mirror of
https://github.com/esphome/esphome.git
synced 2026-06-24 14:34:49 +00:00
[audio] Use the microMP3 library instead of esp-audio-libs (#16236)
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@@ -395,6 +395,7 @@ async def to_code(config):
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)
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if data.mp3_support:
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cg.add_define("USE_AUDIO_MP3_SUPPORT")
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add_idf_component(name="esphome/micro-mp3", ref="0.2.0")
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_emit_memory_pair(
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data.mp3.buffer_memory,
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"CONFIG_MP3_DECODER_PREFER_PSRAM",
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@@ -20,14 +20,6 @@ AudioDecoder::AudioDecoder(size_t input_buffer_size, size_t output_buffer_size)
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this->output_transfer_buffer_ = AudioSinkTransferBuffer::create(output_buffer_size);
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}
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AudioDecoder::~AudioDecoder() {
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#ifdef USE_AUDIO_MP3_SUPPORT
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if (this->audio_file_type_ == AudioFileType::MP3) {
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esp_audio_libs::helix_decoder::MP3FreeDecoder(this->mp3_decoder_);
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}
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#endif
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}
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esp_err_t AudioDecoder::add_source(std::weak_ptr<RingBuffer> &input_ring_buffer) {
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auto source = AudioSourceTransferBuffer::create(this->input_buffer_size_);
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if (source == nullptr) {
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@@ -92,13 +84,10 @@ esp_err_t AudioDecoder::start(AudioFileType audio_file_type) {
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#endif
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#ifdef USE_AUDIO_MP3_SUPPORT
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case AudioFileType::MP3:
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this->mp3_decoder_ = esp_audio_libs::helix_decoder::MP3InitDecoder();
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// MP3 always has 1152 samples per chunk
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this->free_buffer_required_ = 1152 * sizeof(int16_t) * 2; // samples * size per sample * channels
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// Always reallocate the output transfer buffer to the smallest necessary size
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this->output_transfer_buffer_->reallocate(this->free_buffer_required_);
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this->mp3_decoder_ = make_unique<micro_mp3::Mp3Decoder>();
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this->free_buffer_required_ =
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this->output_transfer_buffer_->capacity(); // Adjusted and reallocated after reading the header
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this->decoder_buffers_internally_ = true;
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break;
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#endif
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#ifdef USE_AUDIO_OPUS_SUPPORT
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@@ -312,51 +301,56 @@ FileDecoderState AudioDecoder::decode_flac_() {
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#ifdef USE_AUDIO_MP3_SUPPORT
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FileDecoderState AudioDecoder::decode_mp3_() {
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// Look for the next sync word
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int buffer_length = (int) this->input_buffer_->available();
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int32_t offset = esp_audio_libs::helix_decoder::MP3FindSyncWord(this->input_buffer_->data(), buffer_length);
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// microMP3's samples_decoded value is samples per channel; e.g., what ESPHome typically calls an audio frame.
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// microMP3 uses the term frame to refer to an MP3 frame: an encoded packet that contains multiple audio frames.
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size_t bytes_consumed = 0;
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size_t samples_decoded = 0;
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if (offset < 0) {
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// New data may have the sync word
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this->input_buffer_->consume(buffer_length);
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// microMP3 buffers internally: it consumes from our input buffer at its own pace, emits MP3_STREAM_INFO_READY once
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// the first frame header is parsed, and only then produces PCM. It handles sync-word search and ID3v2 tag skipping.
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micro_mp3::Mp3Result result = this->mp3_decoder_->decode(
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this->input_buffer_->data(), this->input_buffer_->available(), this->output_transfer_buffer_->get_buffer_end(),
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this->output_transfer_buffer_->free(), bytes_consumed, samples_decoded);
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this->input_buffer_->consume(bytes_consumed);
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if (result == micro_mp3::MP3_OK) {
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if (samples_decoded > 0 && this->audio_stream_info_.has_value()) {
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this->output_transfer_buffer_->increase_buffer_length(
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this->audio_stream_info_.value().frames_to_bytes(samples_decoded));
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}
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} else if (result == micro_mp3::MP3_STREAM_INFO_READY) {
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// First successful header parse: capture stream info and resize the output buffer to fit one full frame.
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// microMP3 always outputs 16-bit PCM.
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this->audio_stream_info_ =
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audio::AudioStreamInfo(16, this->mp3_decoder_->get_channels(), this->mp3_decoder_->get_sample_rate());
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this->free_buffer_required_ =
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this->mp3_decoder_->get_samples_per_frame() * this->mp3_decoder_->get_channels() * sizeof(int16_t);
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if (!this->output_transfer_buffer_->reallocate(this->free_buffer_required_)) {
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return FileDecoderState::FAILED;
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}
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} else if (result == micro_mp3::MP3_NEED_MORE_DATA) {
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return FileDecoderState::MORE_TO_PROCESS;
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} else if (result == micro_mp3::MP3_OUTPUT_BUFFER_TOO_SMALL) {
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// Reallocate to decode the frame on the next call
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if (this->mp3_decoder_->get_channels() > 0) {
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this->free_buffer_required_ =
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this->mp3_decoder_->get_samples_per_frame() * this->mp3_decoder_->get_channels() * sizeof(int16_t);
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} else {
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// Fallback to worst-case size if channel info isn't available
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this->free_buffer_required_ = this->mp3_decoder_->get_min_output_buffer_bytes();
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}
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if (!this->output_transfer_buffer_->reallocate(this->free_buffer_required_)) {
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return FileDecoderState::FAILED;
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}
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} else if (result == micro_mp3::MP3_DECODE_ERROR) {
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// Corrupt frame skipped; recoverable, retry on next call
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ESP_LOGW(TAG, "MP3 decoder skipped a corrupt frame");
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return FileDecoderState::POTENTIALLY_FAILED;
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}
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// Advance read pointer to match the offset for the syncword
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this->input_buffer_->consume(offset);
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const uint8_t *buffer_start = this->input_buffer_->data();
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buffer_length = (int) this->input_buffer_->available();
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int err = esp_audio_libs::helix_decoder::MP3Decode(this->mp3_decoder_, &buffer_start, &buffer_length,
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(int16_t *) this->output_transfer_buffer_->get_buffer_end(), 0);
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size_t consumed = this->input_buffer_->available() - buffer_length;
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this->input_buffer_->consume(consumed);
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if (err) {
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switch (err) {
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case esp_audio_libs::helix_decoder::ERR_MP3_OUT_OF_MEMORY:
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[[fallthrough]];
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case esp_audio_libs::helix_decoder::ERR_MP3_NULL_POINTER:
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return FileDecoderState::FAILED;
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break;
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default:
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// Most errors are recoverable by moving on to the next frame, so mark as potentailly failed
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return FileDecoderState::POTENTIALLY_FAILED;
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break;
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}
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} else {
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esp_audio_libs::helix_decoder::MP3FrameInfo mp3_frame_info;
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esp_audio_libs::helix_decoder::MP3GetLastFrameInfo(this->mp3_decoder_, &mp3_frame_info);
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if (mp3_frame_info.outputSamps > 0) {
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int bytes_per_sample = (mp3_frame_info.bitsPerSample / 8);
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this->output_transfer_buffer_->increase_buffer_length(mp3_frame_info.outputSamps * bytes_per_sample);
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if (!this->audio_stream_info_.has_value()) {
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this->audio_stream_info_ =
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audio::AudioStreamInfo(mp3_frame_info.bitsPerSample, mp3_frame_info.nChans, mp3_frame_info.samprate);
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}
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}
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// MP3_ALLOCATION_FAILED, MP3_INPUT_INVALID, or any future error -- not recoverable
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ESP_LOGE(TAG, "MP3 decoder failed: %d", static_cast<int>(result));
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return FileDecoderState::FAILED;
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}
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return FileDecoderState::MORE_TO_PROCESS;
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@@ -16,9 +16,6 @@
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#include "esp_err.h"
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// esp-audio-libs
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#ifdef USE_AUDIO_MP3_SUPPORT
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#include <mp3_decoder.h>
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#endif
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#include <wav_decoder.h>
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// micro-flac
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@@ -26,6 +23,11 @@
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#include <micro_flac/flac_decoder.h>
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#endif
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// micro-mp3
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#ifdef USE_AUDIO_MP3_SUPPORT
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#include <micro_mp3/mp3_decoder.h>
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#endif
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// micro-opus
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#ifdef USE_AUDIO_OPUS_SUPPORT
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#include <micro_opus/ogg_opus_decoder.h>
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@@ -62,8 +64,7 @@ class AudioDecoder {
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/// @param output_buffer_size Size of the output transfer buffer in bytes.
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AudioDecoder(size_t input_buffer_size, size_t output_buffer_size);
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/// @brief Deallocates the MP3 decoder (the flac, opus, and wav decoders are deallocated automatically)
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~AudioDecoder();
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~AudioDecoder() = default;
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/// @brief Adds a source ring buffer for raw file data. Takes ownership of the ring buffer in a shared_ptr.
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/// @param input_ring_buffer weak_ptr of a shared_ptr of the sink ring buffer to transfer ownership
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@@ -125,7 +126,7 @@ class AudioDecoder {
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#endif
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#ifdef USE_AUDIO_MP3_SUPPORT
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FileDecoderState decode_mp3_();
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esp_audio_libs::helix_decoder::HMP3Decoder mp3_decoder_;
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std::unique_ptr<micro_mp3::Mp3Decoder> mp3_decoder_;
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#endif
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#ifdef USE_AUDIO_OPUS_SUPPORT
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FileDecoderState decode_opus_();
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@@ -9,6 +9,8 @@ dependencies:
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version: 0.2.0
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esphome/micro-flac:
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version: 0.1.1
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esphome/micro-mp3:
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version: 0.2.0
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esphome/micro-opus:
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version: 0.4.0
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espressif/esp-dsp:
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