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# Number of days of inactivity before an issue becomes stale
daysUntilStale: 60
# Number of days of inactivity before a stale issue is closed
daysUntilClose: 7
# Issues with these labels will never be considered stale
exemptLabels:
- pinned
- security
# Label to use when marking an issue as stale
staleLabel: wontfix
# Comment to post when marking an issue as stale. Set to `false` to disable
markComment: >
This issue has been automatically marked as stale because it has not had
recent activity. It will be closed if no further activity occurs. Thank you
for your contributions.
# Comment to post when closing a stale issue. Set to `false` to disable
closeComment: false

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{"type": "library", "name": "ESP32-audioI2S", "version": "2.1.0", "spec": {"owner": "esphome", "id": 14055, "name": "ESP32-audioI2S", "requirements": null, "uri": null}}

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GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <https://www.gnu.org/licenses/>.
Also add information on how to contact you by electronic and paper mail.
If the program does terminal interaction, make it output a short
notice like this when it starts in an interactive mode:
<program> Copyright (C) <year> <name of author>
This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
This is free software, and you are welcome to redistribute it
under certain conditions; type `show c' for details.
The hypothetical commands `show w' and `show c' should show the appropriate
parts of the General Public License. Of course, your program's commands
might be different; for a GUI interface, you would use an "about box".
You should also get your employer (if you work as a programmer) or school,
if any, to sign a "copyright disclaimer" for the program, if necessary.
For more information on this, and how to apply and follow the GNU GPL, see
<https://www.gnu.org/licenses/>.
The GNU General Public License does not permit incorporating your program
into proprietary programs. If your program is a subroutine library, you
may consider it more useful to permit linking proprietary applications with
the library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License. But first, please read
<https://www.gnu.org/licenses/why-not-lgpl.html>.

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# ESP32-audioI2S
Plays mp3, m4a and wav files from SD card via I2S with external hardware.
HELIX-mp3 and -aac decoder is included.
Works with MAX98357A (3 Watt amplifier with DAC), connected three lines (DOUT, BLCK, LRC) to I2S.
For stereo are two MAX98357A necessary. AudioI2S works with UDA1334A (Adafruit I2S Stereo Decoder Breakout Board) and PCM5102A.
Other HW may work but not tested. Plays also icy-streams and GoogleTTS. Can be compiled with Arduino IDE. [WIKI](https://github.com/schreibfaul1/ESP32-audioI2S/wiki)
```` c++
#include "Arduino.h"
#include "WiFi.h"
#include "Audio.h"
#include "SD.h"
#include "FS.h"
// Digital I/O used
#define SD_CS 5
#define SPI_MOSI 23
#define SPI_MISO 19
#define SPI_SCK 18
#define I2S_DOUT 25
#define I2S_BCLK 27
#define I2S_LRC 26
Audio audio;
String ssid = "*******";
String password = "*******";
void setup() {
pinMode(SD_CS, OUTPUT); digitalWrite(SD_CS, HIGH);
SPI.begin(SPI_SCK, SPI_MISO, SPI_MOSI);
Serial.begin(115200);
SD.begin(SD_CS);
WiFi.disconnect();
WiFi.mode(WIFI_STA);
WiFi.begin(ssid.c_str(), password.c_str());
while (WiFi.status() != WL_CONNECTED) delay(1500);
audio.setPinout(I2S_BCLK, I2S_LRC, I2S_DOUT);
audio.setVolume(21); // 0...21
// audio.connecttoFS(SD, "/320k_test.mp3");
// audio.connecttohost("http://www.wdr.de/wdrlive/media/einslive.m3u");
// audio.connecttohost("http://macslons-irish-pub-radio.com/media.asx");
// audio.connecttohost("http://mp3.ffh.de/radioffh/hqlivestream.aac"); // 128k aac
audio.connecttohost("http://mp3.ffh.de/radioffh/hqlivestream.mp3"); // 128k mp3
// audio.connecttohost("https://github.com/schreibfaul1/ESP32-audioI2S/raw/master/additional_info/Testfiles/sample1.m4a"); // m4a
// audio.connecttohost("https://github.com/schreibfaul1/ESP32-audioI2S/raw/master/additional_info/Testfiles/test_16bit_stereo.wav"); // wav
// audio.connecttospeech("Wenn die Hunde schlafen, kann der Wolf gut Schafe stehlen.", "de");
}
void loop()
{
audio.loop();
}
// optional
void audio_info(const char *info){
Serial.print("info "); Serial.println(info);
}
void audio_id3data(const char *info){ //id3 metadata
Serial.print("id3data ");Serial.println(info);
}
void audio_eof_mp3(const char *info){ //end of file
Serial.print("eof_mp3 ");Serial.println(info);
}
void audio_showstation(const char *info){
Serial.print("station ");Serial.println(info);
}
void audio_showstreamtitle(const char *info){
Serial.print("streamtitle ");Serial.println(info);
}
void audio_bitrate(const char *info){
Serial.print("bitrate ");Serial.println(info);
}
void audio_commercial(const char *info){ //duration in sec
Serial.print("commercial ");Serial.println(info);
}
void audio_icyurl(const char *info){ //homepage
Serial.print("icyurl ");Serial.println(info);
}
void audio_lasthost(const char *info){ //stream URL played
Serial.print("lasthost ");Serial.println(info);
}
void audio_eof_speech(const char *info){
Serial.print("eof_speech ");Serial.println(info);
}
````
Breadboard
![Breadboard](https://github.com/schreibfaul1/ESP32-audioI2S/blob/master/additional_info/Breadboard.jpg)
Wiring
![Wiring](https://github.com/schreibfaul1/ESP32-audioI2S/blob/master/additional_info/ESP32_I2S_PCM5102A.JPG)
Impulse diagram
![Impulse diagram](https://github.com/schreibfaul1/ESP32-audioI2S/blob/master/additional_info/Impulsdiagramm.jpg)

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// found this sketch on https://www.mikrocontroller.net/topic/474383
// thanks to Michael U. (amiga)
#include "Arduino.h"
#include "WiFi.h"
#include "SPI.h"
#include "SD.h"
#include "FS.h"
#include "Wire.h"
#include "AC101.h" //https://github.com/schreibfaul1/AC101
// #include "ES8388.h" // https://github.com/maditnerd/es8388
#include "Audio.h" //https://github.com/schreibfaul1/ESP32-audioI2S
// SPI GPIOs
#define SD_CS 13
#define SPI_MOSI 15
#define SPI_MISO 2
#define SPI_SCK 14
// I2S GPIOs, the names refer on AC101, AS1 Audio Kit V2.2 2379
#define I2S_DSIN 25
#define I2S_BCLK 27
#define I2S_LRC 26
#define I2S_MCLK 0
#define I2S_DOUT 35
// I2S GPIOs, the names refer on ES8388, AS1 Audio Kit V2.2 3378
//#define I2S_DSIN 35
//#define I2S_BCLK 27
//#define I2S_LRC 25
//#define I2S_MCLK 0
//#define I2S_DOUT 26
// I2C GPIOs
#define IIC_CLK 32
#define IIC_DATA 33
// buttons
// #define BUTTON_2_PIN 13 // shared mit SPI_CS
#define BUTTON_3_PIN 19
#define BUTTON_4_PIN 23
#define BUTTON_5_PIN 18 // Stop
#define BUTTON_6_PIN 5 // Play
// amplifier enable
#define GPIO_PA_EN 21
//Switch S1: 1-OFF, 2-ON, 3-ON, 4-OFF, 5-OFF
String ssid = "xxxxxxxxx";
String password = "xxxxxxxxx";
static AC101 dac; // AC101
// ES8388 dac; // ES8388 (new board)
int volume = 40; // 0...100
Audio audio;
//#####################################################################
void setup()
{
Serial.begin(115200);
Serial.println("\r\nReset");
Serial.printf_P(PSTR("Free mem=%d\n"), ESP.getFreeHeap());
pinMode(SD_CS, OUTPUT);
digitalWrite(SD_CS, HIGH);
SPI.begin(SPI_SCK, SPI_MISO, SPI_MOSI);
SPI.setFrequency(1000000);
SD.begin(SD_CS);
WiFi.mode(WIFI_STA);
WiFi.begin(ssid.c_str(), password.c_str());
while (WiFi.status() != WL_CONNECTED)
{
Serial.print(".");
delay(100);
}
Serial.printf_P(PSTR("Connected\r\nRSSI: "));
Serial.print(WiFi.RSSI());
Serial.print(" IP: ");
Serial.println(WiFi.localIP());
Serial.printf("Connect to DAC codec... ");
while (not dac.begin(IIC_DATA, IIC_CLK))
{
Serial.printf("Failed!\n");
delay(1000);
}
Serial.printf("OK\n");
dac.SetVolumeSpeaker(volume);
dac.SetVolumeHeadphone(volume);
// ac.DumpRegisters();
// Enable amplifier
pinMode(GPIO_PA_EN, OUTPUT);
digitalWrite(GPIO_PA_EN, HIGH);
// set I2S_MasterClock
audio.i2s_mclk_pin_select(I2S_MCLK);
audio.setPinout(I2S_BCLK, I2S_LRC, I2S_DSIN);
audio.setVolume(10); // 0...21
audio.connecttoFS(SD, "/320k_test.mp3");
// audio.connecttoSD("/Banana Boat Song - Harry Belafonte.mp3");
// audio.connecttohost("http://mp3channels.webradio.antenne.de:80/oldies-but-goldies");
// audio.connecttohost("http://dg-rbb-http-dus-dtag-cdn.cast.addradio.de/rbb/antennebrandenburg/live/mp3/128/stream.mp3");
// audio.connecttospeech("Wenn die Hunde schlafen, kann der Wolf gut Schafe stehlen.", "de");
}
//-----------------------------------------------------------------------
void loop()
{
audio.loop();
}
// optional
void audio_info(const char *info){
Serial.print("info "); Serial.println(info);
}
void audio_id3data(const char *info){ //id3 metadata
Serial.print("id3data ");Serial.println(info);
}
void audio_eof_mp3(const char *info){ //end of file
Serial.print("eof_mp3 ");Serial.println(info);
}
void audio_showstation(const char *info){
Serial.print("station ");Serial.println(info);
}
void audio_showstreamtitle(const char *info){
Serial.print("streamtitle ");Serial.println(info);
}
void audio_bitrate(const char *info){
Serial.print("bitrate ");Serial.println(info);
}
void audio_commercial(const char *info){ //duration in sec
Serial.print("commercial ");Serial.println(info);
}
void audio_icyurl(const char *info){ //homepage
Serial.print("icyurl ");Serial.println(info);
}
void audio_lasthost(const char *info){ //stream URL played
Serial.print("lasthost ");Serial.println(info);
}
void audio_eof_speech(const char *info){
Serial.print("eof_speech ");Serial.println(info);
}

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// the pin assignment matches the Olimex ADF board
#include "Arduino.h"
#include "WiFi.h"
#include "SPI.h"
#include "SD.h"
#include "FS.h"
#include "Wire.h"
#include "ES8388.h" // https://github.com/maditnerd/es8388
#include "Audio.h" // https://github.com/schreibfaul1/ESP32-audioI2S
#define SD_CS 21
// GPIOs for SPI
#define SPI_MOSI 13
#define SPI_MISO 12
#define SPI_SCK 14
// I2S GPIOs
#define I2S_SDOUT 26
#define I2S_BCLK 5
#define I2S_LRCK 25
#define I2S_MCLK 0
// I2C GPIOs
#define IIC_CLK 23
#define IIC_DATA 18
// Amplifier enable
#define GPIO_PA_EN 19
char ssid[] = "xxxxxxxxx";
char password[] = "xxxxxxxxx";
int volume = 80; // 0...100
ES8388 es;
Audio audio;
//----------------------------------------------------------------------------------------------------------------------
void setup()
{
Serial.begin(115200);
Serial.println("\r\nReset");
Serial.printf_P(PSTR("Free mem=%d\n"), ESP.getFreeHeap());
SPI.begin(SPI_SCK, SPI_MISO, SPI_MOSI);
SPI.setFrequency(1000000);
SD.begin(SD_CS);
WiFi.mode(WIFI_STA);
WiFi.begin(ssid, password);
while(WiFi.status() != WL_CONNECTED) {
Serial.print(".");
delay(100);
}
Serial.printf_P(PSTR("Connected\r\nRSSI: "));
Serial.print(WiFi.RSSI());
Serial.print(" IP: ");
Serial.println(WiFi.localIP());
Serial.printf("Connect to ES8388 codec... ");
while (not es.begin(IIC_DATA, IIC_CLK))
{
Serial.printf("Failed!\n");
delay(1000);
}
Serial.printf("OK\n");
es.volume(ES8388::ES_MAIN, volume);
es.volume(ES8388::ES_OUT1, volume);
es.volume(ES8388::ES_OUT2, volume);
es.mute(ES8388::ES_OUT1, false);
es.mute(ES8388::ES_OUT2, false);
es.mute(ES8388::ES_MAIN, false);
// Enable amplifier
pinMode(GPIO_PA_EN, OUTPUT);
digitalWrite(GPIO_PA_EN, HIGH);
audio.setPinout(I2S_BCLK, I2S_LRCK, I2S_SDOUT);
audio.i2s_mclk_pin_select(I2S_MCLK);
audio.setVolume(21); // 0...21
audio.connecttohost("http://mp3channels.webradio.antenne.de:80/oldies-but-goldies");
// audio.connecttoFS(SD, "320k_test.mp3");
// audio.connecttospeech("Wenn die Hunde schlafen, kann der Wolf gut Schafe stehlen.", "de");
}
//----------------------------------------------------------------------------------------------------------------------
void loop()
{
audio.loop();
}
//----------------------------------------------------------------------------------------------------------------------
// optional
void audio_info(const char *info){
Serial.print("info "); Serial.println(info);
}
void audio_id3data(const char *info){ //id3 metadata
Serial.print("id3data ");Serial.println(info);
}
void audio_eof_mp3(const char *info){ //end of file
Serial.print("eof_mp3 ");Serial.println(info);
}
void audio_showstation(const char *info){
Serial.print("station ");Serial.println(info);
}
void audio_showstreaminfo(const char *info){
Serial.print("streaminfo ");Serial.println(info);
}
void audio_showstreamtitle(const char *info){
Serial.print("streamtitle ");Serial.println(info);
}
void audio_bitrate(const char *info){
Serial.print("bitrate ");Serial.println(info);
}
void audio_commercial(const char *info){ //duration in sec
Serial.print("commercial ");Serial.println(info);
}
void audio_icyurl(const char *info){ //homepage
Serial.print("icyurl ");Serial.println(info);
}
void audio_lasthost(const char *info){ //stream URL played
Serial.print("lasthost ");Serial.println(info);
}
void audio_eof_speech(const char *info){
Serial.print("eof_speech ");Serial.println(info);
}

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//**********************************************************************************************************
//* audioI2S-- I2S audiodecoder for ESP32, *
//**********************************************************************************************************
//
// first release on 11/2018
// Version 3 , Jul.02/2020
//
//
// THE SOFTWARE IS PROVIDED "AS IS" FOR PRIVATE USE ONLY, IT IS NOT FOR COMMERCIAL USE IN WHOLE OR PART OR CONCEPT.
// FOR PERSONAL USE IT IS SUPPLIED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE
// WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHOR
// OR COPYRIGHT HOLDER BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
// OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE
//
#include "Arduino.h"
#include "WiFiMulti.h"
#include "Audio.h"
#include "SPI.h"
#include "SD.h"
#include "FS.h"
// Digital I/O used
#define SD_CS 5
#define SPI_MOSI 23
#define SPI_MISO 19
#define SPI_SCK 18
#define I2S_DOUT 25
#define I2S_BCLK 27
#define I2S_LRC 26
Audio audio;
WiFiMulti wifiMulti;
String ssid = "xxxxx";
String password = "xxxxx";
void setup() {
pinMode(SD_CS, OUTPUT); digitalWrite(SD_CS, HIGH);
SPI.begin(SPI_SCK, SPI_MISO, SPI_MOSI);
SPI.setFrequency(1000000);
Serial.begin(115200);
SD.begin(SD_CS);
WiFi.mode(WIFI_STA);
wifiMulti.addAP(ssid.c_str(), password.c_str());
wifiMulti.run();
if(WiFi.status() != WL_CONNECTED){
WiFi.disconnect(true);
wifiMulti.run();
}
audio.setPinout(I2S_BCLK, I2S_LRC, I2S_DOUT);
audio.setVolume(12); // 0...21
// audio.connecttoFS(SD, "/320k_test.mp3");
// audio.connecttoFS(SD, "test.wav");
// audio.connecttohost("http://www.wdr.de/wdrlive/media/einslive.m3u");
// audio.connecttohost("http://macslons-irish-pub-radio.com/media.asx");
// audio.connecttohost("http://mp3.ffh.de/radioffh/hqlivestream.aac"); // 128k aac
audio.connecttohost("http://mp3.ffh.de/radioffh/hqlivestream.mp3"); // 128k mp3
}
void loop()
{
audio.loop();
if(Serial.available()){ // put streamURL in serial monitor
audio.stopSong();
String r=Serial.readString(); r.trim();
if(r.length()>5) audio.connecttohost(r.c_str());
log_i("free heap=%i", ESP.getFreeHeap());
}
}
// optional
void audio_info(const char *info){
Serial.print("info "); Serial.println(info);
}
void audio_id3data(const char *info){ //id3 metadata
Serial.print("id3data ");Serial.println(info);
}
void audio_eof_mp3(const char *info){ //end of file
Serial.print("eof_mp3 ");Serial.println(info);
}
void audio_showstation(const char *info){
Serial.print("station ");Serial.println(info);
}
void audio_showstreamtitle(const char *info){
Serial.print("streamtitle ");Serial.println(info);
}
void audio_bitrate(const char *info){
Serial.print("bitrate ");Serial.println(info);
}
void audio_commercial(const char *info){ //duration in sec
Serial.print("commercial ");Serial.println(info);
}
void audio_icyurl(const char *info){ //homepage
Serial.print("icyurl ");Serial.println(info);
}
void audio_lasthost(const char *info){ //stream URL played
Serial.print("lasthost ");Serial.println(info);
}

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//**********************************************************************************************************
//* audioI2S-- I2S audiodecoder for M5Stack Core2 *
//**********************************************************************************************************
//
// first release on May.12/2021
//
//
// THE SOFTWARE IS PROVIDED "AS IS" FOR PRIVATE USE ONLY, IT IS NOT FOR COMMERCIAL USE IN WHOLE OR PART OR CONCEPT.
// FOR PERSONAL USE IT IS SUPPLIED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE
// WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHOR
// OR COPYRIGHT HOLDER BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
// OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE
//
#include <M5Core2.h>
#include "Audio.h"
// Digital I/O used
#define SD_CS 4
#define SD_MOSI 23
#define SD_MISO 38
#define SD_SCK 18
#define I2S_DOUT 2
#define I2S_BCLK 12
#define I2S_LRC 0
Audio audio;
String ssid = "xxxxxx";
String password = "xxxxxx";
void setup() {
M5.begin(true, true, true, true);
M5.Axp.SetSpkEnable(true);
M5.Lcd.fillScreen(BLACK);
M5.Lcd.setTextColor(WHITE);
M5.Lcd.setTextSize(2);
pinMode(SD_CS, OUTPUT);
digitalWrite(SD_CS, HIGH);
SPI.begin(SD_SCK, SD_MISO, SD_MOSI);
SPI.setFrequency(1000000);
SD.begin(SD_CS);
audio.setPinout(I2S_BCLK, I2S_LRC, I2S_DOUT);
audio.setVolume(15); // 0...21
WiFi.mode(WIFI_STA);
WiFi.begin(ssid.c_str(), password.c_str());
while (!WiFi.isConnected()) {
delay(10);
}
ESP_LOGI(TAG, "Connected");
ESP_LOGI(TAG, "Starting MP3...\n");
// audio.connecttoFS(SD, "/320k_test.mp3");
// audio.connecttoFS(SD, "test.wav");
audio.connecttohost("http://air.ofr.fm:8008/jazz/mp3/128");
// audio.connecttospeech("Миска вареників з картоплею та шкварками, змащених салом!", "uk-UA");
}
void loop() {
audio.loop();
if(Serial.available()){ // put streamURL in serial monitor
audio.stopSong();
String r=Serial.readString();
r.trim();
if(r.length()>5) audio.connecttohost(r.c_str());
log_i("free heap=%i", ESP.getFreeHeap());
}
}
// optional
void audio_info(const char *info){
Serial.print("info "); Serial.println(info);
}
void audio_id3data(const char *info){ //id3 metadata
Serial.print("id3data ");Serial.println(info);
}
void audio_eof_mp3(const char *info){ //end of file
Serial.print("eof_mp3 ");Serial.println(info);
}
void audio_showstation(const char *info){
Serial.print("station ");Serial.println(info);
}
void audio_showstreamtitle(const char *info){
Serial.print("streamtitle ");Serial.println(info);
}
void audio_bitrate(const char *info){
Serial.print("bitrate ");Serial.println(info);
}
void audio_commercial(const char *info){ //duration in sec
Serial.print("commercial ");Serial.println(info);
}
void audio_icyurl(const char *info){ //homepage
Serial.print("icyurl ");Serial.println(info);
}
void audio_lasthost(const char *info){ //stream URL played
Serial.print("lasthost ");Serial.println(info);
}
void audio_eof_speech(const char *info){
Serial.print("eof_speech ");Serial.println(info);
}

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// M5Stack Node support
// thanks to Cellie - issue #35 25.Apr.2020
// M5Stack board with Node base also need a MCLK signal on GPIO0.
#include <wm8978.h> /* https://github.com/CelliesProjects/wm8978-esp32 */
#include <Audio.h> /* https://github.com/schreibfaul1/ESP32-audioI2S */
/* M5Stack Node WM8978 I2C pins */
#define I2C_SDA 21
#define I2C_SCL 22
/* M5Stack Node I2S pins */
#define I2S_BCK 5
#define I2S_WS 13
#define I2S_DOUT 2
#define I2S_DIN 34
/* M5Stack WM8978 MCLK gpio number */
#define I2S_MCLKPIN 0
WM8978 dac;
Audio audio;
void setup() {
/* Setup wm8978 I2C interface */
if (!dac.begin(I2C_SDA, I2C_SCL)) {
ESP_LOGE(TAG, "Error setting up dac. System halted");
while (1) delay(100);
}
/* Setup wm8978 I2S interface */
audio.setPinout(I2S_BCK, I2S_WS, I2S_DOUT, I2S_DIN);
/* Setup wm8978 MCLK - for example M5Stack Node needs MCLK on GPIO 0 */
audio.i2s_mclk_pin_select(I2S_MCLKPIN);
WiFi.begin("xxx", "xxx");
while (!WiFi.isConnected()) {
delay(10);
}
ESP_LOGI(TAG, "Connected");
ESP_LOGI(TAG, "Starting MP3...\n");
audio.connecttohost("http://icecast.omroep.nl/3fm-bb-mp3");
dac.setSPKvol(40); /* max 63 */
dac.setHPvol(32, 32);
}
void loop() {
audio.loop();
}

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//**********************************************************************************************************
//* audioI2S-- I2S audiodecoder for M5StickC Plus and SPK HAT *
//**********************************************************************************************************
//
// first release on May.12/2021
//
//
// THE SOFTWARE IS PROVIDED "AS IS" FOR PRIVATE USE ONLY, IT IS NOT FOR COMMERCIAL USE IN WHOLE OR PART OR CONCEPT.
// FOR PERSONAL USE IT IS SUPPLIED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE
// WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHOR
// OR COPYRIGHT HOLDER BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
// OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE
//
#include <M5StickCPlus.h>
#include "Audio.h"
Audio audio = Audio(true);
String ssid = "xxxxxxxx";
String password = "xxxxxxxx";
void setup() {
M5.begin(false); //Lcd disabled to reduce noise
M5.Axp.ScreenBreath(1); //Lower Lcd backlight
pinMode(36, INPUT);
gpio_pulldown_dis(GPIO_NUM_25);
gpio_pullup_dis(GPIO_NUM_25);
M5.Beep.tone(44100); //Built-in buzzer tone
M5.Beep.end(); //disabled
audio.setVolume(15); // 0...21
WiFi.mode(WIFI_STA);
WiFi.begin(ssid.c_str(), password.c_str());
while (!WiFi.isConnected()) {
delay(10);
}
ESP_LOGI(TAG, "Connected");
ESP_LOGI(TAG, "Starting MP3...\n");
audio.connecttohost("http://air.ofr.fm:8008/jazz/mp3/128");
// audio.connecttospeech("Миска вареників з картоплею та шкварками, змащених салом!", "uk-UA");
}
void loop() {
audio.loop();
if(Serial.available()){ // put streamURL in serial monitor
audio.stopSong();
String r=Serial.readString();
r.trim();
if(r.length()>5) audio.connecttohost(r.c_str());
log_i("free heap=%i", ESP.getFreeHeap());
}
}
// optional
void audio_info(const char *info){
Serial.print("info "); Serial.println(info);
}
void audio_id3data(const char *info){ //id3 metadata
Serial.print("id3data ");Serial.println(info);
}
void audio_eof_mp3(const char *info){ //end of file
Serial.print("eof_mp3 ");Serial.println(info);
}
void audio_showstation(const char *info){
Serial.print("station ");Serial.println(info);
}
void audio_showstreamtitle(const char *info){
Serial.print("streamtitle ");Serial.println(info);
}
void audio_bitrate(const char *info){
Serial.print("bitrate ");Serial.println(info);
}
void audio_commercial(const char *info){ //duration in sec
Serial.print("commercial ");Serial.println(info);
}
void audio_icyurl(const char *info){ //homepage
Serial.print("icyurl ");Serial.println(info);
}
void audio_lasthost(const char *info){ //stream URL played
Serial.print("lasthost ");Serial.println(info);
}
void audio_eof_speech(const char *info){
Serial.print("eof_speech ");Serial.println(info);
}

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#include "Arduino.h"
#include "WiFi.h"
#include "Audio.h"
// Digital I/O used
#define I2S_DOUT 25
#define I2S_BCLK 27
#define I2S_LRC 26
Audio audio;
String ssid = "******";
String password = "******";
//****************************************************************************************
// A U D I O _ T A S K *
//****************************************************************************************
struct audioMessage{
uint8_t cmd;
const char* txt;
uint32_t value;
uint32_t ret;
} audioTxMessage, audioRxMessage;
enum : uint8_t { SET_VOLUME, GET_VOLUME, CONNECTTOHOST, CONNECTTOSD };
QueueHandle_t audioSetQueue = NULL;
QueueHandle_t audioGetQueue = NULL;
void CreateQueues(){
audioSetQueue = xQueueCreate(10, sizeof(struct audioMessage));
audioGetQueue = xQueueCreate(10, sizeof(struct audioMessage));
}
void audioTask(void *parameter) {
CreateQueues();
if(!audioSetQueue || !audioGetQueue){
log_e("queues are not initialized");
while(true){;} // endless loop
}
struct audioMessage audioRxTaskMessage;
struct audioMessage audioTxTaskMessage;
audio.setPinout(I2S_BCLK, I2S_LRC, I2S_DOUT);
audio.setVolume(15); // 0...21
while(true){
if(xQueueReceive(audioSetQueue, &audioRxTaskMessage, 1) == pdPASS) {
if(audioRxTaskMessage.cmd == SET_VOLUME){
audioTxTaskMessage.cmd = SET_VOLUME;
audio.setVolume(audioRxTaskMessage.value);
audioTxTaskMessage.ret = 1;
xQueueSend(audioGetQueue, &audioTxTaskMessage, portMAX_DELAY);
}
else if(audioRxTaskMessage.cmd == CONNECTTOHOST){
audioTxTaskMessage.cmd = CONNECTTOHOST;
audioTxTaskMessage.ret = audio.connecttohost(audioRxTaskMessage.txt);
xQueueSend(audioGetQueue, &audioTxTaskMessage, portMAX_DELAY);
}
else if(audioRxTaskMessage.cmd == CONNECTTOSD){
audioTxTaskMessage.cmd = CONNECTTOSD;
audioTxTaskMessage.ret = audio.connecttoSD(audioRxTaskMessage.txt);
xQueueSend(audioGetQueue, &audioTxTaskMessage, portMAX_DELAY);
}
else if(audioRxTaskMessage.cmd == GET_VOLUME){
audioTxTaskMessage.cmd = GET_VOLUME;
audioTxTaskMessage.ret = audio.getVolume();
xQueueSend(audioGetQueue, &audioTxTaskMessage, portMAX_DELAY);
}
else{
log_i("error");
}
}
audio.loop();
}
}
void audioInit() {
xTaskCreatePinnedToCore(
audioTask, /* Function to implement the task */
"audioplay", /* Name of the task */
5000, /* Stack size in words */
NULL, /* Task input parameter */
2 | portPRIVILEGE_BIT, /* Priority of the task */
NULL, /* Task handle. */
1 /* Core where the task should run */
);
}
audioMessage transmitReceive(audioMessage msg){
xQueueSend(audioSetQueue, &msg, portMAX_DELAY);
if(xQueueReceive(audioGetQueue, &audioRxMessage, portMAX_DELAY) == pdPASS){
if(msg.cmd != audioRxMessage.cmd){
log_e("wrong reply from message queue");
}
}
return audioRxMessage;
}
void audioSetVolume(uint8_t vol){
audioTxMessage.cmd = SET_VOLUME;
audioTxMessage.value = vol;
audioMessage RX = transmitReceive(audioTxMessage);
}
uint8_t audioGetVolume(){
audioTxMessage.cmd = GET_VOLUME;
audioMessage RX = transmitReceive(audioTxMessage);
return RX.ret;
}
bool audioConnecttohost(const char* host){
audioTxMessage.cmd = CONNECTTOHOST;
audioTxMessage.txt = host;
audioMessage RX = transmitReceive(audioTxMessage);
return RX.ret;
}
bool audioConnecttoSD(const char* filename){
audioTxMessage.cmd = CONNECTTOSD;
audioTxMessage.txt = filename;
audioMessage RX = transmitReceive(audioTxMessage);
return RX.ret;
}
//****************************************************************************************
// S E T U P *
//****************************************************************************************
void setup() {
Serial.begin(115200);
WiFi.begin(ssid.c_str(), password.c_str());
while (WiFi.status() != WL_CONNECTED) delay(1500);
audioInit();
audioConnecttohost("http://mp3.ffh.de/radioffh/hqlivestream.mp3");
audioSetVolume(15);
log_i("current volume is: %d", audioGetVolume());
}
//****************************************************************************************
// L O O P *
//****************************************************************************************
void loop(){
// your own code here
}
//*****************************************************************************************
// E V E N T S *
//*****************************************************************************************
void audio_info(const char *info){
Serial.print("info "); Serial.println(info);
}

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//**********************************************************************************************************
//* audioI2S-- I2S audiodecoder for ESP32, SdFat example *
//**********************************************************************************************************
//
// first release on 05/2020
// updated on Sep. 27, 2021
/*
⒈ install SdFat V2 from https://github.com/greiman/SdFat
⒉ activate "SDFATFS_USED" in Audio.h
⒊ activate "#define USE_UTF8_LONG_NAMES 1" in SdFatConfig.h
*/
#include "Arduino.h"
#include "Audio.h"
#include "SPI.h"
// Digital I/O used
#define SD_CS 5
#define SPI_MOSI 23
#define SPI_MISO 19
#define SPI_SCK 18
#define I2S_DOUT 25
#define I2S_BCLK 27
#define I2S_LRC 26
Audio audio;
void setup() {
pinMode(SD_CS, OUTPUT); digitalWrite(SD_CS, HIGH);
SPI.begin(SPI_SCK, SPI_MISO, SPI_MOSI);
SPI.setFrequency(1000000);
Serial.begin(115200);
SD.begin(SD_CS);
audio.setPinout(I2S_BCLK, I2S_LRC, I2S_DOUT);
audio.setVolume(12); // 0...21
// audio.connecttoFS(SD, "test.mp3");
audio.connecttoFS(SD, "良い一日私の友達.mp3");
}
void loop()
{
audio.loop();
}
// optional
void audio_info(const char *info){
Serial.print("info "); Serial.println(info);
}
void audio_id3data(const char *info){ //id3 metadata
Serial.print("id3data ");Serial.println(info);
}
void audio_eof_mp3(const char *info){ //end of file
Serial.print("eof_mp3 ");Serial.println(info);
}

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#include <Arduino.h>
#include <Preferences.h>
#include <SPI.h>
#include <WiFi.h>
#include "tft.h" //see my repository at github "https://github.com/schreibfaul1/ESP32-TFT-Library-ILI9486"
#include "Audio.h" //see my repository at github "https://github.com/schreibfaul1/ESP32-audioI2S"
#define TFT_CS 22
#define TFT_DC 21
#define TP_CS 14 //16
#define TP_IRQ 39
#define SPI_MOSI 23
#define SPI_MISO 19
#define SPI_SCK 18
#define I2S_DOUT 25
#define I2S_BCLK 27
#define I2S_LRC 26
Preferences pref;
TFT tft;
TP tp(TP_CS, TP_IRQ);
Audio audio;
String ssid = "*****";
String password = "*****";
String stations[] ={
"0n-80s.radionetz.de:8000/0n-70s.mp3",
"mediaserv30.live-streams.nl:8000/stream",
"www.surfmusic.de/m3u/100-5-das-hitradio,4529.m3u",
"stream.1a-webradio.de/deutsch/mp3-128/vtuner-1a",
"mp3.ffh.de/radioffh/hqlivestream.aac", // 128k aac
"www.antenne.de/webradio/antenne.m3u",
"listen.rusongs.ru/ru-mp3-128",
"edge.audio.3qsdn.com/senderkw-mp3",
"macslons-irish-pub-radio.com/media.asx",
};
//some global variables
uint8_t max_volume = 21;
uint8_t max_stations = 0; //will be set later
uint8_t cur_station = 0; //current station(nr), will be set later
uint8_t cur_volume = 0; //will be set from stored preferences
int8_t cur_btn =-1; //current button (, -1 means idle)
enum action{VOLUME_UP=0, VOLUME_DOWN=1, STATION_UP=2, STATION_DOWN=3};
enum staus {RELEASED=0, PRESSED=1};
struct _btns{
uint16_t x; //PosX
uint16_t y; //PosY
uint16_t w; //Width
uint16_t h; //Hight
uint8_t a; //Action
uint8_t s; //Status
};
typedef _btns btns;
btns btn[4];
//**************************************************************************************************
// G U I *
//**************************************************************************************************
void draw_button(btns b){
uint16_t color=TFT_BLACK;
uint8_t r=4, o=r*3;
if(b.s==RELEASED) color=TFT_GOLD;
if(b.s==PRESSED) color=TFT_DEEPSKYBLUE;
tft.drawRoundRect(b.x, b.y, b.w, b.h, r, color);
switch(b.a){
case VOLUME_UP:
tft.fillTriangle(b.x+b.w/2, b.y+o, b.x+o, b.y+b.h-o, b.x+b.w-o, b.y+b.h-o, color); break;
case VOLUME_DOWN:
tft.fillTriangle(b.x+o, b.y+o, b.x+b.w/2, b.y+b.h-o, b.x+b.w-o, b.y+o, color); break;
case STATION_UP:
tft.fillTriangle(b.x+o, b.y+o, b.x+o, b.y+b.h-o, b.x+b.w-o, b.y+b.h/2, color); break;
case STATION_DOWN:
tft.fillTriangle(b.x+b.w-o, b.y+o, b.x+o, b.y+b.h/2, b.x+b.w-o, b.y+b.h-o, color); break;
}
}
void write_stationNr(uint8_t nr){
tft.fillRect(80, 250, 80, 60, TFT_BLACK);
String snr = String(nr);
if(snr.length()<2) snr = "0"+snr;
tft.setCursor(98, 255);
tft.setFont(Times_New_Roman66x53);
tft.setTextColor(TFT_YELLOW);
tft.print(snr);
}
void write_volume(uint8_t vol){
tft.fillRect(320, 250, 80, 60, TFT_BLACK);
String svol = String(vol);
if(svol.length()<2) svol = "0"+svol;
tft.setCursor(338, 255);
tft.setFont(Times_New_Roman66x53);
tft.setTextColor(TFT_YELLOW);
tft.print(svol);
}
void write_stationName(String sName){
tft.fillRect(0, 0, 480, 100, TFT_BLACK);
tft.setFont(Times_New_Roman43x35);
tft.setTextColor(TFT_CORNSILK);
tft.setCursor(20, 20);
tft.print(sName);
}
void write_streamTitle(String sTitle){
tft.fillRect(0, 100, 480, 150, TFT_BLACK);
tft.setFont(Times_New_Roman43x35);
tft.setTextColor(TFT_LIGHTBLUE);
tft.setCursor(20, 100);
int l = tft.writeText((const uint8_t*) sTitle.c_str(), 100 + 150); // do not write under y=250
if(l < sTitle.length()){
// sTitle has been shortened, is too long for the display
}
}
//**************************************************************************************************
// S E T U P *
//**************************************************************************************************
void setup() {
btn[0].x= 20; btn[0].y=250; btn[0].w=60; btn[0].h=60; btn[0].a=STATION_DOWN; btn[0].s=RELEASED;
btn[1].x=160; btn[1].y=250; btn[1].w=60; btn[1].h=60; btn[1].a=STATION_UP; btn[1].s=RELEASED;
btn[2].x=260; btn[2].y=250; btn[2].w=60; btn[2].h=60; btn[2].a=VOLUME_UP; btn[2].s=RELEASED;
btn[3].x=400; btn[3].y=250; btn[3].w=60; btn[3].h=60; btn[3].a=VOLUME_DOWN; btn[3].s=RELEASED;
max_stations= sizeof(stations)/sizeof(stations[0]); log_i("max stations %i", max_stations);
Serial.begin(115200);
SPI.begin(SPI_SCK, SPI_MISO, SPI_MOSI);
pref.begin("WebRadio", false); // instance of preferences for defaults (station, volume ...)
if(pref.getShort("volume", 1000) == 1000){ // if that: pref was never been initialized
pref.putShort("volume", 10);
pref.putShort("station", 0);
}
else{ // get the stored values
cur_station = pref.getShort("station");
cur_volume = pref.getShort("volume");
}
WiFi.begin(ssid.c_str(), password.c_str());
while (WiFi.status() != WL_CONNECTED){
delay(2000);
Serial.print(".");
}
log_i("Connect to %s", WiFi.SSID().c_str());
tft.begin(TFT_CS, TFT_DC, SPI_MOSI, SPI_MISO, SPI_SCK);
tft.setRotation(3);
tp.setRotation(3);
tft.setFont(Times_New_Roman43x35);
tft.fillScreen(TFT_BLACK);
audio.setPinout(I2S_BCLK, I2S_LRC, I2S_DOUT);
audio.setVolume(cur_volume); // 0...21
audio.connecttohost(stations[cur_station].c_str());
for(uint8_t i=0; i<(sizeof(btn)/sizeof(*btn)); i++) draw_button(btn[i]);
write_volume(cur_volume);
write_stationNr(cur_station);
}
//**************************************************************************************************
// L O O P *
//**************************************************************************************************
void loop()
{
audio.loop();
tp.loop();
}
//**************************************************************************************************
// E V E N T S *
//**************************************************************************************************
void audio_info(const char *info){
Serial.print("audio_info: "); Serial.println(info);
}
void audio_showstation(const char *info){
write_stationName(String(info));
}
void audio_showstreamtitle(const char *info){
String sinfo=String(info);
sinfo.replace("|", "\n");
write_streamTitle(sinfo);
}
void tp_pressed(uint16_t x, uint16_t y){
for(uint8_t i=0; i<(sizeof(btn)/sizeof(*btn)); i++){
if(x>btn[i].x && (x<btn[i].x+btn[i].w)){
if(y>btn[i].y && (y<btn[i].y+btn[i].h)){
cur_btn=i;
btn[cur_btn].s=PRESSED;
draw_button(btn[cur_btn]);
}
}
}
}
void tp_released(){
if(cur_btn !=-1){
btn[cur_btn].s=RELEASED;
draw_button(btn[cur_btn]);
switch(btn[cur_btn].a){
case VOLUME_UP: if(cur_volume<max_volume){
cur_volume++;
write_volume(cur_volume);
audio.setVolume(cur_volume);
pref.putShort("volume", cur_volume);} // store the current volume in nvs
break;
case VOLUME_DOWN: if(cur_volume>0){
cur_volume-- ;
write_volume(cur_volume);
audio.setVolume(cur_volume);
pref.putShort("volume", cur_volume);} // store the current volume in nvs
break;
case STATION_UP: if(cur_station<max_stations-1){
cur_station++;
write_stationNr(cur_station);
tft.fillRect(0, 0, 480, 250, TFT_BLACK);
audio.connecttohost(stations[cur_station].c_str());
pref.putShort("station", cur_station);} // store the current station in nvs
break;
case STATION_DOWN:if(cur_station>0){
cur_station--;
write_stationNr(cur_station);
tft.fillRect(0, 0, 480, 250, TFT_BLACK);
audio.connecttohost(stations[cur_station].c_str());
pref.putShort("station", cur_station);} // store the current station in nvs
break;
}
}
cur_btn=-1;
}

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// this is the same example as Webradio_I2S.ino only with an IR remote control
#include <Arduino.h>
#include <Preferences.h>
#include <SPI.h>
#include <WiFi.h>
#include "tft.h" //see my repository at github "https://github.com/schreibfaul1/ESP32-TFT-Library-ILI9486"
#include "Audio.h" //see my repository at github "https://github.com/schreibfaul1/ESP32-audioI2S"
#include "IR.h" //see my repository at github "ESP32-IR-Remote-Control"
#define TFT_CS 22
#define TFT_DC 21
#define TP_CS 16
#define TP_IRQ 39
#define SPI_MOSI 23
#define SPI_MISO 19
#define SPI_SCK 18
#define I2S_DOUT 25
#define I2S_BCLK 27
#define I2S_LRC 26
#define IR_PIN 34
Preferences pref;
TFT tft; // @suppress("Abstract class cannot be instantiated")
TP tp(TP_CS, TP_IRQ);
Audio audio;
IR ir(IR_PIN); // do not change the objectname, it must be "ir"
String ssid = "*********";
String password = "*********";
String stations[] ={
"0n-80s.radionetz.de:8000/0n-70s.mp3",
"mediaserv30.live-streams.nl:8000/stream",
"www.surfmusic.de/m3u/100-5-das-hitradio,4529.m3u",
"stream.1a-webradio.de/deutsch/mp3-128/vtuner-1a",
"mp3.ffh.de/radioffh/hqlivestream.aac", // 128k aac
"www.antenne.de/webradio/antenne.m3u",
"listen.rusongs.ru/ru-mp3-128",
"edge.audio.3qsdn.com/senderkw-mp3",
"macslons-irish-pub-radio.com/media.asx",
};
//some global variables
uint8_t max_volume = 21;
uint8_t max_stations = 0; //will be set later
uint8_t cur_station = 0; //current station(nr), will be set later
uint8_t cur_volume = 0; //will be set from stored preferences
int8_t cur_btn =-1; //current button (, -1 means idle)
enum action{VOLUME_UP=0, VOLUME_DOWN=1, STATION_UP=2, STATION_DOWN=3};
enum staus {RELEASED=0, PRESSED=1};
struct _btns{
uint16_t x; //PosX
uint16_t y; //PosY
uint16_t w; //Width
uint16_t h; //Hight
uint8_t a; //Action
uint8_t s; //Status
};
typedef _btns btns;
btns btn[4];
//**************************************************************************************************
// G U I *
//**************************************************************************************************
void draw_button(btns b){
uint16_t color=TFT_BLACK;
uint8_t r=4, o=r*3;
if(b.s==RELEASED) color=TFT_GOLD;
if(b.s==PRESSED) color=TFT_DEEPSKYBLUE;
tft.drawRoundRect(b.x, b.y, b.w, b.h, r, color);
switch(b.a){
case VOLUME_UP:
tft.fillTriangle(b.x+b.w/2, b.y+o, b.x+o, b.y+b.h-o, b.x+b.w-o, b.y+b.h-o, color); break;
case VOLUME_DOWN:
tft.fillTriangle(b.x+o, b.y+o, b.x+b.w/2, b.y+b.h-o, b.x+b.w-o, b.y+o, color); break;
case STATION_UP:
tft.fillTriangle(b.x+o, b.y+o, b.x+o, b.y+b.h-o, b.x+b.w-o, b.y+b.h/2, color); break;
case STATION_DOWN:
tft.fillTriangle(b.x+b.w-o, b.y+o, b.x+o, b.y+b.h/2, b.x+b.w-o, b.y+b.h-o, color); break;
}
}
void write_stationNr(uint8_t nr){
tft.fillRect(80, 250, 80, 60, TFT_BLACK);
String snr = String(nr);
if(snr.length()<2) snr = "0"+snr;
tft.setCursor(98, 255);
tft.setFont(Times_New_Roman66x53);
tft.setTextColor(TFT_YELLOW);
tft.print(snr);
}
void write_volume(uint8_t vol){
tft.fillRect(320, 250, 80, 60, TFT_BLACK);
String svol = String(vol);
if(svol.length()<2) svol = "0"+svol;
tft.setCursor(338, 255);
tft.setFont(Times_New_Roman66x53);
tft.setTextColor(TFT_YELLOW);
tft.print(svol);
}
void write_stationName(String sName){
tft.fillRect(0, 0, 480, 100, TFT_BLACK);
tft.setFont(Times_New_Roman43x35);
tft.setTextColor(TFT_CORNSILK);
tft.setCursor(20, 20);
tft.print(sName);
}
void write_streamTitle(String sTitle){
tft.fillRect(0, 100, 480, 150, TFT_BLACK);
tft.setFont(Times_New_Roman43x35);
tft.setTextColor(TFT_LIGHTBLUE);
tft.setCursor(20, 100);
tft.print(sTitle);
}
void volume_up(){
if(cur_volume < max_volume){
cur_volume++;
write_volume(cur_volume);
audio.setVolume(cur_volume);
pref.putShort("volume", cur_volume);} // store the current volume in nvs
}
void volume_down(){
if(cur_volume>0){
cur_volume-- ;
write_volume(cur_volume);
audio.setVolume(cur_volume);
pref.putShort("volume", cur_volume);} // store the current volume in nvs
}
void station_up(){
if(cur_station < max_stations-1){
cur_station++;
write_stationNr(cur_station);
tft.fillRect(0, 0, 480, 250, TFT_BLACK);
audio.connecttohost(stations[cur_station].c_str());
pref.putShort("station", cur_station);} // store the current station in nvs
}
void station_down(){
if(cur_station > 0){
cur_station--;
write_stationNr(cur_station);
tft.fillRect(0, 0, 480, 250, TFT_BLACK);
audio.connecttohost(stations[cur_station].c_str());
pref.putShort("station", cur_station);} // store the current station in nvs
}
//**************************************************************************************************
// S E T U P *
//**************************************************************************************************
void setup() {
btn[0].x= 20; btn[0].y=250; btn[0].w=60; btn[0].h=60; btn[0].a=STATION_DOWN; btn[0].s=RELEASED;
btn[1].x=160; btn[1].y=250; btn[1].w=60; btn[1].h=60; btn[1].a=STATION_UP; btn[1].s=RELEASED;
btn[2].x=260; btn[2].y=250; btn[2].w=60; btn[2].h=60; btn[2].a=VOLUME_UP; btn[2].s=RELEASED;
btn[3].x=400; btn[3].y=250; btn[3].w=60; btn[3].h=60; btn[3].a=VOLUME_DOWN; btn[3].s=RELEASED;
max_stations= sizeof(stations)/sizeof(stations[0]); log_i("max stations %i", max_stations);
Serial.begin(115200);
SPI.begin(SPI_SCK, SPI_MISO, SPI_MOSI);
pref.begin("WebRadio", false); // instance of preferences for defaults (station, volume ...)
if(pref.getShort("volume", 1000) == 1000){ // if that: pref was never been initialized
pref.putShort("volume", 10);
pref.putShort("station", 0);
}
else{ // get the stored values
cur_station = pref.getShort("station");
cur_volume = pref.getShort("volume");
}
WiFi.disconnect();
WiFi.begin(ssid.c_str(), password.c_str());
while (WiFi.status() != WL_CONNECTED) {delay(1500); Serial.print(".");}
log_i("Connected to %s", WiFi.SSID().c_str());
tft.begin(TFT_CS, TFT_DC, SPI_MOSI, SPI_MISO, SPI_SCK);
tft.setFrequency(20000000);
tft.setRotation(3);
tp.setRotation(3);
tft.setFont(Times_New_Roman43x35);
tft.fillScreen(TFT_BLACK);
ir.begin(); // Init InfraredDecoder
audio.setPinout(I2S_BCLK, I2S_LRC, I2S_DOUT);
audio.setVolume(cur_volume); // 0...21
audio.connecttohost(stations[cur_station].c_str());
for(uint8_t i=0; i<(sizeof(btn)/sizeof(*btn)); i++) draw_button(btn[i]);
write_volume(cur_volume);
write_stationNr(cur_station);
}
//**************************************************************************************************
// L O O P *
//**************************************************************************************************
void loop()
{
audio.loop();
tp.loop();
ir.loop();
}
//**************************************************************************************************
// E V E N T S *
//**************************************************************************************************
void audio_info(const char *info){
Serial.print("audio_info: "); Serial.println(info);
}
void audio_showstation(const char *info){
write_stationName(String(info));
}
void audio_showstreamtitle(const char *info){
String sinfo=String(info);
sinfo.replace("|", "\n");
write_streamTitle(sinfo);
}
void tp_pressed(uint16_t x, uint16_t y){
for(uint8_t i=0; i<(sizeof(btn)/sizeof(*btn)); i++){
if(x>btn[i].x && (x<btn[i].x+btn[i].w)){
if(y>btn[i].y && (y<btn[i].y+btn[i].h)){
cur_btn=i;
btn[cur_btn].s=PRESSED;
draw_button(btn[cur_btn]);
}
}
}
}
void tp_released(){
if(cur_btn !=-1){
btn[cur_btn].s=RELEASED;
draw_button(btn[cur_btn]);
switch(btn[cur_btn].a){
case VOLUME_UP: volume_up(); break;
case VOLUME_DOWN: volume_down(); break;
case STATION_UP: station_up(); break;
case STATION_DOWN: station_down(); break;
}
}
cur_btn=-1;
}
// Events from IR Library
void ir_res(uint32_t res){
if(res < max_stations){
cur_station = res;
write_stationNr(cur_station);
tft.fillRect(0, 0, 480, 250, TFT_BLACK);
audio.connecttohost(stations[cur_station].c_str());
pref.putShort("station", cur_station);} // store the current station in nvs
else{
tft.fillRect(0, 0, 480, 250, TFT_BLACK);
audio.connecttohost(stations[cur_station].c_str());
}
}
void ir_number(const char* num){
tft.fillRect(0, 0, 480, 250, TFT_BLACK);
tft.setTextSize(7);
tft.setTextColor(TFT_CORNSILK);
tft.setCursor(50, 70);
tft.print(num);
}
void ir_key(const char* key){
switch(key[0]){
case 'k': break; // OK
case 'r': volume_up(); break; // right
case 'l': volume_down(); break; // left
case 'u': station_up(); break; // up
case 'd': station_down(); break; // down
case '#': break; // #
case '*': break; // *
default: break;
}
}

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.pio/libdeps/esp32-s3-devkitc-1/ESP32-audioI2S/examples/TTGO-TAudio/ESP32_TTGO-TAudio.ino Normal file
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// Copied from https://github.com/LilyGO/TTGO-TAudio/issues/12
// Required Libraries (Download zips and add to the Arduino IDE library).
#include "Arduino.h"
#include <WM8978.h> // https://github.com/CelliesProjects/wm8978-esp32
#include <Audio.h> // https://github.com/schreibfaul1/ESP32-audioI2S
// T-Audio 1.6 WM8978 I2C pins.
#define I2C_SDA 19
#define I2C_SCL 18
// T-Audio 1.6 WM8978 I2S pins.
#define I2S_BCK 33
#define I2S_WS 25
#define I2S_DOUT 26
// T-Audio 1.6 WM8978 MCLK gpio number
#define I2S_MCLKPIN 0
Audio audio;
WM8978 dac;
void setup() {
Serial.begin(115200);
// Setup wm8978 I2C interface.
if (!dac.begin(I2C_SDA, I2C_SCL)) {
ESP_LOGE(TAG, "Error setting up dac: System halted.");
while (1) delay(100);
}
// Select I2S pins
audio.setPinout(I2S_BCK, I2S_WS, I2S_DOUT);
audio.i2s_mclk_pin_select(I2S_MCLKPIN);
// WiFi Settings here.
WiFi.begin("EnterSSIDHere", "EnterPasswordHere");
while (!WiFi.isConnected()) {
delay(10);
}
ESP_LOGI(TAG, "Connected. Starting MP3...");
// Enter your Icecast station URL here.
audio.setVolume(21);
audio.connecttohost("http://hestia2.cdnstream.com/1458_128");
// Volume control.
dac.setSPKvol(63); // Change volume here for board speaker output (Max 63).
dac.setHPvol(63, 63); // Change volume here for headphone jack left, right channel.
}
void loop() {
// Start the stream.
audio.loop();
}
// optional
void audio_info(const char *info){
Serial.print("info "); Serial.println(info);
}
void audio_id3data(const char *info){ //id3 metadata
Serial.print("id3data ");Serial.println(info);
}
void audio_eof_mp3(const char *info){ //end of file
Serial.print("eof_mp3 ");Serial.println(info);
}
void audio_showstation(const char *info){
Serial.print("station ");Serial.println(info);
}
void audio_showstreamtitle(const char *info){
Serial.print("streamtitle ");Serial.println(info);
}
void audio_bitrate(const char *info){
Serial.print("bitrate ");Serial.println(info);
}
void audio_commercial(const char *info){ //duration in sec
Serial.print("commercial ");Serial.println(info);
}
void audio_icyurl(const char *info){ //homepage
Serial.print("icyurl ");Serial.println(info);
}
void audio_lasthost(const char *info){ //stream URL played
Serial.print("lasthost ");Serial.println(info);
}
void audio_eof_speech(const char *info){
Serial.print("eof_speech ");Serial.println(info);
}

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#include "Arduino.h"
#include "Audio.h"
#include "SD.h"
#include "SPI.h"
#include "FS.h"
#include "Ticker.h"
// Digital I/O used
#define SD_CS 5
#define SPI_MOSI 23
#define SPI_MISO 19
#define SPI_SCK 18
#define I2S_DOUT 25
#define I2S_BCLK 27
#define I2S_LRC 26
Audio audio;
Ticker ticker;
struct tm timeinfo;
time_t now;
uint8_t hour = 6;
uint8_t minute = 59;
uint8_t sec = 45;
bool f_time = false;
int8_t timefile = -1;
char chbuf[100];
void tckr1s(){
sec++;
if(sec > 59) {sec = 0; minute++;}
if(minute > 59){minute = 0; hour++;}
if(hour > 23) {hour = 0;}
if(minute == 59 && sec == 50) f_time = true; // flag will be set 10s before full hour
Serial.printf("%02d:%02d:%02d\n", hour, minute, sec);
}
void setup() {
Serial.begin(115200);
SPI.begin(SPI_SCK, SPI_MISO, SPI_MOSI);
SD.begin(SD_CS);
audio.setPinout(I2S_BCLK, I2S_LRC, I2S_DOUT);
audio.setVolume(10); // 0...21
ticker.attach(1, tckr1s);
}
void loop(){
audio.loop();
if(f_time == true){
f_time = false;
timefile = 3;
uint8_t next_hour = hour + 1;
if(next_hour == 25) next_hour = 1;
sprintf(chbuf, "/voice_time/%03d.mp3", next_hour);
audio.connecttoFS(SD, chbuf);
}
}
void audio_eof_mp3(const char *info){ //end of file
//Serial.printf("file :%s\n", info);
if(timefile>0){
if(timefile==1){audio.connecttoFS(SD, "/voice_time/080.mp3"); timefile--;} // stroke
if(timefile==2){audio.connecttoFS(SD, "/voice_time/200.mp3"); timefile--;} // precisely
if(timefile==3){audio.connecttoFS(SD, "/voice_time/O'clock.mp3"); timefile--;}
}
}

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#######################################
# Syntax Coloring Map For Ultrasound
#######################################
#######################################
# Datatypes (KEYWORD1)
#######################################
Audio KEYWORD1
#######################################
# Methods and Functions (KEYWORD2)
#######################################
audio_bitrate KEYWORD2
audio_commercial KEYWORD2
audio_eof_mp3 KEYWORD2
audio_eof_speech KEYWORD2
audio_icyurl KEYWORD2
audio_info KEYWORD2
audio_id3data KEYWORD2
audio_lasthost KEYWORD2
audio_showstreamtitle KEYWORD2
audio_showstation KEYWORD2
audio_eof_stream KEYWORD2
audio_id3image KEYWORD2
connecttoSD KEYWORD2
connecttoFS KEYWORD2
connecttohost KEYWORD2
getAudioCurrentTime KEYWORD2
getAudioFileDuration KEYWORD2
getDatamode KEYWORD2
getFilePos KEYWORD2
getFileSize KEYWORD2
getVolume KEYWORD2
loop KEYWORD2
pauseResume KEYWORD2
setDatamode KEYWORD2
setFilePos KEYWORD2
setPinout KEYWORD2
setVolume KEYWORD2
stopSong KEYWORD2
streamavail KEYWORD2
isRunning KEYWORD2
inBufferFilled KEYWORD2
inBufferFree KEYWORD2
i2s_mclk_pin_select KEYWORD2
setFileLoop KEYWORD2
setTone KEYWORD2
setBalance KEYWORD2
forceMono KEYWORD2
setInternalDAC KEYWORD2
setI2SCommFMT_LSB KEYWORD2
setTimeOffset KEYWORD2
#######################################
# Constants (LITERAL1)
#######################################

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{
"name": "ESP32-audioI2S",
"version": "2.1.0",
"description": "With this library You can easily build a WebRadio with a ESP32 board and a I2S-module",
"keywords": "audio, i2s, esp32",
"repository": {
"type": "git",
"url": "https://github.com/esphome/ESP32-audioI2S.git"
},
"authors": [
{
"name": "schreibfaul1"
},
{
"name": "jesserockz"
}
],
"license": "GPL-3.0",
"homepage": "https://github.com/esphome/ESP32-audioI2S",
"dependencies": {},
"frameworks": "arduino",
"platforms": "espressif32"
}

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.pio/libdeps/esp32-s3-devkitc-1/ESP32-audioI2S/library.properties Normal file
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name=ESP32-audioI2S-master
version=2.1.0
author=schreibfaul1
maintainer=schreibfaul1
sentence=With this library You can easily build a WebRadio with a ESP32 board and a I2S-module.
paragraph=Plays webradio, playlists can be m3u, pls or asx. Data format can be only mp3, aac, flac or m4a. It can also play files from a SD Card.
category=Device Control
url=https://github.com/schreibfaul1/ESP32-audioI2S
architectures=esp32

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/*
* Audio.h
*
* Created on: Oct 26,2018
* Updated on: May 19,2022
* Author: Wolle (schreibfaul1)
*/
//#define SDFATFS_USED // activate for SdFat
#pragma once
#pragma GCC optimize ("Ofast")
#include <Arduino.h>
#include <WiFi.h>
#include <WiFiClientSecure.h>
#include <libb64/cencode.h>
#include <driver/i2s.h>
#ifndef AUDIO_NO_SD_FS
#include <SPI.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
#endif // AUDIO_NO_SD_FS
extern __attribute__((weak)) void audio_info(const char*);
extern __attribute__((weak)) void audio_id3data(const char*); //ID3 metadata
#ifndef AUDIO_NO_SD_FS
extern __attribute__((weak)) void audio_id3image(File& file, const size_t pos, const size_t size); //ID3 metadata image
#endif // AUDIO_NO_SD_FS
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
//----------------------------------------------------------------------------------------------------------------------
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);
#ifndef AUDIO_NO_SD_FS
bool connecttoFS(fs::FS &fs, const char* path, uint32_t resumeFilePos = 0);
bool connecttoSD(const char* path, uint32_t resumeFilePos = 0);
#endif // AUDIO_NO_SD_FS
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 pauseResume();
bool isRunning() {return m_f_running;}
void loop();
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();
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];}
enum : int { CODEC_NONE, CODEC_WAV, CODEC_MP3, CODEC_AAC, CODEC_M4A, CODEC_FLAC, CODEC_OGG,
CODEC_OGG_FLAC, CODEC_OGG_OPUS};
private:
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();
#ifndef AUDIO_NO_SD_FS
void processLocalFile();
#endif // AUDIO_NO_SD_FS
void processWebStream();
void processPlayListData();
void processM3U8entries(uint8_t nrOfEntries = 0, uint32_t seqNr = 0, uint8_t pos = 0, uint16_t targetDuration = 0);
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);
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_M4A_Header(uint8_t* data, size_t len);
int read_OGG_Header(uint8_t *data, size_t len);
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]) ;
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);
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);
// 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) {
//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) {
//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;
}
size_t bigEndian(uint8_t* base, uint8_t numBytes, uint8_t shiftLeft = 8){
size_t result = 0;
if(numBytes < 1 or numBytes > 4) return 0;
for (int i = 0; i < numBytes; i++) {
result += *(base + i) << (numBytes -i - 1) * shiftLeft;
}
return 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;
}
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 { FORMAT_NONE = 0, FORMAT_M3U = 1, FORMAT_PLS = 2, FORMAT_ASX = 3, FORMAT_M3U8 = 4};
enum : int { AUDIO_NONE, AUDIO_HEADER, AUDIO_DATA,
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};
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;
#ifndef AUDIO_NO_SD_FS
File audiofile; // @suppress("Abstract class cannot be instantiated")
#endif // AUDIO_NO_SD_FS
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;
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;
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
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
int m_metalen=0; // Number of bytes in 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_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_filterType[2]; // lowpass, highpass
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_streamUrlHash = 0; // remember streamURL, 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
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_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_continue = false; // next m3u8 chunk is available
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;
};
//----------------------------------------------------------------------------------------------------------------------

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// based on helix aac decoder
#pragma once
//#pragma GCC optimize ("O3")
//#pragma GCC diagnostic ignored "-Wnarrowing"
#include "Arduino.h"
#define AAC_ENABLE_MPEG4
#if (defined CONFIG_IDF_TARGET_ESP32S3 && defined BOARD_HAS_PSRAM)
#define AAC_ENABLE_SBR // needs additional 60KB DRAM,
#endif
#define ASSERT(x) /* do nothing */
#ifndef MAX
#define MAX(a,b) std::max(a,b)
#endif
#ifndef MIN
#define MIN(a,b) std::min(a,b)
#endif
/* AAC file format */
enum {
AAC_FF_Unknown = 0, /* should be 0 on init */
AAC_FF_ADTS = 1,
AAC_FF_ADIF = 2,
AAC_FF_RAW = 3
};
/* syntactic element type */
enum {
AAC_ID_INVALID = -1,
AAC_ID_SCE = 0,
AAC_ID_CPE = 1,
AAC_ID_CCE = 2,
AAC_ID_LFE = 3,
AAC_ID_DSE = 4,
AAC_ID_PCE = 5,
AAC_ID_FIL = 6,
AAC_ID_END = 7
};
enum {
ERR_AAC_NONE = 0,
ERR_AAC_INDATA_UNDERFLOW = -1,
ERR_AAC_NULL_POINTER = -2,
ERR_AAC_INVALID_ADTS_HEADER = -3,
ERR_AAC_INVALID_ADIF_HEADER = -4,
ERR_AAC_INVALID_FRAME = -5,
ERR_AAC_MPEG4_UNSUPPORTED = -6,
ERR_AAC_CHANNEL_MAP = -7,
ERR_AAC_SYNTAX_ELEMENT = -8,
ERR_AAC_DEQUANT = -9,
ERR_AAC_STEREO_PROCESS = -10,
ERR_AAC_PNS = -11,
ERR_AAC_SHORT_BLOCK_DEINT = -12,
ERR_AAC_TNS = -13,
ERR_AAC_IMDCT = -14,
ERR_AAC_NCHANS_TOO_HIGH = -15,
ERR_AAC_SBR_INIT = -16,
ERR_AAC_SBR_BITSTREAM = -17,
ERR_AAC_SBR_DATA = -18,
ERR_AAC_SBR_PCM_FORMAT = -19,
ERR_AAC_SBR_NCHANS_TOO_HIGH = -20,
ERR_AAC_SBR_SINGLERATE_UNSUPPORTED = -21,
ERR_AAC_RAWBLOCK_PARAMS = -22,
ERR_AAC_UNKNOWN = -9999
};
enum {
SBR_GRID_FIXFIX = 0,
SBR_GRID_FIXVAR = 1,
SBR_GRID_VARFIX = 2,
SBR_GRID_VARVAR = 3
};
enum {
HuffTabSBR_tEnv15 = 0,
HuffTabSBR_fEnv15 = 1,
HuffTabSBR_tEnv15b = 2,
HuffTabSBR_fEnv15b = 3,
HuffTabSBR_tEnv30 = 4,
HuffTabSBR_fEnv30 = 5,
HuffTabSBR_tEnv30b = 6,
HuffTabSBR_fEnv30b = 7,
HuffTabSBR_tNoise30 = 8,
HuffTabSBR_fNoise30 = 5,
HuffTabSBR_tNoise30b = 9,
HuffTabSBR_fNoise30b = 7
};
typedef struct _AACDecInfo_t {
/* raw decoded data, before rounding to 16-bit PCM (for postprocessing such as SBR) */
void *rawSampleBuf[2];
int rawSampleBytes;
int rawSampleFBits;
/* fill data (can be used for processing SBR or other extensions) */
uint8_t *fillBuf;
int fillCount;
int fillExtType;
int prevBlockID; /* block information */
int currBlockID;
int currInstTag;
int sbDeinterleaveReqd[2]; // [MAX_NCHANS_ELEM]
int adtsBlocksLeft;
int bitRate; /* user-accessible info */
int nChans;
int sampRate;
float compressionRatio;
int id; /* 0: MPEG-4, 1: MPEG2 */
int profile; /* 0: Main profile, 1: LowComplexity (LC), 2: ScalableSamplingRate (SSR), 3: reserved */
int format;
int sbrEnabled;
int tnsUsed;
int pnsUsed;
int frameCount;
} AACDecInfo_t;
typedef struct _aac_BitStreamInfo_t {
uint8_t *bytePtr;
unsigned int iCache;
int cachedBits;
int nBytes;
} aac_BitStreamInfo_t;
typedef union _U64 {
int64_t w64;
struct {
unsigned int lo32;
signed int hi32;
} r;
} U64;
typedef struct _AACFrameInfo_t {
int bitRate;
int nChans;
int sampRateCore;
int sampRateOut;
int bitsPerSample;
int outputSamps;
int profile;
int tnsUsed;
int pnsUsed;
} AACFrameInfo_t;
typedef struct _HuffInfo_t {
int maxBits; /* number of bits in longest codeword */
uint8_t count[20]; /* count[MAX_HUFF_BITS] = number of codes with length i+1 bits */
int offset; /* offset into symbol table */
} HuffInfo_t;
typedef struct _PulseInfo_t {
uint8_t pulseDataPresent;
uint8_t numPulse;
uint8_t startSFB;
uint8_t offset[4]; // [MAX_PULSES]
uint8_t amp[4]; // [MAX_PULSES]
} PulseInfo_t;
typedef struct _TNSInfo_t {
uint8_t tnsDataPresent;
uint8_t numFilt[8]; // [MAX_TNS_FILTERS] max 1 filter each for 8 short windows, or 3 filters for 1 long window
uint8_t coefRes[8]; // [MAX_TNS_FILTERS]
uint8_t length[8]; // [MAX_TNS_FILTERS]
uint8_t order[8]; // [MAX_TNS_FILTERS]
uint8_t dir[8]; // [MAX_TNS_FILTERS]
int8_t coef[60]; // [MAX_TNS_COEFS] max 3 filters * 20 coefs for 1 long window,
// or 1 filter * 7 coefs for each of 8 short windows
} TNSInfo_t;
typedef struct _GainControlInfo_t {
uint8_t gainControlDataPresent;
uint8_t maxBand;
uint8_t adjNum[3][8]; // [MAX_GAIN_BANDS][MAX_GAIN_WIN]
uint8_t alevCode[3][8][7]; // [MAX_GAIN_BANDS][MAX_GAIN_WIN][MAX_GAIN_ADJUST]
uint8_t alocCode[3][8][7]; // [MAX_GAIN_BANDS][MAX_GAIN_WIN][MAX_GAIN_ADJUST]
} GainControlInfo_t;
typedef struct _ICSInfo_t {
uint8_t icsResBit;
uint8_t winSequence;
uint8_t winShape;
uint8_t maxSFB;
uint8_t sfGroup;
uint8_t predictorDataPresent;
uint8_t predictorReset;
uint8_t predictorResetGroupNum;
uint8_t predictionUsed[41]; // [MAX_PRED_SFB]
uint8_t numWinGroup;
uint8_t winGroupLen[8]; // [MAX_WIN_GROUPS]
} ICSInfo_t;
typedef struct _ADTSHeader_t {
/* fixed */
uint8_t id; /* MPEG bit - should be 1 */
uint8_t layer; /* MPEG layer - should be 0 */
uint8_t protectBit; /* 0 = CRC word follows, 1 = no CRC word */
uint8_t profile; /* 0 = main, 1 = LC, 2 = SSR, 3 = reserved */
uint8_t sampRateIdx; /* sample rate index range = [0, 11] */
uint8_t privateBit; /* ignore */
uint8_t channelConfig; /* 0 = implicit, >0 = use default table */
uint8_t origCopy; /* 0 = copy, 1 = original */
uint8_t home; /* ignore */
/* variable */
uint8_t copyBit; /* 1 bit of the 72-bit copyright ID (transmitted as 1 bit per frame) */
uint8_t copyStart; /* 1 = this bit starts the 72-bit ID, 0 = it does not */
int frameLength; /* length of frame */
int bufferFull; /* number of 32-bit words left in enc buffer, 0x7FF = VBR */
uint8_t numRawDataBlocks; /* number of raw data blocks in frame */
/* CRC */
int crcCheckWord; /* 16-bit CRC check word (present if protectBit == 0) */
} ADTSHeader_t;
typedef struct _ADIFHeader_t {
uint8_t copyBit; /* 0 = no copyright ID, 1 = 72-bit copyright ID follows immediately */
uint8_t origCopy; /* 0 = copy, 1 = original */
uint8_t home; /* ignore */
uint8_t bsType; /* bitstream type: 0 = CBR, 1 = VBR */
int bitRate; /* bitRate: CBR = bits/sec, VBR = peak bits/frame, 0 = unknown */
uint8_t numPCE; /* number of program config elements (max = 16) */
int bufferFull; /* bits left in bit reservoir */
uint8_t copyID[9]; /* [ADIF_COPYID_SIZE] optional 72-bit copyright ID */
} ADIFHeader_t;
/* sizeof(ProgConfigElement_t) = 82 bytes (if KEEP_PCE_COMMENTS not defined) */
typedef struct _ProgConfigElement_t {
uint8_t elemInstTag; /* element instance tag */
uint8_t profile; /* 0 = main, 1 = LC, 2 = SSR, 3 = reserved */
uint8_t sampRateIdx; /* sample rate index range = [0, 11] */
uint8_t numFCE; /* number of front channel elements (max = 15) */
uint8_t numSCE; /* number of side channel elements (max = 15) */
uint8_t numBCE; /* number of back channel elements (max = 15) */
uint8_t numLCE; /* number of LFE channel elements (max = 3) */
uint8_t numADE; /* number of associated data elements (max = 7) */
uint8_t numCCE; /* number of valid channel coupling elements (max = 15) */
uint8_t monoMixdown; /* mono mixdown: bit 4 = present flag, bits 3-0 = element number */
uint8_t stereoMixdown; /* stereo mixdown: bit 4 = present flag, bits 3-0 = element number */
uint8_t matrixMixdown; /* bit 4 = present flag, bit 3 = unused,bits 2-1 = index, bit 0 = pseudo-surround enable */
uint8_t fce[15]; /* [MAX_NUM_FCE] front element channel pair: bit 4 = SCE/CPE flag, bits 3-0 = inst tag */
uint8_t sce[15]; /* [MAX_NUM_SCE] side element channel pair: bit 4 = SCE/CPE flag, bits 3-0 = inst tag */
uint8_t bce[15]; /* [MAX_NUM_BCE] back element channel pair: bit 4 = SCE/CPE flag, bits 3-0 = inst tag */
uint8_t lce[3]; /* [MAX_NUM_LCE] instance tag for LFE elements */
uint8_t ade[7]; /* [MAX_NUM_ADE] instance tag for ADE elements */
uint8_t cce[15]; /* [MAX_NUM_BCE] channel coupling elements: bit 4 = switching flag, bits 3-0 = inst tag */
} ProgConfigElement_t;
typedef struct _SBRHeader {
int count;
uint8_t ampRes;
uint8_t startFreq;
uint8_t stopFreq;
uint8_t crossOverBand;
uint8_t resBitsHdr;
uint8_t hdrExtra1;
uint8_t hdrExtra2;
uint8_t freqScale;
uint8_t alterScale;
uint8_t noiseBands;
uint8_t limiterBands;
uint8_t limiterGains;
uint8_t interpFreq;
uint8_t smoothMode;
} SBRHeader;
/* need one SBRGrid per channel, updated every frame */
typedef struct _SBRGrid {
uint8_t frameClass;
uint8_t ampResFrame;
uint8_t pointer;
uint8_t numEnv; /* L_E */
uint8_t envTimeBorder[5 + 1]; // [MAX_NUM_ENV+1] /* t_E */
uint8_t freqRes[5]; // [MAX_NUM_ENV]/* r */
uint8_t numNoiseFloors; /* L_Q */
uint8_t noiseTimeBorder[2 + 1]; // [MAX_NUM_NOISE_FLOORS+1] /* t_Q */
uint8_t numEnvPrev;
uint8_t numNoiseFloorsPrev;
uint8_t freqResPrev;
} SBRGrid;
/* need one SBRFreq per element (SCE/CPE/LFE), updated only on header reset */
typedef struct _SBRFreq {
int kStart; /* k_x */
int nMaster;
int nHigh;
int nLow;
int nLimiter; /* N_l */
int numQMFBands; /* M */
int numNoiseFloorBands; /* Nq */
int kStartPrev;
int numQMFBandsPrev;
uint8_t freqMaster[48 + 1]; // [MAX_QMF_BANDS + 1] /* not necessary to save this after derived tables are generated */
uint8_t freqHigh[48 + 1]; // [MAX_QMF_BANDS + 1]
uint8_t freqLow[48 / 2 + 1]; // [MAX_QMF_BANDS / 2 + 1] /* nLow = nHigh - (nHigh >> 1) */
uint8_t freqNoise[5 + 1]; // [MAX_NUM_NOISE_FLOOR_BANDS+1]
uint8_t freqLimiter[48 / 2 + 5];// [MAX_QMF_BANDS / 2 + MAX_NUM_PATCHES] /* max (intermediate) size = nLow + numPatches - 1 */
uint8_t numPatches;
uint8_t patchNumSubbands[5 + 1]; // [MAX_NUM_PATCHES + 1]
uint8_t patchStartSubband[5 + 1]; // [MAX_NUM_PATCHES + 1]
} SBRFreq;
typedef struct _SBRChan {
int reset;
uint8_t deltaFlagEnv[5]; // [MAX_NUM_ENV]
uint8_t deltaFlagNoise[2]; // [MAX_NUM_NOISE_FLOORS]
int8_t envDataQuant[5][48]; // [MAX_NUM_ENV][MAX_QMF_BANDS] /* range = [0, 127] */
int8_t noiseDataQuant[2][5]; // [MAX_NUM_NOISE_FLOORS][MAX_NUM_NOISE_FLOOR_BANDS]
uint8_t invfMode[2][5]; // [2][MAX_NUM_NOISE_FLOOR_BANDS] /* invfMode[0/1][band] = prev/curr */
int chirpFact[5]; // [MAX_NUM_NOISE_FLOOR_BANDS] /* bwArray */
uint8_t addHarmonicFlag[2]; /* addHarmonicFlag[0/1] = prev/curr */
uint8_t addHarmonic[2][64]; /* addHarmonic[0/1][band] = prev/curr */
int gbMask[2]; /* gbMask[0/1] = XBuf[0-31]/XBuf[32-39] */
int8_t laPrev;
int noiseTabIndex;
int sinIndex;
int gainNoiseIndex;
int gTemp[5][48]; // [MAX_NUM_SMOOTH_COEFS][MAX_QMF_BANDS]
int qTemp[5][48]; // [MAX_NUM_SMOOTH_COEFS][MAX_QMF_BANDS]
} SBRChan;
/* state info struct for baseline (MPEG-4 LC) decoding */
typedef struct _PSInfoBase_t {
int dataCount;
uint8_t dataBuf[510]; // [DATA_BUF_SIZE]
int fillCount;
uint8_t fillBuf[269]; //[FILL_BUF_SIZE]
/* state information which is the same throughout whole frame */
int nChans;
int useImpChanMap;
int sampRateIdx;
/* state information which can be overwritten by subsequent elements within frame */
ICSInfo_t icsInfo[2]; // [MAX_NCHANS_ELEM]
int commonWin;
short scaleFactors[2][15*8]; // [MAX_NCHANS_ELEM][MAX_SF_BANDS]
uint8_t sfbCodeBook[2][15*8]; // [MAX_NCHANS_ELEM][MAX_SF_BANDS]
int msMaskPresent;
uint8_t msMaskBits[(15 * 8 + 7) >> 3]; // [MAX_MS_MASK_BYTES]
int pnsUsed[2]; // [MAX_NCHANS_ELEM]
int pnsLastVal;
int intensityUsed[2]; // [MAX_NCHANS_ELEM]
// PulseInfo_t pulseInfo[2]; // [MAX_NCHANS_ELEM]
TNSInfo_t tnsInfo[2]; // [MAX_NCHANS_ELEM]
int tnsLPCBuf[20]; // [MAX_TNS_ORDER]
int tnsWorkBuf[20]; //[MAX_TNS_ORDER]
GainControlInfo_t gainControlInfo[2]; // [MAX_NCHANS_ELEM]
int gbCurrent[2]; // [MAX_NCHANS_ELEM]
int coef[2][1024]; // [MAX_NCHANS_ELEM][AAC_MAX_NSAMPS]
#ifdef AAC_ENABLE_SBR
int sbrWorkBuf[2][1024]; // [MAX_NCHANS_ELEM][AAC_MAX_NSAMPS];
#endif
/* state information which must be saved for each element and used in next frame */
int overlap[2][1024]; // [AAC_MAX_NCHANS][AAC_MAX_NSAMPS]
int prevWinShape[2]; // [AAC_MAX_NCHANS]
} PSInfoBase_t;
typedef struct _PSInfoSBR {
/* save for entire file */
int frameCount;
int sampRateIdx;
/* state info that must be saved for each channel */
SBRHeader sbrHdr[2];
SBRGrid sbrGrid[2];
SBRFreq sbrFreq[2];
SBRChan sbrChan[2];
/* temp variables, no need to save between blocks */
uint8_t dataExtra;
uint8_t resBitsData;
uint8_t extendedDataPresent;
int extendedDataSize;
int8_t envDataDequantScale[2][5]; // [MAX_NCHANS_ELEM][MAX_NUM_ENV
int envDataDequant[2][5][48]; // [MAX_NCHANS_ELEM][MAX_NUM_ENV][MAX_QMF_BANDS
int noiseDataDequant[2][2][5]; // [MAX_NCHANS_ELEM][MAX_NUM_NOISE_FLOORS][MAX_NUM_NOISE_FLOOR_BANDS]
int eCurr[48]; // [MAX_QMF_BANDS]
uint8_t eCurrExp[48]; // [MAX_QMF_BANDS]
uint8_t eCurrExpMax;
int8_t la;
int crcCheckWord;
int couplingFlag;
int envBand;
int eOMGainMax;
int gainMax;
int gainMaxFBits;
int noiseFloorBand;
int qp1Inv;
int qqp1Inv;
int sMapped;
int sBand;
int highBand;
int sumEOrigMapped;
int sumECurrGLim;
int sumSM;
int sumQM;
int gLimBoost[48];
int qmLimBoost[48];
int smBoost[48];
int smBuf[48];
int qmLimBuf[48];
int gLimBuf[48];
int gLimFbits[48];
int gFiltLast[48];
int qFiltLast[48];
/* large buffers */
int delayIdxQMFA[2]; // [AAC_MAX_NCHANS]
int delayQMFA[2][10 * 32]; // [AAC_MAX_NCHANS][DELAY_SAMPS_QMFA]
int delayIdxQMFS[2]; // [AAC_MAX_NCHANS]
int delayQMFS[2][10 * 128]; // [AAC_MAX_NCHANS][DELAY_SAMPS_QMFS]
int XBufDelay[2][8][64][2]; // [AAC_MAX_NCHANS][HF_GEN][64][2]
int XBuf[32+8][64][2];
} PSInfoSBR_t;
bool AACDecoder_AllocateBuffers(void);
int AACFlushCodec();
void AACDecoder_FreeBuffers(void);
bool AACDecoder_IsInit(void);
int AACFindSyncWord(uint8_t *buf, int nBytes);
int AACSetRawBlockParams(int copyLast, int nChans, int sampRateCore, int profile);
int AACDecode(uint8_t *inbuf, int *bytesLeft, short *outbuf);
int AACGetSampRate();
int AACGetChannels();
int AACGetID(); // 0-MPEG4, 1-MPEG2
uint8_t AACGetProfile(); // 0-Main, 1-LC, 2-SSR, 3-reserved
uint8_t AACGetFormat(); // 0-unknown 1-ADTS 2-ADIF, 3-RAW
int AACGetBitsPerSample();
int AACGetBitrate();
int AACGetOutputSamps();
int AACGetBitrate();
void DecodeLPCCoefs(int order, int res, int8_t *filtCoef, int *a, int *b);
int FilterRegion(int size, int dir, int order, int *audioCoef, int *a, int *hist);
int TNSFilter(int ch);
int DecodeSingleChannelElement();
int DecodeChannelPairElement();
int DecodeLFEChannelElement();
int DecodeDataStreamElement();
int DecodeProgramConfigElement(uint8_t idx);
int DecodeFillElement();
int DecodeNextElement(uint8_t **buf, int *bitOffset, int *bitsAvail);
void PreMultiply(int tabidx, int *zbuf1);
void PostMultiply(int tabidx, int *fft1);
void PreMultiplyRescale(int tabidx, int *zbuf1, int es);
void PostMultiplyRescale(int tabidx, int *fft1, int es);
void DCT4(int tabidx, int *coef, int gb);
void BitReverse(int *inout, int tabidx);
void R4FirstPass(int *x, int bg);
void R8FirstPass(int *x, int bg);
void R4Core(int *x, int bg, int gp, int *wtab);
void R4FFT(int tabidx, int *x);
void UnpackZeros(int nVals, int *coef);
void UnpackQuads(int cb, int nVals, int *coef);
void UnpackPairsNoEsc(int cb, int nVals, int *coef);
void UnpackPairsEsc(int cb, int nVals, int *coef);
void DecodeSpectrumLong(int ch);
void DecodeSpectrumShort(int ch);
void DecWindowOverlap(int *buf0, int *over0, short *pcm0, int nChans, int winTypeCurr, int winTypePrev);
void DecWindowOverlapLongStart(int *buf0, int *over0, short *pcm0, int nChans, int winTypeCurr, int winTypePrev);
void DecWindowOverlapLongStop(int *buf0, int *over0, short *pcm0, int nChans, int winTypeCurr, int winTypePrev);
void DecWindowOverlapShort(int *buf0, int *over0, short *pcm0, int nChans, int winTypeCurr, int winTypePrev);
int IMDCT(int ch, int chOut, short *outbuf);
void DecodeICSInfo(ICSInfo_t *icsInfo, int sampRateIdx);
void DecodeSectionData(int winSequence, int numWinGrp, int maxSFB, uint8_t *sfbCodeBook);
int DecodeOneScaleFactor();
void DecodeScaleFactors(int numWinGrp, int maxSFB, int globalGain, uint8_t *sfbCodeBook, short *scaleFactors);
void DecodePulseInfo(uint8_t ch);
void DecodeTNSInfo(int winSequence, TNSInfo_t *ti, int8_t *tnsCoef);
void DecodeGainControlInfo(int winSequence, GainControlInfo_t *gi);
void DecodeICS(int ch);
int DecodeNoiselessData(uint8_t **buf, int *bitOffset, int *bitsAvail, int ch);
int DecodeHuffmanScalar(const signed short *huffTab, const HuffInfo_t *huffTabInfo, unsigned int bitBuf, int32_t *val);
int UnpackADTSHeader(uint8_t **buf, int *bitOffset, int *bitsAvail);
int GetADTSChannelMapping(uint8_t *buf, int bitOffset, int bitsAvail);
int GetNumChannelsADIF(int nPCE);
int GetSampleRateIdxADIF(int nPCE);
int UnpackADIFHeader(uint8_t **buf, int *bitOffset, int *bitsAvail);
int SetRawBlockParams(int copyLast, int nChans, int sampRate, int profile);
int PrepareRawBlock();
int DequantBlock(int *inbuf, int nSamps, int scale);
int AACDequantize(int ch);
int DeinterleaveShortBlocks(int ch);
unsigned int Get32BitVal(unsigned int *last);
int InvRootR(int r);
int ScaleNoiseVector(int *coef, int nVals, int sf);
void GenerateNoiseVector(int *coef, int *last, int nVals);
void CopyNoiseVector(int *coefL, int *coefR, int nVals);
int PNS(int ch);
int GetSampRateIdx(int sampRate);
void StereoProcessGroup(int *coefL, int *coefR, const uint16_t *sfbTab, int msMaskPres, uint8_t *msMaskPtr,
int msMaskOffset, int maxSFB, uint8_t *cbRight, short *sfRight, int *gbCurrent);
int StereoProcess();
int RatioPowInv(int a, int b, int c);
int SqrtFix(int q, int fBitsIn, int *fBitsOut);
int InvRNormalized(int r);
void BitReverse32(int *inout);
void R8FirstPass32(int *r0);
void R4Core32(int *r0);
void FFT32C(int *x);
void CVKernel1(int *XBuf, int *accBuf);
void CVKernel2(int *XBuf, int *accBuf);
void SetBitstreamPointer(int nBytes, uint8_t *buf);
inline void RefillBitstreamCache();
unsigned int GetBits(int nBits);
unsigned int GetBitsNoAdvance(int nBits);
void AdvanceBitstream(int nBits);
int CalcBitsUsed(uint8_t *startBuf, int startOffset);
void ByteAlignBitstream();
// SBR
void InitSBRState();
int DecodeSBRBitstream(int chBase);
int DecodeSBRData(int chBase, short *outbuf);
int FlushCodecSBR();
void BubbleSort(uint8_t *v, int nItems);
uint8_t VMin(uint8_t *v, int nItems);
uint8_t VMax(uint8_t *v, int nItems);
int CalcFreqMasterScaleZero(uint8_t *freqMaster, int alterScale, int k0, int k2);
int CalcFreqMaster(uint8_t *freqMaster, int freqScale, int alterScale, int k0, int k2);
int CalcFreqHigh(uint8_t *freqHigh, uint8_t *freqMaster, int nMaster, int crossOverBand);
int CalcFreqLow(uint8_t *freqLow, uint8_t *freqHigh, int nHigh);
int CalcFreqNoise(uint8_t *freqNoise, uint8_t *freqLow, int nLow, int kStart, int k2, int noiseBands);
int BuildPatches(uint8_t *patchNumSubbands, uint8_t *patchStartSubband, uint8_t *freqMaster, int nMaster, int k0,
int kStart, int numQMFBands, int sampRateIdx);
int FindFreq(uint8_t *freq, int nFreq, uint8_t val);
void RemoveFreq(uint8_t *freq, int nFreq, int removeIdx);
int CalcFreqLimiter(uint8_t *freqLimiter, uint8_t *patchNumSubbands, uint8_t *freqLow, int nLow, int kStart,
int limiterBands, int numPatches);
int CalcFreqTables(SBRHeader *sbrHdr, SBRFreq *sbrFreq, int sampRateIdx);
void EstimateEnvelope(SBRHeader *sbrHdr, SBRGrid *sbrGrid, SBRFreq *sbrFreq, int env);
int GetSMapped(SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChan, int env, int band, int la);
void CalcMaxGain(SBRHeader *sbrHdr, SBRGrid *sbrGrid, SBRFreq *sbrFreq, int ch, int env, int lim, int fbitsDQ);
void CalcNoiseDivFactors(int q, int *qp1Inv, int *qqp1Inv);
void CalcComponentGains(SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChan, int ch, int env, int lim, int fbitsDQ);
void ApplyBoost(SBRFreq *sbrFreq, int lim, int fbitsDQ);
void CalcGain(SBRHeader *sbrHdr, SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChan, int ch, int env);
void MapHF(SBRHeader *sbrHdr, SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChan, int env, int hfReset);
void AdjustHighFreq(SBRHeader *sbrHdr, SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChan, int ch);
int CalcCovariance1(int *XBuf, int *p01reN, int *p01imN, int *p12reN, int *p12imN, int *p11reN, int *p22reN);
int CalcCovariance2(int *XBuf, int *p02reN, int *p02imN);
void CalcLPCoefs(int *XBuf, int *a0re, int *a0im, int *a1re, int *a1im, int gb);
void GenerateHighFreq(SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChan, int ch);
int DecodeHuffmanScalar(const signed int *huffTab, const HuffInfo_t *huffTabInfo, unsigned int bitBuf, signed int *val);
int DecodeOneSymbol(int huffTabIndex);
int DequantizeEnvelope(int nBands, int ampRes, int8_t *envQuant, int *envDequant);
void DequantizeNoise(int nBands, int8_t *noiseQuant, int *noiseDequant);
void DecodeSBREnvelope(SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChan, int ch);
void DecodeSBRNoise(SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChan, int ch);
void UncoupleSBREnvelope(SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChanR);
void UncoupleSBRNoise(SBRGrid *sbrGrid, SBRFreq *sbrFreq, SBRChan *sbrChanR);
void DecWindowOverlapNoClip(int *buf0, int *over0, int *out0, int winTypeCurr, int winTypePrev);
void DecWindowOverlapLongStartNoClip(int *buf0, int *over0, int *out0, int winTypeCurr, int winTypePrev);
void DecWindowOverlapLongStopNoClip(int *buf0, int *over0, int *out0, int winTypeCurr, int winTypePrev);
void DecWindowOverlapShortNoClip(int *buf0, int *over0, int *out0, int winTypeCurr, int winTypePrev);
void PreMultiply64(int *zbuf1);
void PostMultiply64(int *fft1, int nSampsOut);
void QMFAnalysisConv(int *cTab, int *delay, int dIdx, int *uBuf);
int QMFAnalysis(int *inbuf, int *delay, int *XBuf, int fBitsIn, int *delayIdx, int qmfaBands);
void QMFSynthesisConv(int *cPtr, int *delay, int dIdx, short *outbuf, int nChans);
void QMFSynthesis(int *inbuf, int *delay, int *delayIdx, int qmfsBands, short *outbuf, int nChans);
int UnpackSBRHeader(SBRHeader *sbrHdr);
void UnpackSBRGrid(SBRHeader *sbrHdr, SBRGrid *sbrGrid);
void UnpackDeltaTimeFreq(int numEnv, uint8_t *deltaFlagEnv, int numNoiseFloors, uint8_t *deltaFlagNoise);
void UnpackInverseFilterMode(int numNoiseFloorBands, uint8_t *mode);
void UnpackSinusoids(int nHigh, int addHarmonicFlag, uint8_t *addHarmonic);
void CopyCouplingGrid(SBRGrid *sbrGridLeft, SBRGrid *sbrGridRight);
void CopyCouplingInverseFilterMode(int numNoiseFloorBands, uint8_t *modeLeft, uint8_t *modeRight);
void UnpackSBRSingleChannel(int chBase);
void UnpackSBRChannelPair(int chBase);

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/*
* flac_decoder.cpp
* Java source code from https://www.nayuki.io/page/simple-flac-implementation
* adapted to ESP32
*
* Created on: Jul 03,2020
* Updated on: Jul 03,2021
*
* Author: Wolle
*
*
*/
#include "flac_decoder.h"
#include "vector"
using namespace std;
FLACFrameHeader_t *FLACFrameHeader;
FLACMetadataBlock_t *FLACMetadataBlock;
FLACsubFramesBuff_t *FLACsubFramesBuff;
vector<int32_t>coefs;
const uint16_t outBuffSize = 2048;
uint16_t m_blockSize=0;
uint16_t m_blockSizeLeft = 0;
uint16_t m_validSamples = 0;
uint8_t m_status = 0;
uint8_t* m_inptr;
int16_t m_bytesAvail;
int16_t m_bytesDecoded = 0;
float m_compressionRatio = 0;
uint16_t m_rIndex=0;
uint64_t m_bitBuffer = 0;
uint8_t m_bitBufferLen = 0;
bool m_f_OggS_found = false;
//----------------------------------------------------------------------------------------------------------------------
// FLAC INI SECTION
//----------------------------------------------------------------------------------------------------------------------
bool FLACDecoder_AllocateBuffers(void){
if(psramFound()) {
// PSRAM found, Buffer will be allocated in PSRAM
if(!FLACFrameHeader) {FLACFrameHeader = (FLACFrameHeader_t*) ps_malloc(sizeof(FLACFrameHeader_t));}
if(!FLACMetadataBlock) {FLACMetadataBlock = (FLACMetadataBlock_t*) ps_malloc(sizeof(FLACMetadataBlock_t));}
if(!FLACsubFramesBuff) {FLACsubFramesBuff = (FLACsubFramesBuff_t*) ps_malloc(sizeof(FLACsubFramesBuff_t));}
}
else {
if(!FLACFrameHeader) {FLACFrameHeader = (FLACFrameHeader_t*) malloc(sizeof(FLACFrameHeader_t));}
if(!FLACMetadataBlock) {FLACMetadataBlock = (FLACMetadataBlock_t*) malloc(sizeof(FLACMetadataBlock_t));}
if(!FLACsubFramesBuff) {FLACsubFramesBuff = (FLACsubFramesBuff_t*) malloc(sizeof(FLACsubFramesBuff_t));}
}
if(!FLACFrameHeader || !FLACMetadataBlock || !FLACsubFramesBuff ){
log_e("not enough memory to allocate flacdecoder buffers");
return false;
}
FLACDecoder_ClearBuffer();
return true;
}
//----------------------------------------------------------------------------------------------------------------------
void FLACDecoder_ClearBuffer(){
memset(FLACFrameHeader, 0, sizeof(FLACFrameHeader_t));
memset(FLACMetadataBlock, 0, sizeof(FLACMetadataBlock_t));
memset(FLACsubFramesBuff, 0, sizeof(FLACsubFramesBuff_t));
m_status = DECODE_FRAME;
return;
}
//----------------------------------------------------------------------------------------------------------------------
void FLACDecoder_FreeBuffers(){
if(FLACFrameHeader) {free(FLACFrameHeader); FLACFrameHeader = NULL;}
if(FLACMetadataBlock) {free(FLACMetadataBlock); FLACMetadataBlock = NULL;}
if(FLACsubFramesBuff) {free(FLACsubFramesBuff); FLACsubFramesBuff = NULL;}
}
//----------------------------------------------------------------------------------------------------------------------
// B I T R E A D E R
//----------------------------------------------------------------------------------------------------------------------
uint32_t readUint(uint8_t nBits){
while (m_bitBufferLen < nBits){
uint8_t temp = *(m_inptr + m_rIndex);
m_rIndex++;
m_bytesAvail--;
if(m_bytesAvail < 0) { log_i("error in bitreader"); }
m_bitBuffer = (m_bitBuffer << 8) | temp;
m_bitBufferLen += 8;
}
m_bitBufferLen -= nBits;
uint32_t result = m_bitBuffer >> m_bitBufferLen;
if (nBits < 32)
result &= (1 << nBits) - 1;
return result;
}
int32_t readSignedInt(int nBits){
int32_t temp = readUint(nBits) << (32 - nBits);
temp = temp >> (32 - nBits); // The C++ compiler uses the sign bit to fill vacated bit positions
return temp;
}
int64_t readRiceSignedInt(uint8_t param){
long val = 0;
while (readUint(1) == 0)
val++;
val = (val << param) | readUint(param);
return (val >> 1) ^ -(val & 1);
}
void alignToByte() {
m_bitBufferLen -= m_bitBufferLen % 8;
}
//----------------------------------------------------------------------------------------------------------------------
// F L A C - D E C O D E R
//----------------------------------------------------------------------------------------------------------------------
void FLACSetRawBlockParams(uint8_t Chans, uint32_t SampRate, uint8_t BPS, uint32_t tsis, uint32_t AuDaLength){
FLACMetadataBlock->numChannels = Chans;
FLACMetadataBlock->sampleRate = SampRate;
FLACMetadataBlock->bitsPerSample = BPS;
FLACMetadataBlock->totalSamples = tsis; // total samples in stream
FLACMetadataBlock->audioDataLength = AuDaLength;
}
//----------------------------------------------------------------------------------------------------------------------
void FLACDecoderReset(){ // set var to default
m_status = DECODE_FRAME;
m_bitBuffer = 0;
m_bitBufferLen = 0;
}
//----------------------------------------------------------------------------------------------------------------------
int FLACFindSyncWord(unsigned char *buf, int nBytes) {
int i;
/* find byte-aligned syncword - need 13 matching bits */
for (i = 0; i < nBytes - 1; i++) {
if ((buf[i + 0] & 0xFF) == 0xFF && (buf[i + 1] & 0xF8) == 0xF8) {
FLACDecoderReset();
return i;
}
}
return -1;
}
//----------------------------------------------------------------------------------------------------------------------
int FLACFindOggSyncWord(unsigned char *buf, int nBytes){
int i;
/* find byte-aligned syncword - need 13 matching bits */
for (i = 0; i < nBytes - 1; i++) {
if ((buf[i + 0] & 0xFF) == 0xFF && (buf[i + 1] & 0xF8) == 0xF8) {
FLACDecoderReset();
log_i("FLAC sync found");
return i;
}
}
/* find byte-aligned OGG Magic - OggS */
for (i = 0; i < nBytes - 1; i++) {
if ((buf[i + 0] == 'O') && (buf[i + 1] == 'g') && (buf[i + 2] == 'g') && (buf[i + 3] == 'S')) {
FLACDecoderReset();
log_i("OggS found");
m_f_OggS_found = true;
return i;
}
}
return -1;
}
//----------------------------------------------------------------------------------------------------------------------
int FLACparseOggHeader(unsigned char *buf){
uint8_t i = 0;
uint8_t ssv = *(buf + i); // stream_structure_version
(void)ssv;
i++;
uint8_t htf = *(buf + i); // header_type_flag
(void)htf;
i++;
uint32_t tmp = 0; // absolute granule position
for (int j = 0; j < 4; j++) {
tmp += *(buf + j + i) << (4 -j - 1) * 8;
}
i += 4;
uint64_t agp = (uint64_t) tmp << 32;
for (int j = 0; j < 4; j++) {
agp += *(buf + j + i) << (4 -j - 1) * 8;
}
i += 4;
uint32_t ssnr = 0; // stream serial number
for (int j = 0; j < 4; j++) {
ssnr += *(buf + j + i) << (4 -j - 1) * 8;
}
i += 4;
uint32_t psnr = 0; // page sequence no
for (int j = 0; j < 4; j++) {
psnr += *(buf + j + i) << (4 -j - 1) * 8;
}
i += 4;
uint32_t pchk = 0; // page checksum
for (int j = 0; j < 4; j++) {
pchk += *(buf + j + i) << (4 -j - 1) * 8;
}
i += 4;
uint8_t psegm = *(buf + i);
i++;
uint8_t psegmBuff[256];
uint32_t pageLen = 0;
for(uint8_t j = 0; j < psegm; j++){
psegmBuff[j] = *(buf + i);
pageLen += psegmBuff[j];
i++;
}
return i;
}
//----------------------------------------------------------------------------------------------------------------------
int8_t FLACDecode(uint8_t *inbuf, int *bytesLeft, short *outbuf){
if(m_f_OggS_found == true){
m_f_OggS_found = false;
*bytesLeft -= FLACparseOggHeader(inbuf);
return ERR_FLAC_NONE;
}
if(m_status != OUT_SAMPLES){
m_rIndex = 0;
m_bytesAvail = (*bytesLeft);
m_inptr = inbuf;
}
if(m_status == DECODE_FRAME){ // Read a ton of header fields, and ignore most of them
if ((inbuf[0] == 'O') && (inbuf[1] == 'g') && (inbuf[2] == 'g') && (inbuf[3] == 'S')){
*bytesLeft -= 4;
m_f_OggS_found = true;
return ERR_FLAC_NONE;
}
uint32_t temp = readUint(8);
uint16_t sync = temp << 6 |readUint(6);
if (sync != 0x3FFE){
log_i("Sync code expected 0x3FFE but received %X", sync);
return ERR_FLAC_SYNC_CODE_NOT_FOUND;
}
readUint(1);
FLACFrameHeader->blockingStrategy = readUint(1);
FLACFrameHeader->blockSizeCode = readUint(4);
FLACFrameHeader->sampleRateCode = readUint(4);
FLACFrameHeader->chanAsgn = readUint(4);
FLACFrameHeader->sampleSizeCode = readUint(3);
if(!FLACMetadataBlock->numChannels){
if(FLACFrameHeader->chanAsgn == 0) FLACMetadataBlock->numChannels = 1;
if(FLACFrameHeader->chanAsgn == 1) FLACMetadataBlock->numChannels = 2;
if(FLACFrameHeader->chanAsgn > 7) FLACMetadataBlock->numChannels = 2;
}
if(FLACMetadataBlock->numChannels < 1) return ERR_FLAC_UNKNOWN_CHANNEL_ASSIGNMENT;
if(!FLACMetadataBlock->bitsPerSample){
if(FLACFrameHeader->sampleSizeCode == 1) FLACMetadataBlock->bitsPerSample = 8;
if(FLACFrameHeader->sampleSizeCode == 2) FLACMetadataBlock->bitsPerSample = 12;
if(FLACFrameHeader->sampleSizeCode == 4) FLACMetadataBlock->bitsPerSample = 16;
if(FLACFrameHeader->sampleSizeCode == 5) FLACMetadataBlock->bitsPerSample = 20;
if(FLACFrameHeader->sampleSizeCode == 6) FLACMetadataBlock->bitsPerSample = 24;
}
if(FLACMetadataBlock->bitsPerSample > 16) return ERR_FLAC_BITS_PER_SAMPLE_TOO_BIG;
if(FLACMetadataBlock->bitsPerSample < 8 ) return ERR_FLAG_BITS_PER_SAMPLE_UNKNOWN;
if(!FLACMetadataBlock->sampleRate){
if(FLACFrameHeader->sampleRateCode == 1) FLACMetadataBlock->sampleRate = 88200;
if(FLACFrameHeader->sampleRateCode == 2) FLACMetadataBlock->sampleRate = 176400;
if(FLACFrameHeader->sampleRateCode == 3) FLACMetadataBlock->sampleRate = 192000;
if(FLACFrameHeader->sampleRateCode == 4) FLACMetadataBlock->sampleRate = 8000;
if(FLACFrameHeader->sampleRateCode == 5) FLACMetadataBlock->sampleRate = 16000;
if(FLACFrameHeader->sampleRateCode == 6) FLACMetadataBlock->sampleRate = 22050;
if(FLACFrameHeader->sampleRateCode == 7) FLACMetadataBlock->sampleRate = 24000;
if(FLACFrameHeader->sampleRateCode == 8) FLACMetadataBlock->sampleRate = 32000;
if(FLACFrameHeader->sampleRateCode == 9) FLACMetadataBlock->sampleRate = 44100;
if(FLACFrameHeader->sampleRateCode == 10) FLACMetadataBlock->sampleRate = 48000;
if(FLACFrameHeader->sampleRateCode == 11) FLACMetadataBlock->sampleRate = 96000;
}
readUint(1);
temp = (readUint(8) << 24);
temp = ~temp;
uint32_t shift = 0x80000000; // Number of leading zeros
int8_t count = 0;
for(int i=0; i<32; i++){
if((temp & shift) == 0) {count++; shift >>= 1;}
else break;
}
count--;
for (int i = 0; i < count; i++) readUint(8);
m_blockSize = 0;
if (FLACFrameHeader->blockSizeCode == 1)
m_blockSize = 192;
else if (2 <= FLACFrameHeader->blockSizeCode && FLACFrameHeader->blockSizeCode <= 5)
m_blockSize = 576 << (FLACFrameHeader->blockSizeCode - 2);
else if (FLACFrameHeader->blockSizeCode == 6)
m_blockSize = readUint(8) + 1;
else if (FLACFrameHeader->blockSizeCode == 7)
m_blockSize = readUint(16) + 1;
else if (8 <= FLACFrameHeader->blockSizeCode && FLACFrameHeader->blockSizeCode <= 15)
m_blockSize = 256 << (FLACFrameHeader->blockSizeCode - 8);
else{
return ERR_FLAC_RESERVED_BLOCKSIZE_UNSUPPORTED;
}
if(m_blockSize > 8192){
log_e("Error: blockSize too big");
return ERR_FLAC_BLOCKSIZE_TOO_BIG;
}
if(FLACFrameHeader->sampleRateCode == 12)
readUint(8);
else if (FLACFrameHeader->sampleRateCode == 13 || FLACFrameHeader->sampleRateCode == 14){
readUint(16);
}
readUint(8);
m_status = DECODE_SUBFRAMES;
*bytesLeft = m_bytesAvail;
m_blockSizeLeft = m_blockSize;
return ERR_FLAC_NONE;
}
if(m_status == DECODE_SUBFRAMES){
// Decode each channel's subframe, then skip footer
int ret = decodeSubframes();
if(ret != 0) return ret;
m_status = OUT_SAMPLES;
}
if(m_status == OUT_SAMPLES){ // Write the decoded samples
// blocksize can be much greater than outbuff, so we can't stuff all in once
// therefore we need often more than one loop (split outputblock into pieces)
uint16_t blockSize;
static uint16_t offset = 0;
if(m_blockSize < outBuffSize + offset) blockSize = m_blockSize - offset;
else blockSize = outBuffSize;
for (int i = 0; i < blockSize; i++) {
for (int j = 0; j < FLACMetadataBlock->numChannels; j++) {
int val = FLACsubFramesBuff->samplesBuffer[j][i + offset];
if (FLACMetadataBlock->bitsPerSample == 8) val += 128;
outbuf[2*i+j] = val;
}
}
m_validSamples = blockSize * FLACMetadataBlock->numChannels;
offset += blockSize;
if(offset != m_blockSize) return GIVE_NEXT_LOOP;
offset = 0;
if(offset > m_blockSize) { log_e("offset has a wrong value"); }
}
alignToByte();
readUint(16);
m_bytesDecoded = *bytesLeft - m_bytesAvail;
// log_i("m_bytesDecoded %i", m_bytesDecoded);
// m_compressionRatio = (float)m_bytesDecoded / (float)m_blockSize * FLACMetadataBlock->numChannels * (16/8);
// log_i("m_compressionRatio % f", m_compressionRatio);
*bytesLeft = m_bytesAvail;
m_status = DECODE_FRAME;
return ERR_FLAC_NONE;
}
//----------------------------------------------------------------------------------------------------------------------
uint16_t FLACGetOutputSamps(){
int vs = m_validSamples;
m_validSamples=0;
return vs;
}
//----------------------------------------------------------------------------------------------------------------------
uint64_t FLACGetTotoalSamplesInStream(){
return FLACMetadataBlock->totalSamples;
}
//----------------------------------------------------------------------------------------------------------------------
uint8_t FLACGetBitsPerSample(){
return FLACMetadataBlock->bitsPerSample;
}
//----------------------------------------------------------------------------------------------------------------------
uint8_t FLACGetChannels(){
return FLACMetadataBlock->numChannels;
}
//----------------------------------------------------------------------------------------------------------------------
uint32_t FLACGetSampRate(){
return FLACMetadataBlock->sampleRate;
}
//----------------------------------------------------------------------------------------------------------------------
uint32_t FLACGetBitRate(){
if(FLACMetadataBlock->totalSamples){
float BitsPerSamp = (float)FLACMetadataBlock->audioDataLength / (float)FLACMetadataBlock->totalSamples * 8;
return ((uint32_t)BitsPerSamp * FLACMetadataBlock->sampleRate);
}
return 0;
}
//----------------------------------------------------------------------------------------------------------------------
uint32_t FLACGetAudioFileDuration() {
if(FLACGetSampRate()){
uint32_t afd = FLACGetTotoalSamplesInStream()/ FLACGetSampRate(); // AudioFileDuration
return afd;
}
return 0;
}
//----------------------------------------------------------------------------------------------------------------------
int8_t decodeSubframes(){
if(FLACFrameHeader->chanAsgn <= 7) {
for (int ch = 0; ch < FLACMetadataBlock->numChannels; ch++)
decodeSubframe(FLACMetadataBlock->bitsPerSample, ch);
}
else if (8 <= FLACFrameHeader->chanAsgn && FLACFrameHeader->chanAsgn <= 10) {
decodeSubframe(FLACMetadataBlock->bitsPerSample + (FLACFrameHeader->chanAsgn == 9 ? 1 : 0), 0);
decodeSubframe(FLACMetadataBlock->bitsPerSample + (FLACFrameHeader->chanAsgn == 9 ? 0 : 1), 1);
if(FLACFrameHeader->chanAsgn == 8) {
for (int i = 0; i < m_blockSize; i++)
FLACsubFramesBuff->samplesBuffer[1][i] = (
FLACsubFramesBuff->samplesBuffer[0][i] -
FLACsubFramesBuff->samplesBuffer[1][i]);
}
else if (FLACFrameHeader->chanAsgn == 9) {
for (int i = 0; i < m_blockSize; i++)
FLACsubFramesBuff->samplesBuffer[0][i] += FLACsubFramesBuff->samplesBuffer[1][i];
}
else if (FLACFrameHeader->chanAsgn == 10) {
for (int i = 0; i < m_blockSize; i++) {
long side = FLACsubFramesBuff->samplesBuffer[1][i];
long right = FLACsubFramesBuff->samplesBuffer[0][i] - (side >> 1);
FLACsubFramesBuff->samplesBuffer[1][i] = right;
FLACsubFramesBuff->samplesBuffer[0][i] = right + side;
}
}
else {
log_e("unknown channel assignment");
return ERR_FLAC_UNKNOWN_CHANNEL_ASSIGNMENT;
}
}
else{
log_e("Reserved channel assignment");
return ERR_FLAC_RESERVED_CHANNEL_ASSIGNMENT;
}
return ERR_FLAC_NONE;
}
//----------------------------------------------------------------------------------------------------------------------
int8_t decodeSubframe(uint8_t sampleDepth, uint8_t ch) {
int8_t ret = 0;
readUint(1);
uint8_t type = readUint(6);
int shift = readUint(1);
if (shift == 1) {
while (readUint(1) == 0)
shift++;
}
sampleDepth -= shift;
if(type == 0){ // Constant coding
int16_t s= readSignedInt(sampleDepth);
for(int i=0; i < m_blockSize; i++){
FLACsubFramesBuff->samplesBuffer[ch][i] = s;
}
}
else if (type == 1) { // Verbatim coding
for (int i = 0; i < m_blockSize; i++)
FLACsubFramesBuff->samplesBuffer[ch][i] = readSignedInt(sampleDepth);
}
else if (8 <= type && type <= 12){
ret = decodeFixedPredictionSubframe(type - 8, sampleDepth, ch);
if(ret) return ret;
}
else if (32 <= type && type <= 63){
ret = decodeLinearPredictiveCodingSubframe(type - 31, sampleDepth, ch);
if(ret) return ret;
}
else{
return ERR_FLAC_RESERVED_SUB_TYPE;
}
if(shift>0){
for (int i = 0; i < m_blockSize; i++){
FLACsubFramesBuff->samplesBuffer[ch][i] <<= shift;
}
}
return ERR_FLAC_NONE;
}
//----------------------------------------------------------------------------------------------------------------------
int8_t decodeFixedPredictionSubframe(uint8_t predOrder, uint8_t sampleDepth, uint8_t ch) {
uint8_t ret = 0;
for(uint8_t i = 0; i < predOrder; i++)
FLACsubFramesBuff->samplesBuffer[ch][i] = readSignedInt(sampleDepth);
ret = decodeResiduals(predOrder, ch);
if(ret) return ret;
coefs.clear();
if(predOrder == 0) coefs.resize(0);
if(predOrder == 1) coefs.push_back(1); // FIXED_PREDICTION_COEFFICIENTS
if(predOrder == 2){coefs.push_back(2); coefs.push_back(-1);}
if(predOrder == 3){coefs.push_back(3); coefs.push_back(-3); coefs.push_back(1);}
if(predOrder == 4){coefs.push_back(4); coefs.push_back(-6); coefs.push_back(4); coefs.push_back(-1);}
if(predOrder > 4) return ERR_FLAC_PREORDER_TOO_BIG; // Error: preorder > 4"
restoreLinearPrediction(ch, 0);
return ERR_FLAC_NONE;
}
//----------------------------------------------------------------------------------------------------------------------
int8_t decodeLinearPredictiveCodingSubframe(int lpcOrder, int sampleDepth, uint8_t ch){
int8_t ret = 0;
for (int i = 0; i < lpcOrder; i++)
FLACsubFramesBuff->samplesBuffer[ch][i] = readSignedInt(sampleDepth);
int precision = readUint(4) + 1;
int shift = readSignedInt(5);
coefs.resize(0);
for (uint8_t i = 0; i < lpcOrder; i++)
coefs.push_back(readSignedInt(precision));
ret = decodeResiduals(lpcOrder, ch);
if(ret) return ret;
restoreLinearPrediction(ch, shift);
return ERR_FLAC_NONE;
}
//----------------------------------------------------------------------------------------------------------------------
int8_t decodeResiduals(uint8_t warmup, uint8_t ch) {
int method = readUint(2);
if (method >= 2)
return ERR_FLAC_RESERVED_RESIDUAL_CODING; // Reserved residual coding method
uint8_t paramBits = method == 0 ? 4 : 5;
int escapeParam = (method == 0 ? 0xF : 0x1F);
int partitionOrder = readUint(4);
int numPartitions = 1 << partitionOrder;
if (m_blockSize % numPartitions != 0)
return ERR_FLAC_WRONG_RICE_PARTITION_NR; //Error: Block size not divisible by number of Rice partitions
int partitionSize = m_blockSize/ numPartitions;
for (int i = 0; i < numPartitions; i++) {
int start = i * partitionSize + (i == 0 ? warmup : 0);
int end = (i + 1) * partitionSize;
int param = readUint(paramBits);
if (param < escapeParam) {
for (int j = start; j < end; j++){
FLACsubFramesBuff->samplesBuffer[ch][j] = readRiceSignedInt(param);
}
} else {
int numBits = readUint(5);
for (int j = start; j < end; j++){
FLACsubFramesBuff->samplesBuffer[ch][j] = readSignedInt(numBits);
}
}
}
return ERR_FLAC_NONE;
}
//----------------------------------------------------------------------------------------------------------------------
void restoreLinearPrediction(uint8_t ch, uint8_t shift) {
for (int i = coefs.size(); i < m_blockSize; i++) {
int32_t sum = 0;
for (int j = 0; j < coefs.size(); j++){
sum += FLACsubFramesBuff->samplesBuffer[ch][i - 1 - j] * coefs[j];
}
FLACsubFramesBuff->samplesBuffer[ch][i] += (sum >> shift);
}
}
//----------------------------------------------------------------------------------------------------------------------

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/*
* flac_decoder.h
*
* Created on: Jul 03,2020
* Updated on: Apr 27,2021
*
* Author: wolle
*
* Restrictions:
* blocksize must not exceed 8192
* bits per sample must be 8 or 16
* num Channels must be 1 or 2
*
*
*/
#pragma once
#pragma GCC optimize ("Ofast")
#include "Arduino.h"
#define MAX_CHANNELS 2
#define MAX_BLOCKSIZE 8192
#define APLL_DISABLE 0
#define EXTERNAL_I2S 0
typedef struct FLACsubFramesBuff_t{
int32_t samplesBuffer[MAX_CHANNELS][MAX_BLOCKSIZE];
}FLACsubframesBuffer_t;
enum : uint8_t {FLACDECODER_INIT, FLACDECODER_READ_IN, FLACDECODER_WRITE_OUT};
enum : uint8_t {DECODE_FRAME, DECODE_SUBFRAMES, OUT_SAMPLES};
enum : int8_t {GIVE_NEXT_LOOP = +1,
ERR_FLAC_NONE = 0,
ERR_FLAC_BLOCKSIZE_TOO_BIG = -1,
ERR_FLAC_RESERVED_BLOCKSIZE_UNSUPPORTED = -2,
ERR_FLAC_SYNC_CODE_NOT_FOUND = -3,
ERR_FLAC_UNKNOWN_CHANNEL_ASSIGNMENT = -4,
ERR_FLAC_RESERVED_CHANNEL_ASSIGNMENT = -5,
ERR_FLAC_RESERVED_SUB_TYPE = -6,
ERR_FLAC_PREORDER_TOO_BIG = -7,
ERR_FLAC_RESERVED_RESIDUAL_CODING = -8,
ERR_FLAC_WRONG_RICE_PARTITION_NR = -9,
ERR_FLAC_BITS_PER_SAMPLE_TOO_BIG = -10,
ERR_FLAG_BITS_PER_SAMPLE_UNKNOWN = 11};
typedef struct FLACMetadataBlock_t{
// METADATA_BLOCK_STREAMINFO
uint16_t minblocksize; // The minimum block size (in samples) used in the stream.
//----------------------------------------------------------------------------------------
// The maximum block size (in samples) used in the stream.
uint16_t maxblocksize; // (Minimum blocksize == maximum blocksize) implies a fixed-blocksize stream.
//----------------------------------------------------------------------------------------
// The minimum frame size (in bytes) used in the stream.
uint32_t minframesize; // May be 0 to imply the value is not known.
//----------------------------------------------------------------------------------------
// The maximum frame size (in bytes) used in the stream.
uint32_t maxframesize; // May be 0 to imply the value is not known.
//----------------------------------------------------------------------------------------
// Sample rate in Hz. Though 20 bits are available,
// the maximum sample rate is limited by the structure of frame headers to 655350Hz.
uint32_t sampleRate; // Also, a value of 0 is invalid.
//----------------------------------------------------------------------------------------
// Number of channels FLAC supports from 1 to 8 channels
uint8_t numChannels; // 000 : 1 channel .... 111 : 8 channels
//----------------------------------------------------------------------------------------
// Sample size in bits:
// 000 : get from STREAMINFO metadata block
// 001 : 8 bits per sample
// 010 : 12 bits per sample
// 011 : reserved
// 100 : 16 bits per sample
// 101 : 20 bits per sample
// 110 : 24 bits per sample
uint8_t bitsPerSample; // 111 : reserved
//----------------------------------------------------------------------------------------
// Total samples in stream. 'Samples' means inter-channel sample,
// i.e. one second of 44.1Khz audio will have 44100 samples regardless of the number
uint64_t totalSamples; // of channels. A value of zero here means the number of total samples is unknown.
//----------------------------------------------------------------------------------------
uint32_t audioDataLength;// is not the filelength, is only the length of the audio datablock in bytes
}FLACMetadataBlock_t;
typedef struct FLACFrameHeader_t {
// 0 : fixed-blocksize stream; frame header encodes the frame number
uint8_t blockingStrategy; // 1 : variable-blocksize stream; frame header encodes the sample number
//----------------------------------------------------------------------------------------
// Block size in inter-channel samples:
// 0000 : reserved
// 0001 : 192 samples
// 0010-0101 : 576 * (2^(n-2)) samples, i.e. 576/1152/2304/4608
// 0110 : get 8 bit (blocksize-1) from end of header
// 0111 : get 16 bit (blocksize-1) from end of header
uint8_t blockSizeCode; // 1000-1111 : 256 * (2^(n-8)) samples, i.e. 256/512/1024/2048/4096/8192/16384/32768
//----------------------------------------------------------------------------------------
// 0000 : get from STREAMINFO metadata block
// 0001 : 88.2kHz
// 0010 : 176.4kHz
// 0011 : 192kHz
// 0100 : 8kHz
// 0101 : 16kHz
// 0110 : 22.05kHz
// 0111 : 24kHz
// 1000 : 32kHz
// 1001 : 44.1kHz
// 1010 : 48kHz
// 1011 : 96kHz
// 1100 : get 8 bit sample rate (in kHz) from end of header
// 1101 : get 16 bit sample rate (in Hz) from end of header
// 1110 : get 16 bit sample rate (in tens of Hz) from end of header
uint8_t sampleRateCode; // 1111 : invalid, to prevent sync-fooling string of 1s
//----------------------------------------------------------------------------------------
// Channel assignment
// 0000 1 channel: mono
// 0001 2 channels: left, right
// 0010 3 channels
// 0011 4 channels
// 0100 5 channels
// 0101 6 channels
// 0110 7 channels
// 0111 8 channels
// 1000 : left/side stereo: channel 0 is the left channel, channel 1 is the side(difference) channel
// 1001 : right/side stereo: channel 0 is the side(difference) channel, channel 1 is the right channel
// 1010 : mid/side stereo: channel 0 is the mid(average) channel, channel 1 is the side(difference) channel
uint8_t chanAsgn; // 1011-1111 : reserved
//----------------------------------------------------------------------------------------
// Sample size in bits:
// 000 : get from STREAMINFO metadata block
// 001 : 8 bits per sample
// 010 : 12 bits per sample
// 011 : reserved
// 100 : 16 bits per sample
// 101 : 20 bits per sample
// 110 : 24 bits per sample
uint8_t sampleSizeCode; // 111 : reserved
//----------------------------------------------------------------------------------------
uint32_t totalSamples; // totalSamplesInStream
//----------------------------------------------------------------------------------------
uint32_t bitrate; // bitrate
}FLACFrameHeader_t;
int FLACFindSyncWord(unsigned char *buf, int nBytes);
int FLACFindOggSyncWord(unsigned char *buf, int nBytes);
int FLACparseOggHeader(unsigned char *buf);
bool FLACDecoder_AllocateBuffers(void);
void FLACDecoder_ClearBuffer();
void FLACDecoder_FreeBuffers();
void FLACSetRawBlockParams(uint8_t Chans, uint32_t SampRate, uint8_t BPS, uint32_t tsis, uint32_t AuDaLength);
void FLACDecoderReset();
int8_t FLACDecode(uint8_t *inbuf, int *bytesLeft, short *outbuf);
uint16_t FLACGetOutputSamps();
uint64_t FLACGetTotoalSamplesInStream();
uint8_t FLACGetBitsPerSample();
uint8_t FLACGetChannels();
uint32_t FLACGetSampRate();
uint32_t FLACGetBitRate();
uint32_t FLACGetAudioFileDuration();
uint32_t readUint(uint8_t nBits);
int32_t readSignedInt(int nBits);
int64_t readRiceSignedInt(uint8_t param);
void alignToByte();
int8_t decodeSubframes();
int8_t decodeSubframe(uint8_t sampleDepth, uint8_t ch);
int8_t decodeFixedPredictionSubframe(uint8_t predOrder, uint8_t sampleDepth, uint8_t ch);
int8_t decodeLinearPredictiveCodingSubframe(int lpcOrder, int sampleDepth, uint8_t ch);
int8_t decodeResiduals(uint8_t warmup, uint8_t ch);
void restoreLinearPrediction(uint8_t ch, uint8_t shift);

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// based om helix mp3 decoder
#pragma once
#include "Arduino.h"
#include "assert.h"
static const uint8_t m_HUFF_PAIRTABS =32;
static const uint8_t m_BLOCK_SIZE =18;
static const uint8_t m_NBANDS =32;
static const uint8_t m_MAX_REORDER_SAMPS =(192-126)*3; // largest critical band for short blocks (see sfBandTable)
static const uint16_t m_VBUF_LENGTH =17*2* m_NBANDS; // for double-sized vbuf FIFO
static const uint8_t m_MAX_SCFBD =4; // max scalefactor bands per channel
static const uint16_t m_MAINBUF_SIZE =1940;
static const uint8_t m_MAX_NGRAN =2; // max granules
static const uint8_t m_MAX_NCHAN =2; // max channels
static const uint16_t m_MAX_NSAMP =576; // max samples per channel, per granule
enum {
ERR_MP3_NONE = 0,
ERR_MP3_INDATA_UNDERFLOW = -1,
ERR_MP3_MAINDATA_UNDERFLOW = -2,
ERR_MP3_FREE_BITRATE_SYNC = -3,
ERR_MP3_OUT_OF_MEMORY = -4,
ERR_MP3_NULL_POINTER = -5,
ERR_MP3_INVALID_FRAMEHEADER = -6,
ERR_MP3_INVALID_SIDEINFO = -7,
ERR_MP3_INVALID_SCALEFACT = -8,
ERR_MP3_INVALID_HUFFCODES = -9,
ERR_MP3_INVALID_DEQUANTIZE = -10,
ERR_MP3_INVALID_IMDCT = -11,
ERR_MP3_INVALID_SUBBAND = -12,
ERR_UNKNOWN = -9999
};
typedef struct MP3FrameInfo {
int bitrate;
int nChans;
int samprate;
int bitsPerSample;
int outputSamps;
int layer;
int version;
} MP3FrameInfo_t;
typedef struct SFBandTable {
int/*short*/ l[23];
int/*short*/ s[14];
} SFBandTable_t;
typedef struct BitStreamInfo {
unsigned char *bytePtr;
unsigned int iCache;
int cachedBits;
int nBytes;
} BitStreamInfo_t;
typedef enum { /* map these to the corresponding 2-bit values in the frame header */
Stereo = 0x00, /* two independent channels, but L and R frames might have different # of bits */
Joint = 0x01, /* coupled channels - layer III: mix of M-S and intensity, Layers I/II: intensity and direct coding only */
Dual = 0x02, /* two independent channels, L and R always have exactly 1/2 the total bitrate */
Mono = 0x03 /* one channel */
} StereoMode_t;
typedef enum { /* map to 0,1,2 to make table indexing easier */
MPEG1 = 0,
MPEG2 = 1,
MPEG25 = 2
} MPEGVersion_t;
typedef struct FrameHeader {
int layer; /* layer index (1, 2, or 3) */
int crc; /* CRC flag: 0 = disabled, 1 = enabled */
int brIdx; /* bitrate index (0 - 15) */
int srIdx; /* sample rate index (0 - 2) */
int paddingBit; /* padding flag: 0 = no padding, 1 = single pad byte */
int privateBit; /* unused */
int modeExt; /* used to decipher joint stereo mode */
int copyFlag; /* copyright flag: 0 = no, 1 = yes */
int origFlag; /* original flag: 0 = copy, 1 = original */
int emphasis; /* deemphasis mode */
int CRCWord; /* CRC word (16 bits, 0 if crc not enabled) */
} FrameHeader_t;
typedef struct SideInfoSub {
int part23Length; /* number of bits in main data */
int nBigvals; /* 2x this = first set of Huffman cw's (maximum amplitude can be > 1) */
int globalGain; /* overall gain for dequantizer */
int sfCompress; /* unpacked to figure out number of bits in scale factors */
int winSwitchFlag; /* window switching flag */
int blockType; /* block type */
int mixedBlock; /* 0 = regular block (all short or long), 1 = mixed block */
int tableSelect[3]; /* index of Huffman tables for the big values regions */
int subBlockGain[3]; /* subblock gain offset, relative to global gain */
int region0Count; /* 1+region0Count = num scale factor bands in first region of bigvals */
int region1Count; /* 1+region1Count = num scale factor bands in second region of bigvals */
int preFlag; /* for optional high frequency boost */
int sfactScale; /* scaling of the scalefactors */
int count1TableSelect; /* index of Huffman table for quad codewords */
} SideInfoSub_t;
typedef struct SideInfo {
int mainDataBegin;
int privateBits;
int scfsi[m_MAX_NCHAN][m_MAX_SCFBD]; /* 4 scalefactor bands per channel */
} SideInfo_t;
typedef struct {
int cbType; /* pure long = 0, pure short = 1, mixed = 2 */
int cbEndS[3]; /* number nonzero short cb's, per subbblock */
int cbEndSMax; /* max of cbEndS[] */
int cbEndL; /* number nonzero long cb's */
} CriticalBandInfo_t;
typedef struct DequantInfo {
int workBuf[m_MAX_REORDER_SAMPS]; /* workbuf for reordering short blocks */
} DequantInfo_t;
typedef struct HuffmanInfo {
int huffDecBuf[m_MAX_NCHAN][m_MAX_NSAMP]; /* used both for decoded Huffman values and dequantized coefficients */
int nonZeroBound[m_MAX_NCHAN]; /* number of coeffs in huffDecBuf[ch] which can be > 0 */
int gb[m_MAX_NCHAN]; /* minimum number of guard bits in huffDecBuf[ch] */
} HuffmanInfo_t;
typedef enum HuffTabType {
noBits,
oneShot,
loopNoLinbits,
loopLinbits,
quadA,
quadB,
invalidTab
} HuffTabType_t;
typedef struct HuffTabLookup {
int linBits;
int tabType; /*HuffTabType*/
} HuffTabLookup_t;
typedef struct IMDCTInfo {
int outBuf[m_MAX_NCHAN][m_BLOCK_SIZE][m_NBANDS]; /* output of IMDCT */
int overBuf[m_MAX_NCHAN][m_MAX_NSAMP / 2]; /* overlap-add buffer (by symmetry, only need 1/2 size) */
int numPrevIMDCT[m_MAX_NCHAN]; /* how many IMDCT's calculated in this channel on prev. granule */
int prevType[m_MAX_NCHAN];
int prevWinSwitch[m_MAX_NCHAN];
int gb[m_MAX_NCHAN];
} IMDCTInfo_t;
typedef struct BlockCount {
int nBlocksLong;
int nBlocksTotal;
int nBlocksPrev;
int prevType;
int prevWinSwitch;
int currWinSwitch;
int gbIn;
int gbOut;
} BlockCount_t;
typedef struct ScaleFactorInfoSub { /* max bits in scalefactors = 5, so use char's to save space */
char l[23]; /* [band] */
char s[13][3]; /* [band][window] */
} ScaleFactorInfoSub_t;
typedef struct ScaleFactorJS { /* used in MPEG 2, 2.5 intensity (joint) stereo only */
int intensityScale;
int slen[4];
int nr[4];
} ScaleFactorJS_t;
/* NOTE - could get by with smaller vbuf if memory is more important than speed
* (in Subband, instead of replicating each block in FDCT32 you would do a memmove on the
* last 15 blocks to shift them down one, a hardware style FIFO)
*/
typedef struct SubbandInfo {
int vbuf[m_MAX_NCHAN * m_VBUF_LENGTH]; /* vbuf for fast DCT-based synthesis PQMF - double size for speed (no modulo indexing) */
int vindex; /* internal index for tracking position in vbuf */
} SubbandInfo_t;
typedef struct MP3DecInfo {
/* buffer which must be large enough to hold largest possible main_data section */
unsigned char mainBuf[m_MAINBUF_SIZE];
/* special info for "free" bitrate files */
int freeBitrateFlag;
int freeBitrateSlots;
/* user-accessible info */
int bitrate;
int nChans;
int samprate;
int nGrans; /* granules per frame */
int nGranSamps; /* samples per granule */
int nSlots;
int layer;
int mainDataBegin;
int mainDataBytes;
int part23Length[m_MAX_NGRAN][m_MAX_NCHAN];
} MP3DecInfo_t;
/* format = Q31
* #define M_PI 3.14159265358979323846
* double u = 2.0 * M_PI / 9.0;
* float c0 = sqrt(3.0) / 2.0;
* float c1 = cos(u);
* float c2 = cos(2*u);
* float c3 = sin(u);
* float c4 = sin(2*u);
*/
const int c9_0 = 0x6ed9eba1;
const int c9_1 = 0x620dbe8b;
const int c9_2 = 0x163a1a7e;
const int c9_3 = 0x5246dd49;
const int c9_4 = 0x7e0e2e32;
const int c3_0 = 0x6ed9eba1; /* format = Q31, cos(pi/6) */
const int c6[3] = { 0x7ba3751d, 0x5a82799a, 0x2120fb83 }; /* format = Q31, cos(((0:2) + 0.5) * (pi/6)) */
/* format = Q31
* cos(((0:8) + 0.5) * (pi/18))
*/
const uint32_t c18[9] = { 0x7f834ed0, 0x7ba3751d, 0x7401e4c1, 0x68d9f964, 0x5a82799a, 0x496af3e2, 0x36185aee, 0x2120fb83, 0x0b27eb5c};
/* scale factor lengths (num bits) */
const char m_SFLenTab[16][2] = { {0, 0}, {0, 1}, {0, 2}, {0, 3}, {3, 0}, {1, 1}, {1, 2}, {1, 3},
{2, 1}, {2, 2}, {2, 3}, {3, 1}, {3, 2}, {3, 3}, {4, 2}, {4, 3}};
/* NRTab[size + 3*is_right][block type][partition]
* block type index: 0 = (bt0,bt1,bt3), 1 = bt2 non-mixed, 2 = bt2 mixed
* partition: scale factor groups (sfb1 through sfb4)
* for block type = 2 (mixed or non-mixed) / by 3 is rolled into this table
* (for 3 short blocks per long block)
* see 2.4.3.2 in MPEG 2 (low sample rate) spec
* stuff rolled into this table:
* NRTab[x][1][y] --> (NRTab[x][1][y]) / 3
* NRTab[x][2][>=1] --> (NRTab[x][2][>=1]) / 3 (first partition is long block)
*/
const char NRTab[6][3][4] = {
{{ 6, 5, 5, 5}, {3, 3, 3, 3}, {6, 3, 3, 3}},
{{ 6, 5, 7, 3}, {3, 3, 4, 2}, {6, 3, 4, 2}},
{{11, 10, 0, 0}, {6, 6, 0, 0}, {6, 3, 6, 0}},
{{ 7, 7, 7, 0}, {4, 4, 4, 0}, {6, 5, 4, 0}},
{{ 6, 6, 6, 3}, {4, 3, 3, 2}, {6, 4, 3, 2}},
{{ 8, 8, 5, 0}, {5, 4, 3, 0}, {6, 6, 3, 0}}
};
/* optional pre-emphasis for high-frequency scale factor bands */
const char preTab[22] = { 0,0,0,0,0,0,0,0,0,0,0,1,1,1,1,2,2,3,3,3,2,0 };
/* pow(2,-i/4) for i=0..3, Q31 format */
const int pow14[4] PROGMEM = {
0x7fffffff, 0x6ba27e65, 0x5a82799a, 0x4c1bf829
};
/*
* Minimax polynomial approximation to pow(x, 4/3), over the range
* poly43lo: x = [0.5, 0.7071]
* poly43hi: x = [0.7071, 1.0]
*
* Relative error < 1E-7
* Coefs are scaled by 4, 2, 1, 0.5, 0.25
*/
const unsigned int poly43lo[5] PROGMEM = { 0x29a0bda9, 0xb02e4828, 0x5957aa1b, 0x236c498d, 0xff581859 };
const unsigned int poly43hi[5] PROGMEM = { 0x10852163, 0xd333f6a4, 0x46e9408b, 0x27c2cef0, 0xfef577b4 };
/* pow(2, i*4/3) as exp and frac */
const int pow2exp[8] PROGMEM = { 14, 13, 11, 10, 9, 7, 6, 5 };
const int pow2frac[8] PROGMEM = {
0x6597fa94, 0x50a28be6, 0x7fffffff, 0x6597fa94,
0x50a28be6, 0x7fffffff, 0x6597fa94, 0x50a28be6
};
const uint16_t m_HUFF_OFFSET_01= 0;
const uint16_t m_HUFF_OFFSET_02= 9 + m_HUFF_OFFSET_01;
const uint16_t m_HUFF_OFFSET_03= 65 + m_HUFF_OFFSET_02;
const uint16_t m_HUFF_OFFSET_05= 65 + m_HUFF_OFFSET_03;
const uint16_t m_HUFF_OFFSET_06=257 + m_HUFF_OFFSET_05;
const uint16_t m_HUFF_OFFSET_07=129 + m_HUFF_OFFSET_06;
const uint16_t m_HUFF_OFFSET_08=110 + m_HUFF_OFFSET_07;
const uint16_t m_HUFF_OFFSET_09=280 + m_HUFF_OFFSET_08;
const uint16_t m_HUFF_OFFSET_10= 93 + m_HUFF_OFFSET_09;
const uint16_t m_HUFF_OFFSET_11=320 + m_HUFF_OFFSET_10;
const uint16_t m_HUFF_OFFSET_12=296 + m_HUFF_OFFSET_11;
const uint16_t m_HUFF_OFFSET_13=185 + m_HUFF_OFFSET_12;
const uint16_t m_HUFF_OFFSET_15=497 + m_HUFF_OFFSET_13;
const uint16_t m_HUFF_OFFSET_16=580 + m_HUFF_OFFSET_15;
const uint16_t m_HUFF_OFFSET_24=651 + m_HUFF_OFFSET_16;
const int huffTabOffset[m_HUFF_PAIRTABS] PROGMEM = {
0, m_HUFF_OFFSET_01, m_HUFF_OFFSET_02, m_HUFF_OFFSET_03,
0, m_HUFF_OFFSET_05, m_HUFF_OFFSET_06, m_HUFF_OFFSET_07,
m_HUFF_OFFSET_08, m_HUFF_OFFSET_09, m_HUFF_OFFSET_10, m_HUFF_OFFSET_11,
m_HUFF_OFFSET_12, m_HUFF_OFFSET_13, 0, m_HUFF_OFFSET_15,
m_HUFF_OFFSET_16, m_HUFF_OFFSET_16, m_HUFF_OFFSET_16, m_HUFF_OFFSET_16,
m_HUFF_OFFSET_16, m_HUFF_OFFSET_16, m_HUFF_OFFSET_16, m_HUFF_OFFSET_16,
m_HUFF_OFFSET_24, m_HUFF_OFFSET_24, m_HUFF_OFFSET_24, m_HUFF_OFFSET_24,
m_HUFF_OFFSET_24, m_HUFF_OFFSET_24, m_HUFF_OFFSET_24, m_HUFF_OFFSET_24,};
const HuffTabLookup_t huffTabLookup[m_HUFF_PAIRTABS] PROGMEM = {
{ 0, noBits },
{ 0, oneShot },
{ 0, oneShot },
{ 0, oneShot },
{ 0, invalidTab },
{ 0, oneShot },
{ 0, oneShot },
{ 0, loopNoLinbits },
{ 0, loopNoLinbits },
{ 0, loopNoLinbits },
{ 0, loopNoLinbits },
{ 0, loopNoLinbits },
{ 0, loopNoLinbits },
{ 0, loopNoLinbits },
{ 0, invalidTab },
{ 0, loopNoLinbits },
{ 1, loopLinbits },
{ 2, loopLinbits },
{ 3, loopLinbits },
{ 4, loopLinbits },
{ 6, loopLinbits },
{ 8, loopLinbits },
{ 10, loopLinbits },
{ 13, loopLinbits },
{ 4, loopLinbits },
{ 5, loopLinbits },
{ 6, loopLinbits },
{ 7, loopLinbits },
{ 8, loopLinbits },
{ 9, loopLinbits },
{ 11, loopLinbits },
{ 13, loopLinbits },
};
const int quadTabOffset[2] PROGMEM = {0, 64};
const int quadTabMaxBits[2] PROGMEM = {6, 4};
/* indexing = [version][samplerate index]
* sample rate of frame (Hz)
*/
const int samplerateTab[3][3] PROGMEM = {
{ 44100, 48000, 32000 }, /* MPEG-1 */
{ 22050, 24000, 16000 }, /* MPEG-2 */
{ 11025, 12000, 8000 }, /* MPEG-2.5 */
};
/* indexing = [version][layer]
* number of samples in one frame (per channel)
*/
const int/*short*/samplesPerFrameTab[3][3] PROGMEM = { { 384, 1152, 1152 }, /* MPEG1 */
{ 384, 1152, 576 }, /* MPEG2 */
{ 384, 1152, 576 }, /* MPEG2.5 */
};
/* layers 1, 2, 3 */
const short bitsPerSlotTab[3] = { 32, 8, 8 };
/* indexing = [version][mono/stereo]
* number of bytes in side info section of bitstream
*/
const int/*short*/sideBytesTab[3][2] PROGMEM = { { 17, 32 }, /* MPEG-1: mono, stereo */
{ 9, 17 }, /* MPEG-2: mono, stereo */
{ 9, 17 }, /* MPEG-2.5: mono, stereo */
};
/* indexing = [version][sampleRate][long (.l) or short (.s) block]
* sfBandTable[v][s].l[cb] = index of first bin in critical band cb (long blocks)
* sfBandTable[v][s].s[cb] = index of first bin in critical band cb (short blocks)
*/
const SFBandTable_t sfBandTable[3][3] PROGMEM = {
{ /* MPEG-1 (44, 48, 32 kHz) */
{ {0, 4, 8, 12, 16, 20, 24, 30, 36, 44, 52, 62, 74, 90, 110, 134, 162, 196, 238, 288, 342, 418, 576 },
{0, 4, 8, 12, 16, 22, 30, 40, 52, 66, 84, 106, 136, 192} },
{ {0, 4, 8, 12, 16, 20, 24, 30, 36, 42, 50, 60, 72, 88, 106, 128, 156, 190, 230, 276, 330, 384, 576 },
{0, 4, 8, 12, 16, 22, 28, 38, 50, 64, 80, 100, 126, 192} },
{ {0, 4, 8, 12, 16, 20, 24, 30, 36, 44, 54, 66, 82, 102, 126, 156, 194, 240, 296, 364, 448, 550, 576 },
{0, 4, 8, 12, 16, 22, 30, 42, 58, 78, 104, 138, 180, 192} } },
{ /* MPEG-2 (22, 24, 16 kHz) */
{ {0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 116, 140, 168, 200, 238, 284, 336, 396, 464, 522, 576 },
{0, 4, 8, 12, 18, 24, 32, 42, 56, 74, 100, 132, 174, 192} },
{ {0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 114, 136, 162, 194, 232, 278, 332, 394, 464, 540, 576 },
{0, 4, 8, 12, 18, 26, 36, 48, 62, 80, 104, 136, 180, 192} },
{ {0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 116, 140, 168, 200, 238, 284, 336, 396, 464, 522, 576 },
{0, 4, 8, 12, 18, 26, 36, 48, 62, 80, 104, 134, 174, 192} }, },
{ /* MPEG-2.5 (11, 12, 8 kHz) */
{ {0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 116, 140, 168, 200, 238, 284, 336, 396, 464, 522, 576 },
{0, 4, 8, 12, 18, 26, 36, 48, 62, 80, 104, 134, 174, 192 } },
{ {0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 116, 140, 168, 200, 238, 284, 336, 396, 464, 522, 576 },
{0, 4, 8, 12, 18, 26, 36, 48, 62, 80, 104, 134, 174, 192 } },
{ {0, 12, 24, 36, 48, 60, 72, 88, 108, 132, 160, 192, 232, 280, 336, 400, 476, 566, 568, 570, 572, 574, 576 },
{0, 8, 16, 24, 36, 52, 72, 96, 124, 160, 162, 164, 166, 192 } }, },
};
/* indexing = [intensity scale on/off][left/right]
* format = Q30, range = [0.0, 1.414]
*
* illegal intensity position scalefactors (see comments on ISFMpeg1)
*/
const int ISFIIP[2][2] PROGMEM = {
{0x40000000, 0x00000000}, /* mid-side off */
{0x40000000, 0x40000000}, /* mid-side on */
};
const unsigned char uniqueIDTab[8] = {0x5f, 0x4b, 0x43, 0x5f, 0x5f, 0x4a, 0x52, 0x5f};
/* anti-alias coefficients - see spec Annex B, table 3-B.9
* csa[0][i] = CSi, csa[1][i] = CAi
* format = Q31
*/
const uint32_t csa[8][2] PROGMEM = {
{0x6dc253f0, 0xbe2500aa},
{0x70dcebe4, 0xc39e4949},
{0x798d6e73, 0xd7e33f4a},
{0x7ddd40a7, 0xe8b71176},
{0x7f6d20b7, 0xf3e4fe2f},
{0x7fe47e40, 0xfac1a3c7},
{0x7ffcb263, 0xfe2ebdc6},
{0x7fffc694, 0xff86c25d},
};
/* format = Q30, right shifted by 12 (sign bits only in top 12 - undo this when rounding to short)
* this is to enable early-terminating multiplies on ARM
* range = [-1.144287109, 1.144989014]
* max gain of filter (per output sample) ~= 2.731
*
* new (properly sign-flipped) values
* - these actually are correct to 32 bits, (floating-pt coefficients in spec
* chosen such that only ~20 bits are required)
*
* Reordering - see table 3-B.3 in spec (appendix B)
*
* polyCoef[i] =
* D[ 0, 32, 64, ... 480], i = [ 0, 15]
* D[ 1, 33, 65, ... 481], i = [ 16, 31]
* D[ 2, 34, 66, ... 482], i = [ 32, 47]
* ...
* D[15, 47, 79, ... 495], i = [240,255]
*
* also exploits symmetry: D[i] = -D[512 - i], for i = [1, 255]
*
* polyCoef[256, 257, ... 263] are for special case of sample 16 (out of 0)
* see PolyphaseStereo() and PolyphaseMono()
*/
// prototypes
bool MP3Decoder_AllocateBuffers(void);
void MP3Decoder_FreeBuffers();
int MP3Decode( unsigned char *inbuf, int *bytesLeft, short *outbuf, int useSize);
void MP3GetLastFrameInfo();
int MP3GetNextFrameInfo(unsigned char *buf);
int MP3FindSyncWord(unsigned char *buf, int nBytes);
int MP3GetSampRate();
int MP3GetChannels();
int MP3GetBitsPerSample();
int MP3GetBitrate();
int MP3GetOutputSamps();
//internally used
void MP3Decoder_ClearBuffer(void);
void PolyphaseMono(short *pcm, int *vbuf, const uint32_t *coefBase);
void PolyphaseStereo(short *pcm, int *vbuf, const uint32_t *coefBase);
void SetBitstreamPointer(BitStreamInfo_t *bsi, int nBytes, unsigned char *buf);
unsigned int GetBits(BitStreamInfo_t *bsi, int nBits);
int CalcBitsUsed(BitStreamInfo_t *bsi, unsigned char *startBuf, int startOffset);
int DequantChannel(int *sampleBuf, int *workBuf, int *nonZeroBound, SideInfoSub_t *sis, ScaleFactorInfoSub_t *sfis, CriticalBandInfo_t *cbi);
void MidSideProc(int x[m_MAX_NCHAN][m_MAX_NSAMP], int nSamps, int mOut[2]);
void IntensityProcMPEG1(int x[m_MAX_NCHAN][m_MAX_NSAMP], int nSamps, ScaleFactorInfoSub_t *sfis, CriticalBandInfo_t *cbi, int midSideFlag, int mixFlag, int mOut[2]);
void IntensityProcMPEG2(int x[m_MAX_NCHAN][m_MAX_NSAMP], int nSamps, ScaleFactorInfoSub_t *sfis, CriticalBandInfo_t *cbi, ScaleFactorJS_t *sfjs, int midSideFlag, int mixFlag, int mOut[2]);
void FDCT32(int *x, int *d, int offset, int oddBlock, int gb);// __attribute__ ((section (".data")));
void FreeBuffers();
int CheckPadBit();
int UnpackFrameHeader(unsigned char *buf);
int UnpackSideInfo(unsigned char *buf);
int DecodeHuffman( unsigned char *buf, int *bitOffset, int huffBlockBits, int gr, int ch);
int MP3Dequantize( int gr);
int IMDCT( int gr, int ch);
int UnpackScaleFactors( unsigned char *buf, int *bitOffset, int bitsAvail, int gr, int ch);
int Subband(short *pcmBuf);
short ClipToShort(int x, int fracBits);
void RefillBitstreamCache(BitStreamInfo_t *bsi);
void UnpackSFMPEG1(BitStreamInfo_t *bsi, SideInfoSub_t *sis, ScaleFactorInfoSub_t *sfis, int *scfsi, int gr, ScaleFactorInfoSub_t *sfisGr0);
void UnpackSFMPEG2(BitStreamInfo_t *bsi, SideInfoSub_t *sis, ScaleFactorInfoSub_t *sfis, int gr, int ch, int modeExt, ScaleFactorJS_t *sfjs);
int MP3FindFreeSync(unsigned char *buf, unsigned char firstFH[4], int nBytes);
void MP3ClearBadFrame( short *outbuf);
int DecodeHuffmanPairs(int *xy, int nVals, int tabIdx, int bitsLeft, unsigned char *buf, int bitOffset);
int DecodeHuffmanQuads(int *vwxy, int nVals, int tabIdx, int bitsLeft, unsigned char *buf, int bitOffset);
int DequantBlock(int *inbuf, int *outbuf, int num, int scale);
void AntiAlias(int *x, int nBfly);
void WinPrevious(int *xPrev, int *xPrevWin, int btPrev);
int FreqInvertRescale(int *y, int *xPrev, int blockIdx, int es);
void idct9(int *x);
int IMDCT36(int *xCurr, int *xPrev, int *y, int btCurr, int btPrev, int blockIdx, int gb);
void imdct12(int *x, int *out);
int IMDCT12x3(int *xCurr, int *xPrev, int *y, int btPrev, int blockIdx, int gb);
int HybridTransform(int *xCurr, int *xPrev, int y[m_BLOCK_SIZE][m_NBANDS], SideInfoSub_t *sis, BlockCount_t *bc);
inline uint64_t SAR64(uint64_t x, int n) {return x >> n;}
inline int MULSHIFT32(int x, int y) { int z; z = (uint64_t) x * (uint64_t) y >> 32; return z;}
inline uint64_t MADD64(uint64_t sum64, int x, int y) {sum64 += (uint64_t) x * (uint64_t) y; return sum64;}/* returns 64-bit value in [edx:eax] */
inline uint64_t xSAR64(uint64_t x, int n){return x >> n;}
inline int FASTABS(int x){ return __builtin_abs(x);} //xtensa has a fast abs instruction //fb
#define CLZ(x) __builtin_clz(x) //fb