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Update WiFi SoftAP password to love, update touch pin mappings, and remove unused decoders to resolve compile crash.

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drjones
2026-07-02 15:49:41 -07:00
parent 7606b9d5df
commit bac1af80fe
35 changed files with 867 additions and 58932 deletions
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49
libraries/ESP32-audioI2S/src/Audio.cpp
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@@ -16,12 +16,12 @@ char audioI2SVers[] = "\
*****************************************************************************************************************************************************/
#include "Audio.h"
#include "aac_decoder/aac_decoder.h"
#include "flac_decoder/flac_decoder.h"
// #include "aac_decoder/aac_decoder.h"
// #include "flac_decoder/flac_decoder.h"
#include "mp3_decoder/mp3_decoder.h"
#include "opus_decoder/opus_decoder.h"
// #include "opus_decoder/opus_decoder.h"
#include "psram_unique_ptr.hpp"
#include "vorbis_decoder/vorbis_decoder.h"
// #include "vorbis_decoder/vorbis_decoder.h"
#include "wav_decoder/wav_decoder.h"
// constants
@@ -376,10 +376,10 @@ void Audio::destroy_decoder() {
std::unique_ptr<Decoder> Audio::createDecoder(const std::string& type) {
destroy_decoder();
if (type == "MP3") return std::make_unique<MP3Decoder>(*this);
if (type == "FLAC") return std::make_unique<FlacDecoder>(*this);
if (type == "OPUS") return std::make_unique<OpusDecoder>(*this);
if (type == "AAC") return std::make_unique<AACDecoder>(*this);
if (type == "VORBIS") return std::make_unique<VorbisDecoder>(*this);
// if (type == "FLAC") return std::make_unique<FlacDecoder>(*this);
// if (type == "OPUS") return std::make_unique<OpusDecoder>(*this);
// if (type == "AAC") return std::make_unique<AACDecoder>(*this);
// if (type == "VORBIS") return std::make_unique<VorbisDecoder>(*this);
if (type == "WAV") return std::make_unique<WavDecoder>(*this);
return nullptr;
}
@@ -5555,29 +5555,6 @@ uint32_t Audio::decodeError(int8_t res, uint8_t* data, int32_t bytesDecoded) {
}
// —————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————————
uint32_t Audio::decodeContinue(int8_t res, uint8_t* data, int32_t bytesDecoded, int32_t* bytesLeft) {
// if(m_codec == CODEC_MP3){ if(res == MAD_ERROR_CONTINUE) return bytesDecoded;} // nothing to play, mybe eof
if (m_codec == CODEC_AAC) {
if (res == AACDecoder::AAC_ID3_HDR) {
uint32_t size = ((data[6 + bytesDecoded] & 0x7F) << 21) | ((data[7 + bytesDecoded] & 0x7F) << 14) | ((data[8 + bytesDecoded] & 0x7F) << 7) | (data[9 + bytesDecoded] & 0x7F);
size += 10; // skip payload
return bytesDecoded + size;
if (bytesDecoded > *bytesLeft) AUDIO_LOG_ERROR("AAC input data too small bytesDecoded {} > bytesLeft {}", bytesDecoded, *bytesLeft);
}
}
if (m_codec == CODEC_FLAC) {
if (res == FlacDecoder::FLAC_PARSE_OGG_DONE) return bytesDecoded;
if (res == FlacDecoder::FLAC_DECODE_FRAMES_LOOP) return bytesDecoded;
} // nothing to play
if (m_codec == CODEC_OPUS) {
if (res == OpusDecoder::OPUS_PARSE_OGG_DONE) return bytesDecoded;
if (res == OpusDecoder::OPUS_END) return bytesDecoded;
} // nothing to play
if (m_codec == CODEC_VORBIS) {
if (res == VorbisDecoder::VORBIS_PARSE_OGG_DONE) return bytesDecoded;
if (res == VorbisDecoder::VORBIS_COMMENT_DONE) return bytesDecoded;
if (res == VorbisDecoder::VORBIS_COMMENT_NEED_MORE) return bytesDecoded;
} // nothing to play
if (m_codec == CODEC_MP3) {
if (res == MP3Decoder::MP3_NEXT_FRAME) return bytesDecoded;
}
@@ -5812,8 +5789,8 @@ bool Audio::i2s_config() {
m_i2s_chan_cfg.dma_desc_num = settings.DMA_DESC_NUM; // number of DMA buffer
m_i2s_chan_cfg.dma_frame_num = settings.DMA_FRAME_NUM; // I2S frame number in one DMA buffer.
m_i2s_chan_cfg.auto_clear = true; // i2s will always send zero automatically if no data to send
m_i2s_chan_cfg.allow_pd = false;
m_i2s_chan_cfg.intr_priority = 2;
// m_i2s_chan_cfg.allow_pd = false;
// m_i2s_chan_cfg.intr_priority = 2;
result = i2s_new_channel(&m_i2s_chan_cfg, &m_i2s_tx_handle, NULL);
if (result != ESP_OK) { // ESP_ERR_INVALID_ARG?
AUDIO_LOG_ERROR("I2S channel: invalid argument");
@@ -6694,9 +6671,9 @@ void Audio::IIR_filter(int32_t* sample) {
float s[2];
s[LEFTCHANNEL] = (float)(s32[LEFTCHANNEL] * m_audio_items.pre_gain);
s[RIGHTCHANNEL] = (float)(s32[RIGHTCHANNEL] * m_audio_items.pre_gain);
dsps_biquad_sf32(s, s, 1, m_audio_items.coeffs[0], m_audio_items.state_biquad[0]);
dsps_biquad_sf32(s, s, 1, m_audio_items.coeffs[1], m_audio_items.state_biquad[1]);
dsps_biquad_sf32(s, s, 1, m_audio_items.coeffs[2], m_audio_items.state_biquad[2]);
dsps_biquad_f32(s, s, 1, m_audio_items.coeffs[0], m_audio_items.state_biquad[0]);
dsps_biquad_f32(s, s, 1, m_audio_items.coeffs[1], m_audio_items.state_biquad[1]);
dsps_biquad_f32(s, s, 1, m_audio_items.coeffs[2], m_audio_items.state_biquad[2]);
s32[LEFTCHANNEL] = (int32_t)std::clamp(s[LEFTCHANNEL], -2147483648.0f, 2147483647.0f);
s32[RIGHTCHANNEL] = (int32_t)std::clamp(s[RIGHTCHANNEL], -2147483648.0f, 2147483647.0f);
return;