Skip to content

GitLab

E
ESP32-audioI2S
Unverified Commit dda1071b authored by Wolle's avatar Wolle Committed by GitHub
Browse files
Options

Add files via upload

parent cf8dea21
Hide whitespace changes
Inline Side-by-side
Showing with 64 additions and 9 deletions
src/Audio.cpp
View file @ dda1071b
......@@ -4,7 +4,7 @@
* Created on: Oct 26.2018
*
* Version 2.0.8d
* Updated on: Jan 13.2023
* Updated on: Jan 27.2023
* Author: Wolle (schreibfaul1)
*
*/
......@@ -12,6 +12,7 @@
#include "mp3_decoder/mp3_decoder.h"
#include "aac_decoder/aac_decoder.h"
#include "flac_decoder/flac_decoder.h"
#include "opus_decoder/opus_decoder.h"
#ifdef SDFATFS_USED
fs::SDFATFS SD_SDFAT;
......@@ -310,6 +311,7 @@ void Audio::setDefaults() {
MP3Decoder_FreeBuffers();
FLACDecoder_FreeBuffers();
AACDecoder_FreeBuffers();
OPUSDecoder_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);
......@@ -503,6 +505,8 @@ bool Audio::connecttohost(const char* host, const char* user, const char* pwd) {
if(endsWith(extension, ".wav")) m_expectedCodec = CODEC_WAV;
if(endsWith(extension, ".m4a")) m_expectedCodec = CODEC_M4A;
if(endsWith(extension, ".flac")) m_expectedCodec = CODEC_FLAC;
if(endsWith(extension, ".opus")) m_expectedCodec = CODEC_OPUS;
if(endsWith(extension, "/opus")) m_expectedCodec = CODEC_OPUS;
if(endsWith(extension, ".asx")) m_expectedPlsFmt = FORMAT_ASX;
if(endsWith(extension, ".m3u")) m_expectedPlsFmt = FORMAT_M3U;
if(endsWith(extension, ".m3u8")) m_expectedPlsFmt = FORMAT_M3U8;
......@@ -738,6 +742,7 @@ bool Audio::connecttoFS(fs::FS &fs, const char* path, uint32_t resumeFilePos) {
if(endsWith(afn, ".aac")) m_codec = CODEC_AAC;
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(m_codec == CODEC_NONE) AUDIO_INFO("The %s format is not supported", afn + dotPos);
......@@ -1191,6 +1196,9 @@ size_t Audio::readAudioHeader(uint32_t bytes){
m_controlCounter = 100;
}
}
if(m_codec == CODEC_OPUS){
m_controlCounter = 100;
}
if(!isRunning()){
log_e("Processing stopped due to invalid audio header");
return 0;
......@@ -2811,6 +2819,7 @@ void Audio::processLocalFile() {
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();
AUDIO_INFO("End of file \"%s\"", afn);
if(audio_eof_mp3) audio_eof_mp3(afn);
if(afn) {free(afn); afn = NULL;}
......@@ -3282,6 +3291,7 @@ bool Audio::parseHttpResponseHeader() { // this is the response to a GET / reque
}
else if(startsWith(rhl, "content-type:")){ // content-type: text/html; charset=UTF-8
log_i("cT: %s", rhl);
int idx = indexOf(rhl + 13, ";");
if(idx >0) rhl[13 + idx] = '\0';
if(parseContentType(rhl + 13)) ct_seen = true;
......@@ -3445,20 +3455,29 @@ bool Audio::parseHttpResponseHeader() { // this is the response to a GET / reque
bool Audio:: initializeDecoder(){
switch(m_codec){
case CODEC_MP3:
if(!MP3Decoder_AllocateBuffers()) goto exit;
if(!MP3Decoder_AllocateBuffers()){
AUDIO_INFO("The MP3Decoder could not be initialized");
goto exit;
}
AUDIO_INFO("MP3Decoder has been initialized, free Heap: %u bytes", ESP.getFreeHeap());
InBuff.changeMaxBlockSize(m_frameSizeMP3);
break;
case CODEC_AAC:
if(!AACDecoder_IsInit()){
if(!AACDecoder_AllocateBuffers()) goto exit;
if(!AACDecoder_AllocateBuffers()){
AUDIO_INFO("The AACDecoder could not be initialized");
goto exit;
}
AUDIO_INFO("AACDecoder has been initialized, free Heap: %u bytes", ESP.getFreeHeap());
InBuff.changeMaxBlockSize(m_frameSizeAAC);
}
break;
case CODEC_M4A:
if(!AACDecoder_IsInit()){
if(!AACDecoder_AllocateBuffers()) goto exit;
if(!AACDecoder_AllocateBuffers()){
AUDIO_INFO("The AACDecoder could not be initialized");
goto exit;
}
AUDIO_INFO("AACDecoder has been initialized, free Heap: %u bytes", ESP.getFreeHeap());
InBuff.changeMaxBlockSize(m_frameSizeAAC);
}
......@@ -3468,10 +3487,22 @@ bool Audio:: initializeDecoder(){
AUDIO_INFO("FLAC works only with PSRAM!");
goto exit;
}
if(!FLACDecoder_AllocateBuffers()) goto exit;
if(!FLACDecoder_AllocateBuffers()){
AUDIO_INFO("The FLACDecoder could not be initialized");
goto exit;
}
InBuff.changeMaxBlockSize(m_frameSizeFLAC);
AUDIO_INFO("FLACDecoder has been initialized, free Heap: %u bytes", ESP.getFreeHeap());
break;
case CODEC_OPUS:
if(!OPUSDecoder_AllocateBuffers()){
AUDIO_INFO("The OPUSDecoder could not be initialized");
goto exit;
}
AUDIO_INFO("OPUSDecoder has been initialized, free Heap: %u bytes", ESP.getFreeHeap());
InBuff.changeMaxBlockSize(m_frameSizeOPUS);
break;
case CODEC_WAV:
InBuff.changeMaxBlockSize(m_frameSizeWav);
break;
......@@ -3489,7 +3520,7 @@ 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_M3U8, CT_TXT, CT_AACP, CT_OPUS};
strlwr(ct);
trim(ct);
......@@ -3526,8 +3557,8 @@ bool Audio::parseContentType(char* ct) {
else if(!strcmp(ct, "video/x-ms-asf")) ct_val = CT_ASX;
else if(!strcmp(ct, "audio/x-ms-asx")) ct_val = CT_ASX; // #413
else if(!strcmp(ct, "application/ogg")) ct_val = CT_FLAC; // assume ogg wrapper
else if(!strcmp(ct, "audio/ogg")) ct_val = CT_FLAC; // auusme ogg wrapper
else if(!strcmp(ct, "application/ogg")) {ct_val = CT_FLAC; if(m_expectedCodec == CODEC_OPUS) ct_val = CT_OPUS;}
else if(!strcmp(ct, "audio/ogg")) {ct_val = CT_FLAC; if(m_expectedCodec == CODEC_OPUS) ct_val = CT_OPUS;}
else if(!strcmp(ct, "application/vnd.apple.mpegurl")) ct_val = CT_M3U8;
else if(!strcmp(ct, "application/x-mpegurl")) ct_val =CT_M3U8;
......@@ -3559,6 +3590,10 @@ bool Audio::parseContentType(char* ct) {
m_codec = CODEC_FLAC;
if(m_f_Log) { log_i("ContentType %s, format is flac", ct); }
break;
case CT_OPUS:
m_codec = CODEC_OPUS;
if(m_f_Log) { log_i("ContentType %s, format is opus", ct); }
break;
case CT_WAV:
m_codec = CODEC_WAV;
if(m_f_Log) { log_i("ContentType %s, format is wav", ct); }
......@@ -3757,6 +3792,9 @@ int Audio::findNextSync(uint8_t* data, size_t len){
m_flacBitsPerSample, m_flacTotalSamplesInStream, m_audioDataSize);
nextSync = FLACFindSyncWord(data, len);
}
if(m_codec == CODEC_OPUS) {
nextSync = OPUSFindSyncWord(data, len);
}
if(nextSync == -1) {
if(audio_info && swnf == 0) audio_info("syncword not found");
else {
......@@ -3803,6 +3841,7 @@ int Audio::sendBytes(uint8_t* data, size_t len) {
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;
default: {log_e("no valid codec found codec = %d", m_codec); stopSong();}
}
......@@ -3853,6 +3892,13 @@ int Audio::sendBytes(uint8_t* data, size_t len) {
setBitsPerSample(FLACGetBitsPerSample());
setBitrate(FLACGetBitRate());
}
if(m_codec == CODEC_OPUS){
setChannels(OPUSGetChannels());
setSampleRate(OPUSGetSampRate());
setBitsPerSample(OPUSGetBitsPerSample());
setBitrate(OPUSGetBitRate());
}
showCodecParams();
}
if(m_codec == CODEC_MP3){
......@@ -3869,6 +3915,14 @@ int Audio::sendBytes(uint8_t* data, size_t len) {
if(audio_showstreamtitle) audio_showstreamtitle(st);
}
}
if(m_codec == CODEC_OPUS){
m_validSamples = OPUSGetOutputSamps() / getChannels();
char* st = OPUSgetStreamTitle();
if(st){
AUDIO_INFO(st);
if(audio_showstreamtitle) audio_showstreamtitle(st);
}
}
}
compute_audioCurrentTime(bytesDecoded);
......
src/Audio.h
View file @ dda1071b
......@@ -442,7 +442,7 @@ 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_AACP = 6, CODEC_OPUS = 7};
enum : int { ST_NONE = 0, ST_WEBFILE = 1, ST_WEBSTREAM = 2};
typedef enum { LEFTCHANNEL=0, RIGHTCHANNEL=1 } SampleIndex;
typedef enum { LOWSHELF = 0, PEAKEQ = 1, HIFGSHELF =2 } FilterType;
......@@ -478,6 +478,7 @@ private:
const size_t m_frameSizeMP3 = 1600;
const size_t m_frameSizeAAC = 1600;
const size_t m_frameSizeFLAC = 4096 * 4;
const size_t m_frameSizeOPUS = 1024;
static const uint8_t m_tsPacketSize = 188;
static const uint8_t m_tsHeaderSize = 4;
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment