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v0.7.355

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This commit is contained in:
e2002
2022-08-17 14:11:17 +03:00
parent 442d970fd9
commit 8fa32f1587
19 changed files with 3348 additions and 2611 deletions
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3846
yoRadio/src/audioI2S/Audio.cpp
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File diff suppressed because it is too large Load Diff

200
yoRadio/src/audioI2S/AudioEx.h
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@@ -1,8 +1,8 @@
/*
* Audio.h
*
* Created on: Oct 26,2018
* Updated on: Jan 05,2022
* Created on: Oct 28,2018
* Updated on: Aug 12,2022
* Author: Wolle (schreibfaul1)
*/
@@ -11,13 +11,15 @@
#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
@@ -31,11 +33,12 @@
#endif // SDFATFS_USED
#ifndef AUDIOBUFFER_MULTIPLIER
#define AUDIOBUFFER_MULTIPLIER 13
#define AUDIOBUFFER_MULTIPLIER 12
#endif
#ifdef SDFATFS_USED
typedef File32 File;
//typedef File32 File;
typedef FsFile File;
namespace fs {
class FS : public SdFat {
@@ -65,7 +68,7 @@ using namespace fs;
#define SD SD_SDFAT
#endif //SDFATFS_USED
using namespace std;
extern __attribute__((weak)) void audio_info(const char*);
@@ -83,6 +86,8 @@ 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
#define AUDIO_INFO(...) {char buff[512 + 64]; sprintf(buff,__VA_ARGS__); if(audio_info) audio_info(buff);}
//----------------------------------------------------------------------------------------------------------------------
class AudioBuffer {
@@ -116,6 +121,8 @@ 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
@@ -128,24 +135,26 @@ public:
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:
const size_t m_buffSizePSRAM = 300000; // most webstreams limit the advance to 100...300Kbytes
//const size_t m_buffSizeRAM = 1600 * 5 * AUDIOBUFFER_MULTIPLIER;
const size_t m_buffSizeRAM = 1600 * AUDIOBUFFER_MULTIPLIER;
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 * AUDIOBUFFER_MULTIPLIER; // reserved buffspace, >= one mp3 frame
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;
size_t m_buffSizePSRAM = 300000; // most webstreams limit the advance to 100...300Kbytes
//size_t m_buffSizeRAM = 1600 * 5;
size_t m_buffSizeRAM = 1600 * AUDIOBUFFER_MULTIPLIER;
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...)
};
//----------------------------------------------------------------------------------------------------------------------
@@ -154,26 +163,30 @@ class Audio : private AudioBuffer{
AudioBuffer InBuff; // instance of input buffer
public:
Audio(bool internalDAC = false, i2s_dac_mode_t channelEnabled = I2S_DAC_CHANNEL_LEFT_EN); // #99
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 connecttoFS(fs::FS &fs, const char* path);
bool connecttoSD(const char* path);
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);
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();
void stopSong();
uint32_t stopSong();
void forceMono(bool m);
void setBalance(int8_t bal = 0);
void setVolume(uint8_t vol);
uint8_t getVolume();
uint8_t getI2sPort();
uint32_t getAudioDataStartPos();
uint32_t getFileSize();
@@ -181,10 +194,11 @@ public:
uint32_t getSampleRate();
uint8_t getBitsPerSample();
uint8_t getChannels();
uint32_t getBitRate();
uint32_t getBitRate(bool avg = false);
uint32_t getAudioFileDuration();
uint32_t getAudioCurrentTime();
uint32_t getTotalPlayingTime();
void setDefaults();
/* VU METER */
void setVUmeter() {};
@@ -196,19 +210,34 @@ public:
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);
[[deprecated]]void setInternalDAC(bool internalDAC = true, i2s_dac_mode_t channelEnabled = I2S_DAC_CHANNEL_LEFT_EN);
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 httpPrint(const char* url);
//void setDefaults(); // free buffers and set defaults
void initInBuff();
bool httpPrint(const char* host);
void processLocalFile();
void processWebStream();
void processPlayListData();
void processM3U8entries(uint8_t nrOfEntries = 0, uint32_t seqNr = 0, uint8_t pos = 0, uint16_t targetDuration = 0);
void processWebStreamTS();
void processWebStreamHLS();
void playAudioData();
size_t chunkedDataTransfer();
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);
@@ -217,24 +246,27 @@ private:
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_MP3_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);
int read_OGG_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]) ;
bool playI2Sremains();
void playI2Sremains();
int32_t Gain(int16_t s[2]);
bool fill_InputBuf();
void showstreamtitle(const char* ml);
bool parseContentType(const char* ct);
void processAudioHeaderData();
bool readMetadata(uint8_t b, bool first = false);
bool parseContentType(char* ct);
bool parseHttpResponseHeader();
bool initializeDecoder();
uint16_t readMetadata(uint16_t b, bool first = false);
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);
@@ -243,8 +275,9 @@ private:
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() {if(m_f_ssl==false) return client.available(); else return clientsecure.available();}
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);
// implement several function with respect to the index of string
void trim(char *s) {
@@ -283,7 +316,7 @@ private:
return (strncmp(p, str, slen) == 0);
}
int indexOf (const char* base, const char* str, int startIndex) {
int indexOf (const char* base, const char* str, int startIndex = 0) {
//fb
const char *p = base;
for (; startIndex > 0; startIndex--)
@@ -293,7 +326,7 @@ private:
return pos - base;
}
int indexOf (const char* base, char ch, int startIndex) {
int indexOf (const char* base, char ch, int startIndex = 0) {
//fb
const char *p = base;
for (; startIndex > 0; startIndex--)
@@ -369,21 +402,42 @@ private:
}
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", "OGG", "OGG FLAC", "OPUS"};
enum : int { APLL_AUTO = -1, APLL_ENABLE = 1, APLL_DISABLE = 0 };
enum : int { EXTERNAL_I2S = 0, INTERNAL_DAC = 1, INTERNAL_PDM = 2 };
enum : int { CODEC_NONE, CODEC_WAV, CODEC_MP3, CODEC_AAC, CODEC_M4A, CODEC_FLAC, CODEC_OGG,
CODEC_OGG_FLAC, CODEC_OGG_OPUS};
enum : int { FORMAT_NONE = 0, FORMAT_M3U = 1, FORMAT_PLS = 2, FORMAT_ASX = 3, FORMAT_M3U8 = 4};
enum : int { AUDIO_NONE, AUDIO_HEADER, AUDIO_DATA,
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 { OGG_BEGIN = 0, OGG_MAGIC = 1, OGG_HEADER = 2, OGG_FIRST = 3, OGG_AMRDY = 99, OGG_OKAY = 100};
enum : int { CODEC_NONE = 0, CODEC_WAV = 1, CODEC_MP3 = 2, CODEC_AAC = 3, CODEC_M4A = 4, CODEC_FLAC = 5,
CODEC_OGG = 6, CODEC_OGG_FLAC = 7, CODEC_OGG_OPUS = 8, CODEC_AACP = 9};
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;
@@ -398,18 +452,29 @@ private:
float b2;
} filter_t;
File audiofile; // @suppress("Abstract class cannot be instantiated")
WiFiClient client; // @suppress("Abstract class cannot be instantiated")
WiFiClientSecure clientsecure; // @suppress("Abstract class cannot be instantiated")
WiFiUDP udpclient; // @suppress("Abstract class cannot be instantiated")
i2s_config_t m_i2s_config; // stores values for I2S driver
i2s_pin_config_t m_pin_config;
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;
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 = 1600;
const size_t m_frameSizeMP3 = 1600;
const size_t m_frameSizeAAC = 1600;
const size_t m_frameSizeFLAC = 4096 * 4;
static const uint8_t m_tsPacketSize = 188;
static const uint8_t m_tsHeaderSize = 4;
char chbuf[512 + 128]; // must be greater than m_lastHost #254
char m_lastHost[512]; // Store the last URL to a webstream
char* m_playlistBuff = NULL; // stores playlistdata
@@ -419,23 +484,28 @@ private:
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
int m_metalen=0; // Number of bytes in metadata
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)
uint8_t m_vol=64; // volume
uint8_t m_bitsPerSample = 16; // bitsPerSample
uint8_t m_channels=2;
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_m3u8codec = CODEC_NONE; // M4A
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_st_remember = 0; // Save hash from the last streamtitle
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
@@ -448,30 +518,27 @@ private:
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; //
bool m_f_swm = true; // 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_localfile = false ; // Play from local mp3-file
bool m_f_webstream = false ; // Play from URL
bool m_f_ssl = false;
bool m_f_running = false;
bool m_f_firstCall = false; // InitSequence for processWebstream and processLokalFile
bool m_f_ctseen = false; // First line of header seen or not
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_webfile = false; // assume it's a radiostream, not a podcast
bool m_f_tts = false; // text to speech
bool m_f_psram = false; // set if PSRAM is availabe
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 = true; // if m3u8: log is cancelled
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_initInbuffOnce = false; // init InBuff only once
i2s_dac_mode_t m_f_channelEnabled = I2S_DAC_CHANNEL_LEFT_EN; // internal DAC on GPIO26 for M5StickC/Plus
bool m_f_ts = true; // transport stream
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
@@ -483,6 +550,11 @@ private:
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;
};
//----------------------------------------------------------------------------------------------------------------------

57
yoRadio/src/audioI2S/aac_decoder/aac_decoder.cpp
View File

@@ -3,7 +3,7 @@
* libhelix_HAACDECODER
*
* Created on: 26.10.2018
* Updated on: 10.09.2021
* Updated on: 27.05.2022
************************************************************************************/
#include "aac_decoder.h"
@@ -1654,47 +1654,42 @@ static const int8_t negMask[3] = {~0x03, ~0x07, ~0x0f};
* Return: false if not enough memory, otherwise true
*
**********************************************************************************************************************/
#ifdef CONFIG_IDF_TARGET_ESP32S3
// ESP32-S3: If there is PSRAM, prefer it
#define __malloc_heap_psram(size) \
heap_caps_malloc_prefer(size, 2, MALLOC_CAP_DEFAULT|MALLOC_CAP_SPIRAM, MALLOC_CAP_DEFAULT|MALLOC_CAP_INTERNAL)
#else
// ESP32, PSRAM is too slow, prefer SRAM
#define __malloc_heap_psram(size) \
heap_caps_malloc_prefer(size, 2, MALLOC_CAP_DEFAULT|MALLOC_CAP_INTERNAL, MALLOC_CAP_DEFAULT|MALLOC_CAP_SPIRAM)
#endif
bool AACDecoder_AllocateBuffers(void){
if(!m_AACDecInfo) {m_AACDecInfo = (AACDecInfo_t*) malloc(sizeof(AACDecInfo_t));}
if(!m_PSInfoBase) {m_PSInfoBase = (PSInfoBase_t*) malloc(sizeof(PSInfoBase_t));}
if(!m_pce[0]) {m_pce[0] = (ProgConfigElement_t*) malloc(sizeof(ProgConfigElement_t)*16);}
/* here, sizes are: AACDecInfo_t:96 PSInfoBase_t:27364 ProgConfigElement_t*16:1312 PSInfoSBR_t:50788 */
#ifdef AAC_ENABLE_SBR
if(!m_PSInfoSBR) {m_PSInfoSBR = (PSInfoSBR_t*)__malloc_heap_psram(sizeof(PSInfoSBR_t));}
if(!m_AACDecInfo || !m_PSInfoBase || !m_pce[0]) {
log_i("heap is too small, try PSRAM");
AACDecoder_FreeBuffers();
if(!m_PSInfoSBR) {
log_e("OOM in SBR, can't allocate %d bytes\n", sizeof(PSInfoSBR_t));
return false; // ERR_AAC_SBR_INIT;
}
else{
goto nextStep;
else {
log_d("AAC Spectral Band Replication enabled, %d additional bytes allocated", sizeof(PSInfoSBR_t));
}
#endif
if(psramFound()) {
// PSRAM found, Buffer will be allocated in PSRAM
if(!m_AACDecInfo) {m_AACDecInfo = (AACDecInfo_t*) ps_calloc(sizeof(AACDecInfo_t), sizeof(uint8_t));}
if(!m_PSInfoBase) {m_PSInfoBase = (PSInfoBase_t*) ps_calloc(sizeof(PSInfoBase_t), sizeof(uint8_t));}
if(!m_pce[0]) {m_pce[0] = (ProgConfigElement_t*) ps_calloc(sizeof(ProgConfigElement_t)*16, sizeof(uint8_t));}
}
/* these could fall back to PSRAM if not enough heap available */
if(!m_AACDecInfo) {m_AACDecInfo = (AACDecInfo_t*) __malloc_heap_psram(sizeof(AACDecInfo_t));}
if(!m_PSInfoBase) {m_PSInfoBase = (PSInfoBase_t*) __malloc_heap_psram(sizeof(PSInfoBase_t));}
if(!m_pce[0]) {m_pce[0] = (ProgConfigElement_t*) __malloc_heap_psram(sizeof(ProgConfigElement_t)*16);}
if(!m_AACDecInfo || !m_PSInfoBase || !m_pce[0]) {
log_e("not enough memory to allocate aacdecoder buffers");
AACDecoder_FreeBuffers();
return false;
}
log_i("AAC buffers allocated in PSRAM");
nextStep:
#ifdef AAC_ENABLE_SBR
// can't allocated in PSRAM, because PSRAM ist too slow
if(!m_PSInfoSBR) {m_PSInfoSBR = (PSInfoSBR_t*)malloc(sizeof(PSInfoSBR_t));}
if(!m_PSInfoSBR) {
log_e("OOM in SBR, can't allocate %d bytes\n", sizeof(PSInfoSBR_t));
return false; // ERR_AAC_SBR_INIT;
}
#endif
// Clear Buffer
memset( m_AACDecInfo, 0, sizeof(AACDecInfo_t)); //Clear AACDecInfo
@@ -1777,7 +1772,7 @@ void AACDecoder_FreeBuffers(void) {
if(m_AACDecInfo) {free(m_AACDecInfo); m_AACDecInfo=NULL;}
if(m_PSInfoBase) {free(m_PSInfoBase); m_PSInfoBase=NULL;}
if(m_pce[0]) {for(int i=0; i<16; i++) free(m_pce[i]); m_pce[0]=NULL;}
if(m_pce[0]) {free(m_pce[0]); m_pce[0]=NULL;}
#ifdef AAC_ENABLE_SBR
if(m_PSInfoSBR) {free(m_PSInfoSBR); m_PSInfoSBR=NULL;} //Clear AACDecInfo

5
yoRadio/src/audioI2S/aac_decoder/aac_decoder.h
View File

@@ -6,7 +6,10 @@
#include "Arduino.h"
#define AAC_ENABLE_MPEG4
//#define AAC_ENABLE_SBR // needs additional 60KB Heap,
#if (defined CONFIG_IDF_TARGET_ESP32S3 && defined BOARD_HAS_PSRAM)
#define AAC_ENABLE_SBR // needs additional 60KB DRAM,
#endif
#define ASSERT(x) /* do nothing */

66
yoRadio/src/audioI2S/mp3_decoder/mp3_decoder.cpp
View File

@@ -3,7 +3,7 @@
* libhelix_HMP3DECODER
*
* Created on: 26.10.2018
* Updated on: 03.01.2022
* Updated on: 27.05.2022
*/
#include "mp3_decoder.h"
/* clip to range [-2^n, 2^n - 1] */
@@ -1490,8 +1490,6 @@ int MP3Decode( unsigned char *inbuf, int *bytesLeft, short *outbuf, int useSize)
*
* Return: none
*
* Notes: if one or more mallocs fail, function frees any buffers already
* allocated before returning
**********************************************************************************************************************/
void MP3Decoder_ClearBuffer(void) {
@@ -1525,54 +1523,38 @@ void MP3Decoder_ClearBuffer(void) {
* Return: pointer to MP3DecInfo structure (initialized with pointers to all
* the internal buffers needed for decoding)
*
* Notes: if one or more mallocs fail, function frees any buffers already
* allocated before returning
*
**********************************************************************************************************************/
#ifdef CONFIG_IDF_TARGET_ESP32S3
// ESP32-S3: If there is PSRAM, prefer it
#define __malloc_heap_psram(size) \
heap_caps_malloc_prefer(size, 2, MALLOC_CAP_DEFAULT|MALLOC_CAP_SPIRAM, MALLOC_CAP_DEFAULT|MALLOC_CAP_INTERNAL)
#else
// ESP32, PSRAM is too slow, prefer SRAM
#define __malloc_heap_psram(size) \
heap_caps_malloc_prefer(size, 2, MALLOC_CAP_DEFAULT|MALLOC_CAP_INTERNAL, MALLOC_CAP_DEFAULT|MALLOC_CAP_SPIRAM)
#endif
bool MP3Decoder_AllocateBuffers(void) {
// try first SRAM because its faster than PSRAM
if(!m_MP3DecInfo) {m_MP3DecInfo = (MP3DecInfo_t*) malloc(sizeof(MP3DecInfo_t) );}
if(!m_FrameHeader) {m_FrameHeader = (FrameHeader_t*) malloc(sizeof(FrameHeader_t) );}
if(!m_SideInfo) {m_SideInfo = (SideInfo_t*) malloc(sizeof(SideInfo_t) );}
if(!m_ScaleFactorJS) {m_ScaleFactorJS = (ScaleFactorJS_t*) malloc(sizeof(ScaleFactorJS_t));}
if(!m_HuffmanInfo) {m_HuffmanInfo = (HuffmanInfo_t*) malloc(sizeof(HuffmanInfo_t) );}
if(!m_DequantInfo) {m_DequantInfo = (DequantInfo_t*) malloc(sizeof(DequantInfo_t) );}
if(!m_IMDCTInfo) {m_IMDCTInfo = (IMDCTInfo_t*) malloc(sizeof(IMDCTInfo_t) );}
if(!m_SubbandInfo) {m_SubbandInfo = (SubbandInfo_t*) malloc(sizeof(SubbandInfo_t) );}
if(!m_MP3FrameInfo) {m_MP3FrameInfo = (MP3FrameInfo_t*) malloc(sizeof(MP3FrameInfo_t) );}
if(!m_MP3DecInfo) {m_MP3DecInfo = (MP3DecInfo_t*) __malloc_heap_psram(sizeof(MP3DecInfo_t) );}
if(!m_FrameHeader) {m_FrameHeader = (FrameHeader_t*) __malloc_heap_psram(sizeof(FrameHeader_t) );}
if(!m_SideInfo) {m_SideInfo = (SideInfo_t*) __malloc_heap_psram(sizeof(SideInfo_t) );}
if(!m_ScaleFactorJS) {m_ScaleFactorJS = (ScaleFactorJS_t*) __malloc_heap_psram(sizeof(ScaleFactorJS_t));}
if(!m_HuffmanInfo) {m_HuffmanInfo = (HuffmanInfo_t*) __malloc_heap_psram(sizeof(HuffmanInfo_t) );}
if(!m_DequantInfo) {m_DequantInfo = (DequantInfo_t*) __malloc_heap_psram(sizeof(DequantInfo_t) );}
if(!m_IMDCTInfo) {m_IMDCTInfo = (IMDCTInfo_t*) __malloc_heap_psram(sizeof(IMDCTInfo_t) );}
if(!m_SubbandInfo) {m_SubbandInfo = (SubbandInfo_t*) __malloc_heap_psram(sizeof(SubbandInfo_t) );}
if(!m_MP3FrameInfo) {m_MP3FrameInfo = (MP3FrameInfo_t*) __malloc_heap_psram(sizeof(MP3FrameInfo_t) );}
if(!m_MP3DecInfo || !m_FrameHeader || !m_SideInfo || !m_ScaleFactorJS || !m_HuffmanInfo ||
!m_DequantInfo || !m_IMDCTInfo || !m_SubbandInfo || !m_MP3FrameInfo) {
log_i("heap is too small, try PSRAM");
MP3Decoder_FreeBuffers();
}
else{
MP3Decoder_ClearBuffer();
return true; // success, all buffers allocated in SRAM (Heap)
}
if(psramFound()) {
// PSRAM found, Buffer will be allocated in PSRAM
if(!m_MP3DecInfo) {m_MP3DecInfo = (MP3DecInfo_t*) ps_calloc(sizeof(MP3DecInfo_t) , sizeof(uint8_t));}
if(!m_FrameHeader) {m_FrameHeader = (FrameHeader_t*) ps_calloc(sizeof(FrameHeader_t) , sizeof(uint8_t));}
if(!m_SideInfo) {m_SideInfo = (SideInfo_t*) ps_calloc(sizeof(SideInfo_t) , sizeof(uint8_t));}
if(!m_ScaleFactorJS) {m_ScaleFactorJS = (ScaleFactorJS_t*) ps_calloc(sizeof(ScaleFactorJS_t), sizeof(uint8_t));}
if(!m_HuffmanInfo) {m_HuffmanInfo = (HuffmanInfo_t*) ps_calloc(sizeof(HuffmanInfo_t) , sizeof(uint8_t));}
if(!m_DequantInfo) {m_DequantInfo = (DequantInfo_t*) ps_calloc(sizeof(DequantInfo_t) , sizeof(uint8_t));}
if(!m_IMDCTInfo) {m_IMDCTInfo = (IMDCTInfo_t*) ps_calloc(sizeof(IMDCTInfo_t) , sizeof(uint8_t));}
if(!m_SubbandInfo) {m_SubbandInfo = (SubbandInfo_t*) ps_calloc(sizeof(SubbandInfo_t) , sizeof(uint8_t));}
if(!m_MP3FrameInfo) {m_MP3FrameInfo = (MP3FrameInfo_t*) ps_calloc(sizeof(MP3FrameInfo_t) , sizeof(uint8_t));}
}
else {
log_e("not enough memory to allocate mp3decoder buffers");
return false;
}
if(!m_MP3DecInfo || !m_FrameHeader || !m_SideInfo || !m_ScaleFactorJS || !m_HuffmanInfo ||
!m_DequantInfo || !m_IMDCTInfo || !m_SubbandInfo || !m_MP3FrameInfo) {
log_e("not enough memory to allocate mp3decoder buffers in PSRAM");
return false;
}
MP3Decoder_ClearBuffer();
return true;
}