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ESP32-audioI2S
Unverified Commit b1a3ffbe authored by Wolle's avatar Wolle Committed by GitHub
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better error recognition 

Introduction of a variable for the error code of the decoder,
Avoiding an overflow of the InBuffer by corrupted audio files
parent cb576f29
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Showing with 43 additions and 44 deletions
src/Audio.cpp
View file @ b1a3ffbe
......@@ -3,8 +3,8 @@
*
* Created on: Oct 26.2018
*
* Version 3.0.1o
* Updated on: Apr 07.2023
* Version 3.0.1p
* Updated on: Apr 19.2023
* Author: Wolle (schreibfaul1)
*
*/
......@@ -2847,31 +2847,28 @@ void Audio::processLocalFile() {
// end of file reached? - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
if(f_fileDataComplete && InBuff.bufferFilled() < InBuff.getMaxBlockSize()){
if(InBuff.bufferFilled()){
if(!readID3V1Tag()){
int bytesDecoded = sendBytes(InBuff.getReadPtr(), InBuff.bufferFilled());
if(m_f_playing){
if(bytesDecoded > 2){InBuff.bytesWasRead(bytesDecoded); return;}
if(bytesDecoded <= InBuff.bufferFilled()){ // avoid InBuff overrun (can be if file is corrupt)
if(m_f_playing){
if(bytesDecoded > 2){InBuff.bytesWasRead(bytesDecoded); return;}
}
}
AUDIO_INFO("audio file is corrupt --> send EOF"); // no return, fall through
}
}
if(m_f_loop && f_stream){ //eof
AUDIO_INFO("loop from: %u to: %u", getFilePos(), m_audioDataStart); //TEST loop
AUDIO_INFO("loop from: %u to: %u", getFilePos(), m_audioDataStart); // loop
setFilePos(m_audioDataStart);
if(m_codec == CODEC_FLAC) FLACDecoderReset();
/*
The current time of the loop mode is not reset,
which will cause the total audio duration to be exceeded.
For example: current time ====progress bar====> total audio duration
3:43 ====================> 3:33
*/
m_audioCurrentTime = 0;
byteCounter = m_audioDataStart;
f_fileDataComplete = false;
return;
} //TEST loop
} //loop
#ifdef SDFATFS_USED
audiofile.getName(m_chbuf, m_chbufSize); // #426
......@@ -3996,7 +3993,7 @@ int Audio::sendBytes(uint8_t* data, size_t len) {
}
// m_f_playing is true at this pos
bytesLeft = len;
int ret = 0;
m_decodeError= 0;
int bytesDecoded = 0;
switch(m_codec){
......@@ -4004,40 +4001,41 @@ int Audio::sendBytes(uint8_t* data, size_t len) {
if(getBitsPerSample() == 16) m_validSamples = len / (2 * getChannels());
if(getBitsPerSample() == 8 ) m_validSamples = len / 2;
bytesLeft = 0; break;
case CODEC_MP3: ret = MP3Decode( data, &bytesLeft, m_outBuff, 0); break;
case CODEC_AAC: ret = AACDecode( data, &bytesLeft, m_outBuff); break;
case CODEC_M4A: ret = AACDecode( data, &bytesLeft, m_outBuff); break;
case CODEC_FLAC: ret = FLACDecode( data, &bytesLeft, m_outBuff); break;
case CODEC_OPUS: ret = OPUSDecode( data, &bytesLeft, m_outBuff); break;
case CODEC_VORBIS: ret = VORBISDecode(data, &bytesLeft, m_outBuff); break;
case CODEC_MP3: m_decodeError = MP3Decode( data, &bytesLeft, m_outBuff, 0); break;
case CODEC_AAC: m_decodeError = AACDecode( data, &bytesLeft, m_outBuff); break;
case CODEC_M4A: m_decodeError = AACDecode( data, &bytesLeft, m_outBuff); break;
case CODEC_FLAC: m_decodeError = FLACDecode( data, &bytesLeft, m_outBuff); break;
case CODEC_OPUS: m_decodeError = OPUSDecode( data, &bytesLeft, m_outBuff); break;
case CODEC_VORBIS: m_decodeError = VORBISDecode(data, &bytesLeft, m_outBuff); break;
default: {log_e("no valid codec found codec = %d", m_codec); stopSong();}
}
bytesDecoded = len - bytesLeft;
if(bytesDecoded == 0 && ret == 0){ // unlikely framesize
if(audio_info) audio_info("framesize is 0, start decoding again");
m_f_playing = false; // seek for new syncword
// we're here because there was a wrong sync word
// so skip two sync bytes and seek for next
return 1;
}
if(ret < 0) { // Error, skip the frame...
if(m_f_Log) if(m_codec == CODEC_M4A){log_i("begin not found"); return 1;}
// m_decodeError - possible values are:
// 0: okay, no error
// 100: the decoder needs more data
// < 0: there has been an error
if(m_decodeError < 0){ // Error, skip the frame...
i2s_zero_dma_buffer((i2s_port_t)m_i2s_num);
if(!getChannels() && (ret == -2)) {
; // suppress errorcode MAINDATA_UNDERFLOW
if(!getChannels() && m_codec == CODEC_MP3 && (m_decodeError == -2)) {
; // at the beginning this doesn't have to be a mistake, suppress errorcode MAINDATA_UNDERFLOW
}
else {
if(m_codec == CODEC_MP3 && ret == -2){ ; } // do nothing (suppress MAINDATA_UNDERFLOW)
else{
printDecodeError(ret);
m_f_playing = false; // seek for new syncword
}
printDecodeError(m_decodeError);
m_f_playing = false; // seek for new syncword
}
if(!bytesDecoded) bytesDecoded = 2;
return bytesDecoded;
return 1; // skip one byte and seek for the next sync word
}
bytesDecoded = len - bytesLeft;
if(bytesDecoded == 0 && m_decodeError == 0){ // unlikely framesize
if(audio_info) audio_info("framesize is 0, start decoding again");
m_f_playing = false; // seek for new syncword
// we're here because there was a wrong sync word so skip one byte and seek for the next
return 1;
}
else{ // ret>=0
// status: bytesDecoded > 0 and m_decodeError >= 0
{
if(m_codec == CODEC_MP3){
m_validSamples = MP3GetOutputSamps() / getChannels();
}
......@@ -4046,7 +4044,7 @@ int Audio::sendBytes(uint8_t* data, size_t len) {
}
if(m_codec == CODEC_FLAC){
const uint8_t FLAC_PARSE_OGG_DONE = 100;
if(ret == FLAC_PARSE_OGG_DONE) return bytesDecoded; // nothing to play
if(m_decodeError == FLAC_PARSE_OGG_DONE) return bytesDecoded; // nothing to play
m_validSamples = FLACGetOutputSamps() / getChannels();
char* st = FLACgetStreamTitle();
if(st){
......@@ -4056,7 +4054,7 @@ int Audio::sendBytes(uint8_t* data, size_t len) {
}
if(m_codec == CODEC_OPUS){
const uint8_t OPUS_PARSE_OGG_DONE = 100;
if(ret == OPUS_PARSE_OGG_DONE) return bytesDecoded; // nothing to play
if(m_decodeError == OPUS_PARSE_OGG_DONE) return bytesDecoded; // nothing to play
m_validSamples = OPUSGetOutputSamps();
char* st = OPUSgetStreamTitle();
if(st){
......@@ -4066,7 +4064,7 @@ int Audio::sendBytes(uint8_t* data, size_t len) {
}
if(m_codec == CODEC_VORBIS){
const uint8_t VORBIS_PARSE_OGG_DONE = 100;
if(ret == VORBIS_PARSE_OGG_DONE) return bytesDecoded; // nothing to play
if(m_decodeError == VORBIS_PARSE_OGG_DONE) return bytesDecoded; // nothing to play
m_validSamples = VORBISGetOutputSamps();
char* st = VORBISgetStreamTitle();
if(st){
......
src/Audio.h
View file @ b1a3ffbe
......@@ -3,8 +3,8 @@
*
* Created on: Oct 28,2018
*
* Version 3.0.1n
* Updated on: Apr 07,2023
* Version 3.0.1p
* Updated on: Apr 19,2023
* Author: Wolle (schreibfaul1)
*/
......@@ -523,6 +523,7 @@ private:
int16_t* m_outBuff = NULL; // Interleaved L/R
int16_t m_validSamples = 0;
int16_t m_curSample = 0;
int16_t m_decodeError = 0; // Stores the return value of the decoder
uint16_t m_datamode = 0; // Statemaschine
uint16_t m_streamTitleHash = 0; // remember streamtitle, ignore multiple occurence in metadata
uint16_t m_timeout_ms = 250;
......
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