From eefad194d06241f0594d1ae97a18c6cad618863a Mon Sep 17 00:00:00 2001 From: Kevin Ahrendt Date: Wed, 18 Feb 2026 21:51:33 -0600 Subject: [PATCH] [audio, speaker] Add support for decoding Ogg Opus files (#13967) --- esphome/components/audio/__init__.py | 42 ++++++++++ esphome/components/audio/audio.cpp | 4 + esphome/components/audio/audio.h | 3 + esphome/components/audio/audio_decoder.cpp | 60 +++++++++++++- esphome/components/audio/audio_decoder.h | 15 +++- esphome/components/audio/audio_reader.cpp | 13 +++ .../speaker/media_player/__init__.py | 82 ++++++++++++++++--- .../speaker/media_player/audio_pipeline.cpp | 10 +++ esphome/core/defines.h | 1 + esphome/idf_component.yml | 2 + .../speaker/audio_dac.esp32-ard.yaml | 10 --- .../speaker/common-media_player.yaml | 7 ++ .../speaker/media_player.esp32-s3-idf.yaml | 9 -- ...idf.yaml => test-audio_dac.esp32-idf.yaml} | 0 ....yaml => test-media_player.esp32-idf.yaml} | 0 tests/components/speaker/test.esp32-ard.yaml | 10 --- 16 files changed, 222 insertions(+), 46 deletions(-) delete mode 100644 tests/components/speaker/audio_dac.esp32-ard.yaml delete mode 100644 tests/components/speaker/media_player.esp32-s3-idf.yaml rename tests/components/speaker/{audio_dac.esp32-idf.yaml => test-audio_dac.esp32-idf.yaml} (100%) rename tests/components/speaker/{media_player.esp32-idf.yaml => test-media_player.esp32-idf.yaml} (100%) delete mode 100644 tests/components/speaker/test.esp32-ard.yaml diff --git a/esphome/components/audio/__init__.py b/esphome/components/audio/__init__.py index f48b776dddb..d8d426ec63a 100644 --- a/esphome/components/audio/__init__.py +++ b/esphome/components/audio/__init__.py @@ -1,10 +1,14 @@ +from dataclasses import dataclass + import esphome.codegen as cg from esphome.components.esp32 import add_idf_component, include_builtin_idf_component import esphome.config_validation as cv from esphome.const import CONF_BITS_PER_SAMPLE, CONF_NUM_CHANNELS, CONF_SAMPLE_RATE +from esphome.core import CORE import esphome.final_validate as fv CODEOWNERS = ["@kahrendt"] +DOMAIN = "audio" audio_ns = cg.esphome_ns.namespace("audio") AudioFile = audio_ns.struct("AudioFile") @@ -14,9 +18,38 @@ AUDIO_FILE_TYPE_ENUM = { "WAV": AudioFileType.WAV, "MP3": AudioFileType.MP3, "FLAC": AudioFileType.FLAC, + "OPUS": AudioFileType.OPUS, } +@dataclass +class AudioData: + flac_support: bool = False + mp3_support: bool = False + opus_support: bool = False + + +def _get_data() -> AudioData: + if DOMAIN not in CORE.data: + CORE.data[DOMAIN] = AudioData() + return CORE.data[DOMAIN] + + +def request_flac_support() -> None: + """Request FLAC codec support for audio decoding.""" + _get_data().flac_support = True + + +def request_mp3_support() -> None: + """Request MP3 codec support for audio decoding.""" + _get_data().mp3_support = True + + +def request_opus_support() -> None: + """Request Opus codec support for audio decoding.""" + _get_data().opus_support = True + + CONF_MIN_BITS_PER_SAMPLE = "min_bits_per_sample" CONF_MAX_BITS_PER_SAMPLE = "max_bits_per_sample" CONF_MIN_CHANNELS = "min_channels" @@ -173,3 +206,12 @@ async def to_code(config): name="esphome/esp-audio-libs", ref="2.0.3", ) + + data = _get_data() + if data.flac_support: + cg.add_define("USE_AUDIO_FLAC_SUPPORT") + if data.mp3_support: + cg.add_define("USE_AUDIO_MP3_SUPPORT") + if data.opus_support: + cg.add_define("USE_AUDIO_OPUS_SUPPORT") + add_idf_component(name="esphome/micro-opus", ref="0.3.3") diff --git a/esphome/components/audio/audio.cpp b/esphome/components/audio/audio.cpp index 9cc9b7d0da0..40592f6107a 100644 --- a/esphome/components/audio/audio.cpp +++ b/esphome/components/audio/audio.cpp @@ -46,6 +46,10 @@ const char *audio_file_type_to_string(AudioFileType file_type) { #ifdef USE_AUDIO_MP3_SUPPORT case AudioFileType::MP3: return "MP3"; +#endif +#ifdef USE_AUDIO_OPUS_SUPPORT + case AudioFileType::OPUS: + return "OPUS"; #endif case AudioFileType::WAV: return "WAV"; diff --git a/esphome/components/audio/audio.h b/esphome/components/audio/audio.h index e01d7eb101e..7d7db9e9444 100644 --- a/esphome/components/audio/audio.h +++ b/esphome/components/audio/audio.h @@ -112,6 +112,9 @@ enum class AudioFileType : uint8_t { #endif #ifdef USE_AUDIO_MP3_SUPPORT MP3, +#endif +#ifdef USE_AUDIO_OPUS_SUPPORT + OPUS, #endif WAV, }; diff --git a/esphome/components/audio/audio_decoder.cpp b/esphome/components/audio/audio_decoder.cpp index 8f514468c48..ee6d7d0a153 100644 --- a/esphome/components/audio/audio_decoder.cpp +++ b/esphome/components/audio/audio_decoder.cpp @@ -3,10 +3,13 @@ #ifdef USE_ESP32 #include "esphome/core/hal.h" +#include "esphome/core/log.h" namespace esphome { namespace audio { +static const char *const TAG = "audio.decoder"; + static const uint32_t DECODING_TIMEOUT_MS = 50; // The decode function will yield after this duration static const uint32_t READ_WRITE_TIMEOUT_MS = 20; // Timeout for transferring audio data @@ -79,6 +82,14 @@ esp_err_t AudioDecoder::start(AudioFileType audio_file_type) { // Always reallocate the output transfer buffer to the smallest necessary size this->output_transfer_buffer_->reallocate(this->free_buffer_required_); break; +#endif +#ifdef USE_AUDIO_OPUS_SUPPORT + case AudioFileType::OPUS: + this->opus_decoder_ = make_unique(); + this->free_buffer_required_ = + this->output_transfer_buffer_->capacity(); // Adjusted and reallocated after reading the header + this->decoder_buffers_internally_ = true; + break; #endif case AudioFileType::WAV: this->wav_decoder_ = make_unique(); @@ -158,8 +169,9 @@ AudioDecoderState AudioDecoder::decode(bool stop_gracefully) { // Decode more audio // Only shift data on the first loop iteration to avoid unnecessary, slow moves - size_t bytes_read = this->input_transfer_buffer_->transfer_data_from_source(pdMS_TO_TICKS(READ_WRITE_TIMEOUT_MS), - first_loop_iteration); + // If the decoder buffers internally, then never shift + size_t bytes_read = this->input_transfer_buffer_->transfer_data_from_source( + pdMS_TO_TICKS(READ_WRITE_TIMEOUT_MS), first_loop_iteration && !this->decoder_buffers_internally_); if (!first_loop_iteration && (this->input_transfer_buffer_->available() < bytes_processed)) { // Less data is available than what was processed in last iteration, so don't attempt to decode. @@ -195,6 +207,11 @@ AudioDecoderState AudioDecoder::decode(bool stop_gracefully) { case AudioFileType::MP3: state = this->decode_mp3_(); break; +#endif +#ifdef USE_AUDIO_OPUS_SUPPORT + case AudioFileType::OPUS: + state = this->decode_opus_(); + break; #endif case AudioFileType::WAV: state = this->decode_wav_(); @@ -339,6 +356,45 @@ FileDecoderState AudioDecoder::decode_mp3_() { } #endif +#ifdef USE_AUDIO_OPUS_SUPPORT +FileDecoderState AudioDecoder::decode_opus_() { + bool processed_header = this->opus_decoder_->is_initialized(); + + size_t bytes_consumed, samples_decoded; + + micro_opus::OggOpusResult result = this->opus_decoder_->decode( + this->input_transfer_buffer_->get_buffer_start(), this->input_transfer_buffer_->available(), + this->output_transfer_buffer_->get_buffer_end(), this->output_transfer_buffer_->free(), bytes_consumed, + samples_decoded); + + if (result == micro_opus::OGG_OPUS_OK) { + if (!processed_header && this->opus_decoder_->is_initialized()) { + // Header processed and stream info is available + this->audio_stream_info_ = + audio::AudioStreamInfo(this->opus_decoder_->get_bit_depth(), this->opus_decoder_->get_channels(), + this->opus_decoder_->get_sample_rate()); + } + if (samples_decoded > 0 && this->audio_stream_info_.has_value()) { + // Some audio was processed + this->output_transfer_buffer_->increase_buffer_length( + this->audio_stream_info_.value().frames_to_bytes(samples_decoded)); + } + this->input_transfer_buffer_->decrease_buffer_length(bytes_consumed); + } else if (result == micro_opus::OGG_OPUS_OUTPUT_BUFFER_TOO_SMALL) { + // Reallocate to decode the packet on the next call + this->free_buffer_required_ = this->opus_decoder_->get_required_output_buffer_size(); + if (!this->output_transfer_buffer_->reallocate(this->free_buffer_required_)) { + // Couldn't reallocate output buffer + return FileDecoderState::FAILED; + } + } else { + ESP_LOGE(TAG, "Opus decoder failed: %" PRId8, result); + return FileDecoderState::POTENTIALLY_FAILED; + } + return FileDecoderState::MORE_TO_PROCESS; +} +#endif + FileDecoderState AudioDecoder::decode_wav_() { if (!this->audio_stream_info_.has_value()) { // Header hasn't been processed diff --git a/esphome/components/audio/audio_decoder.h b/esphome/components/audio/audio_decoder.h index 2ca1d623fe1..cad16110aea 100644 --- a/esphome/components/audio/audio_decoder.h +++ b/esphome/components/audio/audio_decoder.h @@ -24,6 +24,11 @@ #endif #include +// micro-opus +#ifdef USE_AUDIO_OPUS_SUPPORT +#include +#endif + namespace esphome { namespace audio { @@ -47,7 +52,7 @@ class AudioDecoder { * @brief Class that facilitates decoding an audio file. * The audio file is read from a ring buffer source, decoded, and sent to an audio sink (ring buffer or speaker * component). - * Supports wav, flac, and mp3 formats. + * Supports wav, flac, mp3, and ogg opus formats. */ public: /// @brief Allocates the input and output transfer buffers @@ -55,7 +60,7 @@ class AudioDecoder { /// @param output_buffer_size Size of the output transfer buffer in bytes. AudioDecoder(size_t input_buffer_size, size_t output_buffer_size); - /// @brief Deallocates the MP3 decoder (the flac and wav decoders are deallocated automatically) + /// @brief Deallocates the MP3 decoder (the flac, opus, and wav decoders are deallocated automatically) ~AudioDecoder(); /// @brief Adds a source ring buffer for raw file data. Takes ownership of the ring buffer in a shared_ptr. @@ -108,6 +113,10 @@ class AudioDecoder { #ifdef USE_AUDIO_MP3_SUPPORT FileDecoderState decode_mp3_(); esp_audio_libs::helix_decoder::HMP3Decoder mp3_decoder_; +#endif +#ifdef USE_AUDIO_OPUS_SUPPORT + FileDecoderState decode_opus_(); + std::unique_ptr opus_decoder_; #endif FileDecoderState decode_wav_(); @@ -124,6 +133,8 @@ class AudioDecoder { bool end_of_file_{false}; bool wav_has_known_end_{false}; + bool decoder_buffers_internally_{false}; + bool pause_output_{false}; uint32_t accumulated_frames_written_{0}; diff --git a/esphome/components/audio/audio_reader.cpp b/esphome/components/audio/audio_reader.cpp index 4e4bd31f9bb..78d69d7a39d 100644 --- a/esphome/components/audio/audio_reader.cpp +++ b/esphome/components/audio/audio_reader.cpp @@ -197,6 +197,11 @@ esp_err_t AudioReader::start(const std::string &uri, AudioFileType &file_type) { else if (str_endswith_ignore_case(url, ".flac")) { file_type = AudioFileType::FLAC; } +#endif +#ifdef USE_AUDIO_OPUS_SUPPORT + else if (str_endswith_ignore_case(url, ".opus")) { + file_type = AudioFileType::OPUS; + } #endif else { file_type = AudioFileType::NONE; @@ -241,6 +246,14 @@ AudioFileType AudioReader::get_audio_type(const char *content_type) { if (strcasecmp(content_type, "audio/flac") == 0 || strcasecmp(content_type, "audio/x-flac") == 0) { return AudioFileType::FLAC; } +#endif +#ifdef USE_AUDIO_OPUS_SUPPORT + // Match "audio/ogg" with a codecs parameter containing "opus" + // Valid forms: audio/ogg;codecs=opus, audio/ogg; codecs="opus", etc. + // Plain "audio/ogg" without a codecs parameter is not matched, as those are almost always Ogg Vorbis streams + if (strncasecmp(content_type, "audio/ogg", 9) == 0 && strcasestr(content_type + 9, "opus") != nullptr) { + return AudioFileType::OPUS; + } #endif return AudioFileType::NONE; } diff --git a/esphome/components/speaker/media_player/__init__.py b/esphome/components/speaker/media_player/__init__.py index 034312236c6..b302bd9b23c 100644 --- a/esphome/components/speaker/media_player/__init__.py +++ b/esphome/components/speaker/media_player/__init__.py @@ -26,7 +26,6 @@ from esphome.const import ( from esphome.core import CORE, HexInt from esphome.core.entity_helpers import inherit_property_from from esphome.external_files import download_content -from esphome.final_validate import full_config _LOGGER = logging.getLogger(__name__) @@ -37,6 +36,10 @@ DEPENDENCIES = ["network"] CODEOWNERS = ["@kahrendt", "@synesthesiam"] DOMAIN = "media_player" +CODEC_SUPPORT_ALL = "all" +CODEC_SUPPORT_NEEDED = "needed" +CODEC_SUPPORT_NONE = "none" + TYPE_LOCAL = "local" TYPE_WEB = "web" @@ -110,6 +113,8 @@ def _get_supported_format_struct(pipeline, type): args.append(("format", "flac")) elif pipeline[CONF_FORMAT] == "MP3": args.append(("format", "mp3")) + elif pipeline[CONF_FORMAT] == "OPUS": + args.append(("format", "opus")) elif pipeline[CONF_FORMAT] == "WAV": args.append(("format", "wav")) @@ -173,6 +178,13 @@ def _read_audio_file_and_type(file_config): media_file_type = audio.AUDIO_FILE_TYPE_ENUM["MP3"] elif file_type in ("flac"): media_file_type = audio.AUDIO_FILE_TYPE_ENUM["FLAC"] + elif ( + file_type in ("ogg") + and len(data) >= 36 + and data.startswith(b"OggS") + and data[28:36] == b"OpusHead" + ): + media_file_type = audio.AUDIO_FILE_TYPE_ENUM["OPUS"] return data, media_file_type @@ -199,6 +211,10 @@ def _validate_pipeline(config): inherit_property_from(CONF_NUM_CHANNELS, CONF_SPEAKER)(config) inherit_property_from(CONF_SAMPLE_RATE, CONF_SPEAKER)(config) + # Opus only supports 48 kHz + if config.get(CONF_FORMAT) == "OPUS" and config.get(CONF_SAMPLE_RATE) != 48000: + raise cv.Invalid("Opus only supports a sample rate of 48000 Hz") + # Validate the transcoder settings is compatible with the speaker audio.final_validate_audio_schema( "speaker media_player", @@ -225,12 +241,27 @@ def _validate_repeated_speaker(config): def _final_validate(config): - # Default to using codec if psram is enabled - if (use_codec := config.get(CONF_CODEC_SUPPORT_ENABLED)) is None: - use_codec = psram.DOMAIN in full_config.get() - conf_id = config[CONF_ID].id - core_data = CORE.data.setdefault(DOMAIN, {conf_id: {}}) - core_data[conf_id][CONF_CODEC_SUPPORT_ENABLED] = use_codec + # Normalize boolean values to string equivalents + codec_mode = config[CONF_CODEC_SUPPORT_ENABLED] + if codec_mode is True: + codec_mode = CODEC_SUPPORT_ALL + elif codec_mode is False: + codec_mode = CODEC_SUPPORT_NONE + + use_codec = codec_mode != CODEC_SUPPORT_NONE + + # In "needed" mode, collect formats from pipelines and files + needed_formats = set() + need_all = False + if codec_mode == CODEC_SUPPORT_NEEDED: + for pipeline_key in (CONF_ANNOUNCEMENT_PIPELINE, CONF_MEDIA_PIPELINE): + if pipeline := config.get(pipeline_key): + fmt = pipeline[CONF_FORMAT] + if fmt == "NONE": + # No preferred format means any format could arrive + need_all = True + else: + needed_formats.add(fmt) for file_config in config.get(CONF_FILES, []): _, media_file_type = _read_audio_file_and_type(file_config) @@ -243,6 +274,26 @@ def _final_validate(config): raise cv.Invalid( f"Unsupported local media file type, set {CONF_CODEC_SUPPORT_ENABLED} to true or convert the media file to wav" ) + # In "needed" mode, add file format to needed codecs + if codec_mode == CODEC_SUPPORT_NEEDED: + for fmt_name, fmt_enum in audio.AUDIO_FILE_TYPE_ENUM.items(): + if str(media_file_type) == str(fmt_enum): + if fmt_name not in ("WAV", "NONE"): + needed_formats.add(fmt_name) + break + + # Request codec support + if codec_mode == CODEC_SUPPORT_ALL or need_all: + audio.request_flac_support() + audio.request_mp3_support() + audio.request_opus_support() + elif codec_mode == CODEC_SUPPORT_NEEDED: + if "FLAC" in needed_formats: + audio.request_flac_support() + if "MP3" in needed_formats: + audio.request_mp3_support() + if "OPUS" in needed_formats: + audio.request_opus_support() return config @@ -307,7 +358,17 @@ CONFIG_SCHEMA = cv.All( cv.Optional(CONF_BUFFER_SIZE, default=1000000): cv.int_range( min=4000, max=4000000 ), - cv.Optional(CONF_CODEC_SUPPORT_ENABLED): cv.boolean, + cv.Optional( + CONF_CODEC_SUPPORT_ENABLED, default=CODEC_SUPPORT_NEEDED + ): cv.Any( + cv.boolean, + cv.one_of( + CODEC_SUPPORT_ALL, + CODEC_SUPPORT_NEEDED, + CODEC_SUPPORT_NONE, + lower=True, + ), + ), cv.Optional(CONF_FILES): cv.ensure_list(MEDIA_FILE_TYPE_SCHEMA), cv.Optional(CONF_TASK_STACK_IN_PSRAM): cv.All( cv.boolean, cv.requires_component(psram.DOMAIN) @@ -340,11 +401,6 @@ FINAL_VALIDATE_SCHEMA = cv.All( async def to_code(config): - if CORE.data[DOMAIN][config[CONF_ID].id][CONF_CODEC_SUPPORT_ENABLED]: - # Compile all supported audio codecs - cg.add_define("USE_AUDIO_FLAC_SUPPORT", True) - cg.add_define("USE_AUDIO_MP3_SUPPORT", True) - var = await media_player.new_media_player(config) await cg.register_component(var, config) diff --git a/esphome/components/speaker/media_player/audio_pipeline.cpp b/esphome/components/speaker/media_player/audio_pipeline.cpp index 8be37d740ab..177743feb1d 100644 --- a/esphome/components/speaker/media_player/audio_pipeline.cpp +++ b/esphome/components/speaker/media_player/audio_pipeline.cpp @@ -13,7 +13,12 @@ namespace speaker { static const uint32_t INITIAL_BUFFER_MS = 1000; // Start playback after buffering this duration of the file static const uint32_t READ_TASK_STACK_SIZE = 5 * 1024; +// Opus decoding uses more stack than other codecs +#ifdef USE_AUDIO_OPUS_SUPPORT +static const uint32_t DECODE_TASK_STACK_SIZE = 5 * 1024; +#else static const uint32_t DECODE_TASK_STACK_SIZE = 3 * 1024; +#endif static const uint32_t INFO_ERROR_QUEUE_COUNT = 5; @@ -552,6 +557,11 @@ void AudioPipeline::decode_task(void *params) { case audio::AudioFileType::FLAC: initial_bytes_to_buffer /= 2; // Estimate the FLAC compression factor is 2 break; +#endif +#ifdef USE_AUDIO_OPUS_SUPPORT + case audio::AudioFileType::OPUS: + initial_bytes_to_buffer /= 8; // Estimate the Opus compression factor is 8 + break; #endif default: break; diff --git a/esphome/core/defines.h b/esphome/core/defines.h index a70c6cd0d0e..5559ec88d00 100644 --- a/esphome/core/defines.h +++ b/esphome/core/defines.h @@ -130,6 +130,7 @@ #define USE_AUDIO_DAC #define USE_AUDIO_FLAC_SUPPORT #define USE_AUDIO_MP3_SUPPORT +#define USE_AUDIO_OPUS_SUPPORT #define USE_API #define USE_API_CLIENT_CONNECTED_TRIGGER #define USE_API_CLIENT_DISCONNECTED_TRIGGER diff --git a/esphome/idf_component.yml b/esphome/idf_component.yml index f39ea9b3ae4..83b2d9d95c8 100644 --- a/esphome/idf_component.yml +++ b/esphome/idf_component.yml @@ -3,6 +3,8 @@ dependencies: version: "7.4.2" esphome/esp-audio-libs: version: 2.0.3 + esphome/micro-opus: + version: 0.3.3 espressif/esp-tflite-micro: version: 1.3.3~1 espressif/esp32-camera: diff --git a/tests/components/speaker/audio_dac.esp32-ard.yaml b/tests/components/speaker/audio_dac.esp32-ard.yaml deleted file mode 100644 index 3f5d1bba7c1..00000000000 --- a/tests/components/speaker/audio_dac.esp32-ard.yaml +++ /dev/null @@ -1,10 +0,0 @@ -substitutions: - i2s_bclk_pin: GPIO27 - i2s_lrclk_pin: GPIO26 - i2s_mclk_pin: GPIO25 - i2s_dout_pin: GPIO23 - -packages: - i2c: !include ../../test_build_components/common/i2c/esp32-ard.yaml - -<<: !include common-audio_dac.yaml diff --git a/tests/components/speaker/common-media_player.yaml b/tests/components/speaker/common-media_player.yaml index edc9f670fc5..c958c0d9120 100644 --- a/tests/components/speaker/common-media_player.yaml +++ b/tests/components/speaker/common-media_player.yaml @@ -1,5 +1,11 @@ <<: !include common.yaml +wifi: + ap: + +psram: + mode: quad + media_player: - platform: speaker id: speaker_media_player_id @@ -10,3 +16,4 @@ media_player: volume_max: 0.95 volume_min: 0.0 task_stack_in_psram: true + codec_support_enabled: all diff --git a/tests/components/speaker/media_player.esp32-s3-idf.yaml b/tests/components/speaker/media_player.esp32-s3-idf.yaml deleted file mode 100644 index b3eec04d238..00000000000 --- a/tests/components/speaker/media_player.esp32-s3-idf.yaml +++ /dev/null @@ -1,9 +0,0 @@ -substitutions: - scl_pin: GPIO2 - sda_pin: GPIO3 - i2s_bclk_pin: GPIO4 - i2s_lrclk_pin: GPIO5 - i2s_mclk_pin: GPIO6 - i2s_dout_pin: GPIO7 - -<<: !include common-media_player.yaml diff --git a/tests/components/speaker/audio_dac.esp32-idf.yaml b/tests/components/speaker/test-audio_dac.esp32-idf.yaml similarity index 100% rename from tests/components/speaker/audio_dac.esp32-idf.yaml rename to tests/components/speaker/test-audio_dac.esp32-idf.yaml diff --git a/tests/components/speaker/media_player.esp32-idf.yaml b/tests/components/speaker/test-media_player.esp32-idf.yaml similarity index 100% rename from tests/components/speaker/media_player.esp32-idf.yaml rename to tests/components/speaker/test-media_player.esp32-idf.yaml diff --git a/tests/components/speaker/test.esp32-ard.yaml b/tests/components/speaker/test.esp32-ard.yaml deleted file mode 100644 index 13350cd0973..00000000000 --- a/tests/components/speaker/test.esp32-ard.yaml +++ /dev/null @@ -1,10 +0,0 @@ -substitutions: - i2s_bclk_pin: GPIO27 - i2s_lrclk_pin: GPIO26 - i2s_mclk_pin: GPIO25 - i2s_dout_pin: GPIO4 - -packages: - i2c: !include ../../test_build_components/common/i2c/esp32-ard.yaml - -<<: !include common.yaml