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ESP32-audioI2S
Unverified Commit e981b046 authored by Wolle's avatar Wolle Committed by GitHub
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Preparation for a new decoder

parent e2db546f
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Showing with 72 additions and 28 deletions
src/Audio.cpp
View file @ e981b046
......@@ -3,8 +3,8 @@
*
* Created on: Oct 26.2018
*
* Version 3.0.1k
* Updated on: Mar 30.2023
* Version 3.0.1l
* Updated on: Apr 06.2023
* Author: Wolle (schreibfaul1)
*
*/
......@@ -13,6 +13,7 @@
#include "aac_decoder/aac_decoder.h"
#include "flac_decoder/flac_decoder.h"
#include "opus_decoder/opus_decoder.h"
#include "vorbis_decoder/vorbis_decoder.h"
#ifdef SDFATFS_USED
fs::SDFATFS SD_SDFAT;
......@@ -325,6 +326,7 @@ void Audio::setDefaults() {
FLACDecoder_FreeBuffers();
AACDecoder_FreeBuffers();
OPUSDecoder_FreeBuffers();
VORBISDecoder_FreeBuffers();
if(m_playlistBuff) {free(m_playlistBuff); m_playlistBuff = NULL;} // free if stream is not m3u8
vector_clear_and_shrink(m_playlistURL);
vector_clear_and_shrink(m_playlistContent);
......@@ -757,6 +759,7 @@ bool Audio::connecttoFS(fs::FS &fs, const char* path, uint32_t resumeFilePos) {
if(endsWith(afn, ".wav")) m_codec = CODEC_WAV;
if(endsWith(afn, ".flac")) m_codec = CODEC_FLAC;
if(endsWith(afn, ".opus")) m_codec = CODEC_OPUS;
if(endsWith(afn, ".ogg")) m_codec = CODEC_VORBIS;
if(m_codec == CODEC_NONE) AUDIO_INFO("The %s format is not supported", afn + dotPos);
......@@ -1213,6 +1216,9 @@ size_t Audio::readAudioHeader(uint32_t bytes){
if(m_codec == CODEC_OPUS){
m_controlCounter = 100;
}
if(m_codec == CODEC_VORBIS){
m_controlCounter = 100;
}
if(!isRunning()){
log_e("Processing stopped due to invalid audio header");
return 0;
......@@ -2876,11 +2882,12 @@ void Audio::processLocalFile() {
audiofile.close();
AUDIO_INFO("Closing audio file");
if(m_codec == CODEC_MP3) MP3Decoder_FreeBuffers();
if(m_codec == CODEC_AAC) AACDecoder_FreeBuffers();
if(m_codec == CODEC_M4A) AACDecoder_FreeBuffers();
if(m_codec == CODEC_FLAC) FLACDecoder_FreeBuffers();
if(m_codec == CODEC_OPUS) OPUSDecoder_FreeBuffers();
if(m_codec == CODEC_MP3) MP3Decoder_FreeBuffers();
if(m_codec == CODEC_AAC) AACDecoder_FreeBuffers();
if(m_codec == CODEC_M4A) AACDecoder_FreeBuffers();
if(m_codec == CODEC_FLAC) FLACDecoder_FreeBuffers();
if(m_codec == CODEC_OPUS) OPUSDecoder_FreeBuffers();
if(m_codec == CODEC_VORBIS) VORBISDecoder_FreeBuffers();
AUDIO_INFO("End of file \"%s\"", afn);
if(audio_eof_mp3) audio_eof_mp3(afn);
if(afn) {free(afn); afn = NULL;}
......@@ -2954,8 +2961,9 @@ void Audio::processWebStream() {
if(!f_stream) return;
if(m_codec == CODEC_OGG){ // log_i("determine correct codec here");
uint8_t codec = determineOggCodec(InBuff.getReadPtr(), maxFrameSize);
if(codec == CODEC_FLAC) {m_codec = CODEC_FLAC; initializeDecoder(); return;}
if(codec == CODEC_OPUS) {m_codec = CODEC_OPUS; initializeDecoder(); return;}
if(codec == CODEC_FLAC) {m_codec = CODEC_FLAC; initializeDecoder(); return;}
if(codec == CODEC_OPUS) {m_codec = CODEC_OPUS; initializeDecoder(); return;}
if(codec == CODEC_VORBIS) {m_codec = CODEC_VORBIS; initializeDecoder(); return;}
stopSong();
return;
}
......@@ -3026,8 +3034,7 @@ void Audio::processWebFile() {
if(InBuff.bufferFilled() > maxFrameSize && !f_stream) { // waiting for buffer filled
f_stream = true; // ready to play the audio data
uint16_t filltime = millis() - m_t0;
if(m_f_Log) AUDIO_INFO("stream ready");
if(m_f_Log) AUDIO_INFO("buffer filled in %d ms", filltime);
if(m_f_Log) AUDIO_INFO("stream ready\nbuffer filled in %d ms", filltime);
}
if(!f_stream) return;
......@@ -3590,6 +3597,16 @@ bool Audio:: initializeDecoder(){
AUDIO_INFO("OPUSDecoder has been initialized, free Heap: %u bytes , free stack %u DWORDs", gfH, hWM);
InBuff.changeMaxBlockSize(m_frameSizeOPUS);
break;
case CODEC_VORBIS:
if(!VORBISDecoder_AllocateBuffers()){
AUDIO_INFO("The VORBISDecoder could not be initialized");
goto exit;
}
gfH = ESP.getFreeHeap();
hWM = uxTaskGetStackHighWaterMark(NULL);
AUDIO_INFO("VORBISDecoder has been initialized, free Heap: %u bytes, free stack %u DWORDs", gfH, hWM);
InBuff.changeMaxBlockSize(m_frameSizeVORBIS);
break;
case CODEC_WAV:
InBuff.changeMaxBlockSize(m_frameSizeWav);
break;
......@@ -3609,7 +3626,7 @@ bool Audio:: initializeDecoder(){
bool Audio::parseContentType(char* ct) {
enum : int {CT_NONE, CT_MP3, CT_AAC, CT_M4A, CT_WAV, CT_FLAC, CT_PLS, CT_M3U, CT_ASX,
CT_M3U8, CT_TXT, CT_AACP, CT_OPUS, CT_OGG};
CT_M3U8, CT_TXT, CT_AACP, CT_OPUS, CT_OGG, CT_VORBIS};
strlower(ct);
trim(ct);
......@@ -3683,6 +3700,10 @@ bool Audio::parseContentType(char* ct) {
m_codec = CODEC_OPUS;
if(m_f_Log) { log_i("ContentType %s, format is opus", ct); }
break;
case CT_VORBIS:
m_codec = CODEC_VORBIS;
log_i("ContentType %s, format is vorbis", ct);
break;
case CT_WAV:
m_codec = CODEC_WAV;
if(m_f_Log) { log_i("ContentType %s, format is wav", ct); }
......@@ -3888,6 +3909,10 @@ int Audio::findNextSync(uint8_t* data, size_t len){
nextSync = OPUSFindSyncWord(data, len);
if(nextSync == -1) return len; // OggS not found, search next block
}
if(m_codec == CODEC_VORBIS){
nextSync = VORBISFindSyncWord(data, len);
if(nextSync == -1) return len; // OggS not found, search next block
}
if(nextSync == -1) {
if(audio_info && swnf == 0) audio_info("syncword not found");
else {
......@@ -3935,6 +3960,12 @@ void Audio::setDecoderItems(){
setBitsPerSample(OPUSGetBitsPerSample());
setBitrate(OPUSGetBitRate());
}
if(m_codec == CODEC_VORBIS){
setChannels(2);
setSampleRate(VORBISGetSampRate());
setBitsPerSample(VORBISGetBitsPerSample());
setBitrate(VORBISGetBitRate());
}
showCodecParams();
}
//---------------------------------------------------------------------------------------------------------------------
......@@ -3958,11 +3989,12 @@ 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_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;
default: {log_e("no valid codec found codec = %d", m_codec); stopSong();}
}
......@@ -4017,6 +4049,16 @@ int Audio::sendBytes(uint8_t* data, size_t len) {
if(audio_showstreamtitle) audio_showstreamtitle(st);
}
}
if(m_codec == CODEC_VORBIS){
const uint8_t VORBIS_PARSE_OGG_DONE = 100;
if(ret == VORBIS_PARSE_OGG_DONE) return bytesDecoded; // nothing to play
m_validSamples = VORBISGetOutputSamps();
char* st = VORBISgetStreamTitle();
if(st){
AUDIO_INFO(st);
if(audio_showstreamtitle) audio_showstreamtitle(st);
}
}
if(f_setDecodeParamsOnce){
f_setDecodeParamsOnce = false;
setDecoderItems();
......@@ -4233,6 +4275,7 @@ uint32_t Audio::getAudioCurrentTime() { // return current time in seconds
//---------------------------------------------------------------------------------------------------------------------
bool Audio::setAudioPlayPosition(uint16_t sec){
if(m_codec == CODEC_OPUS) return false; // not impl. yet
if(m_codec == CODEC_VORBIS) return false; // not impl. yet
// Jump to an absolute position in time within an audio file
// e.g. setAudioPlayPosition(300) sets the pointer at pos 5 min
if(sec > getAudioFileDuration()) sec = getAudioFileDuration();
......@@ -4261,6 +4304,7 @@ uint32_t Audio::getTotalPlayingTime() {
//---------------------------------------------------------------------------------------------------------------------
bool Audio::setTimeOffset(int sec){
if(m_codec == CODEC_OPUS) return false; // not impl. yet
if(m_codec == CODEC_VORBIS) return false; // not impl. yet
// fast forward or rewind the current position in seconds
// audiosource must be a mp3, aac or wav file
......@@ -4283,13 +4327,12 @@ bool Audio::setTimeOffset(int sec){
}
//---------------------------------------------------------------------------------------------------------------------
bool Audio::setFilePos(uint32_t pos) {
if(m_codec == CODEC_OPUS) return false; // not impl. yet
// xSemaphoreTakeRecursive(mutex_audio, portMAX_DELAY); //#497
if(m_codec == CODEC_OPUS) return false; // not impl. yet
if(m_codec == CODEC_VORBIS) return false; // not impl. yet
if(!audiofile) return false;
if(pos < m_audioDataStart) pos = m_audioDataStart; // issue #96
if(pos > m_file_size) pos = m_file_size;
m_resumeFilePos = pos;
// xSemaphoreGiveRecursive(mutex_audio);
return true;
}
//---------------------------------------------------------------------------------------------------------------------
......@@ -5378,7 +5421,7 @@ uint8_t Audio::determineOggCodec(uint8_t* data, uint16_t len){
idx = specialIndexOf(data, "FLAC", 20);
if(idx >= 0) return CODEC_FLAC;
idx = specialIndexOf(data, "vorbis", 20);
if(idx >= 0) return CODEC_NONE; // CODEC_VORBIS
if(idx >= 0) return CODEC_VORBIS;
return CODEC_NONE;
}
//----------------------------------------------------------------------------------------------------------------------
......
src/Audio.h
View file @ e981b046
......@@ -4,7 +4,7 @@
* Created on: Oct 28,2018
*
* Version 3.0.1k
* Updated on: Mar 30,2023
* Updated on: Apr 06,2023
* Author: Wolle (schreibfaul1)
*/
......@@ -453,7 +453,7 @@ private:
enum : int { M4A_BEGIN = 0, M4A_FTYP = 1, M4A_CHK = 2, M4A_MOOV = 3, M4A_FREE = 4, M4A_TRAK = 5, M4A_MDAT = 6,
M4A_ILST = 7, M4A_MP4A = 8, M4A_AMRDY = 99, M4A_OKAY = 100};
enum : int { CODEC_NONE = 0, CODEC_WAV = 1, CODEC_MP3 = 2, CODEC_AAC = 3, CODEC_M4A = 4, CODEC_FLAC = 5,
CODEC_AACP = 6, CODEC_OPUS = 7, CODEC_OGG = 8};
CODEC_AACP = 6, CODEC_OPUS = 7, CODEC_OGG = 8, CODEC_VORBIS = 9};
enum : int { ST_NONE = 0, ST_WEBFILE = 1, ST_WEBSTREAM = 2};
typedef enum { LEFTCHANNEL=0, RIGHTCHANNEL=1 } SampleIndex;
typedef enum { LOWSHELF = 0, PEAKEQ = 1, HIFGSHELF =2 } FilterType;
......@@ -482,11 +482,12 @@ private:
std::vector<char*> m_playlistURL; // m3u8 streamURLs buffer
std::vector<uint32_t> m_hashQueue;
const size_t m_frameSizeWav = 1024;
const size_t m_frameSizeMP3 = 1600;
const size_t m_frameSizeAAC = 1600;
const size_t m_frameSizeFLAC = 4096 * 4;
const size_t m_frameSizeOPUS = 1024;
const size_t m_frameSizeWav = 1024;
const size_t m_frameSizeMP3 = 1600;
const size_t m_frameSizeAAC = 1600;
const size_t m_frameSizeFLAC = 4096 * 4;
const size_t m_frameSizeOPUS = 1024;
const size_t m_frameSizeVORBIS = 4096 * 2;
static const uint8_t m_tsPacketSize = 188;
static const uint8_t m_tsHeaderSize = 4;
......
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