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ESP32-audioI2S
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/*
* Audio.h
*
* Created on: Oct 28,2018
*
* Version 2.0.8c
* Updated on: Jan 13,2023
* Author: Wolle (schreibfaul1)
*/
//#define SDFATFS_USED // activate for SdFat
#pragma once
#pragma GCC optimize ("Ofast")
#include <vector>
#include <Arduino.h>
#include <libb64/cencode.h>
#include <esp32-hal-log.h>
#include <SPI.h>
#include <WiFi.h>
#include <WiFiClientSecure.h>
#include <vector>
#include <driver/i2s.h>
#ifdef SDFATFS_USED
#include <SdFat.h> // https://github.com/greiman/SdFat
#else
#include <SD.h>
#include <SD_MMC.h>
#include <SPIFFS.h>
#include <FS.h>
#include <FFat.h>
#endif // SDFATFS_USED
#ifdef SDFATFS_USED
//typedef File32 File;
typedef FsFile File;
namespace fs {
class FS : public SdFat {
public:
bool begin(SdCsPin_t csPin = SS, uint32_t maxSck = SD_SCK_MHZ(25)) { return SdFat::begin(csPin, maxSck); }
};
class SDFATFS : public fs::FS {
public:
// sdcard_type_t cardType();
uint64_t cardSize() {
return totalBytes();
}
uint64_t usedBytes() {
// set SdFatConfig MAINTAIN_FREE_CLUSTER_COUNT non-zero. Then only the first call will take time.
return (uint64_t)(clusterCount() - freeClusterCount()) * (uint64_t)bytesPerCluster();
}
uint64_t totalBytes() {
return (uint64_t)clusterCount() * (uint64_t)bytesPerCluster();
}
};
}
extern fs::SDFATFS SD_SDFAT;
using namespace fs;
#define SD SD_SDFAT
#endif //SDFATFS_USED
using namespace std;
extern __attribute__((weak)) void audio_info(const char*);
extern __attribute__((weak)) void audio_id3data(const char*); //ID3 metadata
extern __attribute__((weak)) void audio_id3image(File& file, const size_t pos, const size_t size); //ID3 metadata image
extern __attribute__((weak)) void audio_eof_mp3(const char*); //end of mp3 file
extern __attribute__((weak)) void audio_showstreamtitle(const char*);
extern __attribute__((weak)) void audio_showstation(const char*);
extern __attribute__((weak)) void audio_bitrate(const char*);
extern __attribute__((weak)) void audio_commercial(const char*);
extern __attribute__((weak)) void audio_icyurl(const char*);
extern __attribute__((weak)) void audio_icydescription(const char*);
extern __attribute__((weak)) void audio_lasthost(const char*);
extern __attribute__((weak)) void audio_eof_speech(const char*);
extern __attribute__((weak)) void audio_eof_stream(const char*); // The webstream comes to an end
extern __attribute__((weak)) void audio_process_extern(int16_t* buff, uint16_t len, bool *continueI2S); // record audiodata or send via BT
extern __attribute__((weak)) void audio_process_i2s(uint32_t* sample, bool *continueI2S); // record audiodata or send via BT
#define AUDIO_INFO(...) {char buff[512 + 64]; sprintf(buff,__VA_ARGS__); if(audio_info) audio_info(buff);}
//----------------------------------------------------------------------------------------------------------------------
class AudioBuffer {
// AudioBuffer will be allocated in PSRAM, If PSRAM not available or has not enough space AudioBuffer will be
// allocated in FlashRAM with reduced size
//
// m_buffer m_readPtr m_writePtr m_endPtr
// | |<------dataLength------->|<------ writeSpace ----->|
// ▼ ▼ ▼ ▼
// ---------------------------------------------------------------------------------------------------------------
// | <--m_buffSize--> | <--m_resBuffSize --> |
// ---------------------------------------------------------------------------------------------------------------
// |<-----freeSpace------->| |<------freeSpace-------->|
//
//
//
// if the space between m_readPtr and buffend < m_resBuffSize copy data from the beginning to resBuff
// so that the mp3/aac/flac frame is always completed
//
// m_buffer m_writePtr m_readPtr m_endPtr
// | |<-------writeSpace------>|<--dataLength-->|
// ▼ ▼ ▼ ▼
// ---------------------------------------------------------------------------------------------------------------
// | <--m_buffSize--> | <--m_resBuffSize --> |
// ---------------------------------------------------------------------------------------------------------------
// |<--- ------dataLength-- ------>|<-------freeSpace------->|
//
//
public:
AudioBuffer(size_t maxBlockSize = 0); // constructor
~AudioBuffer(); // frees the buffer
size_t init(); // set default values
bool isInitialized() { return m_f_init; };
void setBufsize(int ram, int psram);
void changeMaxBlockSize(uint16_t mbs); // is default 1600 for mp3 and aac, set 16384 for FLAC
uint16_t getMaxBlockSize(); // returns maxBlockSize
size_t freeSpace(); // number of free bytes to overwrite
size_t writeSpace(); // space fom writepointer to bufferend
size_t bufferFilled(); // returns the number of filled bytes
void bytesWritten(size_t bw); // update writepointer
void bytesWasRead(size_t br); // update readpointer
uint8_t* getWritePtr(); // returns the current writepointer
uint8_t* getReadPtr(); // returns the current readpointer
uint32_t getWritePos(); // write position relative to the beginning
uint32_t getReadPos(); // read position relative to the beginning
void resetBuffer(); // restore defaults
bool havePSRAM() { return m_f_psram; };
protected:
size_t m_buffSizePSRAM = 300000; // most webstreams limit the advance to 100...300Kbytes
size_t m_buffSizeRAM = 1600 * 5;
size_t m_buffSize = 0;
size_t m_freeSpace = 0;
size_t m_writeSpace = 0;
size_t m_dataLength = 0;
size_t m_resBuffSizeRAM = 1600; // reserved buffspace, >= one mp3 frame
size_t m_resBuffSizePSRAM = 4096 * 4; // reserved buffspace, >= one flac frame
size_t m_maxBlockSize = 1600;
uint8_t* m_buffer = NULL;
uint8_t* m_writePtr = NULL;
uint8_t* m_readPtr = NULL;
uint8_t* m_endPtr = NULL;
bool m_f_start = true;
bool m_f_init = false;
bool m_f_psram = false; // PSRAM is available (and used...)
};
//----------------------------------------------------------------------------------------------------------------------
class Audio : private AudioBuffer{
AudioBuffer InBuff; // instance of input buffer
public:
Audio(bool internalDAC = false, uint8_t channelEnabled = 3, uint8_t i2sPort = I2S_NUM_0); // #99
~Audio();
void setBufsize(int rambuf_sz, int psrambuf_sz);
bool connecttohost(const char* host, const char* user = "", const char* pwd = "");
bool connecttospeech(const char* speech, const char* lang);
bool connecttomarytts(const char* speech, const char* lang, const char* voice);
bool connecttoFS(fs::FS &fs, const char* path, uint32_t resumeFilePos = 0);
bool connecttoSD(const char* path, uint32_t resumeFilePos = 0);
bool setFileLoop(bool input);//TEST loop
void setConnectionTimeout(uint16_t timeout_ms, uint16_t timeout_ms_ssl);
bool setAudioPlayPosition(uint16_t sec);
bool setFilePos(uint32_t pos);
bool audioFileSeek(const float speed);
bool setTimeOffset(int sec);
bool setPinout(uint8_t BCLK, uint8_t LRC, uint8_t DOUT, int8_t DIN = I2S_PIN_NO_CHANGE, int8_t MCK = I2S_PIN_NO_CHANGE);
bool pauseResume();
bool isRunning() {return m_f_running;}
void loop();
uint32_t stopSong();
void forceMono(bool m);
void setBalance(int8_t bal = 0);
void setVolumeSteps(uint8_t steps);
void setVolume(uint8_t vol);
uint8_t getVolume();
uint8_t maxVolume();
uint8_t getI2sPort();
uint32_t getAudioDataStartPos();
uint32_t getFileSize();
uint32_t getFilePos();
uint32_t getSampleRate();
uint8_t getBitsPerSample();
uint8_t getChannels();
uint32_t getBitRate(bool avg = false);
uint32_t getAudioFileDuration();
uint32_t getAudioCurrentTime();
uint32_t getTotalPlayingTime();
esp_err_t i2s_mclk_pin_select(const uint8_t pin);
uint32_t inBufferFilled(); // returns the number of stored bytes in the inputbuffer
uint32_t inBufferFree(); // returns the number of free bytes in the inputbuffer
void setTone(int8_t gainLowPass, int8_t gainBandPass, int8_t gainHighPass);
void setI2SCommFMT_LSB(bool commFMT);
int getCodec() {return m_codec;}
const char *getCodecname() {return codecname[m_codec];}
private:
#ifndef ESP_ARDUINO_VERSION_VAL
#define ESP_ARDUINO_VERSION_MAJOR 0
#define ESP_ARDUINO_VERSION_MINOR 0
#define ESP_ARDUINO_VERSION_PATCH 0
#endif
void UTF8toASCII(char* str);
bool latinToUTF8(char* buff, size_t bufflen);
void setDefaults(); // free buffers and set defaults
void initInBuff();
bool httpPrint(const char* host);
void processLocalFile();
void processWebStream();
void processWebFile();
void processWebStreamTS();
void processWebStreamHLS();
void playAudioData();
bool readPlayListData();
const char* parsePlaylist_M3U();
const char* parsePlaylist_PLS();
const char* parsePlaylist_ASX();
const char* parsePlaylist_M3U8();
bool STfromEXTINF(char* str);
void showCodecParams();
int findNextSync(uint8_t* data, size_t len);
int sendBytes(uint8_t* data, size_t len);
void compute_audioCurrentTime(int bd);
void printDecodeError(int r);
void showID3Tag(const char* tag, const char* val);
void unicode2utf8(char* buff, uint32_t len);
size_t readAudioHeader(uint32_t bytes);
int read_WAV_Header(uint8_t* data, size_t len);
int read_FLAC_Header(uint8_t *data, size_t len);
int read_ID3_Header(uint8_t* data, size_t len);
int read_M4A_Header(uint8_t* data, size_t len);
size_t process_m3u8_ID3_Header(uint8_t* packet);
bool setSampleRate(uint32_t hz);
bool setBitsPerSample(int bits);
bool setChannels(int channels);
bool setBitrate(int br);
bool playChunk();
bool playSample(int16_t sample[2]) ;
int32_t Gain(int16_t s[2]);
bool fill_InputBuf();
void showstreamtitle(const char* ml);
bool parseContentType(char* ct);
bool parseHttpResponseHeader();
bool initializeDecoder();
esp_err_t I2Sstart(uint8_t i2s_num);
esp_err_t I2Sstop(uint8_t i2s_num);
void urlencode(char* buff, uint16_t buffLen, bool spacesOnly = false);
int16_t* IIR_filterChain0(int16_t iir_in[2], bool clear = false);
int16_t* IIR_filterChain1(int16_t* iir_in, bool clear = false);
int16_t* IIR_filterChain2(int16_t* iir_in, bool clear = false);
inline void setDatamode(uint8_t dm){m_datamode=dm;}
inline uint8_t getDatamode(){return m_datamode;}
inline uint32_t streamavail(){ return _client ? _client->available() : 0;}
void IIR_calculateCoefficients(int8_t G1, int8_t G2, int8_t G3);
bool ts_parsePacket(uint8_t* packet, uint8_t* packetStart, uint8_t* packetLength);
//+++ W E B S T R E A M - H E L P F U N C T I O N S +++
uint16_t readMetadata(uint16_t b, bool first = false);
size_t chunkedDataTransfer(uint8_t* bytes);
bool readID3V1Tag();
boolean streamDetection(uint32_t bytesAvail);
void seek_m4a_stsz();
void seek_m4a_ilst();
uint32_t m4a_correctResumeFilePos(uint32_t resumeFilePos);
uint32_t flac_correctResumeFilePos(uint32_t resumeFilePos);
uint32_t mp3_correctResumeFilePos(uint32_t resumeFilePos);
//++++ implement several function with respect to the index of string ++++
void trim(char *s) {
//fb trim in place
char *pe;
char *p = s;
while ( isspace(*p) ) p++; //left
pe = p; //right
while ( *pe != '\0' ) pe++;
do {
pe--;
} while ( (pe > p) && isspace(*pe) );
if (p == s) {
*++pe = '\0';
} else { //move
while ( p <= pe ) *s++ = *p++;
*s = '\0';
}
}
bool startsWith (const char* base, const char* str) {
//fb
char c;
while ( (c = *str++) != '\0' )
if (c != *base++) return false;
return true;
}
bool endsWith (const char* base, const char* str) {
//fb
int slen = strlen(str) - 1;
const char *p = base + strlen(base) - 1;
while(p > base && isspace(*p)) p--; // rtrim
p -= slen;
if (p < base) return false;
return (strncmp(p, str, slen) == 0);
}
int indexOf (const char* base, const char* str, int startIndex = 0) {
//fb
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 indexOf (const char* base, char ch, int startIndex = 0) {
//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 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 lastIndexOf(const char* haystack, const char needle) {
//fb
const char *p = strrchr(haystack, needle);
return (p ? p - haystack : -1);
}
int specialIndexOf (uint8_t* base, const char* str, int baselen, bool exact = false){
int result; // 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;
}
// some other functions
size_t bigEndian(uint8_t* base, uint8_t numBytes, uint8_t shiftLeft = 8){
uint64_t result = 0;
if(numBytes < 1 || numBytes > 8) return 0;
for (int i = 0; i < numBytes; i++) {
result += *(base + i) << (numBytes -i - 1) * shiftLeft;
}
if(result > SIZE_MAX) {log_e("range overflow"); result = 0;} // overflow
return (size_t)result;
}
bool 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);
len = base64_encode_blockend((buffer + len), &_state);
memcpy(dest, buffer, strlen(buffer));
dest[strlen(buffer)] = '\0';
free(buffer);
return true;
}
return false;
}
size_t urlencode_expected_len(const char* source){
size_t expectedLen = strlen(source);
for(int i = 0; i < strlen(source); i++) {
if(isalnum(source[i])){;}
else expectedLen += 2;
}
return expectedLen;
}
void vector_clear_and_shrink(vector<char*>&vec){
uint size = vec.size();
for (int i = 0; i < size; i++) {
if(vec[i]){
free(vec[i]);
vec[i] = NULL;
}
}
vec.clear();
vec.shrink_to_fit();
}
uint32_t 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;
}
private:
const char *codecname[9] = {"unknown", "WAV", "MP3", "AAC", "M4A", "FLAC"};
enum : int { APLL_AUTO = -1, APLL_ENABLE = 1, APLL_DISABLE = 0 };
enum : int { EXTERNAL_I2S = 0, INTERNAL_DAC = 1, INTERNAL_PDM = 2 };
enum : int { FORMAT_NONE = 0, FORMAT_M3U = 1, FORMAT_PLS = 2, FORMAT_ASX = 3, FORMAT_M3U8 = 4};
enum : int { AUDIO_NONE, HTTP_RESPONSE_HEADER, AUDIO_DATA, AUDIO_LOCALFILE,
AUDIO_PLAYLISTINIT, AUDIO_PLAYLISTHEADER, AUDIO_PLAYLISTDATA};
enum : int { FLAC_BEGIN = 0, FLAC_MAGIC = 1, FLAC_MBH =2, FLAC_SINFO = 3, FLAC_PADDING = 4, FLAC_APP = 5,
FLAC_SEEK = 6, FLAC_VORBIS = 7, FLAC_CUESHEET = 8, FLAC_PICTURE = 9, FLAC_OKAY = 100};
enum : int { M4A_BEGIN = 0, M4A_FTYP = 1, M4A_CHK = 2, M4A_MOOV = 3, M4A_FREE = 4, M4A_TRAK = 5, M4A_MDAT = 6,
M4A_ILST = 7, M4A_MP4A = 8, M4A_AMRDY = 99, M4A_OKAY = 100};
enum : int { CODEC_NONE = 0, CODEC_WAV = 1, CODEC_MP3 = 2, CODEC_AAC = 3, CODEC_M4A = 4, CODEC_FLAC = 5,
CODEC_AACP = 6, CODEC_OPUS = 7};
enum : int { ST_NONE = 0, ST_WEBFILE = 1, ST_WEBSTREAM = 2};
typedef enum { LEFTCHANNEL=0, RIGHTCHANNEL=1 } SampleIndex;
typedef enum { LOWSHELF = 0, PEAKEQ = 1, HIFGSHELF =2 } FilterType;
const uint8_t volumetable[22]={ 0, 1, 2, 3, 4 , 6 , 8, 10, 12, 14, 17,
20, 23, 27, 30 ,34, 38, 43 ,48, 52, 58, 64}; //22 elements
typedef struct _filter{
float a0;
float a1;
float a2;
float b1;
float b2;
} filter_t;
typedef struct _pis_array{
int number;
int pids[4];
} pid_array;
File audiofile; // @suppress("Abstract class cannot be instantiated")
WiFiClient client; // @suppress("Abstract class cannot be instantiated")
WiFiClientSecure clientsecure; // @suppress("Abstract class cannot be instantiated")
WiFiClient* _client = nullptr;
SemaphoreHandle_t mutex_audio;
i2s_config_t m_i2s_config = {}; // stores values for I2S driver
i2s_pin_config_t m_pin_config = {};
std::vector<char*> m_playlistContent; // m3u8 playlist buffer
std::vector<char*> m_playlistURL; // m3u8 streamURLs buffer
std::vector<uint32_t> m_hashQueue;
const size_t m_frameSizeWav = 1024;
const size_t m_frameSizeMP3 = 1600;
const size_t m_frameSizeAAC = 1600;
const size_t m_frameSizeFLAC = 4096 * 4;
const size_t m_frameSizeOPUS = 1024;
static const uint8_t m_tsPacketSize = 188;
static const uint8_t m_tsHeaderSize = 4;
char* m_chbuf = NULL;
uint16_t m_chbufSize = 0; // will set in constructor (depending on PSRAM)
char m_lastHost[512]; // Store the last URL to a webstream
char* m_playlistBuff = NULL; // stores playlistdata
const uint16_t m_plsBuffEntryLen = 256; // length of each entry in playlistBuff
filter_t m_filter[3]; // digital filters
int m_LFcount = 0; // Detection of end of header
uint32_t m_sampleRate=16000;
uint32_t m_bitRate=0; // current bitrate given fom decoder
uint32_t m_avr_bitrate = 0; // average bitrate, median computed by VBR
int m_readbytes = 0; // bytes read
uint32_t m_metacount = 0; // counts down bytes between metadata
int m_controlCounter = 0; // Status within readID3data() and readWaveHeader()
int8_t m_balance = 0; // -16 (mute left) ... +16 (mute right)
uint16_t m_vol=64; // volume
uint8_t m_vol_steps = 0; // 0 == legacy 21 steps, > 0: number of volume steps
int32_t m_vol_step_div = 64; // max of volumetable[]
uint8_t m_bitsPerSample = 16; // bitsPerSample
uint8_t m_channels = 2;
uint8_t m_i2s_num = I2S_NUM_0; // I2S_NUM_0 or I2S_NUM_1
uint8_t m_playlistFormat = 0; // M3U, PLS, ASX
uint8_t m_codec = CODEC_NONE; //
uint8_t m_expectedCodec = CODEC_NONE; // set in connecttohost (e.g. http://url.mp3 -> CODEC_MP3)
uint8_t m_expectedPlsFmt = FORMAT_NONE; // set in connecttohost (e.g. streaming01.m3u) -> FORMAT_M3U)
uint8_t m_filterType[2]; // lowpass, highpass
uint8_t m_streamType = ST_NONE;
uint8_t m_ID3Size = 0; // lengt of ID3frame - ID3header
int16_t m_outBuff[2048*2]; // Interleaved L/R
int16_t m_validSamples = 0;
int16_t m_curSample = 0;
uint16_t m_datamode = 0; // Statemaschine
uint16_t m_streamTitleHash = 0; // remember streamtitle, ignore multiple occurence in metadata
uint16_t m_timeout_ms = 250;
uint16_t m_timeout_ms_ssl = 2700;
uint8_t m_flacBitsPerSample = 0; // bps should be 16
uint8_t m_flacNumChannels = 0; // can be read out in the FLAC file header
uint32_t m_flacSampleRate = 0; // can be read out in the FLAC file header
uint16_t m_flacMaxFrameSize = 0; // can be read out in the FLAC file header
uint16_t m_flacMaxBlockSize = 0; // can be read out in the FLAC file header
uint32_t m_flacTotalSamplesInStream = 0; // can be read out in the FLAC file header
uint32_t m_metaint = 0; // Number of databytes between metadata
uint32_t m_chunkcount = 0 ; // Counter for chunked transfer
uint32_t m_t0 = 0; // store millis(), is needed for a small delay
uint32_t m_contentlength = 0; // Stores the length if the stream comes from fileserver
uint32_t m_bytesNotDecoded = 0; // pictures or something else that comes with the stream
uint32_t m_PlayingStartTime = 0; // Stores the milliseconds after the start of the audio
uint32_t m_resumeFilePos = 0; // the return value from stopSong() can be entered here
uint16_t m_m3u8_targetDuration = 10; //
uint32_t m_stsz_numEntries = 0; // num of entries inside stsz atom (uint32_t)
uint32_t m_stsz_position = 0; // pos of stsz atom within file
bool m_f_metadata = false; // assume stream without metadata
bool m_f_unsync = false; // set within ID3 tag but not used
bool m_f_exthdr = false; // ID3 extended header
bool m_f_ssl = false;
bool m_f_running = false;
bool m_f_firstCall = false; // InitSequence for processWebstream and processLokalFile
bool m_f_chunked = false ; // Station provides chunked transfer
bool m_f_firstmetabyte = false; // True if first metabyte (counter)
bool m_f_playing = false; // valid mp3 stream recognized
bool m_f_tts = false; // text to speech
bool m_f_loop = false; // Set if audio file should loop
bool m_f_forceMono = false; // if true stereo -> mono
bool m_f_internalDAC = false; // false: output vis I2S, true output via internal DAC
bool m_f_rtsp = false; // set if RTSP is used (m3u8 stream)
bool m_f_m3u8data = false; // used in processM3U8entries
bool m_f_Log = false; // set in platformio.ini -DAUDIO_LOG and -DCORE_DEBUG_LEVEL=3 or 4
bool m_f_continue = false; // next m3u8 chunk is available
bool m_f_ts = true; // transport stream
bool m_f_m4aID3dataAreRead = false; // has the m4a-ID3data already been read?
uint8_t m_f_channelEnabled = 3; // internal DAC, both channels
uint32_t m_audioFileDuration = 0;
float m_audioCurrentTime = 0;
uint32_t m_audioDataStart = 0; // in bytes
size_t m_audioDataSize = 0; //
float m_filterBuff[3][2][2][2]; // IIR filters memory for Audio DSP
size_t m_i2s_bytesWritten = 0; // set in i2s_write() but not used
size_t m_file_size = 0; // size of the file
uint16_t m_filterFrequency[2];
int8_t m_gain0 = 0; // cut or boost filters (EQ)
int8_t m_gain1 = 0;
int8_t m_gain2 = 0;
pid_array m_pidsOfPMT;
int16_t m_pidOfAAC;
uint8_t m_packetBuff[m_tsPacketSize];
int16_t m_pesDataLength = 0;
};
//----------------------------------------------------------------------------------------------------------------------
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