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[audio, i2s_audio, speaker] Media Player Components PR2 (#8164)

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Co-authored-by: Jesse Hills <3060199+jesserockz@users.noreply.github.com>
This commit is contained in:
Kevin Ahrendt 2025-02-02 20:25:41 -06:00 committed by GitHub
parent 2b711e532b
commit f6cf99384b
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9 changed files with 477 additions and 61 deletions
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112
esphome/components/audio/__init__.py
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@ -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

48
esphome/components/i2s_audio/speaker/__init__.py
View File

@ -1,20 +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_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,
@ -58,7 +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()
@ -90,6 +129,7 @@ BASE_SCHEMA = (
.extend(cv.COMPONENT_SCHEMA)
)
CONFIG_SCHEMA = cv.All(
cv.typed_schema(
{
@ -111,7 +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

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

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