesNotDecoded = 0; // counts all not decodable bytes
m_chunkcount = 0; // for chunked streams
m_contentlength = 0; // If Content-Length is known, count it
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_streamTitleHash = 0;
m_fileSize = 0;
m_ID3Size = 0;
m_haveNewFilePos = 0;
m_validSamples = 0;
m_M4A_chConfig = 0;
m_M4A_objectType = 0;
m_M4A_sampleRate = 0;
m_sumBytesDecoded = 0;
m_vuLeft = m_vuRight = 0; // #835
}
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
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.
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, "shimmer", "mp3", "1");
*/
bool Audio::openai_speech(const String& api_key, const String& model, const String& input, const String& voice, const String& response_format, const String& speed) {
char host[] = "api.openai.com";
char path[] = "/v1/audio/speech";
if (input == "") {
AUDIO_INFO("input text is empty");
stopSong();
return false;
}
xSemaphoreTakeRecursive(mutex_playAudioData, 0.3 * configTICK_RATE_HZ);
setDefaults();
m_f_ssl = true;
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 {
input_clean += c;
}
}
String post_body = "{"
"\"model\": \"" + model + "\"," +
"\"input\": \"" + input_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
+ "Host: " + String(host) + "\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: keep-alive\r\n" + "\r\n"
+ post_body + "\r\n"
;
bool res = true;
int port = 443;
_client = static_cast<WiFiClient*>(&clientsecure);
uint32_t t = millis();
AUDIO_INFO("Connect to: \"%s\"", host);
res = _client->connect(host, port, m_timeout_ms_ssl);
if (res) {
uint32_t dt = millis() - t;
strcpy(m_lastHost, host);
AUDIO_INFO("%s has been established in %lu ms, free Heap: %lu bytes", "SSL", (long unsigned int) dt, (long unsigned int) ESP.getFreeHeap());
m_f_running = true;
}
m_expectedCodec = CODEC_NONE;
m_expectedPlsFmt = FORMAT_NONE;
if (res) {
_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_streamType = ST_WEBSTREAM;
} else {
AUDIO_INFO("Request %s failed!", host);
m_lastHost[0] = 0;
}
xSemaphoreGiveRecursive(mutex_playAudioData);
return res;
}
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
bool Audio::connecttohost(const char* host, const char* user, const char* pwd) { // user and pwd for authentification only, can be empty
bool res = false; // return value
uint16_t lenHost = 0; // length of hostname
uint16_t port = 0; // port number
uint16_t authLen = 0; // length of authorization
int16_t pos_slash = 0; // position of "/" in hostname
int16_t pos_colon = 0; // position of ":" in hostname
int16_t pos_ampersand = 0; // position of "&" in hostname
uint32_t timestamp = 0; // timeout surveillance
uint16_t hostwoext_begin = 0;
char* authorization = NULL; // authorization
char* rqh = NULL; // request header
char* toEncode = NULL; // temporary memory for base64 encoding
char* h_host = NULL;
// 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->.......
xSemaphoreTakeRecursive(mutex_playAudioData, 0.3 * configTICK_RATE_HZ);
if (host == NULL) { AUDIO_INFO("Hostaddress is empty"); stopSong(); goto exit;}
if (strlen(host) > 2048) { AUDIO_INFO("Hostaddress is too long"); stopSong(); goto exit;} // max length in Chrome DevTools
h_host = x_ps_strdup(host);
trim(h_host); // remove leading and trailing spaces
lenHost = strlen(h_host);
if(!startsWith(h_host, "http")) { AUDIO_INFO("Hostaddress is not valid"); stopSong(); goto exit;}
if(startsWith(h_host, "https")) {m_f_ssl = true; hostwoext_begin = 8; port = 443;}
else {m_f_ssl = false; hostwoext_begin = 7; port = 80;}
// In the URL there may be an extension, like noisefm.ru:8000/play.m3u&t=.m3u
pos_slash = indexOf(h_host, "/", 10); // position of "/" in hostname
pos_colon = indexOf(h_host, ":", 10); if(isalpha(host[pos_colon + 1])) pos_colon = -1; // no portnumber follows
pos_ampersand = indexOf(h_host, "&", 10); // position of "&" in hostname
if(pos_slash > 0) h_host[pos_slash] = '\0';
if((pos_colon > 0) && ((pos_ampersand == -1) || (pos_ampersand > pos_colon))) {
port = atoi(host + pos_colon + 1); // Get portnumber as integer
h_host[pos_colon] = '\0';
}
// optional basic authorization
if(strlen(user) > 0 && strlen(pwd) > 0) {
authLen = strlen(user) + strlen(pwd);
authorization = x_ps_calloc(base64_encode_expected_len(authLen + 1), 1);
if(!authorization) {AUDIO_INFO("out of memory"); stopSong(); goto exit;}
toEncode = x_ps_calloc(authLen + 4, 1);
if(!toEncode) {AUDIO_INFO("out of memory"); stopSong(); goto exit;}
strcpy(toEncode, user);
strcat(toEncode, ":");
strcat(toEncode, pwd);
b64encode((const char*)toEncode, strlen(toEncode), authorization);
}
else{
authorization = strdup("");
}
setDefaults();
rqh = x_ps_calloc(lenHost + strlen(authorization) + 200, 1); // http request header
if(!rqh) {AUDIO_INFO("out of memory"); stopSong(); goto exit;}
strcat(rqh, "GET /");
if(pos_slash > 0){ strcat(rqh, h_host + pos_slash + 1);}
strcat(rqh, " HTTP/1.1\r\n");
strcat(rqh, "Host: ");
strcat(rqh, h_host + hostwoext_begin);
strcat(rqh, "\r\n");
strcat(rqh, "Icy-MetaData:1\r\n");
strcat(rqh, "Icy-MetaData:2\r\n");
strcat(rqh, "User-Agent: ESP32 audioI2S\r\n");
if(authLen > 0) { strcat(rqh, "Authorization: Basic ");
strcat(rqh, authorization);
strcat(rqh, "\r\n"); }
strcat(rqh, "Accept-Encoding: identity;q=1,*;q=0\r\n");
strcat(rqh, "Connection: keep-alive\r\n\r\n");
if(m_f_ssl) { _client = static_cast<WiFiClient*>(&clientsecure);}
else { _client = static_cast<WiFiClient*>(&client); }
timestamp = millis();
_client->setTimeout(m_f_ssl ? m_timeout_ms_ssl : m_timeout_ms);
AUDIO_INFO("connect to: \"%s\" on port %d path \"/%s\"", h_host + hostwoext_begin, port, h_host + pos_slash + 1);
res = _client->connect(h_host + hostwoext_begin, port);
if(pos_slash > 0) h_host[pos_slash] = '/';
if(pos_colon > 0) h_host[pos_colon] = ':';
strcpy(m_lastHost, h_host);
if(res) {
uint32_t dt = millis() - timestamp;
strcpy(m_lastHost, h_host);
AUDIO_INFO("%s has been established in %lu ms, free Heap: %lu bytes", m_f_ssl ? "SSL" : "Connection", (long unsigned int)dt, (long unsigned int)ESP.getFreeHeap());
m_f_running = true;
}
m_expectedCodec = CODEC_NONE;
m_expectedPlsFmt = FORMAT_NONE;
if(res) {
_client->print(rqh);
if(endsWith(h_host, ".mp3" )) m_expectedCodec = CODEC_MP3;
if(endsWith(h_host, ".aac" )) m_expectedCodec = CODEC_AAC;
if(endsWith(h_host, ".wav" )) m_expectedCodec = CODEC_WAV;
if(endsWith(h_host, ".m4a" )) m_expectedCodec = CODEC_M4A;
if(endsWith(h_host, ".ogg" )) m_expectedCodec = CODEC_OGG;
if(endsWith(h_host, ".flac")) m_expectedCodec = CODEC_FLAC;
if(endsWith(h_host, "-flac")) m_expectedCodec = CODEC_FLAC;
if(endsWith(h_host, ".opus")) m_expectedCodec = CODEC_OPUS;
if(endsWith(h_host, "/opus")) m_expectedCodec = CODEC_OPUS;
if(endsWith(h_host, ".asx" )) m_expectedPlsFmt = FORMAT_ASX;
if(endsWith(h_host, ".m3u" )) m_expectedPlsFmt = FORMAT_M3U;
if(endsWith(h_host, ".pls" )) m_expectedPlsFmt = FORMAT_PLS;
if(endsWith(h_host, ".m3u8")) {
m_expectedPlsFmt = FORMAT_M3U8;
if(audio_lasthost) audio_lasthost(host);
}
m_dataMode = HTTP_RESPONSE_HEADER; // Handle header
m_streamType = ST_WEBSTREAM;
}
else {
AUDIO_INFO("Request %s failed!", host);
if(audio_showstation) audio_showstation("");
if(audio_showstreamtitle) audio_showstreamtitle("");
if(audio_icydescription) audio_icydescription("");
if(audio_icyurl) audio_icyurl("");
m_lastHost[0] = 0;
}
exit:
xSemaphoreGiveRecursive(mutex_playAudioData);
free(h_host);
free(rqh);
free(authorization);
free(toEncode);
return res;
}
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
bool Audio::httpPrint(const char* host) {
// user and pwd for authentification only, can be empty
if(host == NULL) {
AUDIO_INFO("Hostaddress is empty");
stopSong();
return false;
}
char* h_host = NULL; // pointer of l_host without http:// or https://
if(startsWith(host, "https")) m_f_ssl = true;
else m_f_ssl = false;
if(m_f_ssl) h_host = strdup(host + 8);
else h_host = strdup(host + 7);
int16_t pos_slash; // position of "/" in hostname
int16_t pos_colon; // position of ":" in hostname
int16_t pos_ampersand; // position of "&" in hostname
uint16_t port = 80; // port number
// In the URL there may be an extension, like noisefm.ru:8000/play.m3u&t=.m3u
pos_slash = indexOf(h_host, "/", 0);
pos_colon = indexOf(h_host, ":", 0);
if(isalpha(h_host[pos_colon + 1])) pos_colon = -1; // no portnumber follows
pos_ampersand = indexOf(h_host, "&", 0);
char* hostwoext = NULL; // "skonto.ls.lv:8002" in "skonto.ls.lv:8002/mp3"
char* extension = NULL; // "/mp3" in "skonto.ls.lv:8002/mp3"
if(pos_slash > 1) {
hostwoext = (char*)malloc(pos_slash + 1);
memcpy(hostwoext, h_host, pos_slash);
hostwoext[pos_slash] = '\0';
extension = urlencode(h_host + pos_slash, true);
}
else { // url has no extension
hostwoext = strdup(h_host);
extension = strdup("/");
}
if((pos_colon >= 0) && ((pos_ampersand == -1) || (pos_ampersand > pos_colon))) {
port = atoi(h_host + pos_colon + 1); // Get portnumber as integer
hostwoext[pos_colon] = '\0'; // Host without portnumber
}
AUDIO_INFO("connect to: \"%s\"", host);
char rqh[strlen(h_host) + 200]; // http request header
rqh[0] = '\0';
strcat(rqh, "GET ");
strcat(rqh, extension);
strcat(rqh, " HTTP/1.1\r\n");
strcat(rqh, "Host: ");
strcat(rqh, hostwoext);
strcat(rqh, "\r\n");
strcat(rqh, "Accept-Encoding: identity;q=1,*;q=0\r\n");
// strcat(rqh, "User-Agent: Mozilla/5.0\r\n"); #363
strcat(rqh, "Connection: keep-alive\r\n\r\n");
if(m_f_ssl) {
_client = static_cast<WiFiClient*>(&clientsecure);
if(port == 80) port = 443;
}
else { _client = static_cast<WiFiClient*>(&client); }
if(!_client->connected()) {
AUDIO_INFO("The host has disconnected, reconnecting");
if(!_client->connect(hostwoext, port)) {
log_e("connection lost");
stopSong();
return false;
}
}
_client->print(rqh);
if(endsWith(extension, ".mp3")) m_expectedCodec = CODEC_MP3;
if(endsWith(extension, ".aac")) m_expectedCodec = CODEC_AAC;
if(endsWith(extension, ".wav")) m_expectedCodec = CODEC_WAV;
if(endsWith(extension, ".m4a")) m_expectedCodec = CODEC_M4A;
if(endsWith(extension, ".flac")) m_expectedCodec = CODEC_FLAC;
if(endsWith(extension, ".asx")) m_expectedPlsFmt = FORMAT_ASX;
if(endsWith(extension, ".m3u")) m_expectedPlsFmt = FORMAT_M3U;
if(indexOf( extension, ".m3u8") >= 0) m_expectedPlsFmt = FORMAT_M3U8;
if(endsWith(extension, ".pls")) m_expectedPlsFmt = FORMAT_PLS;
m_dataMode = HTTP_RESPONSE_HEADER; // Handle header
m_streamType = ST_WEBSTREAM;
m_contentlength = 0;
m_f_chunked = false;
free(hostwoext);
free(extension);
free(h_host);
return true;
}
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
bool Audio::setFileLoop(bool input) {
if(m_codec == CODEC_M4A) return 0;
m_f_loop = input;
return input;
}
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
// clang-format off
void Audio::UTF8toASCII(char* str) {
const uint8_t ascii[60] = {
//129, 130, 131, 132, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148 // UTF8(C3)
// Ä Å Æ Ç É Ñ // CHAR
000, 000, 000, 142, 143, 146, 128, 000, 144, 000, 000, 000, 000, 000, 000, 000, 165, 000, 000, 000, // ASCII
//149, 150, 151, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168
// Ö Ü ß à ä å æ è
000, 153, 000, 000, 000, 000, 000, 154, 000, 000, 225, 133, 000, 000, 000, 132, 134, 145, 000, 138,
//169, 170, 171, 172. 173. 174. 175, 176, 177, 179, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188
// ê ë ì î ï ñ ò ô ö ù û ü
000, 136, 137, 141, 000, 140, 139, 000, 164, 149, 000, 147, 000, 148, 000, 000, 151, 000, 150, 129};
uint16_t i = 0, j = 0, s = 0;
bool f_C3_seen = false;
while(str[i] != 0) { // convert UTF8 to ASCII
if(str[i] == 195) { // C3
i++;
f_C3_seen = true;
continue;
}
str[j] = str[i];
if(str[j] > 128 && str[j] < 189 && f_C3_seen == true) {
s = ascii[str[j] - 129];
if(s != 0) str[j] = s; // found a related ASCII sign
f_C3_seen = false;
}
i++;
j++;
}
str[j] = 0;
}
// clang-format on
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
bool Audio::connecttoFS(fs::FS& fs, const char* path, int32_t fileStartPos) {
xSemaphoreTakeRecursive(mutex_playAudioData, 0.3 * configTICK_RATE_HZ);
bool res = false;
int16_t dotPos;
char* audioPath = NULL;
m_fileStartPos = fileStartPos;
if(!path) {printProcessLog(AUDIOLOG_PATH_IS_NULL); goto exit;} // guard
dotPos = lastIndexOf(path, ".");
if(dotPos == -1) {AUDIO_INFO("No file extension found"); goto exit;} // guard
setDefaults(); // free buffers an set defaults
if(endsWith(path, ".mp3")) m_codec = CODEC_MP3;
if(endsWith(path, ".m4a")) m_codec = CODEC_M4A;
if(endsWith(path, ".aac")) m_codec = CODEC_AAC;
if(endsWith(path, ".wav")) m_codec = CODEC_WAV;
if(endsWith(path, ".flac")) m_codec = CODEC_FLAC;
if(endsWith(path, ".opus")) m_codec = CODEC_OPUS;
if(endsWith(path, ".ogg")) m_codec = CODEC_OGG;
if(endsWith(path, ".oga")) m_codec = CODEC_OGG;
if(m_codec == CODEC_NONE) {AUDIO_INFO("The %s format is not supported", path + dotPos); goto exit;} // guard
audioPath = (char *)x_ps_calloc(strlen(path) + 2, sizeof(char));
if(!audioPath){printProcessLog(AUDIOLOG_OUT_OF_MEMORY); goto exit;};
if(path[0] != '/')audioPath[0] = '/';
strcat(audioPath, path);
if(!fs.exists(audioPath)) {printProcessLog(AUDIOLOG_FILE_NOT_FOUND, audioPath); goto exit;}
AUDIO_INFO("Reading file: \"%s\"", audioPath);
audiofile = fs.open(audioPath);
m_dataMode = AUDIO_LOCALFILE;
m_fileSize = audiofile.size();
res = initializeDecoder();
if(res) m_f_running = true;
else audiofile.close();
exit:
free(audioPath);
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";
uint16_t speechLen = strlen(speech);
uint16_t speechBuffLen = speechLen + 300;
char* speechBuff = (char*)malloc(speechBuffLen);
if(!speechBuff) {
log_e("out of memory");
xSemaphoreGiveRecursive(mutex_playAudioData);
return false;
}
memcpy(speechBuff, speech, speechLen);
speechBuff[speechLen] = '\0';
char* urlStr = urlencode(speechBuff, false); // percent encoding
if(!urlStr) {
log_e("out of memory");
xSemaphoreGiveRecursive(mutex_playAudioData);
return false;
}
char resp[strlen(urlStr) + 200] = "";
strcat(resp, "GET ");
strcat(resp, path);
strcat(resp, "?ie=UTF-8&tl=");
strcat(resp, lang);
strcat(resp, "&client=tw-ob&q=");
strcat(resp, urlStr);
strcat(resp, " HTTP/1.1\r\n");
strcat(resp, "Host: ");
strcat(resp, host);
strcat(resp, "\r\n");
strcat(resp, "User-Agent: Mozilla/5.0 \r\n");
strcat(resp, "Accept-Encoding: identity\r\n");
strcat(resp, "Accept: text/html\r\n");
strcat(resp, "Connection: close\r\n\r\n");
if(speechBuff) {
free(speechBuff);
speechBuff = NULL;
}
if(urlStr) {
free(urlStr);
urlStr = NULL;
}
_client = static_cast<WiFiClient*>(&client);
AUDIO_INFO("connect to \"%s\"", host);
if(!_client->connect(host, 80)) {
log_e("Connection failed");
xSemaphoreGiveRecursive(mutex_playAudioData);
return false;
}
_client->print(resp);
m_streamType = ST_WEBFILE;
m_f_running = true;
m_f_ssl = false;
m_f_tts = true;
m_dataMode = HTTP_RESPONSE_HEADER;
xSemaphoreGiveRecursive(mutex_playAudioData);
return true;
}
//------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
void Audio::showID3Tag(const char* tag, const char* value) {
m_chbuf[0] = 0;
// V2.2
if(!strcmp(tag, "CNT")) sprintf(m_chbuf, "Play counter: %s", value);
// if(!strcmp(tag, "COM")) sprintf(m_chbuf, "Comments: %s", value);
if(!strcmp(tag, "CRA")) sprintf(m_chbuf, "Audio encryption: %s", value);
if(!strcmp(tag, "CRM")) sprintf(m_chbuf, "Encrypted meta frame: %s", value);
if(!strcmp(tag, "ETC")) sprintf(m_chbuf, "Event timing codes: %s", value);
if(!strcmp(tag, "EQU")) sprintf(m_chbuf, "Equalization: %s", value);
if(!strcmp(tag, "IPL")) sprintf(m_chbuf, "Involved people list: %s", value);
if(!strcmp(tag, "PIC")) sprintf(m_chbuf, "Attached picture: %s", value);
if(!strcmp(tag, "SLT")) sprintf(m_chbuf, "Synchronized lyric/text: %s", value);
if(!strcmp(tag, "TAL")) sprintf(m_chbuf, "Album/Movie/Show title: %s", value);
if(!strcmp(tag, "TBP")) sprintf(m_chbuf, "BPM (Beats Per Minute): %s", value);
if(!strcmp(tag, "TCM")) sprintf(m_chbuf, "Composer: %s", value);
if(!strcmp(tag, "TCO")) sprintf(m_chbuf, "Content type: %s", value);
if(!strcmp(tag, "TCR")) sprintf(m_chbuf, "Copyright message: %s", value);
if(!strcmp(tag, "TDA")) sprintf(m_chbuf, "Date: %s", value);
if(!strcmp(tag, "TDY")) sprintf(m_chbuf, "Playlist delay: %s", value);
if(!strcmp(tag, "TEN")) sprintf(m_chbuf, "Encoded by: %s", value);
if(!strcmp(tag, "TFT")) sprintf(m_chbuf, "File type: %s", value);
if(!strcmp(tag, "TIM")) sprintf(m_chbuf, "Time: %s", value);
if(!strcmp(tag, "TKE")) sprintf(m_chbuf, "Initial key: %s", value);
if(!strcmp(tag, "TLA")) sprintf(m_chbuf, "Language(s): %s", value);
if(!strcmp(tag, "TLE")) sprintf(m_chbuf, "Length: %s", value);
if(!strcmp(tag, "TMT")) sprintf(m_chbuf, "Media type: %s", value);
if(!strcmp(tag, "TOA")) sprintf(m_chbuf, "Original artist(s)/performer(s): %s", value);
if(!strcmp(tag, "TOF")) sprintf(m_chbuf, "Original filename: %s", value);
if(!strcmp(tag, "TOL")) sprintf(m_chbuf, "Original Lyricist(s)/text writer(s): %s", value);
if(!strcmp(tag, "TOR")) sprintf(m_chbuf, "Original release year: %s", value);
if(!strcmp(tag, "TOT")) sprintf(m_chbuf, "Original album/Movie/Show title: %s", value);
if(!strcmp(tag, "TP1")) sprintf(m_chbuf, "Lead artist(s)/Lead performer(s)/Soloist(s)/Performing group: %s", value);
if(!strcmp(tag, "TP2")) sprintf(m_chbuf, "Band/Orchestra/Accompaniment: %s", value);
if(!strcmp(tag, "TP3")) sprintf(m_chbuf, "Conductor/Performer refinement: %s", value);
if(!strcmp(tag, "TP4")) sprintf(m_chbuf, "Interpreted, remixed, or otherwise modified by: %s", value);
if(!strcmp(tag, "TPA")) sprintf(m_chbuf, "Part of a set: %s", value);
if(!strcmp(tag, "TPB")) sprintf(m_chbuf, "Publisher: %s", value);
if(!strcmp(tag, "TRC")) sprintf(m_chbuf, "ISRC (International Standard Recording Code): %s", value);
if(!strcmp(tag, "TRD")) sprintf(m_chbuf, "Recording dates: %s", value);
if(!strcmp(tag, "TRK")) sprintf(m_chbuf, "Track number/Position in set: %s", value);
if(!strcmp(tag, "TSI")) sprintf(m_chbuf, "Size: %s", value);
if(!strcmp(tag, "TSS")) sprintf(m_chbuf, "Software/hardware and settings used for encoding: %s", value);
if(!strcmp(tag, "TT1")) sprintf(m_chbuf, "Content group description: %s", value);
if(!strcmp(tag, "TT2")) sprintf(m_chbuf, "Title/Songname/Content description: %s", value);
if(!strcmp(tag, "TT3")) sprintf(m_chbuf, "Subtitle/Description refinement: %s", value);
if(!strcmp(tag, "TXT")) sprintf(m_chbuf, "Lyricist/text writer: %s", value);
if(!strcmp(tag, "TXX")) sprintf(m_chbuf, "User defined text information frame: %s", value);
if(!strcmp(tag, "TYE")) sprintf(m_chbuf, "Year: %s", value);
if(!strcmp(tag, "UFI")) sprintf(m_chbuf, "Unique file identifier: %s", value);
if(!strcmp(tag, "ULT")) sprintf(m_chbuf, "Unsychronized lyric/text transcription: %s", value);
if(!strcmp(tag, "WAF")) sprintf(m_chbuf, "Official audio file webpage: %s", value);
if(!strcmp(tag, "WAR")) sprintf(m_chbuf, "Official artist/performer webpage: %s", value);
if(!strcmp(tag, "WAS")) sprintf(m_chbuf, "Official audio source webpage: %s", value);
if(!strcmp(tag, "WCM")) sprintf(m_chbuf, "Commercial information: %s", value);
if(!strcmp(tag, "WCP")) sprintf(m_chbuf, "Copyright/Legal information: %s", value);
if(!strcmp(tag, "WPB")) sprintf(m_chbuf, "Publishers official webpage: %s", value);
if(!strcmp(tag, "WXX")) sprintf(m_chbuf, "User defined URL link frame: %s", value);
// V2.3 V2.4 tags
// if(!strcmp(tag, "COMM")) sprintf(m_chbuf, "Comment: %s", value);
if(!strcmp(tag, "OWNE")) sprintf(m_chbuf, "Ownership: %s", value);
// if(!strcmp(tag, "PRIV")) sprintf(m_chbuf, "Private: %s", value);
if(!strcmp(tag, "SYLT")) sprintf(m_chbuf, "SynLyrics: %s", value);
if(!strcmp(tag, "TALB")) sprintf(m_chbuf, "Album: %s", value);
if(!strcmp(tag, "TBPM")) sprintf(m_chbuf, "BeatsPerMinute: %s", value);
if(!strcmp(tag, "TCMP")) sprintf(m_chbuf, "Compilation: %s", value);
if(!strcmp(tag, "TCOM")) sprintf(m_chbuf, "Composer: %s", value);
if(!strcmp(tag, "TCON")) sprintf(m_chbuf, "ContentType: %s", value);
if(!strcmp(tag, "TCOP")) sprintf(m_chbuf, "Copyright: %s", value);
if(!strcmp(tag, "TDAT")) sprintf(m_chbuf, "Date: %s", value);
if(!strcmp(tag, "TEXT")) sprintf(m_chbuf, "Lyricist: %s", value);
if(!strcmp(tag, "TIME")) sprintf(m_chbuf, "Time: %s", value);
if(!strcmp(tag, "TIT1")) sprintf(m_chbuf, "Grouping: %s", value);
if(!strcmp(tag, "TIT2")) sprintf(m_chbuf, "Title: %s", value);
if(!strcmp(tag, "TIT3")) sprintf(m_chbuf, "Subtitle: %s", value);
if(!strcmp(tag, "TLAN")) sprintf(m_chbuf, "Language: %s", value);
if(!strcmp(tag, "TLEN")) sprintf(m_chbuf, "Length (ms): %s", value);
if(!strcmp(tag, "TMED")) sprintf(m_chbuf, "Media: %s", value);
if(!strcmp(tag, "TOAL")) sprintf(m_chbuf, "OriginalAlbum: %s", value);
if(!strcmp(tag, "TOPE")) sprintf(m_chbuf, "OriginalArtist: %s", value);
if(!strcmp(tag, "TORY")) sprintf(m_chbuf, "OriginalReleaseYear: %s", value);
if(!strcmp(tag, "TPE1")) sprintf(m_chbuf, "Artist: %s", value);
if(!strcmp(tag, "TPE2")) sprintf(m_chbuf, "Band: %s", value);