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Code Issues Releases Wiki Activity

Merge branch 'dev' into vornado-ir

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This commit is contained in:
Jordan Zucker 2025-02-04 16:56:37 -08:00
commit d9b8a7bc17
1210 changed files with 12002 additions and 20523 deletions
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2
.github/workflows/ci-docker.yml vendored
View File

@ -37,7 +37,7 @@ jobs:
strategy:
fail-fast: false
matrix:
arch: [amd64, armv7, aarch64]
arch: [amd64, aarch64]
build_type: ["ha-addon", "docker", "lint"]
steps:
- uses: actions/checkout@v4.1.7

1
.github/workflows/release.yml vendored
View File

@ -80,7 +80,6 @@ jobs:
matrix:
platform:
- linux/amd64
- linux/arm/v7
- linux/arm64
steps:
- uses: actions/checkout@v4.1.7

3
CODEOWNERS
View File

@ -277,6 +277,7 @@ esphome/components/mics_4514/* @jesserockz
esphome/components/midea/* @dudanov
esphome/components/midea_ir/* @dudanov
esphome/components/mitsubishi/* @RubyBailey
esphome/components/mixer/speaker/* @kahrendt
esphome/components/mlx90393/* @functionpointer
esphome/components/mlx90614/* @jesserockz
esphome/components/mmc5603/* @benhoff
@ -343,6 +344,7 @@ esphome/components/radon_eye_rd200/* @jeffeb3
esphome/components/rc522/* @glmnet
esphome/components/rc522_i2c/* @glmnet
esphome/components/rc522_spi/* @glmnet
esphome/components/resampler/speaker/* @kahrendt
esphome/components/restart/* @esphome/core
esphome/components/rf_bridge/* @jesserockz
esphome/components/rgbct/* @jesserockz
@ -499,5 +501,6 @@ esphome/components/xiaomi_mhoc401/* @vevsvevs
esphome/components/xiaomi_rtcgq02lm/* @jesserockz
esphome/components/xl9535/* @mreditor97
esphome/components/xpt2046/touchscreen/* @nielsnl68 @numo68
esphome/components/xxtea/* @clydebarrow
esphome/components/zhlt01/* @cfeenstra1024
esphome/components/zio_ultrasonic/* @kahrendt

49
docker/Dockerfile
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@ -51,19 +51,7 @@ ENV \
# Store globally installed pio libs in /piolibs
PLATFORMIO_GLOBALLIB_DIR=/piolibs
# Support legacy binaries on Debian multiarch system. There is no "correct" way
# to do this, other than using properly built toolchains...
# See: https://unix.stackexchange.com/questions/553743/correct-way-to-add-lib-ld-linux-so-3-in-debian
RUN \
if [ "$TARGETARCH$TARGETVARIANT" = "armv7" ]; then \
ln -s /lib/arm-linux-gnueabihf/ld-linux-armhf.so.3 /lib/ld-linux.so.3; \
fi
RUN \
# Ubuntu python3-pip is missing wheel
if [ "$TARGETARCH$TARGETVARIANT" = "armv7" ]; then \
export PIP_EXTRA_INDEX_URL="https://www.piwheels.org/simple"; \
fi; \
pip3 install \
--break-system-packages --no-cache-dir \
# Keep platformio version in sync with requirements.txt
@ -82,14 +70,6 @@ RUN --mount=type=tmpfs,target=/root/.cargo <<END-OF-RUN
# Fail on any non-zero status
set -e
if [ "$TARGETARCH$TARGETVARIANT" = "armv7" ]
then
curl -L https://www.piwheels.org/cp311/cryptography-43.0.0-cp37-abi3-linux_armv7l.whl -o /tmp/cryptography-43.0.0-cp37-abi3-linux_armv7l.whl
pip3 install --break-system-packages --no-cache-dir /tmp/cryptography-43.0.0-cp37-abi3-linux_armv7l.whl
rm /tmp/cryptography-43.0.0-cp37-abi3-linux_armv7l.whl
export PIP_EXTRA_INDEX_URL="https://www.piwheels.org/simple";
fi
# install build tools in case wheels are not available
BUILD_DEPS="
build-essential=12.9
@ -106,7 +86,7 @@ LIB_DEPS="
libtiff6=4.5.0-6+deb12u1
libopenjp2-7=2.5.0-2
"
if [ "$TARGETARCH$TARGETVARIANT" = "arm64" ] || [ "$TARGETARCH$TARGETVARIANT" = "armv7" ]
if [ "$TARGETARCH$TARGETVARIANT" = "arm64" ]
then
apt-get update
apt-get install -y --no-install-recommends $BUILD_DEPS $LIB_DEPS
@ -115,7 +95,7 @@ fi
CARGO_REGISTRIES_CRATES_IO_PROTOCOL=sparse CARGO_HOME=/root/.cargo
pip3 install --break-system-packages --no-cache-dir -r /requirements.txt -r /requirements_optional.txt
if [ "$TARGETARCH$TARGETVARIANT" = "arm64" ] || [ "$TARGETARCH$TARGETVARIANT" = "armv7" ]
if [ "$TARGETARCH$TARGETVARIANT" = "arm64" ]
then
apt-get remove -y --purge --auto-remove $BUILD_DEPS
rm -rf /tmp/* /var/{cache,log}/* /var/lib/apt/lists/*
@ -135,11 +115,7 @@ FROM base AS docker
# Copy esphome and install
COPY . /esphome
RUN if [ "$TARGETARCH$TARGETVARIANT" = "armv7" ]; then \
export PIP_EXTRA_INDEX_URL="https://www.piwheels.org/simple"; \
fi; \
pip3 install \
--break-system-packages --no-cache-dir -e /esphome
RUN pip3 install --break-system-packages --no-cache-dir -e /esphome
# Settings for dashboard
ENV USERNAME="" PASSWORD=""
@ -197,11 +173,7 @@ COPY docker/ha-addon-rootfs/ /
# Copy esphome and install
COPY . /esphome
RUN if [ "$TARGETARCH$TARGETVARIANT" = "armv7" ]; then \
export PIP_EXTRA_INDEX_URL="https://www.piwheels.org/simple"; \
fi; \
pip3 install \
--break-system-packages --no-cache-dir -e /esphome
RUN pip3 install --break-system-packages --no-cache-dir -e /esphome
# Labels
LABEL \
@ -232,21 +204,14 @@ RUN \
nano=7.2-1+deb12u1 \
build-essential=12.9 \
python3-dev=3.11.2-1+b1 \
&& if [ "$TARGETARCH$TARGETVARIANT" != "armv7" ]; then \
# move this up after armv7 is retired
apt-get install -y --no-install-recommends clang-tidy-18=1:18.1.8~++20240731024826+3b5b5c1ec4a3-1~exp1~20240731144843.145 ; \
fi; \
rm -rf \
clang-tidy-18=1:18.1.8~++20240731024826+3b5b5c1ec4a3-1~exp1~20240731144843.145 \
&& rm -rf \
/tmp/* \
/var/{cache,log}/* \
/var/lib/apt/lists/*
COPY requirements_test.txt /
RUN if [ "$TARGETARCH$TARGETVARIANT" = "armv7" ]; then \
export PIP_EXTRA_INDEX_URL="https://www.piwheels.org/simple"; \
fi; \
pip3 install \
--break-system-packages --no-cache-dir -r /requirements_test.txt
RUN pip3 install --break-system-packages --no-cache-dir -r /requirements_test.txt
VOLUME ["/esphome"]
WORKDIR /esphome

12
docker/build.py
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@ -1,22 +1,19 @@
#!/usr/bin/env python3
from dataclasses import dataclass
import subprocess
import argparse
from platform import machine
import shlex
from dataclasses import dataclass
import re
import shlex
import subprocess
import sys
CHANNEL_DEV = "dev"
CHANNEL_BETA = "beta"
CHANNEL_RELEASE = "release"
CHANNELS = [CHANNEL_DEV, CHANNEL_BETA, CHANNEL_RELEASE]
ARCH_AMD64 = "amd64"
ARCH_ARMV7 = "armv7"
ARCH_AARCH64 = "aarch64"
ARCHS = [ARCH_AMD64, ARCH_ARMV7, ARCH_AARCH64]
ARCHS = [ARCH_AMD64, ARCH_AARCH64]
TYPE_DOCKER = "docker"
TYPE_HA_ADDON = "ha-addon"
@ -76,7 +73,6 @@ class DockerParams:
}[build_type]
platform = {
ARCH_AMD64: "linux/amd64",
ARCH_ARMV7: "linux/arm/v7",
ARCH_AARCH64: "linux/arm64",
}[arch]
target = {

112
esphome/components/audio/__init__.py
View File

@ -1,9 +1,121 @@
import esphome.codegen as cg
import esphome.config_validation as cv
from esphome.const import CONF_BITS_PER_SAMPLE, CONF_NUM_CHANNELS, CONF_SAMPLE_RATE
import esphome.final_validate as fv
CODEOWNERS = ["@kahrendt"]
audio_ns = cg.esphome_ns.namespace("audio")
AudioFile = audio_ns.struct("AudioFile")
AudioFileType = audio_ns.enum("AudioFileType", is_class=True)
AUDIO_FILE_TYPE_ENUM = {
"NONE": AudioFileType.NONE,
"WAV": AudioFileType.WAV,
"MP3": AudioFileType.MP3,
"FLAC": AudioFileType.FLAC,
}
CONF_MIN_BITS_PER_SAMPLE = "min_bits_per_sample"
CONF_MAX_BITS_PER_SAMPLE = "max_bits_per_sample"
CONF_MIN_CHANNELS = "min_channels"
CONF_MAX_CHANNELS = "max_channels"
CONF_MIN_SAMPLE_RATE = "min_sample_rate"
CONF_MAX_SAMPLE_RATE = "max_sample_rate"
CONFIG_SCHEMA = cv.All(
cv.Schema({}),
)
AUDIO_COMPONENT_SCHEMA = cv.Schema(
{
cv.Optional(CONF_BITS_PER_SAMPLE): cv.int_range(8, 32),
cv.Optional(CONF_NUM_CHANNELS): cv.int_range(1, 2),
cv.Optional(CONF_SAMPLE_RATE): cv.int_range(8000, 48000),
}
)
_UNDEF = object()
def set_stream_limits(
min_bits_per_sample: int = _UNDEF,
max_bits_per_sample: int = _UNDEF,
min_channels: int = _UNDEF,
max_channels: int = _UNDEF,
min_sample_rate: int = _UNDEF,
max_sample_rate: int = _UNDEF,
):
def set_limits_in_config(config):
if min_bits_per_sample is not _UNDEF:
config[CONF_MIN_BITS_PER_SAMPLE] = min_bits_per_sample
if max_bits_per_sample is not _UNDEF:
config[CONF_MAX_BITS_PER_SAMPLE] = max_bits_per_sample
if min_channels is not _UNDEF:
config[CONF_MIN_CHANNELS] = min_channels
if max_channels is not _UNDEF:
config[CONF_MAX_CHANNELS] = max_channels
if min_sample_rate is not _UNDEF:
config[CONF_MIN_SAMPLE_RATE] = min_sample_rate
if max_sample_rate is not _UNDEF:
config[CONF_MAX_SAMPLE_RATE] = max_sample_rate
return set_limits_in_config
def final_validate_audio_schema(
name: str,
*,
audio_device: str,
bits_per_sample: int,
channels: int,
sample_rate: int,
):
def validate_audio_compatiblity(audio_config):
audio_schema = {}
try:
cv.int_range(
min=audio_config.get(CONF_MIN_BITS_PER_SAMPLE),
max=audio_config.get(CONF_MAX_BITS_PER_SAMPLE),
)(bits_per_sample)
except cv.Invalid as exc:
raise cv.Invalid(
f"Invalid configuration for the {name} component. The {CONF_BITS_PER_SAMPLE} {str(exc)}"
) from exc
try:
cv.int_range(
min=audio_config.get(CONF_MIN_CHANNELS),
max=audio_config.get(CONF_MAX_CHANNELS),
)(channels)
except cv.Invalid as exc:
raise cv.Invalid(
f"Invalid configuration for the {name} component. The {CONF_NUM_CHANNELS} {str(exc)}"
) from exc
try:
cv.int_range(
min=audio_config.get(CONF_MIN_SAMPLE_RATE),
max=audio_config.get(CONF_MAX_SAMPLE_RATE),
)(sample_rate)
return cv.Schema(audio_schema, extra=cv.ALLOW_EXTRA)(audio_config)
except cv.Invalid as exc:
raise cv.Invalid(
f"Invalid configuration for the {name} component. The {CONF_SAMPLE_RATE} {str(exc)}"
) from exc
return cv.Schema(
{
cv.Required(audio_device): fv.id_declaration_match_schema(
validate_audio_compatiblity
)
},
extra=cv.ALLOW_EXTRA,
)
async def to_code(config):
cg.add_library("esphome/esp-audio-libs", "1.1.1")

67
esphome/components/audio/audio.cpp Normal file
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@ -0,0 +1,67 @@
#include "audio.h"
namespace esphome {
namespace audio {
// Euclidean's algorithm for finding the greatest common divisor
static uint32_t gcd(uint32_t a, uint32_t b) {
while (b != 0) {
uint32_t t = b;
b = a % b;
a = t;
}
return a;
}
AudioStreamInfo::AudioStreamInfo(uint8_t bits_per_sample, uint8_t channels, uint32_t sample_rate)
: bits_per_sample_(bits_per_sample), channels_(channels), sample_rate_(sample_rate) {
this->ms_sample_rate_gcd_ = gcd(1000, this->sample_rate_);
this->bytes_per_sample_ = (this->bits_per_sample_ + 7) / 8;
}
uint32_t AudioStreamInfo::frames_to_microseconds(uint32_t frames) const {
return (frames * 1000000 + (this->sample_rate_ >> 1)) / this->sample_rate_;
}
uint32_t AudioStreamInfo::frames_to_milliseconds_with_remainder(uint32_t *total_frames) const {
uint32_t unprocessable_frames = *total_frames % (this->sample_rate_ / this->ms_sample_rate_gcd_);
uint32_t frames_for_ms_calculation = *total_frames - unprocessable_frames;
uint32_t playback_ms = (frames_for_ms_calculation * 1000) / this->sample_rate_;
*total_frames = unprocessable_frames;
return playback_ms;
}
bool AudioStreamInfo::operator==(const AudioStreamInfo &rhs) const {
return (this->bits_per_sample_ == rhs.get_bits_per_sample()) && (this->channels_ == rhs.get_channels()) &&
(this->sample_rate_ == rhs.get_sample_rate());
}
const char *audio_file_type_to_string(AudioFileType file_type) {
switch (file_type) {
#ifdef USE_AUDIO_FLAC_SUPPORT
case AudioFileType::FLAC:
return "FLAC";
#endif
#ifdef USE_AUDIO_MP3_SUPPORT
case AudioFileType::MP3:
return "MP3";
#endif
case AudioFileType::WAV:
return "WAV";
default:
return "unknown";
}
}
void scale_audio_samples(const int16_t *audio_samples, int16_t *output_buffer, int16_t scale_factor,
size_t samples_to_scale) {
// Note the assembly dsps_mulc function has audio glitches if the input and output buffers are the same.
for (int i = 0; i < samples_to_scale; i++) {
int32_t acc = (int32_t) audio_samples[i] * (int32_t) scale_factor;
output_buffer[i] = (int16_t) (acc >> 15);
}
}
} // namespace audio
} // namespace esphome

132
esphome/components/audio/audio.h
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@ -1,21 +1,139 @@
#pragma once
#include "esphome/core/defines.h"
#include <cstddef>
#include <cstdint>
namespace esphome {
namespace audio {
struct AudioStreamInfo {
bool operator==(const AudioStreamInfo &rhs) const {
return (channels == rhs.channels) && (bits_per_sample == rhs.bits_per_sample) && (sample_rate == rhs.sample_rate);
class AudioStreamInfo {
/* Class to respresent important parameters of the audio stream that also provides helper function to convert between
* various audio related units.
*
* - An audio sample represents a unit of audio for one channel.
* - A frame represents a unit of audio with a sample for every channel.
*
* In gneneral, converting between bytes, samples, and frames shouldn't result in rounding errors so long as frames
* are used as the main unit when transferring audio data. Durations may result in rounding for certain sample rates;
* e.g., 44.1 KHz. The ``frames_to_milliseconds_with_remainder`` function should be used for accuracy, as it takes
* into account the remainder rather than just ignoring any rounding.
*/
public:
AudioStreamInfo()
: AudioStreamInfo(16, 1, 16000){}; // Default values represent ESPHome's audio components historical values
AudioStreamInfo(uint8_t bits_per_sample, uint8_t channels, uint32_t sample_rate);
uint8_t get_bits_per_sample() const { return this->bits_per_sample_; }
uint8_t get_channels() const { return this->channels_; }
uint32_t get_sample_rate() const { return this->sample_rate_; }
/// @brief Convert bytes to duration in milliseconds.
/// @param bytes Number of bytes to convert
/// @return Duration in milliseconds that will store `bytes` bytes of audio. May round down for certain sample rates
/// or values of `bytes`.
uint32_t bytes_to_ms(size_t bytes) const {
return bytes * 1000 / (this->sample_rate_ * this->bytes_per_sample_ * this->channels_);
}
/// @brief Convert bytes to frames.
/// @param bytes Number of bytes to convert
/// @return Audio frames that will store `bytes` bytes.
uint32_t bytes_to_frames(size_t bytes) const { return (bytes / (this->bytes_per_sample_ * this->channels_)); }
/// @brief Convert bytes to samples.
/// @param bytes Number of bytes to convert
/// @return Audio samples that will store `bytes` bytes.
uint32_t bytes_to_samples(size_t bytes) const { return (bytes / this->bytes_per_sample_); }
/// @brief Converts frames to bytes.
/// @param frames Number of frames to convert.
/// @return Number of bytes that will store `frames` frames of audio.
size_t frames_to_bytes(uint32_t frames) const { return frames * this->bytes_per_sample_ * this->channels_; }
/// @brief Converts samples to bytes.
/// @param samples Number of samples to convert.
/// @return Number of bytes that will store `samples` samples of audio.
size_t samples_to_bytes(uint32_t samples) const { return samples * this->bytes_per_sample_; }
/// @brief Converts duration to frames.
/// @param ms Duration in milliseconds
/// @return Audio frames that will store `ms` milliseconds of audio. May round down for certain sample rates.
uint32_t ms_to_frames(uint32_t ms) const { return (ms * this->sample_rate_) / 1000; }
/// @brief Converts duration to samples.
/// @param ms Duration in milliseconds
/// @return Audio samples that will store `ms` milliseconds of audio. May round down for certain sample rates.
uint32_t ms_to_samples(uint32_t ms) const { return (ms * this->channels_ * this->sample_rate_) / 1000; }
/// @brief Converts duration to bytes. May round down for certain sample rates.
/// @param ms Duration in milliseconds
/// @return Bytes that will store `ms` milliseconds of audio. May round down for certain sample rates.
size_t ms_to_bytes(uint32_t ms) const {
return (ms * this->bytes_per_sample_ * this->channels_ * this->sample_rate_) / 1000;
}
/// @brief Computes the duration, in microseconds, the given amount of frames represents.
/// @param frames Number of audio frames
/// @return Duration in microseconds `frames` respresents. May be slightly inaccurate due to integer divison rounding
/// for certain sample rates.
uint32_t frames_to_microseconds(uint32_t frames) const;
/// @brief Computes the duration, in milliseconds, the given amount of frames represents. Avoids
/// accumulating rounding errors by updating `frames` with the remainder after converting.
/// @param frames Pointer to uint32_t with the number of audio frames. Replaced with the remainder.
/// @return Duration in milliseconds `frames` represents. Always less than or equal to the actual value due to
/// rounding.
uint32_t frames_to_milliseconds_with_remainder(uint32_t *frames) const;
// Class comparison operators
bool operator==(const AudioStreamInfo &rhs) const;
bool operator!=(const AudioStreamInfo &rhs) const { return !operator==(rhs); }
size_t get_bytes_per_sample() const { return bits_per_sample / 8; }
uint8_t channels = 1;
uint8_t bits_per_sample = 16;
uint32_t sample_rate = 16000;
protected:
uint8_t bits_per_sample_;
uint8_t channels_;
uint32_t sample_rate_;
// The greatest common divisor between 1000 ms = 1 second and the sample rate. Used to avoid accumulating error when
// converting from frames to duration. Computed at construction.
uint32_t ms_sample_rate_gcd_;
// Conversion factor derived from the number of bits per sample. Assumes audio data is aligned to the byte. Computed
// at construction.
size_t bytes_per_sample_;
};
enum class AudioFileType : uint8_t {
NONE = 0,
#ifdef USE_AUDIO_FLAC_SUPPORT
FLAC,
#endif
#ifdef USE_AUDIO_MP3_SUPPORT
MP3,
#endif
WAV,
};
struct AudioFile {
const uint8_t *data;
size_t length;
AudioFileType file_type;
};
/// @brief Helper function to convert file type to a const char string
/// @param file_type
/// @return const char pointer to the readable file type
const char *audio_file_type_to_string(AudioFileType file_type);
/// @brief Scales Q15 fixed point audio samples. Scales in place if audio_samples == output_buffer.
/// @param audio_samples PCM int16 audio samples
/// @param output_buffer Buffer to store the scaled samples
/// @param scale_factor Q15 fixed point scaling factor
/// @param samples_to_scale Number of samples to scale
void scale_audio_samples(const int16_t *audio_samples, int16_t *output_buffer, int16_t scale_factor,
size_t samples_to_scale);
} // namespace audio
} // namespace esphome

361
esphome/components/audio/audio_decoder.cpp Normal file
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@ -0,0 +1,361 @@
#include "audio_decoder.h"
#ifdef USE_ESP32
#include "esphome/core/hal.h"
namespace esphome {
namespace audio {
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
static const uint32_t MAX_POTENTIALLY_FAILED_COUNT = 10;
AudioDecoder::AudioDecoder(size_t input_buffer_size, size_t output_buffer_size) {
this->input_transfer_buffer_ = AudioSourceTransferBuffer::create(input_buffer_size);
this->output_transfer_buffer_ = AudioSinkTransferBuffer::create(output_buffer_size);
}
AudioDecoder::~AudioDecoder() {
#ifdef USE_AUDIO_MP3_SUPPORT
if (this->audio_file_type_ == AudioFileType::MP3) {
esp_audio_libs::helix_decoder::MP3FreeDecoder(this->mp3_decoder_);
}
#endif
}
esp_err_t AudioDecoder::add_source(std::weak_ptr<RingBuffer> &input_ring_buffer) {
if (this->input_transfer_buffer_ != nullptr) {
this->input_transfer_buffer_->set_source(input_ring_buffer);
return ESP_OK;
}
return ESP_ERR_NO_MEM;
}
esp_err_t AudioDecoder::add_sink(std::weak_ptr<RingBuffer> &output_ring_buffer) {
if (this->output_transfer_buffer_ != nullptr) {
this->output_transfer_buffer_->set_sink(output_ring_buffer);
return ESP_OK;
}
return ESP_ERR_NO_MEM;
}
#ifdef USE_SPEAKER
esp_err_t AudioDecoder::add_sink(speaker::Speaker *speaker) {
if (this->output_transfer_buffer_ != nullptr) {
this->output_transfer_buffer_->set_sink(speaker);
return ESP_OK;
}
return ESP_ERR_NO_MEM;
}
#endif
esp_err_t AudioDecoder::start(AudioFileType audio_file_type) {
if ((this->input_transfer_buffer_ == nullptr) || (this->output_transfer_buffer_ == nullptr)) {
return ESP_ERR_NO_MEM;
}
this->audio_file_type_ = audio_file_type;
this->potentially_failed_count_ = 0;
this->end_of_file_ = false;
switch (this->audio_file_type_) {
#ifdef USE_AUDIO_FLAC_SUPPORT
case AudioFileType::FLAC:
this->flac_decoder_ = make_unique<esp_audio_libs::flac::FLACDecoder>();
this->free_buffer_required_ =
this->output_transfer_buffer_->capacity(); // We'll revise this after reading the header
break;
#endif
#ifdef USE_AUDIO_MP3_SUPPORT
case AudioFileType::MP3:
this->mp3_decoder_ = esp_audio_libs::helix_decoder::MP3InitDecoder();
this->free_buffer_required_ = 1152 * sizeof(int16_t) * 2; // samples * size per sample * channels
break;
#endif
case AudioFileType::WAV:
this->wav_decoder_ = make_unique<esp_audio_libs::wav_decoder::WAVDecoder>();
this->wav_decoder_->reset();
this->free_buffer_required_ = 1024;
break;
case AudioFileType::NONE:
default:
return ESP_ERR_NOT_SUPPORTED;
break;
}
return ESP_OK;
}
AudioDecoderState AudioDecoder::decode(bool stop_gracefully) {
if (stop_gracefully) {
if (this->output_transfer_buffer_->available() == 0) {
if (this->end_of_file_) {
// The file decoder indicates it reached the end of file
return AudioDecoderState::FINISHED;
}
if (!this->input_transfer_buffer_->has_buffered_data()) {
// If all the internal buffers are empty, the decoding is done
return AudioDecoderState::FINISHED;
}
}
}
if (this->potentially_failed_count_ > MAX_POTENTIALLY_FAILED_COUNT) {
if (stop_gracefully) {
// No more new data is going to come in, so decoding is done
return AudioDecoderState::FINISHED;
}
return AudioDecoderState::FAILED;
}
FileDecoderState state = FileDecoderState::MORE_TO_PROCESS;
uint32_t decoding_start = millis();
while (state == FileDecoderState::MORE_TO_PROCESS) {
// Transfer decoded out
if (!this->pause_output_) {
size_t bytes_written = this->output_transfer_buffer_->transfer_data_to_sink(pdMS_TO_TICKS(READ_WRITE_TIMEOUT_MS));
if (this->audio_stream_info_.has_value()) {
this->accumulated_frames_written_ += this->audio_stream_info_.value().bytes_to_frames(bytes_written);
this->playback_ms_ +=
this->audio_stream_info_.value().frames_to_milliseconds_with_remainder(&this->accumulated_frames_written_);
}
} else {
// If paused, block to avoid wasting CPU resources
delay(READ_WRITE_TIMEOUT_MS);
}
// Verify there is enough space to store more decoded audio and that the function hasn't been running too long
if ((this->output_transfer_buffer_->free() < this->free_buffer_required_) ||
(millis() - decoding_start > DECODING_TIMEOUT_MS)) {
return AudioDecoderState::DECODING;
}
// Decode more audio
size_t bytes_read = this->input_transfer_buffer_->transfer_data_from_source(pdMS_TO_TICKS(READ_WRITE_TIMEOUT_MS));
if ((this->potentially_failed_count_ > 0) && (bytes_read == 0)) {
// Failed to decode in last attempt and there is no new data
if (this->input_transfer_buffer_->free() == 0) {
// The input buffer is full. Since it previously failed on the exact same data, we can never recover
state = FileDecoderState::FAILED;
} else {
// Attempt to get more data next time
state = FileDecoderState::IDLE;
}
} else if (this->input_transfer_buffer_->available() == 0) {
// No data to decode, attempt to get more data next time
state = FileDecoderState::IDLE;
} else {
switch (this->audio_file_type_) {
#ifdef USE_AUDIO_FLAC_SUPPORT
case AudioFileType::FLAC:
state = this->decode_flac_();
break;
#endif
#ifdef USE_AUDIO_MP3_SUPPORT
case AudioFileType::MP3:
state = this->decode_mp3_();
break;
#endif
case AudioFileType::WAV:
state = this->decode_wav_();
break;
case AudioFileType::NONE:
default:
state = FileDecoderState::IDLE;
break;
}
}
if (state == FileDecoderState::POTENTIALLY_FAILED) {
++this->potentially_failed_count_;
} else if (state == FileDecoderState::END_OF_FILE) {
this->end_of_file_ = true;
} else if (state == FileDecoderState::FAILED) {
return AudioDecoderState::FAILED;
} else if (state == FileDecoderState::MORE_TO_PROCESS) {
this->potentially_failed_count_ = 0;
}
}
return AudioDecoderState::DECODING;
}
#ifdef USE_AUDIO_FLAC_SUPPORT
FileDecoderState AudioDecoder::decode_flac_() {
if (!this->audio_stream_info_.has_value()) {
// Header hasn't been read
auto result = this->flac_decoder_->read_header(this->input_transfer_buffer_->get_buffer_start(),
this->input_transfer_buffer_->available());
if (result == esp_audio_libs::flac::FLAC_DECODER_HEADER_OUT_OF_DATA) {
return FileDecoderState::POTENTIALLY_FAILED;
}
if (result != esp_audio_libs::flac::FLAC_DECODER_SUCCESS) {
// Couldn't read FLAC header
return FileDecoderState::FAILED;
}
size_t bytes_consumed = this->flac_decoder_->get_bytes_index();
this->input_transfer_buffer_->decrease_buffer_length(bytes_consumed);
this->free_buffer_required_ = flac_decoder_->get_output_buffer_size_bytes();
if (this->output_transfer_buffer_->capacity() < this->free_buffer_required_) {
// Output buffer is not big enough
if (!this->output_transfer_buffer_->reallocate(this->free_buffer_required_)) {
// Couldn't reallocate output buffer
return FileDecoderState::FAILED;
}
}
this->audio_stream_info_ =
audio::AudioStreamInfo(this->flac_decoder_->get_sample_depth(), this->flac_decoder_->get_num_channels(),
this->flac_decoder_->get_sample_rate());
return FileDecoderState::MORE_TO_PROCESS;
}
uint32_t output_samples = 0;
auto result = this->flac_decoder_->decode_frame(
this->input_transfer_buffer_->get_buffer_start(), this->input_transfer_buffer_->available(),
reinterpret_cast<int16_t *>(this->output_transfer_buffer_->get_buffer_end()), &output_samples);
if (result == esp_audio_libs::flac::FLAC_DECODER_ERROR_OUT_OF_DATA) {
// Not an issue, just needs more data that we'll get next time.
return FileDecoderState::POTENTIALLY_FAILED;
}
size_t bytes_consumed = this->flac_decoder_->get_bytes_index();
this->input_transfer_buffer_->decrease_buffer_length(bytes_consumed);
if (result > esp_audio_libs::flac::FLAC_DECODER_ERROR_OUT_OF_DATA) {
// Corrupted frame, don't retry with current buffer content, wait for new sync
return FileDecoderState::POTENTIALLY_FAILED;
}
// We have successfully decoded some input data and have new output data
this->output_transfer_buffer_->increase_buffer_length(
this->audio_stream_info_.value().samples_to_bytes(output_samples));
if (result == esp_audio_libs::flac::FLAC_DECODER_NO_MORE_FRAMES) {
return FileDecoderState::END_OF_FILE;
}
return FileDecoderState::MORE_TO_PROCESS;
}
#endif
#ifdef USE_AUDIO_MP3_SUPPORT
FileDecoderState AudioDecoder::decode_mp3_() {
// Look for the next sync word
int buffer_length = (int) this->input_transfer_buffer_->available();
int32_t offset =
esp_audio_libs::helix_decoder::MP3FindSyncWord(this->input_transfer_buffer_->get_buffer_start(), buffer_length);
if (offset < 0) {
// New data may have the sync word
this->input_transfer_buffer_->decrease_buffer_length(buffer_length);
return FileDecoderState::POTENTIALLY_FAILED;
}
// Advance read pointer to match the offset for the syncword
this->input_transfer_buffer_->decrease_buffer_length(offset);
uint8_t *buffer_start = this->input_transfer_buffer_->get_buffer_start();
buffer_length = (int) this->input_transfer_buffer_->available();
int err = esp_audio_libs::helix_decoder::MP3Decode(this->mp3_decoder_, &buffer_start, &buffer_length,
(int16_t *) this->output_transfer_buffer_->get_buffer_end(), 0);
size_t consumed = this->input_transfer_buffer_->available() - buffer_length;
this->input_transfer_buffer_->decrease_buffer_length(consumed);
if (err) {
switch (err) {
case esp_audio_libs::helix_decoder::ERR_MP3_OUT_OF_MEMORY:
// Intentional fallthrough
case esp_audio_libs::helix_decoder::ERR_MP3_NULL_POINTER:
return FileDecoderState::FAILED;
break;
default:
// Most errors are recoverable by moving on to the next frame, so mark as potentailly failed
return FileDecoderState::POTENTIALLY_FAILED;
break;
}
} else {
esp_audio_libs::helix_decoder::MP3FrameInfo mp3_frame_info;
esp_audio_libs::helix_decoder::MP3GetLastFrameInfo(this->mp3_decoder_, &mp3_frame_info);
if (mp3_frame_info.outputSamps > 0) {
int bytes_per_sample = (mp3_frame_info.bitsPerSample / 8);
this->output_transfer_buffer_->increase_buffer_length(mp3_frame_info.outputSamps * bytes_per_sample);
if (!this->audio_stream_info_.has_value()) {
this->audio_stream_info_ =
audio::AudioStreamInfo(mp3_frame_info.bitsPerSample, mp3_frame_info.nChans, mp3_frame_info.samprate);
}
}
}
return FileDecoderState::MORE_TO_PROCESS;
}
#endif
FileDecoderState AudioDecoder::decode_wav_() {
if (!this->audio_stream_info_.has_value()) {
// Header hasn't been processed
esp_audio_libs::wav_decoder::WAVDecoderResult result = this->wav_decoder_->decode_header(
this->input_transfer_buffer_->get_buffer_start(), this->input_transfer_buffer_->available());
if (result == esp_audio_libs::wav_decoder::WAV_DECODER_SUCCESS_IN_DATA) {
this->input_transfer_buffer_->decrease_buffer_length(this->wav_decoder_->bytes_processed());
this->audio_stream_info_ = audio::AudioStreamInfo(
this->wav_decoder_->bits_per_sample(), this->wav_decoder_->num_channels(), this->wav_decoder_->sample_rate());
this->wav_bytes_left_ = this->wav_decoder_->chunk_bytes_left();
this->wav_has_known_end_ = (this->wav_bytes_left_ > 0);
return FileDecoderState::MORE_TO_PROCESS;
} else if (result == esp_audio_libs::wav_decoder::WAV_DECODER_WARNING_INCOMPLETE_DATA) {
// Available data didn't have the full header
return FileDecoderState::POTENTIALLY_FAILED;
} else {
return FileDecoderState::FAILED;
}
} else {
if (!this->wav_has_known_end_ || (this->wav_bytes_left_ > 0)) {
size_t bytes_to_copy = this->input_transfer_buffer_->available();
if (this->wav_has_known_end_) {
bytes_to_copy = std::min(bytes_to_copy, this->wav_bytes_left_);
}
bytes_to_copy = std::min(bytes_to_copy, this->output_transfer_buffer_->free());
if (bytes_to_copy > 0) {
std::memcpy(this->output_transfer_buffer_->get_buffer_end(), this->input_transfer_buffer_->get_buffer_start(),
bytes_to_copy);
this->input_transfer_buffer_->decrease_buffer_length(bytes_to_copy);
this->output_transfer_buffer_->increase_buffer_length(bytes_to_copy);
if (this->wav_has_known_end_) {
this->wav_bytes_left_ -= bytes_to_copy;
}
}
return FileDecoderState::IDLE;
}
}
return FileDecoderState::END_OF_FILE;
}
} // namespace audio
} // namespace esphome
#endif

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#pragma once
#ifdef USE_ESP32
#include "audio.h"
#include "audio_transfer_buffer.h"
#include "esphome/core/defines.h"
#include "esphome/core/helpers.h"
#include "esphome/core/ring_buffer.h"
#ifdef USE_SPEAKER
#include "esphome/components/speaker/speaker.h"
#endif
#include "esp_err.h"
// esp-audio-libs
#ifdef USE_AUDIO_FLAC_SUPPORT
#include <flac_decoder.h>
#endif
#ifdef USE_AUDIO_MP3_SUPPORT
#include <mp3_decoder.h>
#endif
#include <wav_decoder.h>
namespace esphome {
namespace audio {
enum class AudioDecoderState : uint8_t {
DECODING = 0, // More data is available to decode
FINISHED, // All file data has been decoded and transferred
FAILED, // Encountered an error
};
// Only used within the AudioDecoder class; conveys the state of the particular file type decoder
enum class FileDecoderState : uint8_t {
MORE_TO_PROCESS, // Successsfully read a file chunk and more data is available to decode
IDLE, // Not enough data to decode, waiting for more to be transferred
POTENTIALLY_FAILED, // Decoder encountered a potentially recoverable error if more file data is available
FAILED, // Decoder encoutnered an uncrecoverable error
END_OF_FILE, // The specific file decoder knows its the end of the file
};
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.
*/
public:
/// @brief Allocates the input and output transfer buffers
/// @param input_buffer_size Size of the input transfer buffer in bytes.
/// @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)
~AudioDecoder();
/// @brief Adds a source ring buffer for raw file data. Takes ownership of the ring buffer in a shared_ptr.
/// @param input_ring_buffer weak_ptr of a shared_ptr of the sink ring buffer to transfer ownership
/// @return ESP_OK if successsful, ESP_ERR_NO_MEM if the transfer buffer wasn't allocated
esp_err_t add_source(std::weak_ptr<RingBuffer> &input_ring_buffer);
/// @brief Adds a sink ring buffer for decoded audio. Takes ownership of the ring buffer in a shared_ptr.
/// @param output_ring_buffer weak_ptr of a shared_ptr of the sink ring buffer to transfer ownership
/// @return ESP_OK if successsful, ESP_ERR_NO_MEM if the transfer buffer wasn't allocated
esp_err_t add_sink(std::weak_ptr<RingBuffer> &output_ring_buffer);
#ifdef USE_SPEAKER
/// @brief Adds a sink speaker for decoded audio.
/// @param speaker pointer to speaker component
/// @return ESP_OK if successsful, ESP_ERR_NO_MEM if the transfer buffer wasn't allocated
esp_err_t add_sink(speaker::Speaker *speaker);
#endif
/// @brief Sets up decoding the file
/// @param audio_file_type AudioFileType of the file
/// @return ESP_OK if successful, ESP_ERR_NO_MEM if the transfer buffers fail to allocate, or ESP_ERR_NOT_SUPPORTED if
/// the format isn't supported.
esp_err_t start(AudioFileType audio_file_type);
/// @brief Decodes audio from the ring buffer source and writes to the sink.
/// @param stop_gracefully If true, it indicates the file source is finished. The decoder will decode all the
/// reamining data and then finish.
/// @return AudioDecoderState
AudioDecoderState decode(bool stop_gracefully);
/// @brief Gets the audio stream information, if it has been decoded from the files header
/// @return optional<AudioStreamInfo> with the audio information. If not available yet, returns no value.
const optional<audio::AudioStreamInfo> &get_audio_stream_info() const { return this->audio_stream_info_; }
/// @brief Returns the duration of audio (in milliseconds) decoded and sent to the sink
/// @return Duration of decoded audio in milliseconds
uint32_t get_playback_ms() const { return this->playback_ms_; }
/// @brief Pauses sending resampled audio to the sink. If paused, it will continue to process internal buffers.
/// @param pause_state If true, audio data is not sent to the sink.
void set_pause_output_state(bool pause_state) { this->pause_output_ = pause_state; }
protected:
std::unique_ptr<esp_audio_libs::wav_decoder::WAVDecoder> wav_decoder_;
#ifdef USE_AUDIO_FLAC_SUPPORT
FileDecoderState decode_flac_();
std::unique_ptr<esp_audio_libs::flac::FLACDecoder> flac_decoder_;
#endif
#ifdef USE_AUDIO_MP3_SUPPORT
FileDecoderState decode_mp3_();
esp_audio_libs::helix_decoder::HMP3Decoder mp3_decoder_;
#endif
FileDecoderState decode_wav_();
std::unique_ptr<AudioSourceTransferBuffer> input_transfer_buffer_;
std::unique_ptr<AudioSinkTransferBuffer> output_transfer_buffer_;
AudioFileType audio_file_type_{AudioFileType::NONE};
optional<AudioStreamInfo> audio_stream_info_{};
size_t free_buffer_required_{0};
size_t wav_bytes_left_{0};
uint32_t potentially_failed_count_{0};
bool end_of_file_{false};
bool wav_has_known_end_{false};
bool pause_output_{false};
uint32_t accumulated_frames_written_{0};
uint32_t playback_ms_{0};
};
} // namespace audio
} // namespace esphome
#endif

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#include "audio_reader.h"
#ifdef USE_ESP_IDF
#include "esphome/core/defines.h"
#include "esphome/core/hal.h"
#include "esphome/core/helpers.h"
#if CONFIG_MBEDTLS_CERTIFICATE_BUNDLE
#include "esp_crt_bundle.h"
#endif
namespace esphome {
namespace audio {
static const uint32_t READ_WRITE_TIMEOUT_MS = 20;
// The number of times the http read times out with no data before throwing an error
static const uint32_t ERROR_COUNT_NO_DATA_READ_TIMEOUT = 100;
static const size_t HTTP_STREAM_BUFFER_SIZE = 2048;
static const uint8_t MAX_REDIRECTION = 5;
// Some common HTTP status codes - borrowed from http_request component accessed 20241224
enum HttpStatus {
HTTP_STATUS_OK = 200,
HTTP_STATUS_NO_CONTENT = 204,
HTTP_STATUS_PARTIAL_CONTENT = 206,
/* 3xx - Redirection */
HTTP_STATUS_MULTIPLE_CHOICES = 300,
HTTP_STATUS_MOVED_PERMANENTLY = 301,
HTTP_STATUS_FOUND = 302,
HTTP_STATUS_SEE_OTHER = 303,
HTTP_STATUS_NOT_MODIFIED = 304,
HTTP_STATUS_TEMPORARY_REDIRECT = 307,
HTTP_STATUS_PERMANENT_REDIRECT = 308,
/* 4XX - CLIENT ERROR */
HTTP_STATUS_BAD_REQUEST = 400,
HTTP_STATUS_UNAUTHORIZED = 401,
HTTP_STATUS_FORBIDDEN = 403,
HTTP_STATUS_NOT_FOUND = 404,
HTTP_STATUS_METHOD_NOT_ALLOWED = 405,
HTTP_STATUS_NOT_ACCEPTABLE = 406,
HTTP_STATUS_LENGTH_REQUIRED = 411,
/* 5xx - Server Error */
HTTP_STATUS_INTERNAL_ERROR = 500
};
AudioReader::~AudioReader() { this->cleanup_connection_(); }
esp_err_t AudioReader::add_sink(const std::weak_ptr<RingBuffer> &output_ring_buffer) {
if (current_audio_file_ != nullptr) {
// A transfer buffer isn't ncessary for a local file
this->file_ring_buffer_ = output_ring_buffer.lock();
return ESP_OK;
}
if (this->output_transfer_buffer_ != nullptr) {
this->output_transfer_buffer_->set_sink(output_ring_buffer);
return ESP_OK;
}
return ESP_ERR_INVALID_STATE;
}
esp_err_t AudioReader::start(AudioFile *audio_file, AudioFileType &file_type) {
file_type = AudioFileType::NONE;
this->current_audio_file_ = audio_file;
this->file_current_ = audio_file->data;
file_type = audio_file->file_type;
return ESP_OK;
}
esp_err_t AudioReader::start(const std::string &uri, AudioFileType &file_type) {
file_type = AudioFileType::NONE;
this->cleanup_connection_();
if (uri.empty()) {
return ESP_ERR_INVALID_ARG;
}
esp_http_client_config_t client_config = {};
client_config.url = uri.c_str();
client_config.cert_pem = nullptr;
client_config.disable_auto_redirect = false;
client_config.max_redirection_count = 10;
client_config.event_handler = http_event_handler;
client_config.user_data = this;
client_config.buffer_size = HTTP_STREAM_BUFFER_SIZE;
client_config.keep_alive_enable = true;
client_config.timeout_ms = 5000; // Shouldn't trigger watchdog resets if caller runs in a task
#if CONFIG_MBEDTLS_CERTIFICATE_BUNDLE
if (uri.find("https:") != std::string::npos) {
client_config.crt_bundle_attach = esp_crt_bundle_attach;
}
#endif
this->client_ = esp_http_client_init(&client_config);
if (this->client_ == nullptr) {
return ESP_FAIL;
}
esp_err_t err = esp_http_client_open(this->client_, 0);
if (err != ESP_OK) {
this->cleanup_connection_();
return err;
}
int64_t header_length = esp_http_client_fetch_headers(this->client_);
if (header_length < 0) {
this->cleanup_connection_();
return ESP_FAIL;
}
int status_code = esp_http_client_get_status_code(this->client_);
if ((status_code < HTTP_STATUS_OK) || (status_code > HTTP_STATUS_PERMANENT_REDIRECT)) {
this->cleanup_connection_();
return ESP_FAIL;
}
ssize_t redirect_count = 0;
while ((esp_http_client_set_redirection(this->client_) == ESP_OK) && (redirect_count < MAX_REDIRECTION)) {
err = esp_http_client_open(this->client_, 0);
if (err != ESP_OK) {
this->cleanup_connection_();
return ESP_FAIL;
}
header_length = esp_http_client_fetch_headers(this->client_);
if (header_length < 0) {
this->cleanup_connection_();
return ESP_FAIL;
}
status_code = esp_http_client_get_status_code(this->client_);
if ((status_code < HTTP_STATUS_OK) || (status_code > HTTP_STATUS_PERMANENT_REDIRECT)) {
this->cleanup_connection_();
return ESP_FAIL;
}
++redirect_count;
}
if (this->audio_file_type_ == AudioFileType::NONE) {
// Failed to determine the file type from the header, fallback to using the url
char url[500];
err = esp_http_client_get_url(this->client_, url, 500);
if (err != ESP_OK) {
this->cleanup_connection_();
return err;
}
std::string url_string = str_lower_case(url);
if (str_endswith(url_string, ".wav")) {
file_type = AudioFileType::WAV;
}
#ifdef USE_AUDIO_MP3_SUPPORT
else if (str_endswith(url_string, ".mp3")) {
file_type = AudioFileType::MP3;
}
#endif
#ifdef USE_AUDIO_FLAC_SUPPORT
else if (str_endswith(url_string, ".flac")) {
file_type = AudioFileType::FLAC;
}
#endif
else {
file_type = AudioFileType::NONE;
this->cleanup_connection_();
return ESP_ERR_NOT_SUPPORTED;
}
} else {
file_type = this->audio_file_type_;
}
this->no_data_read_count_ = 0;
this->output_transfer_buffer_ = AudioSinkTransferBuffer::create(this->buffer_size_);
if (this->output_transfer_buffer_ == nullptr) {
return ESP_ERR_NO_MEM;
}
return ESP_OK;
}
AudioReaderState AudioReader::read() {
if (this->client_ != nullptr) {
return this->http_read_();
} else if (this->current_audio_file_ != nullptr) {
return this->file_read_();
}
return AudioReaderState::FAILED;
}
AudioFileType AudioReader::get_audio_type(const char *content_type) {
#ifdef USE_AUDIO_MP3_SUPPORT
if (strcasecmp(content_type, "mp3") == 0 || strcasecmp(content_type, "audio/mp3") == 0 ||
strcasecmp(content_type, "audio/mpeg") == 0) {
return AudioFileType::MP3;
}
#endif
if (strcasecmp(content_type, "audio/wav") == 0) {
return AudioFileType::WAV;
}
#ifdef USE_AUDIO_FLAC_SUPPORT
if (strcasecmp(content_type, "audio/flac") == 0 || strcasecmp(content_type, "audio/x-flac") == 0) {
return AudioFileType::FLAC;
}
#endif
return AudioFileType::NONE;
}
esp_err_t AudioReader::http_event_handler(esp_http_client_event_t *evt) {
// Based on https://github.com/maroc81/WeatherLily/tree/main/main/net accessed 20241224
AudioReader *this_reader = (AudioReader *) evt->user_data;
switch (evt->event_id) {
case HTTP_EVENT_ON_HEADER:
if (strcasecmp(evt->header_key, "Content-Type") == 0) {
this_reader->audio_file_type_ = get_audio_type(evt->header_value);
}
break;
default:
break;
}
return ESP_OK;
}
AudioReaderState AudioReader::file_read_() {
size_t remaining_bytes = this->current_audio_file_->length - (this->file_current_ - this->current_audio_file_->data);
if (remaining_bytes > 0) {
size_t bytes_written = this->file_ring_buffer_->write_without_replacement(this->file_current_, remaining_bytes,
pdMS_TO_TICKS(READ_WRITE_TIMEOUT_MS));
this->file_current_ += bytes_written;
return AudioReaderState::READING;
}
return AudioReaderState::FINISHED;
}
AudioReaderState AudioReader::http_read_() {
this->output_transfer_buffer_->transfer_data_to_sink(pdMS_TO_TICKS(READ_WRITE_TIMEOUT_MS));
if (esp_http_client_is_complete_data_received(this->client_)) {
if (this->output_transfer_buffer_->available() == 0) {
this->cleanup_connection_();
return AudioReaderState::FINISHED;
}
} else {
size_t bytes_to_read = this->output_transfer_buffer_->free();
int received_len =
esp_http_client_read(this->client_, (char *) this->output_transfer_buffer_->get_buffer_end(), bytes_to_read);
if (received_len > 0) {
this->output_transfer_buffer_->increase_buffer_length(received_len);
this->no_data_read_count_ = 0;
} else if (received_len < 0) {
// HTTP read error
this->cleanup_connection_();
return AudioReaderState::FAILED;
} else {
if (bytes_to_read > 0) {
// Read timed out
++this->no_data_read_count_;
if (this->no_data_read_count_ >= ERROR_COUNT_NO_DATA_READ_TIMEOUT) {
// Timed out with no data read too many times, so the http read has failed
this->cleanup_connection_();
return AudioReaderState::FAILED;
}
delay(READ_WRITE_TIMEOUT_MS);
}
}
}
return AudioReaderState::READING;
}
void AudioReader::cleanup_connection_() {
if (this->client_ != nullptr) {
esp_http_client_close(this->client_);
esp_http_client_cleanup(this->client_);
this->client_ = nullptr;
}
}
} // namespace audio
} // namespace esphome
#endif

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#pragma once
#ifdef USE_ESP_IDF
#include "audio.h"
#include "audio_transfer_buffer.h"
#include "esphome/core/ring_buffer.h"
#include "esp_err.h"
#include <esp_http_client.h>
namespace esphome {
namespace audio {
enum class AudioReaderState : uint8_t {
READING = 0, // More data is available to read
FINISHED, // All data has been read and transferred
FAILED, // Encountered an error
};
class AudioReader {
/*
* @brief Class that facilitates reading a raw audio file.
* Files can be read from flash (stored in a AudioFile struct) or from an http source.
* The file data is sent to a ring buffer sink.
*/
public:
/// @brief Constructs an AudioReader object.
/// The transfer buffer isn't allocated here, but only if necessary (an http source) in the start function.
/// @param buffer_size Transfer buffer size in bytes.
AudioReader(size_t buffer_size) : buffer_size_(buffer_size) {}
~AudioReader();
/// @brief Adds a sink ring buffer for audio data. Takes ownership of the ring buffer in a shared_ptr
/// @param output_ring_buffer weak_ptr of a shared_ptr of the sink ring buffer to transfer ownership
/// @return ESP_OK if successful, ESP_ERR_INVALID_STATE otherwise
esp_err_t add_sink(const std::weak_ptr<RingBuffer> &output_ring_buffer);
/// @brief Starts reading an audio file from an http source. The transfer buffer is allocated here.
/// @param uri Web url to the http file.
/// @param file_type AudioFileType variable passed-by-reference indicating the type of file being read.
/// @return ESP_OK if successful, an ESP_ERR* code otherwise.
esp_err_t start(const std::string &uri, AudioFileType &file_type);
/// @brief Starts reading an audio file from flash. No transfer buffer is allocated.
/// @param audio_file AudioFile struct containing the file.
/// @param file_type AudioFileType variable passed-by-reference indicating the type of file being read.
/// @return ESP_OK
esp_err_t start(AudioFile *audio_file, AudioFileType &file_type);
/// @brief Reads new file data from the source and sends to the ring buffer sink.
/// @return AudioReaderState
AudioReaderState read();
protected:
/// @brief Monitors the http client events to attempt determining the file type from the Content-Type header
static esp_err_t http_event_handler(esp_http_client_event_t *evt);
/// @brief Determines the audio file type from the http header's Content-Type key
/// @param content_type string with the Content-Type key
/// @return AudioFileType of the url, if it can be determined. If not, return AudioFileType::NONE.
static AudioFileType get_audio_type(const char *content_type);
AudioReaderState file_read_();
AudioReaderState http_read_();
std::shared_ptr<RingBuffer> file_ring_buffer_;
std::unique_ptr<AudioSinkTransferBuffer> output_transfer_buffer_;
void cleanup_connection_();
size_t buffer_size_;
uint32_t no_data_read_count_;
esp_http_client_handle_t client_{nullptr};
AudioFile *current_audio_file_{nullptr};
AudioFileType audio_file_type_{AudioFileType::NONE};
const uint8_t *file_current_{nullptr};
};
} // namespace audio
} // namespace esphome
#endif

159
esphome/components/audio/audio_resampler.cpp Normal file
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#include "audio_resampler.h"
#ifdef USE_ESP32
#include "esphome/core/hal.h"
namespace esphome {
namespace audio {
static const uint32_t READ_WRITE_TIMEOUT_MS = 20;
AudioResampler::AudioResampler(size_t input_buffer_size, size_t output_buffer_size)
: input_buffer_size_(input_buffer_size), output_buffer_size_(output_buffer_size) {
this->input_transfer_buffer_ = AudioSourceTransferBuffer::create(input_buffer_size);
this->output_transfer_buffer_ = AudioSinkTransferBuffer::create(output_buffer_size);
}
esp_err_t AudioResampler::add_source(std::weak_ptr<RingBuffer> &input_ring_buffer) {
if (this->input_transfer_buffer_ != nullptr) {
this->input_transfer_buffer_->set_source(input_ring_buffer);
return ESP_OK;
}
return ESP_ERR_NO_MEM;
}
esp_err_t AudioResampler::add_sink(std::weak_ptr<RingBuffer> &output_ring_buffer) {
if (this->output_transfer_buffer_ != nullptr) {
this->output_transfer_buffer_->set_sink(output_ring_buffer);
return ESP_OK;
}
return ESP_ERR_NO_MEM;
}
#ifdef USE_SPEAKER
esp_err_t AudioResampler::add_sink(speaker::Speaker *speaker) {
if (this->output_transfer_buffer_ != nullptr) {
this->output_transfer_buffer_->set_sink(speaker);
return ESP_OK;
}
return ESP_ERR_NO_MEM;
}
#endif
esp_err_t AudioResampler::start(AudioStreamInfo &input_stream_info, AudioStreamInfo &output_stream_info,
uint16_t number_of_taps, uint16_t number_of_filters) {
this->input_stream_info_ = input_stream_info;
this->output_stream_info_ = output_stream_info;
if ((this->input_transfer_buffer_ == nullptr) || (this->output_transfer_buffer_ == nullptr)) {
return ESP_ERR_NO_MEM;
}
if ((input_stream_info.get_bits_per_sample() > 32) || (output_stream_info.get_bits_per_sample() > 32) ||
(input_stream_info_.get_channels() != output_stream_info.get_channels())) {
return ESP_ERR_NOT_SUPPORTED;
}
if ((input_stream_info.get_sample_rate() != output_stream_info.get_sample_rate()) ||
(input_stream_info.get_bits_per_sample() != output_stream_info.get_bits_per_sample())) {
this->resampler_ = make_unique<esp_audio_libs::resampler::Resampler>(
input_stream_info.bytes_to_samples(this->input_buffer_size_),
output_stream_info.bytes_to_samples(this->output_buffer_size_));
// Use cascaded biquad filters when downsampling to avoid aliasing
bool use_pre_filter = output_stream_info.get_sample_rate() < input_stream_info.get_sample_rate();
esp_audio_libs::resampler::ResamplerConfiguration resample_config = {
.source_sample_rate = static_cast<float>(input_stream_info.get_sample_rate()),
.target_sample_rate = static_cast<float>(output_stream_info.get_sample_rate()),
.source_bits_per_sample = input_stream_info.get_bits_per_sample(),
.target_bits_per_sample = output_stream_info.get_bits_per_sample(),
.channels = input_stream_info_.get_channels(),
.use_pre_or_post_filter = use_pre_filter,
.subsample_interpolate = false, // Doubles the CPU load. Using more filters is a better alternative
.number_of_taps = number_of_taps,
.number_of_filters = number_of_filters,
};
if (!this->resampler_->initialize(resample_config)) {
// Failed to allocate the resampler's internal buffers
return ESP_ERR_NO_MEM;
}
}
return ESP_OK;
}
AudioResamplerState AudioResampler::resample(bool stop_gracefully, int32_t *ms_differential) {
if (stop_gracefully) {
if (!this->input_transfer_buffer_->has_buffered_data() && (this->output_transfer_buffer_->available() == 0)) {
return AudioResamplerState::FINISHED;
}
}
if (!this->pause_output_) {
// Move audio data to the sink
this->output_transfer_buffer_->transfer_data_to_sink(pdMS_TO_TICKS(READ_WRITE_TIMEOUT_MS));
} else {
// If paused, block to avoid wasting CPU resources
delay(READ_WRITE_TIMEOUT_MS);
}
this->input_transfer_buffer_->transfer_data_from_source(pdMS_TO_TICKS(READ_WRITE_TIMEOUT_MS));
if (this->input_transfer_buffer_->available() == 0) {
// No samples available to process
return AudioResamplerState::RESAMPLING;
}
const size_t bytes_free = this->output_transfer_buffer_->free();
const uint32_t frames_free = this->output_stream_info_.bytes_to_frames(bytes_free);
const size_t bytes_available = this->input_transfer_buffer_->available();
const uint32_t frames_available = this->input_stream_info_.bytes_to_frames(bytes_available);
if ((this->input_stream_info_.get_sample_rate() != this->output_stream_info_.get_sample_rate()) ||
(this->input_stream_info_.get_bits_per_sample() != this->output_stream_info_.get_bits_per_sample())) {
esp_audio_libs::resampler::ResamplerResults results =
this->resampler_->resample(this->input_transfer_buffer_->get_buffer_start(),
this->output_transfer_buffer_->get_buffer_end(), frames_available, frames_free, -3);
this->input_transfer_buffer_->decrease_buffer_length(this->input_stream_info_.frames_to_bytes(results.frames_used));
this->output_transfer_buffer_->increase_buffer_length(
this->output_stream_info_.frames_to_bytes(results.frames_generated));
// Resampling causes slight differences in the durations used versus generated. Computes the difference in
// millisconds. The callback function passing the played audio duration uses the difference to convert from output
// duration to input duration.
this->accumulated_frames_used_ += results.frames_used;
this->accumulated_frames_generated_ += results.frames_generated;
const int32_t used_ms =
this->input_stream_info_.frames_to_milliseconds_with_remainder(&this->accumulated_frames_used_);
const int32_t generated_ms =
this->output_stream_info_.frames_to_milliseconds_with_remainder(&this->accumulated_frames_generated_);
*ms_differential = used_ms - generated_ms;
} else {
// No resampling required, copy samples directly to the output transfer buffer
*ms_differential = 0;
const size_t bytes_to_transfer = std::min(this->output_stream_info_.frames_to_bytes(frames_free),
this->input_stream_info_.frames_to_bytes(frames_available));
std::memcpy((void *) this->output_transfer_buffer_->get_buffer_end(),
(void *) this->input_transfer_buffer_->get_buffer_start(), bytes_to_transfer);
this->input_transfer_buffer_->decrease_buffer_length(bytes_to_transfer);
this->output_transfer_buffer_->increase_buffer_length(bytes_to_transfer);
}
return AudioResamplerState::RESAMPLING;
}
} // namespace audio
} // namespace esphome
#endif

101
esphome/components/audio/audio_resampler.h Normal file
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#pragma once
#ifdef USE_ESP32
#include "audio.h"
#include "audio_transfer_buffer.h"
#include "esphome/core/defines.h"
#include "esphome/core/ring_buffer.h"
#ifdef USE_SPEAKER
#include "esphome/components/speaker/speaker.h"
#endif
#include "esp_err.h"
#include <resampler.h> // esp-audio-libs
namespace esphome {
namespace audio {
enum class AudioResamplerState : uint8_t {
RESAMPLING, // More data is available to resample
FINISHED, // All file data has been resampled and transferred
FAILED, // Unused state included for consistency among Audio classes
};
class AudioResampler {
/*
* @brief Class that facilitates resampling audio.
* The audio data is read from a ring buffer source, resampled, and sent to an audio sink (ring buffer or speaker
* component). Also supports converting bits per sample.
*/
public:
/// @brief Allocates the input and output transfer buffers
/// @param input_buffer_size Size of the input transfer buffer in bytes.
/// @param output_buffer_size Size of the output transfer buffer in bytes.
AudioResampler(size_t input_buffer_size, size_t output_buffer_size);
/// @brief Adds a source ring buffer for audio data. Takes ownership of the ring buffer in a shared_ptr.
/// @param input_ring_buffer weak_ptr of a shared_ptr of the sink ring buffer to transfer ownership
/// @return ESP_OK if successsful, ESP_ERR_NO_MEM if the transfer buffer wasn't allocated
esp_err_t add_source(std::weak_ptr<RingBuffer> &input_ring_buffer);
/// @brief Adds a sink ring buffer for resampled audio. Takes ownership of the ring buffer in a shared_ptr.
/// @param output_ring_buffer weak_ptr of a shared_ptr of the sink ring buffer to transfer ownership
/// @return ESP_OK if successsful, ESP_ERR_NO_MEM if the transfer buffer wasn't allocated
esp_err_t add_sink(std::weak_ptr<RingBuffer> &output_ring_buffer);
#ifdef USE_SPEAKER
/// @brief Adds a sink speaker for decoded audio.
/// @param speaker pointer to speaker component
/// @return ESP_OK if successsful, ESP_ERR_NO_MEM if the transfer buffer wasn't allocated
esp_err_t add_sink(speaker::Speaker *speaker);
#endif
/// @brief Sets up the class to resample.
/// @param input_stream_info The incoming sample rate, bits per sample, and number of channels
/// @param output_stream_info The desired outgoing sample rate, bits per sample, and number of channels
/// @param number_of_taps Number of taps per FIR filter
/// @param number_of_filters Number of FIR filters
/// @return ESP_OK if it is able to convert the incoming stream,
/// ESP_ERR_NO_MEM if the transfer buffers failed to allocate,
/// ESP_ERR_NOT_SUPPORTED if the stream can't be converted.
esp_err_t start(AudioStreamInfo &input_stream_info, AudioStreamInfo &output_stream_info, uint16_t number_of_taps,
uint16_t number_of_filters);
/// @brief Resamples audio from the ring buffer source and writes to the sink.
/// @param stop_gracefully If true, it indicates the file decoder is finished. The resampler will resample all the
/// remaining audio and then finish.
/// @param ms_differential Pointer to a (int32_t) variable that will store the difference, in milliseconds, between
/// the duration of input audio used and the duration of output audio generated.
/// @return AudioResamplerState
AudioResamplerState resample(bool stop_gracefully, int32_t *ms_differential);
/// @brief Pauses sending resampled audio to the sink. If paused, it will continue to process internal buffers.
/// @param pause_state If true, audio data is not sent to the sink.
void set_pause_output_state(bool pause_state) { this->pause_output_ = pause_state; }
protected:
std::unique_ptr<AudioSourceTransferBuffer> input_transfer_buffer_;
std::unique_ptr<AudioSinkTransferBuffer> output_transfer_buffer_;
size_t input_buffer_size_;
size_t output_buffer_size_;
uint32_t accumulated_frames_used_{0};
uint32_t accumulated_frames_generated_{0};
bool pause_output_{false};
AudioStreamInfo input_stream_info_;
AudioStreamInfo output_stream_info_;
std::unique_ptr<esp_audio_libs::resampler::Resampler> resampler_;
};
} // namespace audio
} // namespace esphome
#endif

165
esphome/components/audio/audio_transfer_buffer.cpp Normal file
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@ -0,0 +1,165 @@
#include "audio_transfer_buffer.h"
#ifdef USE_ESP32
#include "esphome/core/helpers.h"
namespace esphome {
namespace audio {
AudioTransferBuffer::~AudioTransferBuffer() { this->deallocate_buffer_(); };
std::unique_ptr<AudioSinkTransferBuffer> AudioSinkTransferBuffer::create(size_t buffer_size) {
std::unique_ptr<AudioSinkTransferBuffer> sink_buffer = make_unique<AudioSinkTransferBuffer>();
if (!sink_buffer->allocate_buffer_(buffer_size)) {
return nullptr;
}
return sink_buffer;
}
std::unique_ptr<AudioSourceTransferBuffer> AudioSourceTransferBuffer::create(size_t buffer_size) {
std::unique_ptr<AudioSourceTransferBuffer> source_buffer = make_unique<AudioSourceTransferBuffer>();
if (!source_buffer->allocate_buffer_(buffer_size)) {
return nullptr;
}
return source_buffer;
}
size_t AudioTransferBuffer::free() const {
if (this->buffer_size_ == 0) {
return 0;
}
return this->buffer_size_ - (this->buffer_length_ - (this->data_start_ - this->buffer_));
}
void AudioTransferBuffer::decrease_buffer_length(size_t bytes) {
this->buffer_length_ -= bytes;
this->data_start_ += bytes;
}
void AudioTransferBuffer::increase_buffer_length(size_t bytes) { this->buffer_length_ += bytes; }
void AudioTransferBuffer::clear_buffered_data() {
this->buffer_length_ = 0;
if (this->ring_buffer_.use_count() > 0) {
this->ring_buffer_->reset();
}
}
void AudioSinkTransferBuffer::clear_buffered_data() {
this->buffer_length_ = 0;
if (this->ring_buffer_.use_count() > 0) {
this->ring_buffer_->reset();
}
#ifdef USE_SPEAKER
if (this->speaker_ != nullptr) {
this->speaker_->stop();
}
#endif
}
bool AudioTransferBuffer::has_buffered_data() const {
if (this->ring_buffer_.use_count() > 0) {
return ((this->ring_buffer_->available() > 0) || (this->available() > 0));
}
return (this->available() > 0);
}
bool AudioTransferBuffer::reallocate(size_t new_buffer_size) {
if (this->buffer_length_ > 0) {
// Already has data in the buffer, fail
return false;
}
this->deallocate_buffer_();
return this->allocate_buffer_(new_buffer_size);
}
bool AudioTransferBuffer::allocate_buffer_(size_t buffer_size) {
this->buffer_size_ = buffer_size;
RAMAllocator<uint8_t> allocator(ExternalRAMAllocator<uint8_t>::ALLOW_FAILURE);
this->buffer_ = allocator.allocate(this->buffer_size_);
if (this->buffer_ == nullptr) {
return false;
}
this->data_start_ = this->buffer_;
this->buffer_length_ = 0;
return true;
}
void AudioTransferBuffer::deallocate_buffer_() {
if (this->buffer_ != nullptr) {
RAMAllocator<uint8_t> allocator(ExternalRAMAllocator<uint8_t>::ALLOW_FAILURE);
allocator.deallocate(this->buffer_, this->buffer_size_);
this->buffer_ = nullptr;
this->data_start_ = nullptr;
}
this->buffer_size_ = 0;
this->buffer_length_ = 0;
}
size_t AudioSourceTransferBuffer::transfer_data_from_source(TickType_t ticks_to_wait) {
// Shift data in buffer to start
if (this->buffer_length_ > 0) {
memmove(this->buffer_, this->data_start_, this->buffer_length_);
}
this->data_start_ = this->buffer_;
size_t bytes_to_read = this->free();
size_t bytes_read = 0;
if (bytes_to_read > 0) {
if (this->ring_buffer_.use_count() > 0) {
bytes_read = this->ring_buffer_->read((void *) this->get_buffer_end(), bytes_to_read, ticks_to_wait);
}
this->increase_buffer_length(bytes_read);
}
return bytes_read;
}
size_t AudioSinkTransferBuffer::transfer_data_to_sink(TickType_t ticks_to_wait) {
size_t bytes_written = 0;
if (this->available()) {
#ifdef USE_SPEAKER
if (this->speaker_ != nullptr) {
bytes_written = this->speaker_->play(this->data_start_, this->available(), ticks_to_wait);
} else
#endif
if (this->ring_buffer_.use_count() > 0) {
bytes_written =
this->ring_buffer_->write_without_replacement((void *) this->data_start_, this->available(), ticks_to_wait);
}
this->decrease_buffer_length(bytes_written);
// Shift unwritten data to the start of the buffer
memmove(this->buffer_, this->data_start_, this->buffer_length_);
this->data_start_ = this->buffer_;
}
return bytes_written;
}
bool AudioSinkTransferBuffer::has_buffered_data() const {
#ifdef USE_SPEAKER
if (this->speaker_ != nullptr) {
return (this->speaker_->has_buffered_data() || (this->available() > 0));
}
#endif
if (this->ring_buffer_.use_count() > 0) {
return ((this->ring_buffer_->available() > 0) || (this->available() > 0));
}
return (this->available() > 0);
}
} // namespace audio
} // namespace esphome
#endif

139
esphome/components/audio/audio_transfer_buffer.h Normal file
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@ -0,0 +1,139 @@
#pragma once
#ifdef USE_ESP32
#include "esphome/core/defines.h"
#include "esphome/core/ring_buffer.h"
#ifdef USE_SPEAKER
#include "esphome/components/speaker/speaker.h"
#endif
#include "esp_err.h"
#include <freertos/FreeRTOS.h>
namespace esphome {
namespace audio {
class AudioTransferBuffer {
/*
* @brief Class that facilitates tranferring data between a buffer and an audio source or sink.
* The transfer buffer is a typical C array that temporarily holds data for processing in other audio components.
* Both sink and source transfer buffers can use a ring buffer as the sink/source.
* - The ring buffer is stored in a shared_ptr, so destroying the transfer buffer object will release ownership.
*/
public:
/// @brief Destructor that deallocates the transfer buffer
~AudioTransferBuffer();
/// @brief Returns a pointer to the start of the transfer buffer where available() bytes of exisiting data can be read
uint8_t *get_buffer_start() const { return this->data_start_; }
/// @brief Returns a pointer to the end of the transfer buffer where free() bytes of new data can be written
uint8_t *get_buffer_end() const { return this->data_start_ + this->buffer_length_; }
/// @brief Updates the internal state of the transfer buffer. This should be called after reading data
/// @param bytes The number of bytes consumed/read
void decrease_buffer_length(size_t bytes);
/// @brief Updates the internal state of the transfer buffer. This should be called after writing data
/// @param bytes The number of bytes written
void increase_buffer_length(size_t bytes);
/// @brief Returns the transfer buffer's currently available bytes to read
size_t available() const { return this->buffer_length_; }
/// @brief Returns the transfer buffers allocated bytes
size_t capacity() const { return this->buffer_size_; }
/// @brief Returns the transfer buffer's currrently free bytes available to write
size_t free() const;
/// @brief Clears data in the transfer buffer and, if possible, the source/sink.
virtual void clear_buffered_data();
/// @brief Tests if there is any data in the tranfer buffer or the source/sink.
/// @return True if there is data, false otherwise.
virtual bool has_buffered_data() const;
bool reallocate(size_t new_buffer_size);
protected:
/// @brief Allocates the transfer buffer in external memory, if available.
/// @return True is successful, false otherwise.
bool allocate_buffer_(size_t buffer_size);
/// @brief Deallocates the buffer and resets the class variables.
void deallocate_buffer_();
// A possible source or sink for the transfer buffer
std::shared_ptr<RingBuffer> ring_buffer_;
uint8_t *buffer_{nullptr};
uint8_t *data_start_{nullptr};
size_t buffer_size_{0};
size_t buffer_length_{0};
};
class AudioSinkTransferBuffer : public AudioTransferBuffer {
/*
* @brief A class that implements a transfer buffer for audio sinks.
* Supports writing processed data in the transfer buffer to a ring buffer or a speaker component.
*/
public:
/// @brief Creates a new sink transfer buffer.
/// @param buffer_size Size of the transfer buffer in bytes.
/// @return unique_ptr if successfully allocated, nullptr otherwise
static std::unique_ptr<AudioSinkTransferBuffer> create(size_t buffer_size);
/// @brief Writes any available data in the transfer buffer to the sink.
/// @param ticks_to_wait FreeRTOS ticks to block while waiting for the sink to have enough space
/// @return Number of bytes written
size_t transfer_data_to_sink(TickType_t ticks_to_wait);
/// @brief Adds a ring buffer as the transfer buffer's sink.
/// @param ring_buffer weak_ptr to the allocated ring buffer
void set_sink(const std::weak_ptr<RingBuffer> &ring_buffer) { this->ring_buffer_ = ring_buffer.lock(); }
#ifdef USE_SPEAKER
/// @brief Adds a speaker as the transfer buffer's sink.
/// @param speaker Pointer to the speaker component
void set_sink(speaker::Speaker *speaker) { this->speaker_ = speaker; }
#endif
void clear_buffered_data() override;
bool has_buffered_data() const override;
protected:
#ifdef USE_SPEAKER
speaker::Speaker *speaker_{nullptr};
#endif
};
class AudioSourceTransferBuffer : public AudioTransferBuffer {
/*
* @brief A class that implements a transfer buffer for audio sources.
* Supports reading audio data from a ring buffer into the transfer buffer for processing.
*/
public:
/// @brief Creates a new source transfer buffer.
/// @param buffer_size Size of the transfer buffer in bytes.
/// @return unique_ptr if successfully allocated, nullptr otherwise
static std::unique_ptr<AudioSourceTransferBuffer> create(size_t buffer_size);
/// @brief Reads any available data from the sink into the transfer buffer.
/// @param ticks_to_wait FreeRTOS ticks to block while waiting for the source to have enough data
/// @return Number of bytes read
size_t transfer_data_from_source(TickType_t ticks_to_wait);
/// @brief Adds a ring buffer as the transfer buffer's source.
/// @param ring_buffer weak_ptr to the allocated ring buffer
void set_source(const std::weak_ptr<RingBuffer> &ring_buffer) { this->ring_buffer_ = ring_buffer.lock(); };
};
} // namespace audio
} // namespace esphome
#endif

2
esphome/components/ch422g/ch422g.h
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@ -57,6 +57,8 @@ class CH422GGPIOPin : public GPIOPin {
void set_inverted(bool inverted) { inverted_ = inverted; }
void set_flags(gpio::Flags flags);
gpio::Flags get_flags() const override { return this->flags_; }
protected:
CH422GComponent *parent_{};
uint8_t pin_{};

2
esphome/components/esp32/gpio.h
View File

@ -13,6 +13,7 @@ class ESP32InternalGPIOPin : public InternalGPIOPin {
void set_inverted(bool inverted) { inverted_ = inverted; }
void set_drive_strength(gpio_drive_cap_t drive_strength) { drive_strength_ = drive_strength; }
void set_flags(gpio::Flags flags) { flags_ = flags; }
void setup() override;
void pin_mode(gpio::Flags flags) override;
bool digital_read() override;
@ -21,6 +22,7 @@ class ESP32InternalGPIOPin : public InternalGPIOPin {
void detach_interrupt() const override;
ISRInternalGPIOPin to_isr() const override;
uint8_t get_pin() const override { return (uint8_t) pin_; }
gpio::Flags get_flags() const override { return flags_; }
bool is_inverted() const override { return inverted_; }
protected:

1
esphome/components/esp8266/gpio.h
View File

@ -22,6 +22,7 @@ class ESP8266GPIOPin : public InternalGPIOPin {
void detach_interrupt() const override;
ISRInternalGPIOPin to_isr() const override;
uint8_t get_pin() const override { return pin_; }
gpio::Flags get_flags() const override { return flags_; }
bool is_inverted() const override { return inverted_; }
protected:

1
esphome/components/host/gpio.h
View File

@ -21,6 +21,7 @@ class HostGPIOPin : public InternalGPIOPin {
void detach_interrupt() const override;
ISRInternalGPIOPin to_isr() const override;
uint8_t get_pin() const override { return pin_; }
gpio::Flags get_flags() const override { return flags_; }
bool is_inverted() const override { return inverted_; }
protected:

4
esphome/components/i2c/i2c_bus_esp_idf.cpp
View File

@ -39,6 +39,10 @@ void IDFI2CBus::setup() {
conf.scl_io_num = scl_pin_;
conf.scl_pullup_en = scl_pullup_enabled_;
conf.master.clk_speed = frequency_;
#ifdef USE_ESP32_VARIANT_ESP32S2
// workaround for https://github.com/esphome/issues/issues/6718
conf.clk_flags = I2C_SCLK_SRC_FLAG_AWARE_DFS;
#endif
esp_err_t err = i2c_param_config(port_, &conf);
if (err != ESP_OK) {
ESP_LOGW(TAG, "i2c_param_config failed: %s", esp_err_to_name(err));

59
esphome/components/i2s_audio/speaker/__init__.py
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@ -1,13 +1,25 @@
from esphome import pins
import esphome.codegen as cg
from esphome.components import esp32, speaker
from esphome.components import audio, esp32, speaker
import esphome.config_validation as cv
from esphome.const import CONF_CHANNEL, CONF_ID, CONF_MODE, CONF_TIMEOUT
from esphome.const import (
CONF_BITS_PER_SAMPLE,
CONF_BUFFER_DURATION,
CONF_CHANNEL,
CONF_ID,
CONF_MODE,
CONF_NEVER,
CONF_NUM_CHANNELS,
CONF_SAMPLE_RATE,
CONF_TIMEOUT,
)
from .. import (
CONF_I2S_DOUT_PIN,
CONF_I2S_MODE,
CONF_LEFT,
CONF_MONO,
CONF_PRIMARY,
CONF_RIGHT,
CONF_STEREO,
I2SAudioOut,
@ -24,10 +36,8 @@ I2SAudioSpeaker = i2s_audio_ns.class_(
"I2SAudioSpeaker", cg.Component, speaker.Speaker, I2SAudioOut
)
CONF_BUFFER_DURATION = "buffer_duration"
CONF_DAC_TYPE = "dac_type"
CONF_I2S_COMM_FMT = "i2s_comm_fmt"
CONF_NEVER = "never"
i2s_dac_mode_t = cg.global_ns.enum("i2s_dac_mode_t")
INTERNAL_DAC_OPTIONS = {
@ -53,7 +63,41 @@ I2C_COMM_FMT_OPTIONS = {
NO_INTERNAL_DAC_VARIANTS = [esp32.const.VARIANT_ESP32S2]
def validate_esp32_variant(config):
def _set_num_channels_from_config(config):
if config[CONF_CHANNEL] in (CONF_MONO, CONF_LEFT, CONF_RIGHT):
config[CONF_NUM_CHANNELS] = 1
else:
config[CONF_NUM_CHANNELS] = 2
return config
def _set_stream_limits(config):
if config[CONF_I2S_MODE] == CONF_PRIMARY:
# Primary mode has modifiable stream settings
audio.set_stream_limits(
min_bits_per_sample=8,
max_bits_per_sample=32,
min_channels=1,
max_channels=2,
min_sample_rate=16000,
max_sample_rate=48000,
)(config)
else:
# Secondary mode has unmodifiable max bits per sample and min/max sample rates
audio.set_stream_limits(
min_bits_per_sample=8,
max_bits_per_sample=config.get(CONF_BITS_PER_SAMPLE),
min_channels=1,
max_channels=2,
min_sample_rate=config.get(CONF_SAMPLE_RATE),
max_sample_rate=config.get(CONF_SAMPLE_RATE),
)
return config
def _validate_esp32_variant(config):
if config[CONF_DAC_TYPE] != "internal":
return config
variant = esp32.get_esp32_variant()
@ -85,6 +129,7 @@ BASE_SCHEMA = (
.extend(cv.COMPONENT_SCHEMA)
)
CONFIG_SCHEMA = cv.All(
cv.typed_schema(
{
@ -106,7 +151,9 @@ CONFIG_SCHEMA = cv.All(
},
key=CONF_DAC_TYPE,
),
validate_esp32_variant,
_validate_esp32_variant,
_set_num_channels_from_config,
_set_stream_limits,
)

138
esphome/components/i2s_audio/speaker/i2s_audio_speaker.cpp
View File

@ -148,9 +148,11 @@ void I2SAudioSpeaker::loop() {
this->status_set_error("Failed to adjust I2S bus to match the incoming audio");
ESP_LOGE(TAG,
"Incompatible audio format: sample rate = %" PRIu32 ", channels = %" PRIu8 ", bits per sample = %" PRIu8,
this->audio_stream_info_.sample_rate, this->audio_stream_info_.channels,
this->audio_stream_info_.bits_per_sample);
this->audio_stream_info_.get_sample_rate(), this->audio_stream_info_.get_channels(),
this->audio_stream_info_.get_bits_per_sample());
}
xEventGroupClearBits(this->event_group_, ALL_ERR_ESP_BITS);
}
void I2SAudioSpeaker::set_volume(float volume) {
@ -201,6 +203,12 @@ size_t I2SAudioSpeaker::play(const uint8_t *data, size_t length, TickType_t tick
this->start();
}
if ((this->state_ != speaker::STATE_RUNNING) || (this->audio_ring_buffer_.use_count() == 1)) {
// Unable to write data to a running speaker, so delay the max amount of time so it can get ready
vTaskDelay(ticks_to_wait);
ticks_to_wait = 0;
}
size_t bytes_written = 0;
if ((this->state_ == speaker::STATE_RUNNING) && (this->audio_ring_buffer_.use_count() == 1)) {
// Only one owner of the ring buffer (the speaker task), so the ring buffer is allocated and no other components are
@ -223,6 +231,8 @@ bool I2SAudioSpeaker::has_buffered_data() const {
void I2SAudioSpeaker::speaker_task(void *params) {
I2SAudioSpeaker *this_speaker = (I2SAudioSpeaker *) params;
this_speaker->task_created_ = true;
uint32_t event_group_bits =
xEventGroupWaitBits(this_speaker->event_group_,
SpeakerEventGroupBits::COMMAND_START | SpeakerEventGroupBits::COMMAND_STOP |
@ -240,19 +250,20 @@ void I2SAudioSpeaker::speaker_task(void *params) {
audio::AudioStreamInfo audio_stream_info = this_speaker->audio_stream_info_;
const uint32_t bytes_per_ms =
audio_stream_info.channels * audio_stream_info.get_bytes_per_sample() * audio_stream_info.sample_rate / 1000;
const uint32_t dma_buffers_duration_ms = DMA_BUFFER_DURATION_MS * DMA_BUFFERS_COUNT;
// Ensure ring buffer duration is at least the duration of all DMA buffers
const uint32_t ring_buffer_duration = std::max(dma_buffers_duration_ms, this_speaker->buffer_duration_ms_);
const size_t dma_buffers_size = DMA_BUFFERS_COUNT * DMA_BUFFER_DURATION_MS * bytes_per_ms;
// The DMA buffers may have more bits per sample, so calculate buffer sizes based in the input audio stream info
const size_t data_buffer_size = audio_stream_info.ms_to_bytes(dma_buffers_duration_ms);
const size_t ring_buffer_size = audio_stream_info.ms_to_bytes(ring_buffer_duration);
// Ensure ring buffer is at least as large as the total size of the DMA buffers
const size_t ring_buffer_size =
std::max((uint32_t) dma_buffers_size, this_speaker->buffer_duration_ms_ * bytes_per_ms);
const size_t single_dma_buffer_input_size = data_buffer_size / DMA_BUFFERS_COUNT;
if (this_speaker->send_esp_err_to_event_group_(this_speaker->allocate_buffers_(dma_buffers_size, ring_buffer_size))) {
if (this_speaker->send_esp_err_to_event_group_(this_speaker->allocate_buffers_(data_buffer_size, ring_buffer_size))) {
// Failed to allocate buffers
xEventGroupSetBits(this_speaker->event_group_, SpeakerEventGroupBits::ERR_ESP_NO_MEM);
this_speaker->delete_task_(dma_buffers_size);
this_speaker->delete_task_(data_buffer_size);
}
if (!this_speaker->send_esp_err_to_event_group_(this_speaker->start_i2s_driver_(audio_stream_info))) {
@ -262,20 +273,25 @@ void I2SAudioSpeaker::speaker_task(void *params) {
uint32_t last_data_received_time = millis();
bool tx_dma_underflow = false;
while (!this_speaker->timeout_.has_value() ||
this_speaker->accumulated_frames_written_ = 0;
// Keep looping if paused, there is no timeout configured, or data was received more recently than the configured
// timeout
while (this_speaker->pause_state_ || !this_speaker->timeout_.has_value() ||
(millis() - last_data_received_time) <= this_speaker->timeout_.value()) {
event_group_bits = xEventGroupGetBits(this_speaker->event_group_);
if (event_group_bits & SpeakerEventGroupBits::COMMAND_STOP) {
xEventGroupClearBits(this_speaker->event_group_, SpeakerEventGroupBits::COMMAND_STOP);
break;
}
if (event_group_bits & SpeakerEventGroupBits::COMMAND_STOP_GRACEFULLY) {
xEventGroupClearBits(this_speaker->event_group_, SpeakerEventGroupBits::COMMAND_STOP_GRACEFULLY);
stop_gracefully = true;
}
if (this_speaker->audio_stream_info_ != audio_stream_info) {
// Audio stream info has changed, stop the speaker task so it will restart with the proper settings.
// Audio stream info changed, stop the speaker task so it will restart with the proper settings.
break;
}
@ -286,33 +302,64 @@ void I2SAudioSpeaker::speaker_task(void *params) {
}
}
size_t bytes_to_read = dma_buffers_size;
size_t bytes_read = this_speaker->audio_ring_buffer_->read((void *) this_speaker->data_buffer_, bytes_to_read,
if (this_speaker->pause_state_) {
// Pause state is accessed atomically, so thread safe
// Delay so the task can yields, then skip transferring audio data
delay(TASK_DELAY_MS);
continue;
}
size_t bytes_read = this_speaker->audio_ring_buffer_->read((void *) this_speaker->data_buffer_, data_buffer_size,
pdMS_TO_TICKS(TASK_DELAY_MS));
if (bytes_read > 0) {
size_t bytes_written = 0;
if ((audio_stream_info.bits_per_sample == 16) && (this_speaker->q15_volume_factor_ < INT16_MAX)) {
if ((audio_stream_info.get_bits_per_sample() == 16) && (this_speaker->q15_volume_factor_ < INT16_MAX)) {
// Scale samples by the volume factor in place
q15_multiplication((int16_t *) this_speaker->data_buffer_, (int16_t *) this_speaker->data_buffer_,
bytes_read / sizeof(int16_t), this_speaker->q15_volume_factor_);
}
if (audio_stream_info.bits_per_sample == (uint8_t) this_speaker->bits_per_sample_) {
i2s_write(this_speaker->parent_->get_port(), this_speaker->data_buffer_, bytes_read, &bytes_written,
portMAX_DELAY);
} else if (audio_stream_info.bits_per_sample < (uint8_t) this_speaker->bits_per_sample_) {
i2s_write_expand(this_speaker->parent_->get_port(), this_speaker->data_buffer_, bytes_read,
audio_stream_info.bits_per_sample, this_speaker->bits_per_sample_, &bytes_written,
portMAX_DELAY);
}
// Write the audio data to a single DMA buffer at a time to reduce latency for the audio duration played
// callback.
const uint32_t batches = (bytes_read + single_dma_buffer_input_size - 1) / single_dma_buffer_input_size;
if (bytes_written != bytes_read) {
xEventGroupSetBits(this_speaker->event_group_, SpeakerEventGroupBits::ERR_ESP_INVALID_SIZE);
for (uint32_t i = 0; i < batches; ++i) {
size_t bytes_written = 0;
size_t bytes_to_write = std::min(single_dma_buffer_input_size, bytes_read);
if (audio_stream_info.get_bits_per_sample() == (uint8_t) this_speaker->bits_per_sample_) {
i2s_write(this_speaker->parent_->get_port(), this_speaker->data_buffer_ + i * single_dma_buffer_input_size,
bytes_to_write, &bytes_written, pdMS_TO_TICKS(DMA_BUFFER_DURATION_MS * 5));
} else if (audio_stream_info.get_bits_per_sample() < (uint8_t) this_speaker->bits_per_sample_) {
i2s_write_expand(this_speaker->parent_->get_port(),
this_speaker->data_buffer_ + i * single_dma_buffer_input_size, bytes_to_write,
audio_stream_info.get_bits_per_sample(), this_speaker->bits_per_sample_, &bytes_written,
pdMS_TO_TICKS(DMA_BUFFER_DURATION_MS * 5));
}
uint32_t write_timestamp = micros();
if (bytes_written != bytes_to_write) {
xEventGroupSetBits(this_speaker->event_group_, SpeakerEventGroupBits::ERR_ESP_INVALID_SIZE);
}
bytes_read -= bytes_written;
this_speaker->accumulated_frames_written_ += audio_stream_info.bytes_to_frames(bytes_written);
const uint32_t new_playback_ms =
audio_stream_info.frames_to_milliseconds_with_remainder(&this_speaker->accumulated_frames_written_);
const uint32_t remainder_us =
audio_stream_info.frames_to_microseconds(this_speaker->accumulated_frames_written_);
uint32_t pending_frames =
audio_stream_info.bytes_to_frames(bytes_read + this_speaker->audio_ring_buffer_->available());
const uint32_t pending_ms = audio_stream_info.frames_to_milliseconds_with_remainder(&pending_frames);
this_speaker->audio_output_callback_(new_playback_ms, remainder_us, pending_ms, write_timestamp);
tx_dma_underflow = false;
last_data_received_time = millis();
}
tx_dma_underflow = false;
last_data_received_time = millis();
} else {
// No data received
if (stop_gracefully && tx_dma_underflow) {
@ -328,7 +375,7 @@ void I2SAudioSpeaker::speaker_task(void *params) {
this_speaker->parent_->unlock();
}
this_speaker->delete_task_(dma_buffers_size);
this_speaker->delete_task_(data_buffer_size);
}
void I2SAudioSpeaker::start() {
@ -337,16 +384,15 @@ void I2SAudioSpeaker::start() {
if ((this->state_ == speaker::STATE_STARTING) || (this->state_ == speaker::STATE_RUNNING))
return;
if (this->speaker_task_handle_ == nullptr) {
if (!this->task_created_ && (this->speaker_task_handle_ == nullptr)) {
xTaskCreate(I2SAudioSpeaker::speaker_task, "speaker_task", TASK_STACK_SIZE, (void *) this, TASK_PRIORITY,
&this->speaker_task_handle_);
}
if (this->speaker_task_handle_ != nullptr) {
xEventGroupSetBits(this->event_group_, SpeakerEventGroupBits::COMMAND_START);
this->task_created_ = true;
} else {
xEventGroupSetBits(this->event_group_, SpeakerEventGroupBits::ERR_TASK_FAILED_TO_START);
if (this->speaker_task_handle_ != nullptr) {
xEventGroupSetBits(this->event_group_, SpeakerEventGroupBits::COMMAND_START);
} else {
xEventGroupSetBits(this->event_group_, SpeakerEventGroupBits::ERR_TASK_FAILED_TO_START);
}
}
}
@ -416,12 +462,12 @@ esp_err_t I2SAudioSpeaker::allocate_buffers_(size_t data_buffer_size, size_t rin
}
esp_err_t I2SAudioSpeaker::start_i2s_driver_(audio::AudioStreamInfo &audio_stream_info) {
if ((this->i2s_mode_ & I2S_MODE_SLAVE) && (this->sample_rate_ != audio_stream_info.sample_rate)) { // NOLINT
// Can't reconfigure I2S bus, so the sample rate must match the configured value
if ((this->i2s_mode_ & I2S_MODE_SLAVE) && (this->sample_rate_ != audio_stream_info.get_sample_rate())) { // NOLINT
// Can't reconfigure I2S bus, so the sample rate must match the configured value
return ESP_ERR_NOT_SUPPORTED;
}
if ((i2s_bits_per_sample_t) audio_stream_info.bits_per_sample > this->bits_per_sample_) {
if ((i2s_bits_per_sample_t) audio_stream_info.get_bits_per_sample() > this->bits_per_sample_) {
// Currently can't handle the case when the incoming audio has more bits per sample than the configured value
return ESP_ERR_NOT_SUPPORTED;
}
@ -432,21 +478,21 @@ esp_err_t I2SAudioSpeaker::start_i2s_driver_(audio::AudioStreamInfo &audio_strea
i2s_channel_fmt_t channel = this->channel_;
if (audio_stream_info.channels == 1) {
if (audio_stream_info.get_channels() == 1) {
if (this->channel_ == I2S_CHANNEL_FMT_ONLY_LEFT) {
channel = I2S_CHANNEL_FMT_ONLY_LEFT;
} else {
channel = I2S_CHANNEL_FMT_ONLY_RIGHT;
}
} else if (audio_stream_info.channels == 2) {
} else if (audio_stream_info.get_channels() == 2) {
channel = I2S_CHANNEL_FMT_RIGHT_LEFT;
}
int dma_buffer_length = DMA_BUFFER_DURATION_MS * this->sample_rate_ / 1000;
int dma_buffer_length = audio_stream_info.ms_to_frames(DMA_BUFFER_DURATION_MS);
i2s_driver_config_t config = {
.mode = (i2s_mode_t) (this->i2s_mode_ | I2S_MODE_TX),
.sample_rate = audio_stream_info.sample_rate,
.sample_rate = audio_stream_info.get_sample_rate(),
.bits_per_sample = this->bits_per_sample_,
.channel_format = channel,
.communication_format = this->i2s_comm_fmt_,
@ -504,7 +550,7 @@ esp_err_t I2SAudioSpeaker::start_i2s_driver_(audio::AudioStreamInfo &audio_strea
}
void I2SAudioSpeaker::delete_task_(size_t buffer_size) {
this->audio_ring_buffer_.reset(); // Releases onwership of the shared_ptr
this->audio_ring_buffer_.reset(); // Releases ownership of the shared_ptr
if (this->data_buffer_ != nullptr) {
ExternalRAMAllocator<uint8_t> allocator(ExternalRAMAllocator<uint8_t>::ALLOW_FAILURE);

8
esphome/components/i2s_audio/speaker/i2s_audio_speaker.h
View File

@ -40,6 +40,9 @@ class I2SAudioSpeaker : public I2SAudioOut, public speaker::Speaker, public Comp
void stop() override;
void finish() override;
void set_pause_state(bool pause_state) override { this->pause_state_ = pause_state; }
bool get_pause_state() const override { return this->pause_state_; }
/// @brief Plays the provided audio data.
/// Starts the speaker task, if necessary. Writes the audio data to the ring buffer.
/// @param data Audio data in the format set by the parent speaker classes ``set_audio_stream_info`` method.
@ -121,13 +124,18 @@ class I2SAudioSpeaker : public I2SAudioOut, public speaker::Speaker, public Comp
uint8_t dout_pin_;
bool task_created_{false};
bool pause_state_{false};
int16_t q15_volume_factor_{INT16_MAX};
size_t bytes_written_{0};
#if SOC_I2S_SUPPORTS_DAC
i2s_dac_mode_t internal_dac_mode_{I2S_DAC_CHANNEL_DISABLE};
#endif
i2s_comm_format_t i2s_comm_fmt_;
uint32_t accumulated_frames_written_{0};
};
} // namespace i2s_audio

1
esphome/components/libretiny/gpio_arduino.h
View File

@ -20,6 +20,7 @@ class ArduinoInternalGPIOPin : public InternalGPIOPin {
void detach_interrupt() const override;
ISRInternalGPIOPin to_isr() const override;
uint8_t get_pin() const override { return pin_; }
gpio::Flags get_flags() const override { return flags_; }
bool is_inverted() const override { return inverted_; }
protected:

13
esphome/components/lvgl/__init__.py
View File

@ -61,7 +61,14 @@ from .types import (
lv_style_t,
lvgl_ns,
)
from .widgets import Widget, add_widgets, get_scr_act, set_obj_properties, styles_used
from .widgets import (
LvScrActType,
Widget,
add_widgets,
get_scr_act,
set_obj_properties,
styles_used,
)
from .widgets.animimg import animimg_spec
from .widgets.arc import arc_spec
from .widgets.button import button_spec
@ -318,7 +325,7 @@ async def to_code(configs):
config[df.CONF_RESUME_ON_INPUT],
)
await cg.register_component(lv_component, config)
Widget.create(config[CONF_ID], lv_component, obj_spec, config)
Widget.create(config[CONF_ID], lv_component, LvScrActType(), config)
lv_scr_act = get_scr_act(lv_component)
async with LvContext():
@ -391,7 +398,7 @@ FINAL_VALIDATE_SCHEMA = final_validation
LVGL_SCHEMA = (
cv.polling_component_schema("1s")
.extend(obj_schema(obj_spec))
.extend(obj_schema(LvScrActType()))
.extend(
{
cv.GenerateID(CONF_ID): cv.declare_id(LvglComponent),

7
esphome/components/lvgl/defines.py
View File

@ -146,6 +146,8 @@ TYPE_FLEX = "flex"
TYPE_GRID = "grid"
TYPE_NONE = "none"
DIRECTIONS = LvConstant("LV_DIR_", "LEFT", "RIGHT", "BOTTOM", "TOP")
LV_FONTS = list(f"montserrat_{s}" for s in range(8, 50, 2)) + [
"dejavu_16_persian_hebrew",
"simsun_16_cjk",
@ -169,9 +171,13 @@ LV_EVENT_MAP = {
"CANCEL": "CANCEL",
"ALL_EVENTS": "ALL",
"CHANGE": "VALUE_CHANGED",
"GESTURE": "GESTURE",
}
LV_EVENT_TRIGGERS = tuple(f"on_{x.lower()}" for x in LV_EVENT_MAP)
SWIPE_TRIGGERS = tuple(
f"on_swipe_{x.lower()}" for x in DIRECTIONS.choices + ("up", "down")
)
LV_ANIM = LvConstant(
@ -250,7 +256,6 @@ KEYBOARD_MODES = LvConstant(
"NUMBER",
)
ROLLER_MODES = LvConstant("LV_ROLLER_MODE_", "NORMAL", "INFINITE")
DIRECTIONS = LvConstant("LV_DIR_", "LEFT", "RIGHT", "BOTTOM", "TOP")
TILE_DIRECTIONS = DIRECTIONS.extend("HOR", "VER", "ALL")
CHILD_ALIGNMENTS = LvConstant(
"LV_ALIGN_",

5
esphome/components/lvgl/schemas.py
View File

@ -211,10 +211,9 @@ def part_schema(parts):
def automation_schema(typ: LvType):
events = df.LV_EVENT_TRIGGERS + df.SWIPE_TRIGGERS
if typ.has_on_value:
events = df.LV_EVENT_TRIGGERS + (CONF_ON_VALUE,)
else:
events = df.LV_EVENT_TRIGGERS
events = events + (CONF_ON_VALUE,)
args = typ.get_arg_type() if isinstance(typ, LvType) else []
args.append(lv_event_t_ptr)
return {

30
esphome/components/lvgl/trigger.py
View File

@ -7,8 +7,10 @@ from .defines import (
CONF_ALIGN_TO,
CONF_X,
CONF_Y,
DIRECTIONS,
LV_EVENT_MAP,
LV_EVENT_TRIGGERS,
SWIPE_TRIGGERS,
literal,
)
from .lvcode import (
@ -23,7 +25,7 @@ from .lvcode import (
lvgl_static,
)
from .types import LV_EVENT
from .widgets import widget_map
from .widgets import LvScrActType, get_scr_act, widget_map
async def generate_triggers():
@ -33,6 +35,9 @@ async def generate_triggers():
"""
for w in widget_map.values():
if isinstance(w.type, LvScrActType):
w = get_scr_act(w.var)
if w.config:
for event, conf in {
event: conf
@ -43,6 +48,24 @@ async def generate_triggers():
w.add_flag("LV_OBJ_FLAG_CLICKABLE")
event = literal("LV_EVENT_" + LV_EVENT_MAP[event[3:].upper()])
await add_trigger(conf, w, event)
for event, conf in {
event: conf
for event, conf in w.config.items()
if event in SWIPE_TRIGGERS
}.items():
conf = conf[0]
dir = event[9:].upper()
dir = {"UP": "TOP", "DOWN": "BOTTOM"}.get(dir, dir)
dir = DIRECTIONS.mapper(dir)
w.clear_flag("LV_OBJ_FLAG_SCROLLABLE")
selected = literal(
f"lv_indev_get_gesture_dir(lv_indev_get_act()) == {dir}"
)
await add_trigger(
conf, w, literal("LV_EVENT_GESTURE"), is_selected=selected
)
for conf in w.config.get(CONF_ON_VALUE, ()):
await add_trigger(
conf,
@ -61,13 +84,14 @@ async def generate_triggers():
lv.obj_align_to(w.obj, target, align, x, y)
async def add_trigger(conf, w, *events):
async def add_trigger(conf, w, *events, is_selected=None):
is_selected = is_selected or w.is_selected()
tid = conf[CONF_TRIGGER_ID]
trigger = cg.new_Pvariable(tid)
args = w.get_args() + [(lv_event_t_ptr, "event")]
value = w.get_values()
await automation.build_automation(trigger, args, conf)
async with LambdaContext(EVENT_ARG, where=tid) as context:
with LvConditional(w.is_selected()):
with LvConditional(is_selected):
lv_add(trigger.trigger(*value, literal("event")))
lv_add(lvgl_static.add_event_cb(w.obj, await context.get_lambda(), *events))

2
esphome/components/max6956/max6956.h
View File

@ -83,6 +83,8 @@ class MAX6956GPIOPin : public GPIOPin {
void set_inverted(bool inverted) { inverted_ = inverted; }
void set_flags(gpio::Flags flags) { flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
protected:
MAX6956 *parent_;
uint8_t pin_;

2
esphome/components/mcp23016/mcp23016.h
View File

@ -61,6 +61,8 @@ class MCP23016GPIOPin : public GPIOPin {
void set_inverted(bool inverted) { inverted_ = inverted; }
void set_flags(gpio::Flags flags) { flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
protected:
MCP23016 *parent_;
uint8_t pin_;

2
esphome/components/mcp23xxx_base/mcp23xxx_base.h
View File

@ -43,6 +43,8 @@ class MCP23XXXGPIOPin : public GPIOPin {
void set_flags(gpio::Flags flags) { flags_ = flags; }
void set_interrupt_mode(MCP23XXXInterruptMode interrupt_mode) { interrupt_mode_ = interrupt_mode; }
gpio::Flags get_flags() const override { return this->flags_; }
protected:
MCP23XXXBase *parent_;
uint8_t pin_;

0
esphome/components/mixer/__init__.py Normal file
View File

172
esphome/components/mixer/speaker/__init__.py Normal file
View File

@ -0,0 +1,172 @@
from esphome import automation
import esphome.codegen as cg
from esphome.components import audio, esp32, speaker
import esphome.config_validation as cv
from esphome.const import (
CONF_BITS_PER_SAMPLE,
CONF_BUFFER_DURATION,
CONF_DURATION,
CONF_ID,
CONF_NEVER,
CONF_NUM_CHANNELS,
CONF_OUTPUT_SPEAKER,
CONF_SAMPLE_RATE,
CONF_TASK_STACK_IN_PSRAM,
CONF_TIMEOUT,
PLATFORM_ESP32,
)
from esphome.core.entity_helpers import inherit_property_from
import esphome.final_validate as fv
AUTO_LOAD = ["audio"]
CODEOWNERS = ["@kahrendt"]
mixer_speaker_ns = cg.esphome_ns.namespace("mixer_speaker")
MixerSpeaker = mixer_speaker_ns.class_("MixerSpeaker", cg.Component)
SourceSpeaker = mixer_speaker_ns.class_("SourceSpeaker", cg.Component, speaker.Speaker)
CONF_DECIBEL_REDUCTION = "decibel_reduction"
CONF_QUEUE_MODE = "queue_mode"
CONF_SOURCE_SPEAKERS = "source_speakers"
DuckingApplyAction = mixer_speaker_ns.class_(
"DuckingApplyAction", automation.Action, cg.Parented.template(SourceSpeaker)
)
SOURCE_SPEAKER_SCHEMA = speaker.SPEAKER_SCHEMA.extend(
{
cv.GenerateID(): cv.declare_id(SourceSpeaker),
cv.Optional(
CONF_BUFFER_DURATION, default="100ms"
): cv.positive_time_period_milliseconds,
cv.Optional(CONF_TIMEOUT, default="500ms"): cv.Any(
cv.positive_time_period_milliseconds,
cv.one_of(CONF_NEVER, lower=True),
),
cv.Optional(CONF_BITS_PER_SAMPLE, default=16): cv.int_range(16, 16),
}
)
def _set_stream_limits(config):
audio.set_stream_limits(
min_bits_per_sample=16,
max_bits_per_sample=16,
)(config)
return config
def _validate_source_speaker(config):
fconf = fv.full_config.get()
# Get ID for the output speaker and add it to the source speakrs config to easily inherit properties
path = fconf.get_path_for_id(config[CONF_ID])[:-3]
path.append(CONF_OUTPUT_SPEAKER)
output_speaker_id = fconf.get_config_for_path(path)
config[CONF_OUTPUT_SPEAKER] = output_speaker_id
inherit_property_from(CONF_NUM_CHANNELS, CONF_OUTPUT_SPEAKER)(config)
inherit_property_from(CONF_SAMPLE_RATE, CONF_OUTPUT_SPEAKER)(config)
audio.final_validate_audio_schema(
"mixer",
audio_device=CONF_OUTPUT_SPEAKER,
bits_per_sample=config.get(CONF_BITS_PER_SAMPLE),
channels=config.get(CONF_NUM_CHANNELS),
sample_rate=config.get(CONF_SAMPLE_RATE),
)(config)
return config
CONFIG_SCHEMA = cv.All(
cv.Schema(
{
cv.GenerateID(): cv.declare_id(MixerSpeaker),
cv.Required(CONF_OUTPUT_SPEAKER): cv.use_id(speaker.Speaker),
cv.Required(CONF_SOURCE_SPEAKERS): cv.All(
cv.ensure_list(SOURCE_SPEAKER_SCHEMA),
cv.Length(min=2, max=8),
[_set_stream_limits],
),
cv.Optional(CONF_NUM_CHANNELS): cv.int_range(min=1, max=2),
cv.Optional(CONF_QUEUE_MODE, default=False): cv.boolean,
cv.SplitDefault(CONF_TASK_STACK_IN_PSRAM, esp32_idf=False): cv.All(
cv.boolean, cv.only_with_esp_idf
),
}
),
cv.only_on([PLATFORM_ESP32]),
)
FINAL_VALIDATE_SCHEMA = cv.All(
cv.Schema(
{
cv.Optional(CONF_SOURCE_SPEAKERS): [_validate_source_speaker],
},
extra=cv.ALLOW_EXTRA,
),
inherit_property_from(CONF_NUM_CHANNELS, CONF_OUTPUT_SPEAKER),
)
async def to_code(config):
var = cg.new_Pvariable(config[CONF_ID])
await cg.register_component(var, config)
spkr = await cg.get_variable(config[CONF_OUTPUT_SPEAKER])
cg.add(var.set_output_channels(config[CONF_NUM_CHANNELS]))
cg.add(var.set_output_speaker(spkr))
cg.add(var.set_queue_mode(config[CONF_QUEUE_MODE]))
if task_stack_in_psram := config.get(CONF_TASK_STACK_IN_PSRAM):
cg.add(var.set_task_stack_in_psram(task_stack_in_psram))
if task_stack_in_psram:
if config[CONF_TASK_STACK_IN_PSRAM]:
esp32.add_idf_sdkconfig_option(
"CONFIG_SPIRAM_ALLOW_STACK_EXTERNAL_MEMORY", True
)
for speaker_config in config[CONF_SOURCE_SPEAKERS]:
source_speaker = cg.new_Pvariable(speaker_config[CONF_ID])
cg.add(source_speaker.set_buffer_duration(speaker_config[CONF_BUFFER_DURATION]))
if speaker_config[CONF_TIMEOUT] != CONF_NEVER:
cg.add(source_speaker.set_timeout(speaker_config[CONF_TIMEOUT]))
await cg.register_component(source_speaker, speaker_config)
await cg.register_parented(source_speaker, config[CONF_ID])
await speaker.register_speaker(source_speaker, speaker_config)
cg.add(var.add_source_speaker(source_speaker))
@automation.register_action(
"mixer_speaker.apply_ducking",
DuckingApplyAction,
cv.Schema(
{
cv.GenerateID(): cv.use_id(SourceSpeaker),
cv.Required(CONF_DECIBEL_REDUCTION): cv.templatable(
cv.int_range(min=0, max=51)
),
cv.Optional(CONF_DURATION, default="0.0s"): cv.templatable(
cv.positive_time_period_milliseconds
),
}
),
)
async def ducking_set_to_code(config, action_id, template_arg, args):
var = cg.new_Pvariable(action_id, template_arg)
await cg.register_parented(var, config[CONF_ID])
decibel_reduction = await cg.templatable(
config[CONF_DECIBEL_REDUCTION], args, cg.uint8
)
cg.add(var.set_decibel_reduction(decibel_reduction))
duration = await cg.templatable(config[CONF_DURATION], args, cg.uint32)
cg.add(var.set_duration(duration))
return var

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esphome/components/mixer/speaker/automation.h Normal file
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#pragma once
#include "mixer_speaker.h"
#ifdef USE_ESP32
namespace esphome {
namespace mixer_speaker {
template<typename... Ts> class DuckingApplyAction : public Action<Ts...>, public Parented<SourceSpeaker> {
TEMPLATABLE_VALUE(uint8_t, decibel_reduction)
TEMPLATABLE_VALUE(uint32_t, duration)
void play(Ts... x) override {
this->parent_->apply_ducking(this->decibel_reduction_.value(x...), this->duration_.value(x...));
}
};
} // namespace mixer_speaker
} // namespace esphome
#endif

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esphome/components/mixer/speaker/mixer_speaker.cpp Normal file
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#include "mixer_speaker.h"
#ifdef USE_ESP32
#include "esphome/core/hal.h"
#include "esphome/core/helpers.h"
#include "esphome/core/log.h"
#include <algorithm>
#include <cstring>
namespace esphome {
namespace mixer_speaker {
static const UBaseType_t MIXER_TASK_PRIORITY = 10;
static const uint32_t TRANSFER_BUFFER_DURATION_MS = 50;
static const uint32_t TASK_DELAY_MS = 25;
static const size_t TASK_STACK_SIZE = 4096;
static const int16_t MAX_AUDIO_SAMPLE_VALUE = INT16_MAX;
static const int16_t MIN_AUDIO_SAMPLE_VALUE = INT16_MIN;
static const char *const TAG = "speaker_mixer";
// Gives the Q15 fixed point scaling factor to reduce by 0 dB, 1dB, ..., 50 dB
// dB to PCM scaling factor formula: floating_point_scale_factor = 2^(-db/6.014)
// float to Q15 fixed point formula: q15_scale_factor = floating_point_scale_factor * 2^(15)
static const std::vector<int16_t> DECIBEL_REDUCTION_TABLE = {
32767, 29201, 26022, 23189, 20665, 18415, 16410, 14624, 13032, 11613, 10349, 9222, 8218, 7324, 6527, 5816, 5183,
4619, 4116, 3668, 3269, 2913, 2596, 2313, 2061, 1837, 1637, 1459, 1300, 1158, 1032, 920, 820, 731,
651, 580, 517, 461, 411, 366, 326, 291, 259, 231, 206, 183, 163, 146, 130, 116, 103};
enum MixerEventGroupBits : uint32_t {
COMMAND_STOP = (1 << 0), // stops the mixer task
STATE_STARTING = (1 << 10),
STATE_RUNNING = (1 << 11),
STATE_STOPPING = (1 << 12),
STATE_STOPPED = (1 << 13),
ERR_ESP_NO_MEM = (1 << 19),
ALL_BITS = 0x00FFFFFF, // All valid FreeRTOS event group bits
};
void SourceSpeaker::dump_config() {
ESP_LOGCONFIG(TAG, "Mixer Source Speaker");
ESP_LOGCONFIG(TAG, " Buffer Duration: %" PRIu32 " ms", this->buffer_duration_ms_);
if (this->timeout_ms_.has_value()) {
ESP_LOGCONFIG(TAG, " Timeout: %" PRIu32 " ms", this->timeout_ms_.value());
} else {
ESP_LOGCONFIG(TAG, " Timeout: never");
}
}
void SourceSpeaker::setup() {
this->parent_->get_output_speaker()->add_audio_output_callback(
[this](uint32_t new_playback_ms, uint32_t remainder_us, uint32_t pending_ms, uint32_t write_timestamp) {
uint32_t personal_playback_ms = std::min(new_playback_ms, this->pending_playback_ms_);
if (personal_playback_ms > 0) {
this->pending_playback_ms_ -= personal_playback_ms;
this->audio_output_callback_(personal_playback_ms, remainder_us, this->pending_playback_ms_, write_timestamp);
}
});
}
void SourceSpeaker::loop() {
switch (this->state_) {
case speaker::STATE_STARTING: {
esp_err_t err = this->start_();
if (err == ESP_OK) {
this->state_ = speaker::STATE_RUNNING;
this->stop_gracefully_ = false;
this->last_seen_data_ms_ = millis();
this->status_clear_error();
} else {
switch (err) {
case ESP_ERR_NO_MEM:
this->status_set_error("Failed to start mixer: not enough memory");
break;
case ESP_ERR_NOT_SUPPORTED:
this->status_set_error("Failed to start mixer: unsupported bits per sample");
break;
case ESP_ERR_INVALID_ARG:
this->status_set_error("Failed to start mixer: audio stream isn't compatible with the other audio stream.");
break;
case ESP_ERR_INVALID_STATE:
this->status_set_error("Failed to start mixer: mixer task failed to start");
break;
default:
this->status_set_error("Failed to start mixer");
break;
}
this->state_ = speaker::STATE_STOPPING;
}
break;
}
case speaker::STATE_RUNNING:
if (!this->transfer_buffer_->has_buffered_data()) {
if ((this->timeout_ms_.has_value() && ((millis() - this->last_seen_data_ms_) > this->timeout_ms_.value())) ||
this->stop_gracefully_) {
this->state_ = speaker::STATE_STOPPING;
}
}
break;
case speaker::STATE_STOPPING:
this->stop_();
this->stop_gracefully_ = false;
this->state_ = speaker::STATE_STOPPED;
break;
case speaker::STATE_STOPPED:
break;
}
}
size_t SourceSpeaker::play(const uint8_t *data, size_t length, TickType_t ticks_to_wait) {
if (this->is_stopped()) {
this->start();
}
size_t bytes_written = 0;
if (this->ring_buffer_.use_count() == 1) {
std::shared_ptr<RingBuffer> temp_ring_buffer = this->ring_buffer_.lock();
bytes_written = temp_ring_buffer->write_without_replacement(data, length, ticks_to_wait);
if (bytes_written > 0) {
this->last_seen_data_ms_ = millis();
}
}
return bytes_written;
}
void SourceSpeaker::start() { this->state_ = speaker::STATE_STARTING; }
esp_err_t SourceSpeaker::start_() {
const size_t ring_buffer_size = this->audio_stream_info_.ms_to_bytes(this->buffer_duration_ms_);
if (this->transfer_buffer_.use_count() == 0) {
this->transfer_buffer_ =
audio::AudioSourceTransferBuffer::create(this->audio_stream_info_.ms_to_bytes(TRANSFER_BUFFER_DURATION_MS));
if (this->transfer_buffer_ == nullptr) {
return ESP_ERR_NO_MEM;
}
std::shared_ptr<RingBuffer> temp_ring_buffer;
if (!this->ring_buffer_.use_count()) {
temp_ring_buffer = RingBuffer::create(ring_buffer_size);
this->ring_buffer_ = temp_ring_buffer;
}
if (!this->ring_buffer_.use_count()) {
return ESP_ERR_NO_MEM;
} else {
this->transfer_buffer_->set_source(temp_ring_buffer);
}
}
return this->parent_->start(this->audio_stream_info_);
}
void SourceSpeaker::stop() {
if (this->state_ != speaker::STATE_STOPPED) {
this->state_ = speaker::STATE_STOPPING;
}
}
void SourceSpeaker::stop_() {
this->transfer_buffer_.reset(); // deallocates the transfer buffer
}
void SourceSpeaker::finish() { this->stop_gracefully_ = true; }
bool SourceSpeaker::has_buffered_data() const {
return ((this->transfer_buffer_.use_count() > 0) && this->transfer_buffer_->has_buffered_data());
}
void SourceSpeaker::set_mute_state(bool mute_state) {
this->mute_state_ = mute_state;
this->parent_->get_output_speaker()->set_mute_state(mute_state);
}
void SourceSpeaker::set_volume(float volume) {
this->volume_ = volume;
this->parent_->get_output_speaker()->set_volume(volume);
}
size_t SourceSpeaker::process_data_from_source(TickType_t ticks_to_wait) {
if (!this->transfer_buffer_.use_count()) {
return 0;
}
// Store current offset, as these samples are already ducked
const size_t current_length = this->transfer_buffer_->available();
size_t bytes_read = this->transfer_buffer_->transfer_data_from_source(ticks_to_wait);
uint32_t samples_to_duck = this->audio_stream_info_.bytes_to_samples(bytes_read);
if (samples_to_duck > 0) {
int16_t *current_buffer = reinterpret_cast<int16_t *>(this->transfer_buffer_->get_buffer_start() + current_length);
duck_samples(current_buffer, samples_to_duck, &this->current_ducking_db_reduction_,
&this->ducking_transition_samples_remaining_, this->samples_per_ducking_step_,
this->db_change_per_ducking_step_);
}
return bytes_read;
}
void SourceSpeaker::apply_ducking(uint8_t decibel_reduction, uint32_t duration) {
if (this->target_ducking_db_reduction_ != decibel_reduction) {
this->current_ducking_db_reduction_ = this->target_ducking_db_reduction_;
this->target_ducking_db_reduction_ = decibel_reduction;
uint8_t total_ducking_steps = 0;
if (this->target_ducking_db_reduction_ > this->current_ducking_db_reduction_) {
// The dB reduction level is increasing (which results in quieter audio)
total_ducking_steps = this->target_ducking_db_reduction_ - this->current_ducking_db_reduction_ - 1;
this->db_change_per_ducking_step_ = 1;
} else {
// The dB reduction level is decreasing (which results in louder audio)
total_ducking_steps = this->current_ducking_db_reduction_ - this->target_ducking_db_reduction_ - 1;
this->db_change_per_ducking_step_ = -1;
}
if ((duration > 0) && (total_ducking_steps > 0)) {
this->ducking_transition_samples_remaining_ = this->audio_stream_info_.ms_to_samples(duration);
this->samples_per_ducking_step_ = this->ducking_transition_samples_remaining_ / total_ducking_steps;
this->ducking_transition_samples_remaining_ =
this->samples_per_ducking_step_ * total_ducking_steps; // Adjust for integer division rounding
this->current_ducking_db_reduction_ += this->db_change_per_ducking_step_;
} else {
this->ducking_transition_samples_remaining_ = 0;
this->current_ducking_db_reduction_ = this->target_ducking_db_reduction_;
}
}
}
void SourceSpeaker::duck_samples(int16_t *input_buffer, uint32_t input_samples_to_duck,
int8_t *current_ducking_db_reduction, uint32_t *ducking_transition_samples_remaining,
uint32_t samples_per_ducking_step, int8_t db_change_per_ducking_step) {
if (*ducking_transition_samples_remaining > 0) {
// Ducking level is still transitioning
// Takes the ceiling of input_samples_to_duck/samples_per_ducking_step
uint32_t ducking_steps_in_batch =
input_samples_to_duck / samples_per_ducking_step + (input_samples_to_duck % samples_per_ducking_step != 0);
for (uint32_t i = 0; i < ducking_steps_in_batch; ++i) {
uint32_t samples_left_in_step = *ducking_transition_samples_remaining % samples_per_ducking_step;
if (samples_left_in_step == 0) {
samples_left_in_step = samples_per_ducking_step;
}
uint32_t samples_to_duck = std::min(input_samples_to_duck, samples_left_in_step);
samples_to_duck = std::min(samples_to_duck, *ducking_transition_samples_remaining);
// Ensure we only point to valid index in the Q15 scaling factor table
uint8_t safe_db_reduction_index =
clamp<uint8_t>(*current_ducking_db_reduction, 0, DECIBEL_REDUCTION_TABLE.size() - 1);
int16_t q15_scale_factor = DECIBEL_REDUCTION_TABLE[safe_db_reduction_index];
audio::scale_audio_samples(input_buffer, input_buffer, q15_scale_factor, samples_to_duck);
if (samples_left_in_step - samples_to_duck == 0) {
// After scaling the current samples, we are ready to transition to the next step
*current_ducking_db_reduction += db_change_per_ducking_step;
}
input_buffer += samples_to_duck;
*ducking_transition_samples_remaining -= samples_to_duck;
input_samples_to_duck -= samples_to_duck;
}
}
if ((*current_ducking_db_reduction > 0) && (input_samples_to_duck > 0)) {
// Audio is ducked, but its not in the middle of a transition step
uint8_t safe_db_reduction_index =
clamp<uint8_t>(*current_ducking_db_reduction, 0, DECIBEL_REDUCTION_TABLE.size() - 1);
int16_t q15_scale_factor = DECIBEL_REDUCTION_TABLE[safe_db_reduction_index];
audio::scale_audio_samples(input_buffer, input_buffer, q15_scale_factor, input_samples_to_duck);
}
}
void MixerSpeaker::dump_config() {
ESP_LOGCONFIG(TAG, "Speaker Mixer:");
ESP_LOGCONFIG(TAG, " Number of output channels: %u", this->output_channels_);
}
void MixerSpeaker::setup() {
this->event_group_ = xEventGroupCreate();
if (this->event_group_ == nullptr) {
ESP_LOGE(TAG, "Failed to create event group");
this->mark_failed();
return;
}
}
void MixerSpeaker::loop() {
uint32_t event_group_bits = xEventGroupGetBits(this->event_group_);
if (event_group_bits & MixerEventGroupBits::STATE_STARTING) {
ESP_LOGD(TAG, "Starting speaker mixer");
xEventGroupClearBits(this->event_group_, MixerEventGroupBits::STATE_STARTING);
}
if (event_group_bits & MixerEventGroupBits::ERR_ESP_NO_MEM) {
this->status_set_error("Failed to allocate the mixer's internal buffer");
xEventGroupClearBits(this->event_group_, MixerEventGroupBits::ERR_ESP_NO_MEM);
}
if (event_group_bits & MixerEventGroupBits::STATE_RUNNING) {
ESP_LOGD(TAG, "Started speaker mixer");
this->status_clear_error();
xEventGroupClearBits(this->event_group_, MixerEventGroupBits::STATE_RUNNING);
}
if (event_group_bits & MixerEventGroupBits::STATE_STOPPING) {
ESP_LOGD(TAG, "Stopping speaker mixer");
xEventGroupClearBits(this->event_group_, MixerEventGroupBits::STATE_STOPPING);
}
if (event_group_bits & MixerEventGroupBits::STATE_STOPPED) {
if (this->delete_task_() == ESP_OK) {
xEventGroupClearBits(this->event_group_, MixerEventGroupBits::ALL_BITS);
}
}
if (this->task_handle_ != nullptr) {
bool all_stopped = true;
for (auto &speaker : this->source_speakers_) {
all_stopped &= speaker->is_stopped();
}
if (all_stopped) {
this->stop();
}
}
}
esp_err_t MixerSpeaker::start(audio::AudioStreamInfo &stream_info) {
if (!this->audio_stream_info_.has_value()) {
if (stream_info.get_bits_per_sample() != 16) {
// Audio streams that don't have 16 bits per sample are not supported
return ESP_ERR_NOT_SUPPORTED;
}
this->audio_stream_info_ = audio::AudioStreamInfo(stream_info.get_bits_per_sample(), this->output_channels_,
stream_info.get_sample_rate());
this->output_speaker_->set_audio_stream_info(this->audio_stream_info_.value());
} else {
if (!this->queue_mode_ && (stream_info.get_sample_rate() != this->audio_stream_info_.value().get_sample_rate())) {
// The two audio streams must have the same sample rate to mix properly if not in queue mode
return ESP_ERR_INVALID_ARG;
}
}
return this->start_task_();
}
esp_err_t MixerSpeaker::start_task_() {
if (this->task_stack_buffer_ == nullptr) {
if (this->task_stack_in_psram_) {
RAMAllocator<StackType_t> stack_allocator(RAMAllocator<StackType_t>::ALLOC_EXTERNAL);
this->task_stack_buffer_ = stack_allocator.allocate(TASK_STACK_SIZE);
} else {
RAMAllocator<StackType_t> stack_allocator(RAMAllocator<StackType_t>::ALLOC_INTERNAL);
this->task_stack_buffer_ = stack_allocator.allocate(TASK_STACK_SIZE);
}
}
if (this->task_stack_buffer_ == nullptr) {
return ESP_ERR_NO_MEM;
}
if (this->task_handle_ == nullptr) {
this->task_handle_ = xTaskCreateStatic(audio_mixer_task, "mixer", TASK_STACK_SIZE, (void *) this,
MIXER_TASK_PRIORITY, this->task_stack_buffer_, &this->task_stack_);
}
if (this->task_handle_ == nullptr) {
return ESP_ERR_INVALID_STATE;
}
return ESP_OK;
}
esp_err_t MixerSpeaker::delete_task_() {
if (!this->task_created_) {
this->task_handle_ = nullptr;
if (this->task_stack_buffer_ != nullptr) {
if (this->task_stack_in_psram_) {
RAMAllocator<StackType_t> stack_allocator(RAMAllocator<StackType_t>::ALLOC_EXTERNAL);
stack_allocator.deallocate(this->task_stack_buffer_, TASK_STACK_SIZE);
} else {
RAMAllocator<StackType_t> stack_allocator(RAMAllocator<StackType_t>::ALLOC_INTERNAL);
stack_allocator.deallocate(this->task_stack_buffer_, TASK_STACK_SIZE);
}
this->task_stack_buffer_ = nullptr;
}
return ESP_OK;
}
return ESP_ERR_INVALID_STATE;
}
void MixerSpeaker::stop() { xEventGroupSetBits(this->event_group_, MixerEventGroupBits::COMMAND_STOP); }
void MixerSpeaker::copy_frames(const int16_t *input_buffer, audio::AudioStreamInfo input_stream_info,
int16_t *output_buffer, audio::AudioStreamInfo output_stream_info,
uint32_t frames_to_transfer) {
uint8_t input_channels = input_stream_info.get_channels();
uint8_t output_channels = output_stream_info.get_channels();
const uint8_t max_input_channel_index = input_channels - 1;
if (input_channels == output_channels) {
size_t bytes_to_copy = input_stream_info.frames_to_bytes(frames_to_transfer);
memcpy(output_buffer, input_buffer, bytes_to_copy);
return;
}
for (uint32_t frame_index = 0; frame_index < frames_to_transfer; ++frame_index) {
for (uint8_t output_channel_index = 0; output_channel_index < output_channels; ++output_channel_index) {
uint8_t input_channel_index = std::min(output_channel_index, max_input_channel_index);
output_buffer[output_channels * frame_index + output_channel_index] =
input_buffer[input_channels * frame_index + input_channel_index];
}
}
}
void MixerSpeaker::mix_audio_samples(const int16_t *primary_buffer, audio::AudioStreamInfo primary_stream_info,
const int16_t *secondary_buffer, audio::AudioStreamInfo secondary_stream_info,
int16_t *output_buffer, audio::AudioStreamInfo output_stream_info,
uint32_t frames_to_mix) {
const uint8_t primary_channels = primary_stream_info.get_channels();
const uint8_t secondary_channels = secondary_stream_info.get_channels();
const uint8_t output_channels = output_stream_info.get_channels();
const uint8_t max_primary_channel_index = primary_channels - 1;
const uint8_t max_secondary_channel_index = secondary_channels - 1;
for (uint32_t frames_index = 0; frames_index < frames_to_mix; ++frames_index) {
for (uint8_t output_channel_index = 0; output_channel_index < output_channels; ++output_channel_index) {
const uint32_t secondary_channel_index = std::min(output_channel_index, max_secondary_channel_index);
const int32_t secondary_sample = secondary_buffer[frames_index * secondary_channels + secondary_channel_index];
const uint32_t primary_channel_index = std::min(output_channel_index, max_primary_channel_index);
const int32_t primary_sample =
static_cast<int32_t>(primary_buffer[frames_index * primary_channels + primary_channel_index]);
const int32_t added_sample = secondary_sample + primary_sample;
output_buffer[frames_index * output_channels + output_channel_index] =
static_cast<int16_t>(clamp<int32_t>(added_sample, MIN_AUDIO_SAMPLE_VALUE, MAX_AUDIO_SAMPLE_VALUE));
}
}
}
void MixerSpeaker::audio_mixer_task(void *params) {
MixerSpeaker *this_mixer = (MixerSpeaker *) params;
xEventGroupSetBits(this_mixer->event_group_, MixerEventGroupBits::STATE_STARTING);
this_mixer->task_created_ = true;
std::unique_ptr<audio::AudioSinkTransferBuffer> output_transfer_buffer = audio::AudioSinkTransferBuffer::create(
this_mixer->audio_stream_info_.value().ms_to_bytes(TRANSFER_BUFFER_DURATION_MS));
if (output_transfer_buffer == nullptr) {
xEventGroupSetBits(this_mixer->event_group_,
MixerEventGroupBits::STATE_STOPPED | MixerEventGroupBits::ERR_ESP_NO_MEM);
this_mixer->task_created_ = false;
vTaskDelete(nullptr);
}
output_transfer_buffer->set_sink(this_mixer->output_speaker_);
xEventGroupSetBits(this_mixer->event_group_, MixerEventGroupBits::STATE_RUNNING);
bool sent_finished = false;
while (true) {
uint32_t event_group_bits = xEventGroupGetBits(this_mixer->event_group_);
if (event_group_bits & MixerEventGroupBits::COMMAND_STOP) {
break;
}
output_transfer_buffer->transfer_data_to_sink(pdMS_TO_TICKS(TASK_DELAY_MS));
const uint32_t output_frames_free =
this_mixer->audio_stream_info_.value().bytes_to_frames(output_transfer_buffer->free());
std::vector<SourceSpeaker *> speakers_with_data;
std::vector<std::shared_ptr<audio::AudioSourceTransferBuffer>> transfer_buffers_with_data;
for (auto &speaker : this_mixer->source_speakers_) {
if (speaker->get_transfer_buffer().use_count() > 0) {
std::shared_ptr<audio::AudioSourceTransferBuffer> transfer_buffer = speaker->get_transfer_buffer().lock();
speaker->process_data_from_source(0); // Transfers and ducks audio from source ring buffers
if ((transfer_buffer->available() > 0) && !speaker->get_pause_state()) {
// Store the locked transfer buffers in their own vector to avoid releasing ownership until after the loop
transfer_buffers_with_data.push_back(transfer_buffer);
speakers_with_data.push_back(speaker);
}
}
}
if (transfer_buffers_with_data.empty()) {
// No audio available for transferring, block task temporarily
delay(TASK_DELAY_MS);
continue;
}
uint32_t frames_to_mix = output_frames_free;
if ((transfer_buffers_with_data.size() == 1) || this_mixer->queue_mode_) {
// Only one speaker has audio data, just copy samples over
audio::AudioStreamInfo active_stream_info = speakers_with_data[0]->get_audio_stream_info();
if (active_stream_info.get_sample_rate() ==
this_mixer->output_speaker_->get_audio_stream_info().get_sample_rate()) {
// Speaker's sample rate matches the output speaker's, copy directly
const uint32_t frames_available_in_buffer =
active_stream_info.bytes_to_frames(transfer_buffers_with_data[0]->available());
frames_to_mix = std::min(frames_to_mix, frames_available_in_buffer);
copy_frames(reinterpret_cast<int16_t *>(transfer_buffers_with_data[0]->get_buffer_start()), active_stream_info,
reinterpret_cast<int16_t *>(output_transfer_buffer->get_buffer_end()),
this_mixer->audio_stream_info_.value(), frames_to_mix);
// Update source speaker buffer length
transfer_buffers_with_data[0]->decrease_buffer_length(active_stream_info.frames_to_bytes(frames_to_mix));
speakers_with_data[0]->accumulated_frames_read_ += frames_to_mix;
// Add new audio duration to the source speaker pending playback
speakers_with_data[0]->pending_playback_ms_ +=
active_stream_info.frames_to_milliseconds_with_remainder(&speakers_with_data[0]->accumulated_frames_read_);
// Update output transfer buffer length
output_transfer_buffer->increase_buffer_length(
this_mixer->audio_stream_info_.value().frames_to_bytes(frames_to_mix));
} else {
// Speaker's stream info doesn't match the output speaker's, so it's a new source speaker
if (!this_mixer->output_speaker_->is_stopped()) {
if (!sent_finished) {
this_mixer->output_speaker_->finish();
sent_finished = true; // Avoid repeatedly sending the finish command
}
} else {
// Speaker has finished writing the current audio, update the stream information and restart the speaker
this_mixer->audio_stream_info_ =
audio::AudioStreamInfo(active_stream_info.get_bits_per_sample(), this_mixer->output_channels_,
active_stream_info.get_sample_rate());
this_mixer->output_speaker_->set_audio_stream_info(this_mixer->audio_stream_info_.value());
this_mixer->output_speaker_->start();
sent_finished = false;
}
}
} else {
// Determine how many frames to mix
for (int i = 0; i < transfer_buffers_with_data.size(); ++i) {
const uint32_t frames_available_in_buffer =
speakers_with_data[i]->get_audio_stream_info().bytes_to_frames(transfer_buffers_with_data[i]->available());
frames_to_mix = std::min(frames_to_mix, frames_available_in_buffer);
}
int16_t *primary_buffer = reinterpret_cast<int16_t *>(transfer_buffers_with_data[0]->get_buffer_start());
audio::AudioStreamInfo primary_stream_info = speakers_with_data[0]->get_audio_stream_info();
// Mix two streams together
for (int i = 1; i < transfer_buffers_with_data.size(); ++i) {
mix_audio_samples(primary_buffer, primary_stream_info,
reinterpret_cast<int16_t *>(transfer_buffers_with_data[i]->get_buffer_start()),
speakers_with_data[i]->get_audio_stream_info(),
reinterpret_cast<int16_t *>(output_transfer_buffer->get_buffer_end()),
this_mixer->audio_stream_info_.value(), frames_to_mix);
speakers_with_data[i]->pending_playback_ms_ +=
speakers_with_data[i]->get_audio_stream_info().frames_to_milliseconds_with_remainder(
&speakers_with_data[i]->accumulated_frames_read_);
if (i != transfer_buffers_with_data.size() - 1) {
// Need to mix more streams together, point primary buffer and stream info to the already mixed output
primary_buffer = reinterpret_cast<int16_t *>(output_transfer_buffer->get_buffer_end());
primary_stream_info = this_mixer->audio_stream_info_.value();
}
}
// Update source transfer buffer lengths and add new audio durations to the source speaker pending playbacks
for (int i = 0; i < transfer_buffers_with_data.size(); ++i) {
transfer_buffers_with_data[i]->decrease_buffer_length(
speakers_with_data[i]->get_audio_stream_info().frames_to_bytes(frames_to_mix));
speakers_with_data[i]->accumulated_frames_read_ += frames_to_mix;
speakers_with_data[i]->pending_playback_ms_ +=
speakers_with_data[i]->get_audio_stream_info().frames_to_milliseconds_with_remainder(
&speakers_with_data[i]->accumulated_frames_read_);
}
// Update output transfer buffer length
output_transfer_buffer->increase_buffer_length(
this_mixer->audio_stream_info_.value().frames_to_bytes(frames_to_mix));
}
}
xEventGroupSetBits(this_mixer->event_group_, MixerEventGroupBits::STATE_STOPPING);
output_transfer_buffer.reset();
xEventGroupSetBits(this_mixer->event_group_, MixerEventGroupBits::STATE_STOPPED);
this_mixer->task_created_ = false;
vTaskDelete(nullptr);
}
} // namespace mixer_speaker
} // namespace esphome
#endif

207
esphome/components/mixer/speaker/mixer_speaker.h Normal file
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@ -0,0 +1,207 @@
#pragma once
#ifdef USE_ESP32
#include "esphome/components/audio/audio.h"
#include "esphome/components/audio/audio_transfer_buffer.h"
#include "esphome/components/speaker/speaker.h"
#include "esphome/core/component.h"
#include <freertos/event_groups.h>
#include <freertos/FreeRTOS.h>
namespace esphome {
namespace mixer_speaker {
/* Classes for mixing several source speaker audio streams and writing it to another speaker component.
* - Volume controls are passed through to the output speaker
* - Directly handles pausing at the SourceSpeaker level; pause state is not passed through to the output speaker.
* - Audio sent to the SourceSpeaker's must have 16 bits per sample.
* - Audio sent to the SourceSpeaker can have any number of channels. They are duplicated or ignored as needed to match
* the number of channels required for the output speaker.
* - In queue mode, the audio sent to the SoureSpeakers can have different sample rates.
* - In non-queue mode, the audio sent to the SourceSpeakers must have the same sample rates.
* - SourceSpeaker has an internal ring buffer. It also allocates a shared_ptr for an AudioTranserBuffer object.
* - Audio Data Flow:
* - Audio data played on a SourceSpeaker first writes to its internal ring buffer.
* - MixerSpeaker task temporarily takes shared ownership of each SourceSpeaker's AudioTransferBuffer.
* - MixerSpeaker calls SourceSpeaker's `process_data_from_source`, which tranfers audio from the SourceSpeaker's
* ring buffer to its AudioTransferBuffer. Audio ducking is applied at this step.
* - In queue mode, MixerSpeaker prioritizes the earliest configured SourceSpeaker with audio data. Audio data is
* sent to the output speaker.
* - In non-queue mode, MixerSpeaker adds all the audio data in each SourceSpeaker into one stream that is written
* to the output speaker.
*/
class MixerSpeaker;
class SourceSpeaker : public speaker::Speaker, public Component {
public:
void dump_config() override;
void setup() override;
void loop() override;
size_t play(const uint8_t *data, size_t length, TickType_t ticks_to_wait) override;
size_t play(const uint8_t *data, size_t length) override { return this->play(data, length, 0); }
void start() override;
void stop() override;
void finish() override;
bool has_buffered_data() const override;
/// @brief Mute state changes are passed to the parent's output speaker
void set_mute_state(bool mute_state) override;
/// @brief Volume state changes are passed to the parent's output speaker
void set_volume(float volume) override;
void set_pause_state(bool pause_state) override { this->pause_state_ = pause_state; }
bool get_pause_state() const override { return this->pause_state_; }
/// @brief Transfers audio from the ring buffer into the transfer buffer. Ducks audio while transferring.
/// @param ticks_to_wait FreeRTOS ticks to wait while waiting to read from the ring buffer.
/// @return Number of bytes transferred from the ring buffer.
size_t process_data_from_source(TickType_t ticks_to_wait);
/// @brief Sets the ducking level for the source speaker.
/// @param decibel_reduction (uint8_t) The dB reduction level. For example, 0 is no change, 10 is a reduction by 10 dB
/// @param duration (uint32_t) The number of milliseconds to transition from the current level to the new level
void apply_ducking(uint8_t decibel_reduction, uint32_t duration);
void set_buffer_duration(uint32_t buffer_duration_ms) { this->buffer_duration_ms_ = buffer_duration_ms; }
void set_parent(MixerSpeaker *parent) { this->parent_ = parent; }
void set_timeout(uint32_t ms) { this->timeout_ms_ = ms; }
std::weak_ptr<audio::AudioSourceTransferBuffer> get_transfer_buffer() { return this->transfer_buffer_; }
protected:
friend class MixerSpeaker;
esp_err_t start_();
void stop_();
/// @brief Ducks audio samples by a specified amount. When changing the ducking amount, it can transition gradually
/// over a specified amount of samples.
/// @param input_buffer buffer with audio samples to be ducked in place
/// @param input_samples_to_duck number of samples to process in ``input_buffer``
/// @param current_ducking_db_reduction pointer to the current dB reduction
/// @param ducking_transition_samples_remaining pointer to the total number of samples left before the the
/// transition is finished
/// @param samples_per_ducking_step total number of samples per ducking step for the transition
/// @param db_change_per_ducking_step the change in dB reduction per step
static void duck_samples(int16_t *input_buffer, uint32_t input_samples_to_duck, int8_t *current_ducking_db_reduction,
uint32_t *ducking_transition_samples_remaining, uint32_t samples_per_ducking_step,
int8_t db_change_per_ducking_step);
MixerSpeaker *parent_;
std::shared_ptr<audio::AudioSourceTransferBuffer> transfer_buffer_;
std::weak_ptr<RingBuffer> ring_buffer_;
uint32_t buffer_duration_ms_;
uint32_t last_seen_data_ms_{0};
optional<uint32_t> timeout_ms_;
bool stop_gracefully_{false};
bool pause_state_{false};
int8_t target_ducking_db_reduction_{0};
int8_t current_ducking_db_reduction_{0};
int8_t db_change_per_ducking_step_{1};
uint32_t ducking_transition_samples_remaining_{0};
uint32_t samples_per_ducking_step_{0};
uint32_t accumulated_frames_read_{0};
uint32_t pending_playback_ms_{0};
};
class MixerSpeaker : public Component {
public:
void dump_config() override;
void setup() override;
void loop() override;
void add_source_speaker(SourceSpeaker *source_speaker) { this->source_speakers_.push_back(source_speaker); }
/// @brief Starts the mixer task. Called by a source speaker giving the current audio stream information
/// @param stream_info The calling source speakers audio stream information
/// @return ESP_ERR_NOT_SUPPORTED if the incoming stream is incompatible due to unsupported bits per sample
/// ESP_ERR_INVALID_ARG if the incoming stream is incompatible to be mixed with the other input audio stream
/// ESP_ERR_NO_MEM if there isn't enough memory for the task's stack
/// ESP_ERR_INVALID_STATE if the task fails to start
/// ESP_OK if the incoming stream is compatible and the mixer task starts
esp_err_t start(audio::AudioStreamInfo &stream_info);
void stop();
void set_output_channels(uint8_t output_channels) { this->output_channels_ = output_channels; }
void set_output_speaker(speaker::Speaker *speaker) { this->output_speaker_ = speaker; }
void set_queue_mode(bool queue_mode) { this->queue_mode_ = queue_mode; }
void set_task_stack_in_psram(bool task_stack_in_psram) { this->task_stack_in_psram_ = task_stack_in_psram; }
speaker::Speaker *get_output_speaker() const { return this->output_speaker_; }
protected:
/// @brief Copies audio frames from the input buffer to the output buffer taking into account the number of channels
/// in each stream. If the output stream has more channels, the input samples are duplicated. If the output stream has
/// less channels, the extra channel input samples are dropped.
/// @param input_buffer
/// @param input_stream_info
/// @param output_buffer
/// @param output_stream_info
/// @param frames_to_transfer number of frames (consisting of a sample for each channel) to copy from the input buffer
static void copy_frames(const int16_t *input_buffer, audio::AudioStreamInfo input_stream_info, int16_t *output_buffer,
audio::AudioStreamInfo output_stream_info, uint32_t frames_to_transfer);
/// @brief Mixes the primary and secondary streams taking into account the number of channels in each stream. Primary
/// and secondary samples are duplicated or dropped as necessary to ensure the output stream has the configured number
/// of channels. Output samples are clamped to the corresponding int16 min or max values if the mixed sample
/// overflows.
/// @param primary_buffer (int16_t *) samples buffer for the primary stream
/// @param primary_stream_info stream info for the primary stream
/// @param secondary_buffer (int16_t *) samples buffer for secondary stream
/// @param secondary_stream_info stream info for the secondary stream
/// @param output_buffer (int16_t *) buffer for the mixed samples
/// @param output_stream_info stream info for the output buffer
/// @param frames_to_mix number of frames in the primary and secondary buffers to mix together
static void mix_audio_samples(const int16_t *primary_buffer, audio::AudioStreamInfo primary_stream_info,
const int16_t *secondary_buffer, audio::AudioStreamInfo secondary_stream_info,
int16_t *output_buffer, audio::AudioStreamInfo output_stream_info,
uint32_t frames_to_mix);
static void audio_mixer_task(void *params);
/// @brief Starts the mixer task after allocating memory for the task stack.
/// @return ESP_ERR_NO_MEM if there isn't enough memory for the task's stack
/// ESP_ERR_INVALID_STATE if the task didn't start
/// ESP_OK if successful
esp_err_t start_task_();
/// @brief If the task is stopped, it sets the task handle to the nullptr and deallocates its stack
/// @return ESP_OK if the task was stopped, ESP_ERR_INVALID_STATE otherwise.
esp_err_t delete_task_();
EventGroupHandle_t event_group_{nullptr};
std::vector<SourceSpeaker *> source_speakers_;
speaker::Speaker *output_speaker_{nullptr};
uint8_t output_channels_;
bool queue_mode_;
bool task_stack_in_psram_{false};
bool task_created_{false};
TaskHandle_t task_handle_{nullptr};
StaticTask_t task_stack_;
StackType_t *task_stack_buffer_{nullptr};
optional<audio::AudioStreamInfo> audio_stream_info_;
};
} // namespace mixer_speaker
} // namespace esphome
#endif

2
esphome/components/mpr121/mpr121.h
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@ -117,6 +117,8 @@ class MPR121GPIOPin : public GPIOPin {
void set_inverted(bool inverted) { this->inverted_ = inverted; }
void set_flags(gpio::Flags flags) { this->flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
protected:
MPR121Component *parent_;
uint8_t pin_;

2
esphome/components/pca6416a/pca6416a.h
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@ -52,6 +52,8 @@ class PCA6416AGPIOPin : public GPIOPin {
void set_inverted(bool inverted) { inverted_ = inverted; }
void set_flags(gpio::Flags flags) { flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
protected:
PCA6416AComponent *parent_;
uint8_t pin_;

2
esphome/components/pca9554/pca9554.h
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@ -65,6 +65,8 @@ class PCA9554GPIOPin : public GPIOPin {
void set_inverted(bool inverted) { inverted_ = inverted; }
void set_flags(gpio::Flags flags) { flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
protected:
PCA9554Component *parent_;
uint8_t pin_;

2
esphome/components/pcf8574/pcf8574.h
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@ -54,6 +54,8 @@ class PCF8574GPIOPin : public GPIOPin {
void set_inverted(bool inverted) { inverted_ = inverted; }
void set_flags(gpio::Flags flags) { flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
protected:
PCF8574Component *parent_;
uint8_t pin_;

0
esphome/components/resampler/__init__.py Normal file
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103
esphome/components/resampler/speaker/__init__.py Normal file
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@ -0,0 +1,103 @@
import esphome.codegen as cg
from esphome.components import audio, esp32, speaker
import esphome.config_validation as cv
from esphome.const import (
CONF_BITS_PER_SAMPLE,
CONF_BUFFER_DURATION,
CONF_FILTERS,
CONF_ID,
CONF_NUM_CHANNELS,
CONF_OUTPUT_SPEAKER,
CONF_SAMPLE_RATE,
CONF_TASK_STACK_IN_PSRAM,
PLATFORM_ESP32,
)
from esphome.core.entity_helpers import inherit_property_from
AUTO_LOAD = ["audio"]
CODEOWNERS = ["@kahrendt"]
resampler_ns = cg.esphome_ns.namespace("resampler")
ResamplerSpeaker = resampler_ns.class_(
"ResamplerSpeaker", cg.Component, speaker.Speaker
)
CONF_TAPS = "taps"
def _set_stream_limits(config):
audio.set_stream_limits(
min_bits_per_sample=16,
max_bits_per_sample=32,
)(config)
return config
def _validate_audio_compatability(config):
inherit_property_from(CONF_BITS_PER_SAMPLE, CONF_OUTPUT_SPEAKER)(config)
inherit_property_from(CONF_NUM_CHANNELS, CONF_OUTPUT_SPEAKER)(config)
inherit_property_from(CONF_SAMPLE_RATE, CONF_OUTPUT_SPEAKER)(config)
audio.final_validate_audio_schema(
"source_speaker",
audio_device=CONF_OUTPUT_SPEAKER,
bits_per_sample=config.get(CONF_BITS_PER_SAMPLE),
channels=config.get(CONF_NUM_CHANNELS),
sample_rate=config.get(CONF_SAMPLE_RATE),
)(config)
def _validate_taps(taps):
value = cv.int_range(min=16, max=128)(taps)
if value % 4 != 0:
raise cv.Invalid("Number of taps must be divisible by 4")
return value
CONFIG_SCHEMA = cv.All(
speaker.SPEAKER_SCHEMA.extend(
{
cv.GenerateID(): cv.declare_id(ResamplerSpeaker),
cv.Required(CONF_OUTPUT_SPEAKER): cv.use_id(speaker.Speaker),
cv.Optional(
CONF_BUFFER_DURATION, default="100ms"
): cv.positive_time_period_milliseconds,
cv.SplitDefault(CONF_TASK_STACK_IN_PSRAM, esp32_idf=False): cv.All(
cv.boolean, cv.only_with_esp_idf
),
cv.Optional(CONF_FILTERS, default=16): cv.int_range(min=2, max=1024),
cv.Optional(CONF_TAPS, default=16): _validate_taps,
}
).extend(cv.COMPONENT_SCHEMA),
cv.only_on([PLATFORM_ESP32]),
_set_stream_limits,
)
FINAL_VALIDATE_SCHEMA = _validate_audio_compatability
async def to_code(config):
var = cg.new_Pvariable(config[CONF_ID])
await cg.register_component(var, config)
await speaker.register_speaker(var, config)
output_spkr = await cg.get_variable(config[CONF_OUTPUT_SPEAKER])
cg.add(var.set_output_speaker(output_spkr))
cg.add(var.set_buffer_duration(config[CONF_BUFFER_DURATION]))
if task_stack_in_psram := config.get(CONF_TASK_STACK_IN_PSRAM):
cg.add(var.set_task_stack_in_psram(task_stack_in_psram))
if task_stack_in_psram:
if config[CONF_TASK_STACK_IN_PSRAM]:
esp32.add_idf_sdkconfig_option(
"CONFIG_SPIRAM_ALLOW_STACK_EXTERNAL_MEMORY", True
)
cg.add(var.set_target_bits_per_sample(config[CONF_BITS_PER_SAMPLE]))
cg.add(var.set_target_sample_rate(config[CONF_SAMPLE_RATE]))
cg.add(var.set_filters(config[CONF_FILTERS]))
cg.add(var.set_taps(config[CONF_TAPS]))

318
esphome/components/resampler/speaker/resampler_speaker.cpp Normal file
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@ -0,0 +1,318 @@
#include "resampler_speaker.h"
#ifdef USE_ESP32
#include "esphome/components/audio/audio_resampler.h"
#include "esphome/core/helpers.h"
#include "esphome/core/log.h"
#include <algorithm>
#include <cstring>
namespace esphome {
namespace resampler {
static const UBaseType_t RESAMPLER_TASK_PRIORITY = 1;
static const uint32_t TRANSFER_BUFFER_DURATION_MS = 50;
static const uint32_t TASK_DELAY_MS = 20;
static const uint32_t TASK_STACK_SIZE = 3072;
static const char *const TAG = "resampler_speaker";
enum ResamplingEventGroupBits : uint32_t {
COMMAND_STOP = (1 << 0), // stops the resampler task
STATE_STARTING = (1 << 10),
STATE_RUNNING = (1 << 11),
STATE_STOPPING = (1 << 12),
STATE_STOPPED = (1 << 13),
ERR_ESP_NO_MEM = (1 << 19),
ERR_ESP_NOT_SUPPORTED = (1 << 20),
ERR_ESP_FAIL = (1 << 21),
ALL_BITS = 0x00FFFFFF, // All valid FreeRTOS event group bits
};
void ResamplerSpeaker::setup() {
this->event_group_ = xEventGroupCreate();
if (this->event_group_ == nullptr) {
ESP_LOGE(TAG, "Failed to create event group");
this->mark_failed();
return;
}
this->output_speaker_->add_audio_output_callback(
[this](uint32_t new_playback_ms, uint32_t remainder_us, uint32_t pending_ms, uint32_t write_timestamp) {
int32_t adjustment = this->playback_differential_ms_;
this->playback_differential_ms_ -= adjustment;
int32_t adjusted_playback_ms = static_cast<int32_t>(new_playback_ms) + adjustment;
this->audio_output_callback_(adjusted_playback_ms, remainder_us, pending_ms, write_timestamp);
});
}
void ResamplerSpeaker::loop() {
uint32_t event_group_bits = xEventGroupGetBits(this->event_group_);
if (event_group_bits & ResamplingEventGroupBits::STATE_STARTING) {
ESP_LOGD(TAG, "Starting resampler task");
xEventGroupClearBits(this->event_group_, ResamplingEventGroupBits::STATE_STARTING);
}
if (event_group_bits & ResamplingEventGroupBits::ERR_ESP_NO_MEM) {
this->status_set_error("Resampler task failed to allocate the internal buffers");
xEventGroupClearBits(this->event_group_, ResamplingEventGroupBits::ERR_ESP_NO_MEM);
this->state_ = speaker::STATE_STOPPING;
}
if (event_group_bits & ResamplingEventGroupBits::ERR_ESP_NOT_SUPPORTED) {
this->status_set_error("Cannot resample due to an unsupported audio stream");
xEventGroupClearBits(this->event_group_, ResamplingEventGroupBits::ERR_ESP_NOT_SUPPORTED);
this->state_ = speaker::STATE_STOPPING;
}
if (event_group_bits & ResamplingEventGroupBits::ERR_ESP_FAIL) {
this->status_set_error("Resampler task failed");
xEventGroupClearBits(this->event_group_, ResamplingEventGroupBits::ERR_ESP_FAIL);
this->state_ = speaker::STATE_STOPPING;
}
if (event_group_bits & ResamplingEventGroupBits::STATE_RUNNING) {
ESP_LOGD(TAG, "Started resampler task");
this->status_clear_error();
xEventGroupClearBits(this->event_group_, ResamplingEventGroupBits::STATE_RUNNING);
}
if (event_group_bits & ResamplingEventGroupBits::STATE_STOPPING) {
ESP_LOGD(TAG, "Stopping resampler task");
xEventGroupClearBits(this->event_group_, ResamplingEventGroupBits::STATE_STOPPING);
}
if (event_group_bits & ResamplingEventGroupBits::STATE_STOPPED) {
if (this->delete_task_() == ESP_OK) {
ESP_LOGD(TAG, "Stopped resampler task");
xEventGroupClearBits(this->event_group_, ResamplingEventGroupBits::ALL_BITS);
}
}
switch (this->state_) {
case speaker::STATE_STARTING: {
esp_err_t err = this->start_();
if (err == ESP_OK) {
this->status_clear_error();
this->state_ = speaker::STATE_RUNNING;
} else {
switch (err) {
case ESP_ERR_INVALID_STATE:
this->status_set_error("Failed to start resampler: resampler task failed to start");
break;
case ESP_ERR_NO_MEM:
this->status_set_error("Failed to start resampler: not enough memory for task stack");
default:
this->status_set_error("Failed to start resampler");
break;
}
this->state_ = speaker::STATE_STOPPING;
}
break;
}
case speaker::STATE_RUNNING:
if (this->output_speaker_->is_stopped()) {
this->state_ = speaker::STATE_STOPPING;
}
break;
case speaker::STATE_STOPPING:
this->stop_();
this->state_ = speaker::STATE_STOPPED;
break;
case speaker::STATE_STOPPED:
break;
}
}
size_t ResamplerSpeaker::play(const uint8_t *data, size_t length, TickType_t ticks_to_wait) {
if (this->is_stopped()) {
this->start();
}
size_t bytes_written = 0;
if ((this->output_speaker_->is_running()) && (!this->requires_resampling_())) {
bytes_written = this->output_speaker_->play(data, length, ticks_to_wait);
} else {
if (this->ring_buffer_.use_count() == 1) {
std::shared_ptr<RingBuffer> temp_ring_buffer = this->ring_buffer_.lock();
bytes_written = temp_ring_buffer->write_without_replacement(data, length, ticks_to_wait);
}
}
return bytes_written;
}
void ResamplerSpeaker::start() { this->state_ = speaker::STATE_STARTING; }
esp_err_t ResamplerSpeaker::start_() {
this->target_stream_info_ = audio::AudioStreamInfo(
this->target_bits_per_sample_, this->audio_stream_info_.get_channels(), this->target_sample_rate_);
this->output_speaker_->set_audio_stream_info(this->target_stream_info_);
this->output_speaker_->start();
if (this->requires_resampling_()) {
// Start the resampler task to handle converting sample rates
return this->start_task_();
}
return ESP_OK;
}
esp_err_t ResamplerSpeaker::start_task_() {
if (this->task_stack_buffer_ == nullptr) {
if (this->task_stack_in_psram_) {
RAMAllocator<StackType_t> stack_allocator(RAMAllocator<StackType_t>::ALLOC_EXTERNAL);
this->task_stack_buffer_ = stack_allocator.allocate(TASK_STACK_SIZE);
} else {
RAMAllocator<StackType_t> stack_allocator(RAMAllocator<StackType_t>::ALLOC_INTERNAL);
this->task_stack_buffer_ = stack_allocator.allocate(TASK_STACK_SIZE);
}
}
if (this->task_stack_buffer_ == nullptr) {
return ESP_ERR_NO_MEM;
}
if (this->task_handle_ == nullptr) {
this->task_handle_ = xTaskCreateStatic(resample_task, "sample", TASK_STACK_SIZE, (void *) this,
RESAMPLER_TASK_PRIORITY, this->task_stack_buffer_, &this->task_stack_);
}
if (this->task_handle_ == nullptr) {
return ESP_ERR_INVALID_STATE;
}
return ESP_OK;
}
void ResamplerSpeaker::stop() { this->state_ = speaker::STATE_STOPPING; }
void ResamplerSpeaker::stop_() {
if (this->task_handle_ != nullptr) {
xEventGroupSetBits(this->event_group_, ResamplingEventGroupBits::COMMAND_STOP);
}
this->output_speaker_->stop();
}
esp_err_t ResamplerSpeaker::delete_task_() {
if (!this->task_created_) {
this->task_handle_ = nullptr;
if (this->task_stack_buffer_ != nullptr) {
if (this->task_stack_in_psram_) {
RAMAllocator<StackType_t> stack_allocator(RAMAllocator<StackType_t>::ALLOC_EXTERNAL);
stack_allocator.deallocate(this->task_stack_buffer_, TASK_STACK_SIZE);
} else {
RAMAllocator<StackType_t> stack_allocator(RAMAllocator<StackType_t>::ALLOC_INTERNAL);
stack_allocator.deallocate(this->task_stack_buffer_, TASK_STACK_SIZE);
}
this->task_stack_buffer_ = nullptr;
}
return ESP_OK;
}
return ESP_ERR_INVALID_STATE;
}
void ResamplerSpeaker::finish() { this->output_speaker_->finish(); }
bool ResamplerSpeaker::has_buffered_data() const {
bool has_ring_buffer_data = false;
if (this->requires_resampling_() && (this->ring_buffer_.use_count() > 0)) {
has_ring_buffer_data = (this->ring_buffer_.lock()->available() > 0);
}
return (has_ring_buffer_data || this->output_speaker_->has_buffered_data());
}
void ResamplerSpeaker::set_mute_state(bool mute_state) {
this->mute_state_ = mute_state;
this->output_speaker_->set_mute_state(mute_state);
}
void ResamplerSpeaker::set_volume(float volume) {
this->volume_ = volume;
this->output_speaker_->set_volume(volume);
}
bool ResamplerSpeaker::requires_resampling_() const {
return (this->audio_stream_info_.get_sample_rate() != this->target_sample_rate_) ||
(this->audio_stream_info_.get_bits_per_sample() != this->target_bits_per_sample_);
}
void ResamplerSpeaker::resample_task(void *params) {
ResamplerSpeaker *this_resampler = (ResamplerSpeaker *) params;
this_resampler->task_created_ = true;
xEventGroupSetBits(this_resampler->event_group_, ResamplingEventGroupBits::STATE_STARTING);
std::unique_ptr<audio::AudioResampler> resampler =
make_unique<audio::AudioResampler>(this_resampler->audio_stream_info_.ms_to_bytes(TRANSFER_BUFFER_DURATION_MS),
this_resampler->target_stream_info_.ms_to_bytes(TRANSFER_BUFFER_DURATION_MS));
esp_err_t err = resampler->start(this_resampler->audio_stream_info_, this_resampler->target_stream_info_,
this_resampler->taps_, this_resampler->filters_);
if (err == ESP_OK) {
std::shared_ptr<RingBuffer> temp_ring_buffer =
RingBuffer::create(this_resampler->audio_stream_info_.ms_to_bytes(this_resampler->buffer_duration_ms_));
if (temp_ring_buffer.use_count() == 0) {
err = ESP_ERR_NO_MEM;
} else {
this_resampler->ring_buffer_ = temp_ring_buffer;
resampler->add_source(this_resampler->ring_buffer_);
this_resampler->output_speaker_->set_audio_stream_info(this_resampler->target_stream_info_);
resampler->add_sink(this_resampler->output_speaker_);
}
}
if (err == ESP_OK) {
xEventGroupSetBits(this_resampler->event_group_, ResamplingEventGroupBits::STATE_RUNNING);
} else if (err == ESP_ERR_NO_MEM) {
xEventGroupSetBits(this_resampler->event_group_, ResamplingEventGroupBits::ERR_ESP_NO_MEM);
} else if (err == ESP_ERR_NOT_SUPPORTED) {
xEventGroupSetBits(this_resampler->event_group_, ResamplingEventGroupBits::ERR_ESP_NOT_SUPPORTED);
}
this_resampler->playback_differential_ms_ = 0;
while (err == ESP_OK) {
uint32_t event_bits = xEventGroupGetBits(this_resampler->event_group_);
if (event_bits & ResamplingEventGroupBits::COMMAND_STOP) {
break;
}
// Stop gracefully if the decoder is done
int32_t ms_differential = 0;
audio::AudioResamplerState resampler_state = resampler->resample(false, &ms_differential);
this_resampler->playback_differential_ms_ += ms_differential;
if (resampler_state == audio::AudioResamplerState::FINISHED) {
break;
} else if (resampler_state == audio::AudioResamplerState::FAILED) {
xEventGroupSetBits(this_resampler->event_group_, ResamplingEventGroupBits::ERR_ESP_FAIL);
break;
}
}
xEventGroupSetBits(this_resampler->event_group_, ResamplingEventGroupBits::STATE_STOPPING);
resampler.reset();
xEventGroupSetBits(this_resampler->event_group_, ResamplingEventGroupBits::STATE_STOPPED);
this_resampler->task_created_ = false;
vTaskDelete(nullptr);
}
} // namespace resampler
} // namespace esphome
#endif

107
esphome/components/resampler/speaker/resampler_speaker.h Normal file
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@ -0,0 +1,107 @@
#pragma once
#ifdef USE_ESP32
#include "esphome/components/audio/audio.h"
#include "esphome/components/audio/audio_transfer_buffer.h"
#include "esphome/components/speaker/speaker.h"
#include "esphome/core/component.h"
#include <freertos/event_groups.h>
#include <freertos/FreeRTOS.h>
namespace esphome {
namespace resampler {
class ResamplerSpeaker : public Component, public speaker::Speaker {
public:
float get_setup_priority() const override { return esphome::setup_priority::DATA; }
void setup() override;
void loop() override;
size_t play(const uint8_t *data, size_t length, TickType_t ticks_to_wait) override;
size_t play(const uint8_t *data, size_t length) override { return this->play(data, length, 0); }
void start() override;
void stop() override;
void finish() override;
void set_pause_state(bool pause_state) override { this->output_speaker_->set_pause_state(pause_state); }
bool get_pause_state() const override { return this->output_speaker_->get_pause_state(); }
bool has_buffered_data() const override;
/// @brief Mute state changes are passed to the parent's output speaker
void set_mute_state(bool mute_state) override;
/// @brief Volume state changes are passed to the parent's output speaker
void set_volume(float volume) override;
void set_output_speaker(speaker::Speaker *speaker) { this->output_speaker_ = speaker; }
void set_task_stack_in_psram(bool task_stack_in_psram) { this->task_stack_in_psram_ = task_stack_in_psram; }
void set_target_bits_per_sample(uint8_t target_bits_per_sample) {
this->target_bits_per_sample_ = target_bits_per_sample;
}
void set_target_sample_rate(uint32_t target_sample_rate) { this->target_sample_rate_ = target_sample_rate; }
void set_filters(uint16_t filters) { this->filters_ = filters; }
void set_taps(uint16_t taps) { this->taps_ = taps; }
void set_buffer_duration(uint32_t buffer_duration_ms) { this->buffer_duration_ms_ = buffer_duration_ms; }
protected:
/// @brief Starts the output speaker after setting the resampled stream info. If resampling is required, it starts the
/// task.
/// @return ESP_OK if resampling is required
/// return value of start_task_() if resampling is required
esp_err_t start_();
/// @brief Starts the resampler task after allocating the task stack
/// @return ESP_OK if successful,
/// ESP_ERR_NO_MEM if the task stack couldn't be allocated
/// ESP_ERR_INVALID_STATE if the task wasn't created
esp_err_t start_task_();
/// @brief Stops the output speaker. If the resampling task is running, it sends the stop command.
void stop_();
/// @brief Deallocates the task stack and resets the pointers.
/// @return ESP_OK if successful
/// ESP_ERR_INVALID_STATE if the task hasn't stopped itself
esp_err_t delete_task_();
inline bool requires_resampling_() const;
static void resample_task(void *params);
EventGroupHandle_t event_group_{nullptr};
std::weak_ptr<RingBuffer> ring_buffer_;
speaker::Speaker *output_speaker_{nullptr};
bool task_stack_in_psram_{false};
bool task_created_{false};
TaskHandle_t task_handle_{nullptr};
StaticTask_t task_stack_;
StackType_t *task_stack_buffer_{nullptr};
audio::AudioStreamInfo target_stream_info_;
uint16_t taps_;
uint16_t filters_;
uint8_t target_bits_per_sample_;
uint32_t target_sample_rate_;
uint32_t buffer_duration_ms_;
int32_t playback_differential_ms_{0};
};
} // namespace resampler
} // namespace esphome
#endif

1
esphome/components/rp2040/gpio.h
View File

@ -22,6 +22,7 @@ class RP2040GPIOPin : public InternalGPIOPin {
void detach_interrupt() const override;
ISRInternalGPIOPin to_isr() const override;
uint8_t get_pin() const override { return pin_; }
gpio::Flags get_flags() const override { return flags_; }
bool is_inverted() const override { return inverted_; }
protected:

3
esphome/components/sn74hc165/sn74hc165.h
View File

@ -52,6 +52,9 @@ class SN74HC165GPIOPin : public GPIOPin, public Parented<SN74HC165Component> {
void set_pin(uint16_t pin) { pin_ = pin; }
void set_inverted(bool inverted) { inverted_ = inverted; }
/// Always returns `gpio::Flags::FLAG_INPUT`.
gpio::Flags get_flags() const override { return gpio::Flags::FLAG_INPUT; }
protected:
uint16_t pin_;
bool inverted_;

3
esphome/components/sn74hc595/sn74hc595.h
View File

@ -59,6 +59,9 @@ class SN74HC595GPIOPin : public GPIOPin, public Parented<SN74HC595Component> {
void set_pin(uint16_t pin) { pin_ = pin; }
void set_inverted(bool inverted) { inverted_ = inverted; }
/// Always returns `gpio::Flags::FLAG_OUTPUT`.
gpio::Flags get_flags() const override { return gpio::Flags::FLAG_OUTPUT; }
protected:
uint16_t pin_;
bool inverted_;

9
esphome/components/speaker/__init__.py
View File

@ -1,7 +1,6 @@
from esphome import automation
from esphome.automation import maybe_simple_id
import esphome.codegen as cg
from esphome.components import audio_dac
from esphome.components import audio, audio_dac
import esphome.config_validation as cv
from esphome.const import CONF_DATA, CONF_ID, CONF_VOLUME
from esphome.core import CORE
@ -54,13 +53,15 @@ async def register_speaker(var, config):
await setup_speaker_core_(var, config)
SPEAKER_SCHEMA = cv.Schema(
SPEAKER_SCHEMA = cv.Schema.extend(audio.AUDIO_COMPONENT_SCHEMA).extend(
{
cv.Optional(CONF_AUDIO_DAC): cv.use_id(audio_dac.AudioDac),
}
)
SPEAKER_AUTOMATION_SCHEMA = maybe_simple_id({cv.GenerateID(): cv.use_id(Speaker)})
SPEAKER_AUTOMATION_SCHEMA = automation.maybe_simple_id(
{cv.GenerateID(): cv.use_id(Speaker)}
)
async def speaker_action(config, action_id, template_arg, args):

20
esphome/components/speaker/speaker.h
View File

@ -9,6 +9,7 @@
#endif
#include "esphome/core/defines.h"
#include "esphome/core/helpers.h"
#include "esphome/components/audio/audio.h"
#ifdef USE_AUDIO_DAC
@ -56,6 +57,10 @@ class Speaker {
// When finish() is not implemented on the platform component it should just do a normal stop.
virtual void finish() { this->stop(); }
// Pauses processing incoming audio. Needs to be implemented specifically per speaker component
virtual void set_pause_state(bool pause_state) {}
virtual bool get_pause_state() const { return false; }
virtual bool has_buffered_data() const = 0;
bool is_running() const { return this->state_ == STATE_RUNNING; }
@ -95,6 +100,19 @@ class Speaker {
this->audio_stream_info_ = audio_stream_info;
}
audio::AudioStreamInfo &get_audio_stream_info() { return this->audio_stream_info_; }
/// Callback function for sending the duration of the audio written to the speaker since the last callback.
/// Parameters:
/// - Duration in milliseconds. Never rounded and should always be less than or equal to the actual duration.
/// - Remainder duration in microseconds. Rounded duration after subtracting the previous parameter from the actual
/// duration.
/// - Duration of remaining, unwritten audio buffered in the speaker in milliseconds.
/// - System time in microseconds when the last write was completed.
void add_audio_output_callback(std::function<void(uint32_t, uint32_t, uint32_t, uint32_t)> &&callback) {
this->audio_output_callback_.add(std::move(callback));
}
protected:
State state_{STATE_STOPPED};
audio::AudioStreamInfo audio_stream_info_;
@ -104,6 +122,8 @@ class Speaker {
#ifdef USE_AUDIO_DAC
audio_dac::AudioDac *audio_dac_{nullptr};
#endif
CallbackManager<void(uint32_t, uint32_t, uint32_t, uint32_t)> audio_output_callback_{};
};
} // namespace speaker

2
esphome/components/spi/spi.h
View File

@ -114,6 +114,8 @@ class NullPin : public GPIOPin {
void pin_mode(gpio::Flags flags) override {}
gpio::Flags get_flags() const override { return gpio::Flags::FLAG_NONE; }
bool digital_read() override { return false; }
void digital_write(bool value) override {}

2
esphome/components/sx1509/sx1509_gpio_pin.h
View File

@ -20,6 +20,8 @@ class SX1509GPIOPin : public GPIOPin {
void set_inverted(bool inverted) { this->inverted_ = inverted; }
void set_flags(gpio::Flags flags) { this->flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
protected:
SX1509Component *parent_;
uint8_t pin_;

2
esphome/components/tca9555/tca9555.h
View File

@ -54,6 +54,8 @@ class TCA9555GPIOPin : public GPIOPin, public Parented<TCA9555Component> {
void set_inverted(bool inverted) { this->inverted_ = inverted; }
void set_flags(gpio::Flags flags) { this->flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
protected:
uint8_t pin_;
bool inverted_;

2
esphome/components/udp/__init__.py
View File

@ -18,7 +18,7 @@ from esphome.cpp_generator import MockObjClass
CODEOWNERS = ["@clydebarrow"]
DEPENDENCIES = ["network"]
AUTO_LOAD = ["socket"]
AUTO_LOAD = ["socket", "xxtea"]
MULTI_CONF = True
udp_ns = cg.esphome_ns.namespace("udp")

55
esphome/components/udp/udp_component.cpp
View File

@ -3,6 +3,8 @@
#include "esphome/components/network/util.h"
#include "udp_component.h"
#include "esphome/components/xxtea/xxtea.h"
namespace esphome {
namespace udp {
@ -47,54 +49,7 @@ namespace udp {
*/
static const char *const TAG = "udp";
/**
* XXTEA implementation, using 256 bit key.
*/
static const uint32_t DELTA = 0x9e3779b9;
#define MX ((((z >> 5) ^ (y << 2)) + ((y >> 3) ^ (z << 4))) ^ ((sum ^ y) + (k[(p ^ e) & 7] ^ z)))
/**
* Encrypt a block of data in-place
*/
static void xxtea_encrypt(uint32_t *v, size_t n, const uint32_t *k) {
uint32_t z, y, sum, e;
size_t p;
size_t q = 6 + 52 / n;
sum = 0;
z = v[n - 1];
while (q-- != 0) {
sum += DELTA;
e = (sum >> 2);
for (p = 0; p != n - 1; p++) {
y = v[p + 1];
z = v[p] += MX;
}
y = v[0];
z = v[n - 1] += MX;
}
}
static void xxtea_decrypt(uint32_t *v, size_t n, const uint32_t *k) {
uint32_t z, y, sum, e;
size_t p;
size_t q = 6 + 52 / n;
sum = q * DELTA;
y = v[0];
while (q-- != 0) {
e = (sum >> 2);
for (p = n - 1; p != 0; p--) {
z = v[p - 1];
y = v[p] -= MX;
}
z = v[n - 1];
y = v[0] -= MX;
sum -= DELTA;
}
}
inline static size_t round4(size_t value) { return (value + 3) & ~3; }
static size_t round4(size_t value) { return (value + 3) & ~3; }
union FuData {
uint32_t u32;
@ -312,7 +267,7 @@ void UDPComponent::flush_() {
memcpy(buffer, this->header_.data(), this->header_.size());
memcpy(buffer + header_len, this->data_.data(), this->data_.size());
if (this->is_encrypted_()) {
xxtea_encrypt(buffer + header_len, len, (uint32_t *) this->encryption_key_.data());
xxtea::encrypt(buffer + header_len, len, (uint32_t *) this->encryption_key_.data());
}
auto total_len = (header_len + len) * 4;
this->send_packet_(buffer, total_len);
@ -503,7 +458,7 @@ void UDPComponent::process_(uint8_t *buf, const size_t len) {
#endif
if (!provider.encryption_key.empty()) {
xxtea_decrypt((uint32_t *) buf, (end - buf) / 4, (uint32_t *) provider.encryption_key.data());
xxtea::decrypt((uint32_t *) buf, (end - buf) / 4, (uint32_t *) provider.encryption_key.data());
}
byte = *buf++;
if (byte == ROLLING_CODE_KEY) {

2
esphome/components/weikai/weikai.h
View File

@ -275,6 +275,8 @@ class WeikaiGPIOPin : public GPIOPin {
void set_inverted(bool inverted) { this->inverted_ = inverted; }
void set_flags(gpio::Flags flags) { this->flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
void setup() override;
std::string dump_summary() const override;
void pin_mode(gpio::Flags flags) override { this->parent_->set_pin_direction_(this->pin_, flags); }

2
esphome/components/xl9535/xl9535.h
View File

@ -36,6 +36,8 @@ class XL9535GPIOPin : public GPIOPin {
void set_inverted(bool inverted) { this->inverted_ = inverted; }
void set_flags(gpio::Flags flags) { this->flags_ = flags; }
gpio::Flags get_flags() const override { return this->flags_; }
void setup() override;
std::string dump_summary() const override;
void pin_mode(gpio::Flags flags) override;

3
esphome/components/xxtea/__init__.py Normal file
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@ -0,0 +1,3 @@
"""ESPHome XXTEA encryption component."""
CODEOWNERS = ["@clydebarrow"]

46
esphome/components/xxtea/xxtea.cpp Normal file
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@ -0,0 +1,46 @@
#include "xxtea.h"
namespace esphome {
namespace xxtea {
static const uint32_t DELTA = 0x9e3779b9;
#define MX ((((z >> 5) ^ (y << 2)) + ((y >> 3) ^ (z << 4))) ^ ((sum ^ y) + (k[(p ^ e) & 7] ^ z)))
void encrypt(uint32_t *v, size_t n, const uint32_t *k) {
uint32_t z, y, sum, e;
size_t p;
size_t q = 6 + 52 / n;
sum = 0;
z = v[n - 1];
while (q-- != 0) {
sum += DELTA;
e = (sum >> 2);
for (p = 0; p != n - 1; p++) {
y = v[p + 1];
z = v[p] += MX;
}
y = v[0];
z = v[n - 1] += MX;
}
}
void decrypt(uint32_t *v, size_t n, const uint32_t *k) {
uint32_t z, y, sum, e;
size_t p;
size_t q = 6 + 52 / n;
sum = q * DELTA;
y = v[0];
while (q-- != 0) {
e = (sum >> 2);
for (p = n - 1; p != 0; p--) {
z = v[p - 1];
y = v[p] -= MX;
}
z = v[n - 1];
y = v[0] -= MX;
sum -= DELTA;
}
}
} // namespace xxtea
} // namespace esphome

26
esphome/components/xxtea/xxtea.h Normal file
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@ -0,0 +1,26 @@
#pragma once
#include <cstdint>
#include <cstddef>
namespace esphome {
namespace xxtea {
/**
* Encrypt a block of data in-place using XXTEA algorithm with 256-bit key
* @param v Data to encrypt (as array of 32-bit words)
* @param n Number of 32-bit words in data
* @param k Key (array of 8 32-bit words)
*/
void encrypt(uint32_t *v, size_t n, const uint32_t *k);
/**
* Decrypt a block of data in-place using XXTEA algorithm with 256-bit key
* @param v Data to decrypt (as array of 32-bit words)
* @param n Number of 32-bit words in data
* @param k Key (array of 8 32-bit words)
*/
void decrypt(uint32_t *v, size_t n, const uint32_t *k);
} // namespace xxtea
} // namespace esphome

4
esphome/const.py
View File

@ -94,6 +94,7 @@ CONF_BRIGHTNESS = "brightness"
CONF_BRIGHTNESS_LIMITS = "brightness_limits"
CONF_BROKER = "broker"
CONF_BSSID = "bssid"
CONF_BUFFER_DURATION = "buffer_duration"
CONF_BUFFER_SIZE = "buffer_size"
CONF_BUILD_PATH = "build_path"
CONF_BUS_VOLTAGE = "bus_voltage"
@ -527,6 +528,7 @@ CONF_NAME_FONT = "name_font"
CONF_NBITS = "nbits"
CONF_NEC = "nec"
CONF_NETWORKS = "networks"
CONF_NEVER = "never"
CONF_NEW_PASSWORD = "new_password"
CONF_NITROGEN_DIOXIDE = "nitrogen_dioxide"
CONF_NOISE_LEVEL = "noise_level"
@ -615,6 +617,7 @@ CONF_OTA = "ota"
CONF_OUTDOOR_TEMPERATURE = "outdoor_temperature"
CONF_OUTPUT = "output"
CONF_OUTPUT_ID = "output_id"
CONF_OUTPUT_SPEAKER = "output_speaker"
CONF_OUTPUTS = "outputs"
CONF_OVERSAMPLING = "oversampling"
CONF_PACKAGES = "packages"
@ -859,6 +862,7 @@ CONF_TARGET_TEMPERATURE_LOW = "target_temperature_low"
CONF_TARGET_TEMPERATURE_LOW_COMMAND_TOPIC = "target_temperature_low_command_topic"
CONF_TARGET_TEMPERATURE_LOW_STATE_TOPIC = "target_temperature_low_state_topic"
CONF_TARGET_TEMPERATURE_STATE_TOPIC = "target_temperature_state_topic"
CONF_TASK_STACK_IN_PSRAM = "task_stack_in_psram"
CONF_TEMPERATURE = "temperature"
CONF_TEMPERATURE_COMPENSATION = "temperature_compensation"
CONF_TEMPERATURE_OFFSET = "temperature_offset"

7
esphome/core/__init__.py
View File

@ -689,7 +689,7 @@ class EsphomeCore:
_LOGGER.debug("Adding: %s", expression)
return expression
def add_global(self, expression):
def add_global(self, expression, prepend=False):
from esphome.cpp_generator import Expression, Statement, statement
if isinstance(expression, Expression):
@ -698,7 +698,10 @@ class EsphomeCore:
raise ValueError(
f"Add '{expression}' must be expression or statement, not {type(expression)}"
)
self.global_statements.append(expression)
if prepend:
self.global_statements.insert(0, expression)
else:
self.global_statements.append(expression)
_LOGGER.debug("Adding global: %s", expression)
return expression

17
esphome/core/config.py
View File

@ -72,6 +72,9 @@ def validate_hostname(config):
def valid_include(value):
# Look for "<...>" includes
if value.startswith("<") and value.endswith(">"):
return value
try:
return cv.directory(value)
except cv.Invalid:
@ -360,7 +363,19 @@ async def to_code(config):
CORE.add_job(add_arduino_global_workaround)
if config[CONF_INCLUDES]:
CORE.add_job(add_includes, config[CONF_INCLUDES])
# Get the <...> includes
system_includes = []
other_includes = []
for include in config[CONF_INCLUDES]:
if include.startswith("<") and include.endswith(">"):
system_includes.append(include)
else:
other_includes.append(include)
# <...> includes should be at the start
for include in system_includes:
cg.add_global(cg.RawStatement(f"#include {include}"), prepend=True)
# Other includes should be at the end
CORE.add_job(add_includes, other_includes)
if project_conf := config.get(CONF_PROJECT):
cg.add_define("ESPHOME_PROJECT_NAME", project_conf[CONF_NAME])

2
esphome/core/defines.h
View File

@ -16,6 +16,8 @@
// Feature flags
#define USE_ALARM_CONTROL_PANEL
#define USE_AUDIO_FLAC_SUPPORT
#define USE_AUDIO_MP3_SUPPORT
#define USE_API
#define USE_API_NOISE
#define USE_API_PLAINTEXT

7
esphome/core/gpio.h
View File

@ -53,6 +53,13 @@ class GPIOPin {
virtual void pin_mode(gpio::Flags flags) = 0;
/**
* @brief Retrieve GPIO pin flags.
*
* @return The GPIO flags describing the pin mode and properties.
*/
virtual gpio::Flags get_flags() const = 0;
virtual bool digital_read() = 0;
virtual void digital_write(bool value) = 0;

4
esphome/cpp_generator.py
View File

@ -588,9 +588,9 @@ def add(expression: Union[Expression, Statement]):
CORE.add(expression)
def add_global(expression: Union[SafeExpType, Statement]):
def add_global(expression: Union[SafeExpType, Statement], prepend: bool = False):
"""Add an expression to the codegen global storage (above setup())."""
CORE.add_global(expression)
CORE.add_global(expression, prepend)
def add_library(name: str, version: Optional[str], repository: Optional[str] = None):

4
platformio.ini
View File

@ -127,7 +127,8 @@ lib_deps =
ESPmDNS ; mdns (Arduino built-in)
DNSServer ; captive_portal (Arduino built-in)
esphome/ESP32-audioI2S@2.0.7 ; i2s_audio
droscy/esp_wireguard@0.4.2 ; wireguard
droscy/esp_wireguard@0.4.2 ; wireguard
esphome/esp-audio-libs@1.1.1 ; audio
build_flags =
${common:arduino.build_flags}
@ -148,6 +149,7 @@ lib_deps =
${common:idf.lib_deps}
droscy/esp_wireguard@0.4.2 ; wireguard
kahrendt/ESPMicroSpeechFeatures@1.1.0 ; micro_wake_word
esphome/esp-audio-libs@1.1.1 ; audio
build_flags =
${common:idf.build_flags}
-Wno-nonnull-compare

2
requirements.txt
View File

@ -14,7 +14,7 @@ esptool==4.7.0
click==8.1.7
esphome-dashboard==20241217.1
aioesphomeapi==24.6.2
zeroconf==0.132.2
zeroconf==0.143.0
puremagic==1.27
ruamel.yaml==0.18.6 # dashboard_import
glyphsets==1.0.0

11
tests/components/a02yyuw/common.yaml Normal file
View File

@ -0,0 +1,11 @@
uart:
- id: uart_a02yyuw
tx_pin: ${tx_pin}
rx_pin: ${rx_pin}
baud_rate: 9600
sensor:
- platform: a02yyuw
id: a02yyuw_sensor
name: a02yyuw Distance
uart_id: uart_a02yyuw

16
tests/components/a02yyuw/test.esp32-ard.yaml
View File

@ -1,13 +1,5 @@
uart:
- id: uart_a02yyuw
tx_pin:
number: 17
rx_pin:
number: 16
baud_rate: 9600
substitutions:
tx_pin: GPIO17
rx_pin: GPIO16
sensor:
- platform: a02yyuw
id: a02yyuw_sensor
name: a02yyuw Distance
uart_id: uart_a02yyuw
<<: !include common.yaml

16
tests/components/a02yyuw/test.esp32-c3-ard.yaml
View File

@ -1,13 +1,5 @@
uart:
- id: uart_a02yyuw
tx_pin:
number: 4
rx_pin:
number: 5
baud_rate: 9600
substitutions:
tx_pin: GPIO4
rx_pin: GPIO5
sensor:
- platform: a02yyuw
id: a02yyuw_sensor
name: a02yyuw Distance
uart_id: uart_a02yyuw
<<: !include common.yaml

16
tests/components/a02yyuw/test.esp32-c3-idf.yaml
View File

@ -1,13 +1,5 @@
uart:
- id: uart_a02yyuw
tx_pin:
number: 4
rx_pin:
number: 5
baud_rate: 9600
substitutions:
tx_pin: GPIO4
rx_pin: GPIO5
sensor:
- platform: a02yyuw
id: a02yyuw_sensor
name: a02yyuw Distance
uart_id: uart_a02yyuw
<<: !include common.yaml

16
tests/components/a02yyuw/test.esp32-idf.yaml
View File

@ -1,13 +1,5 @@
uart:
- id: uart_a02yyuw
tx_pin:
number: 17
rx_pin:
number: 16
baud_rate: 9600
substitutions:
tx_pin: GPIO17
rx_pin: GPIO16
sensor:
- platform: a02yyuw
id: a02yyuw_sensor
name: a02yyuw Distance
uart_id: uart_a02yyuw
<<: !include common.yaml

16
tests/components/a02yyuw/test.esp8266-ard.yaml
View File

@ -1,13 +1,5 @@
uart:
- id: uart_a02yyuw
tx_pin:
number: 4
rx_pin:
number: 5
baud_rate: 9600
substitutions:
tx_pin: GPIO4
rx_pin: GPIO5
sensor:
- platform: a02yyuw
id: a02yyuw_sensor
name: a02yyuw Distance
uart_id: uart_a02yyuw
<<: !include common.yaml

16
tests/components/a02yyuw/test.rp2040-ard.yaml
View File

@ -1,13 +1,5 @@
uart:
- id: uart_a02yyuw
tx_pin:
number: 4
rx_pin:
number: 5
baud_rate: 9600
substitutions:
tx_pin: GPIO4
rx_pin: GPIO5
sensor:
- platform: a02yyuw
id: a02yyuw_sensor
name: a02yyuw Distance
uart_id: uart_a02yyuw
<<: !include common.yaml

9
tests/components/a4988/common.yaml Normal file
View File

@ -0,0 +1,9 @@
stepper:
- platform: a4988
id: a4988_stepper
step_pin: ${step_pin}
dir_pin: ${dir_pin}
sleep_pin: ${sleep_pin}
max_speed: 250 steps/s
acceleration: 100 steps/s^2
deceleration: 200 steps/s^2

18
tests/components/a4988/test.esp32-ard.yaml
View File

@ -1,12 +1,6 @@
stepper:
- platform: a4988
id: a4988_stepper
step_pin:
number: 22
dir_pin:
number: 23
sleep_pin:
number: 25
max_speed: 250 steps/s
acceleration: 100 steps/s^2
deceleration: 200 steps/s^2
substitutions:
step_pin: GPIO22
dir_pin: GPIO23
sleep_pin: GPIO25
<<: !include common.yaml

18
tests/components/a4988/test.esp32-c3-ard.yaml
View File

@ -1,12 +1,6 @@
stepper:
- platform: a4988
id: a4988_stepper
step_pin:
number: 2
dir_pin:
number: 3
sleep_pin:
number: 5
max_speed: 250 steps/s
acceleration: 100 steps/s^2
deceleration: 200 steps/s^2
substitutions:
step_pin: GPIO2
dir_pin: GPIO3
sleep_pin: GPIO5
<<: !include common.yaml

18
tests/components/a4988/test.esp32-c3-idf.yaml
View File

@ -1,12 +1,6 @@
stepper:
- platform: a4988
id: a4988_stepper
step_pin:
number: 2
dir_pin:
number: 3
sleep_pin:
number: 5
max_speed: 250 steps/s
acceleration: 100 steps/s^2
deceleration: 200 steps/s^2
substitutions:
step_pin: GPIO2
dir_pin: GPIO3
sleep_pin: GPIO5
<<: !include common.yaml

18
tests/components/a4988/test.esp32-idf.yaml
View File

@ -1,12 +1,6 @@
stepper:
- platform: a4988
id: a4988_stepper
step_pin:
number: 22
dir_pin:
number: 23
sleep_pin:
number: 25
max_speed: 250 steps/s
acceleration: 100 steps/s^2
deceleration: 200 steps/s^2
substitutions:
step_pin: GPIO22
dir_pin: GPIO23
sleep_pin: GPIO25
<<: !include common.yaml

18
tests/components/a4988/test.esp8266-ard.yaml
View File

@ -1,12 +1,6 @@
stepper:
- platform: a4988
id: a4988_stepper
step_pin:
number: 1
dir_pin:
number: 2
sleep_pin:
number: 5
max_speed: 250 steps/s
acceleration: 100 steps/s^2
deceleration: 200 steps/s^2
substitutions:
step_pin: GPIO1
dir_pin: GPIO2
sleep_pin: GPIO5
<<: !include common.yaml

18
tests/components/a4988/test.rp2040-ard.yaml
View File

@ -1,12 +1,6 @@
stepper:
- platform: a4988
id: a4988_stepper
step_pin:
number: 2
dir_pin:
number: 3
sleep_pin:
number: 5
max_speed: 250 steps/s
acceleration: 100 steps/s^2
deceleration: 200 steps/s^2
substitutions:
step_pin: GPIO2
dir_pin: GPIO3
sleep_pin: GPIO5
<<: !include common.yaml

5
tests/components/ac_dimmer/common.yaml Normal file
View File

@ -0,0 +1,5 @@
output:
- platform: ac_dimmer
id: ac_dimmer_1
gate_pin: ${gate_pin}
zero_cross_pin: ${zero_cross_pin}

12
tests/components/ac_dimmer/test.esp32-ard.yaml
View File

@ -1,7 +1,5 @@
output:
- platform: ac_dimmer
id: ac_dimmer_1
gate_pin:
number: 12
zero_cross_pin:
number: 13
substitutions:
gate_pin: GPIO18
zero_cross_pin: GPIO19
<<: !include common.yaml

12
tests/components/ac_dimmer/test.esp32-c3-ard.yaml
View File

@ -1,7 +1,5 @@
output:
- platform: ac_dimmer
id: ac_dimmer_1
gate_pin:
number: 5
zero_cross_pin:
number: 6
substitutions:
gate_pin: GPIO5
zero_cross_pin: GPIO4
<<: !include common.yaml

12
tests/components/ac_dimmer/test.esp8266-ard.yaml
View File

@ -1,7 +1,5 @@
output:
- platform: ac_dimmer
id: ac_dimmer_1
gate_pin:
number: 5
zero_cross_pin:
number: 4
substitutions:
gate_pin: GPIO5
zero_cross_pin: GPIO4
<<: !include common.yaml

12
tests/components/ac_dimmer/test.rp2040-ard.yaml
View File

@ -1,7 +1,5 @@
output:
- platform: ac_dimmer
id: ac_dimmer_1
gate_pin:
number: 5
zero_cross_pin:
number: 6
substitutions:
gate_pin: GPIO5
zero_cross_pin: GPIO4
<<: !include common.yaml

2
tests/components/adc/test.esp32-c3-ard.yaml
View File

@ -2,4 +2,4 @@ sensor:
- platform: adc
id: my_sensor
pin: 4
attenuation: 11db
attenuation: 12db

2
tests/components/adc/test.esp32-s2-ard.yaml
View File

@ -2,4 +2,4 @@ sensor:
- platform: adc
id: my_sensor
pin: 1
attenuation: 11db
attenuation: 12db

2
tests/components/adc/test.esp32-s3-ard.yaml
View File

@ -2,4 +2,4 @@ sensor:
- platform: adc
id: my_sensor
pin: 1
attenuation: 11db
attenuation: 12db

14
tests/components/adc128s102/common.yaml Normal file
View File

@ -0,0 +1,14 @@
spi:
- id: spi_adc128s102
clk_pin: ${clk_pin}
mosi_pin: ${mosi_pin}
miso_pin: ${miso_pin}
adc128s102:
cs_pin: ${cs_pin}
id: adc128s102_adc
sensor:
- platform: adc128s102
id: adc128s102_channel_0
channel: 0

19
tests/components/adc128s102/test.esp32-ard.yaml
View File

@ -1,14 +1,7 @@
spi:
- id: spi_adc128s102
clk_pin: 16
mosi_pin: 17
miso_pin: 15
substitutions:
clk_pin: GPIO16
mosi_pin: GPIO17
miso_pin: GPIO15
cs_pin: GPIO12
adc128s102:
cs_pin: 12
id: adc128s102_adc
sensor:
- platform: adc128s102
id: adc128s102_channel_0
channel: 0
<<: !include common.yaml

19
tests/components/adc128s102/test.esp32-c3-ard.yaml
View File

@ -1,14 +1,7 @@
spi:
- id: spi_adc128s102
clk_pin: 6
mosi_pin: 7
miso_pin: 5
substitutions:
clk_pin: GPIO6
mosi_pin: GPIO7
miso_pin: GPIO5
cs_pin: GPIO2
adc128s102:
cs_pin: 8
id: adc128s102_adc
sensor:
- platform: adc128s102
id: adc128s102_channel_0
channel: 0
<<: !include common.yaml

19
tests/components/adc128s102/test.esp32-c3-idf.yaml
View File

@ -1,14 +1,7 @@
spi:
- id: spi_adc128s102
clk_pin: 6
mosi_pin: 7
miso_pin: 5
substitutions:
clk_pin: GPIO6
mosi_pin: GPIO7
miso_pin: GPIO5
cs_pin: GPIO2
adc128s102:
cs_pin: 8
id: adc128s102_adc
sensor:
- platform: adc128s102
id: adc128s102_channel_0
channel: 0
<<: !include common.yaml

19
tests/components/adc128s102/test.esp32-idf.yaml
View File

@ -1,14 +1,7 @@
spi:
- id: spi_adc128s102
clk_pin: 16
mosi_pin: 17
miso_pin: 15
substitutions:
clk_pin: GPIO16
mosi_pin: GPIO17
miso_pin: GPIO15
cs_pin: GPIO12
adc128s102:
cs_pin: 12
id: adc128s102_adc
sensor:
- platform: adc128s102
id: adc128s102_channel_0
channel: 0
<<: !include common.yaml

19
tests/components/adc128s102/test.esp8266-ard.yaml
View File

@ -1,14 +1,7 @@
spi:
- id: spi_adc128s102
clk_pin: 14
mosi_pin: 13
miso_pin: 12
substitutions:
clk_pin: GPIO14
mosi_pin: GPIO13
miso_pin: GPIO12
cs_pin: GPIO15
adc128s102:
cs_pin: 15
id: adc128s102_adc
sensor:
- platform: adc128s102
id: adc128s102_channel_0
channel: 0
<<: !include common.yaml

19
tests/components/adc128s102/test.rp2040-ard.yaml
View File

@ -1,14 +1,7 @@
spi:
- id: spi_adc128s102
clk_pin: 2
mosi_pin: 3
miso_pin: 4
substitutions:
clk_pin: GPIO2
mosi_pin: GPIO3
miso_pin: GPIO4
cs_pin: GPIO5
adc128s102:
cs_pin: 5
id: adc128s102_adc
sensor:
- platform: adc128s102
id: adc128s102_channel_0
channel: 0
<<: !include common.yaml

2
tests/components/addressable_light/test.esp32-c3-ard.yaml → tests/components/addressable_light/common-ard-esp32_rmt_led_strip.yaml
View File

@ -5,7 +5,7 @@ light:
chipset: ws2812
rgb_order: GRB
num_leds: 256
pin: 2
pin: ${pin}
rmt_channel: 0
display:

2
tests/components/addressable_light/test.esp32-ard.yaml → tests/components/addressable_light/common-ard-fastled.yaml
View File

@ -3,7 +3,7 @@ light:
id: led_matrix_32x8
name: led_matrix_32x8
chipset: WS2812B
pin: 2
pin: ${pin}
num_leds: 256
rgb_order: GRB
default_transition_length: 0s

2
tests/components/addressable_light/test.esp32-c3-idf.yaml → tests/components/addressable_light/common-idf-esp32_rmt_led_strip.yaml
View File

@ -5,7 +5,7 @@ light:
chipset: ws2812
rgb_order: GRB
num_leds: 256
pin: 2
pin: ${pin}
display:
- platform: addressable_light

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