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update configs
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@stas-sl
stas-sl committed Oct 8, 2024
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4 changes: 2 additions & 2 deletions README.md
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Expand Up @@ -30,8 +30,8 @@ i2s:

# according to datasheet when L/R pin is connected to GND,
# the mic should output its signal in the left channel,
# however in my experience it's the opposite: when I connect
# L/R to GND then the signal is in the right channel
# but on some boards I've encountered the opposite,
# so you can try both
channel: right # default: right

# right shift samples.
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2 changes: 1 addition & 1 deletion components/sound_level_meter/sound_level_meter.cpp
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Expand Up @@ -116,7 +116,7 @@ void SoundLevelMeter::task(void *param) {
if (process_count >= sr * (this_->update_interval_ / 1000.f)) {
auto t = uint32_t(float(process_time) / process_count * (sr / 1000.f));
auto cpu_util = float(process_time) / 1000 / this_->update_interval_;
ESP_LOGE(TAG, "CPU (Core %u) Utilization: %.1f %%", xPortGetCoreID(), cpu_util * 100);
ESP_LOGD(TAG, "CPU (Core %u) Utilization: %.1f %%", xPortGetCoreID(), cpu_util * 100);
process_time = process_count = 0;
}
}
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302 changes: 150 additions & 152 deletions configs/advanced-example-config.yaml
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Expand Up @@ -18,190 +18,187 @@ i2s:
bck_pin: 23
ws_pin: 18
din_pin: 19
sample_rate: 48000 # default: 48000
bits_per_sample: 32 # default: 32
dma_buf_count: 8 # default: 8
dma_buf_len: 256 # default: 256
use_apll: true # default: false
sample_rate: 48000 # default: 48000
bits_per_sample: 32 # default: 32
dma_buf_count: 8 # default: 8
dma_buf_len: 256 # default: 256
use_apll: true # default: false

# according to datasheet when L/R pin is connected to GND,
# the mic should output its signal in the left channel,
# however in my experience it's the opposite: when I connect
# L/R to GND then the signal is in the right channel
channel: right # default: right
# but on some boards I've encountered the opposite,
# so you can try both
channel: right # default: right

# right shift samples.
# for example if mic has 24 bit resolution, and
# i2s configured as 32 bits, then audio data will be aligned left (MSB)
# and LSB will be padded with zeros, so you might want to shift them right by 8 bits
bits_shift: 8 # default: 0
bits_shift: 8 # default: 0

sound_level_meter:
id: sound_level_meter1

# update_interval specifies over which interval to aggregate audio data
# you can specify default update_interval on top level, but you can also override
# it further by specifying it on sensor level
update_interval: 60s # default: 60s
update_interval: 60s # default: 60s

# you can disable (turn off) component by default (on boot)
# and turn it on later when needed via sound_level_meter.turn_on/toggle actions;
# when used with switch it might conflict/being overriden by
# switch state restoration logic, so you have to either disable it in
# switch config and then is_on property here will have effect,
# or completely rely on switch state restoration/initialization and
# switch config and then is_on property here will have effect,
# or completely rely on switch state restoration/initialization and
# any value set here will be ignored
is_on: true # default: true
is_on: true # default: true

# buffer_size is in samples (not bytes), so for float data type
# number of bytes will be buffer_size * 4
buffer_size: 1024 # default: 1024
buffer_size: 1024 # default: 1024

# ignore audio data at startup for this long
warmup_interval: 500ms # default: 500ms
warmup_interval: 500ms # default: 500ms

# audio processing runs in a separate task, you can change its settings below
task_stack_size: 4096 # default: 4096
task_priority: 2 # default: 2
task_core: 1 # default: 1
task_stack_size: 4096 # default: 4096

# idle task priority is 0,
# main esphome loop priority is 1,
# critical system tasks have priorities 18, 19, 20...
# I set the priority to 2, slightly higher than the main loop,
# because audio processing is highly sensitive to timing.
# Large delays from certain components could cause the
# DMA buffers to overflow, resulting in lost audio data.
task_priority: 2 # default: 2
task_core: 1 # default: 1

# see your mic datasheet to find sensitivity and reference SPL.
# those are used to convert dB FS to db SPL
mic_sensitivity: -26dB # default: empty
mic_sensitivity_ref: 94dB # default: empty
mic_sensitivity: -26dB # default: empty
mic_sensitivity_ref: 94dB # default: empty
# additional offset if needed
offset: 0dB # default: empty

# for flexibility sensors are organized hierarchically into groups. each group
# could have any number of filters, sensors and nested groups.
# for examples if there is a top level group A with filter A and nested group B
# with filter B, then for sensors inside group B filters A and then B will be
# applied:
# groups:
# # group A
# - filters:
# - filter A
# groups:
# # group B
# - filters:
# - filter B
# sensors:
# - sensor X
groups:
# group 1 (mic eq)
- filters:
# for now only SOS filter type is supported, see math/filter-design.ipynb
# to learn how to create or convert other filter types to SOS
- type: sos
coeffs:
# INMP441:
# b0 b1 b2 a1 a2
- [ 1.0019784 , -1.9908513 , 0.9889158 , -1.9951786 , 0.99518436]

# nested groups
groups:
# group 1.1 (no weighting)
- sensors:
# 'eq' type sensor calculates Leq (average) sound level over specified period
- type: eq
name: LZeq_1s
id: LZeq_1s
# you can override updated_interval specified on top level
# individually per each sensor
update_interval: 1s

# you can have as many sensors of same type, but with different
# other parameters (e.g. update_interval) as needed
- type: eq
name: LZeq_1min
id: LZeq_1min
unit_of_measurement: dBZ

# 'max' sensor type calculates Lmax with specified window_size.
# for example, if update_interval is 60s and window_size is 1s
# then it will calculate 60 Leq values for each second of audio data
# and the result will be max of them
- type: max
name: LZmax_1s_1min
id: LZmax_1s_1min
window_size: 1s
unit_of_measurement: dBZ

# same as 'max', but 'min'
- type: min
name: LZmin_1s_1min
id: LZmin_1s_1min
window_size: 1s
unit_of_measurement: dBZ

# it finds max single sample over whole update_interval
- type: peak
name: LZpeak_1min
id: LZpeak_1min
unit_of_measurement: dBZ

# group 1.2 (A-weighting)
- filters:
# for now only SOS filter type is supported, see math/filter-design.ipynb
# to learn how to create or convert other filter types to SOS
- type: sos
coeffs:
# A-weighting:
# b0 b1 b2 a1 a2
- [ 0.16999495 , 0.741029 , 0.52548885 , -0.11321865 , -0.056549273]
- [ 1. , -2.00027 , 1.0002706 , -0.03433284 , -0.79215795 ]
- [ 1. , -0.709303 , -0.29071867 , -1.9822421 , 0.9822986 ]
sensors:
- type: eq
name: LAeq_1min
id: LAeq_1min
unit_of_measurement: dBA
- type: max
name: LAmax_1s_1min
id: LAmax_1s_1min
window_size: 1s
unit_of_measurement: dBA
- type: min
name: LAmin_1s_1min
id: LAmin_1s_1min
window_size: 1s
unit_of_measurement: dBA
- type: peak
name: LApeak_1min
id: LApeak_1min
unit_of_measurement: dBA

# group 1.3 (C-weighting)
- filters:
# for now only SOS filter type is supported, see math/filter-design.ipynb
# to learn how to create or convert other filter types to SOS
- type: sos
coeffs:
# C-weighting:
# b0 b1 b2 a1 a2
- [-0.49651518 , -0.12296628 , -0.0076134163, -0.37165618 , 0.03453208 ]
- [ 1. , 1.3294908 , 0.44188643 , 1.2312505 , 0.37899444 ]
- [ 1. , -2. , 1. , -1.9946145 , 0.9946217 ]
sensors:
- type: eq
name: LCeq_1min
id: LCeq_1min
unit_of_measurement: dBC
- type: max
name: LCmax_1s_1min
id: LCmax_1s_1min
window_size: 1s
unit_of_measurement: dBC
- type: min
name: LCmin_1s_1min
id: LCmin_1s_1min
window_size: 1s
unit_of_measurement: dBC
- type: peak
name: LCpeak_1min
id: LCpeak_1min
unit_of_measurement: dBC

offset: 0dB # default: empty

# under dsp_filters section you can define multiple filters,
# which can be referenced later by each sensor

# for now only SOS filter type is supported, see math/filter-design.ipynb
# to learn how to create or convert other filter types to SOS
dsp_filters:
- id: f_inmp441 # INMP441 mic eq @ 48kHz
type: sos
coeffs:
# b0 b1 b2 a1 a2
- [1.0019784, -1.9908513, 0.9889158, -1.9951786, 0.99518436]
- id: f_a # A weighting @ 48kHz
type: sos
coeffs:
# b0 b1 b2 a1 a2
- [0.16999495, 0.741029, 0.52548885, -0.11321865, -0.056549273]
- [1., -2.00027, 1.0002706, -0.03433284, -0.79215795]
- [1., -0.709303, -0.29071867, -1.9822421, 0.9822986]
- id: f_c # C weighting @ 48kHz
type: sos
coeffs:
# b0 b1 b2 a1 a2
- [-0.49651518, -0.12296628, -0.0076134163, -0.37165618, 0.03453208]
- [1., 1.3294908, 0.44188643, 1.2312505, 0.37899444]
- [1., -2., 1., -1.9946145, 0.9946217]

sensors:
# 'eq' type sensor calculates Leq (average) sound level over specified period
- type: eq
name: LZeq_1s
id: LZeq_1s
# you can override updated_interval specified on top level
# individually per each sensor
update_interval: 1s

# you can have as many sensors of same type, but with different
# other parameters (e.g. update_interval) as needed
- type: eq
name: LZeq_1min
id: LZeq_1min
unit_of_measurement: dBZ

# 'max' sensor type calculates Lmax with specified window_size.
# for example, if update_interval is 60s and window_size is 1s
# then it will calculate 60 Leq values for each second of audio data
# and the result will be max of them
- type: max
name: LZmax_1s_1min
id: LZmax_1s_1min
window_size: 1s
unit_of_measurement: dBZ

# same as 'max', but 'min'
- type: min
name: LZmin_1s_1min
id: LZmin_1s_1min
window_size: 1s
unit_of_measurement: dBZ

# it finds max single sample over whole update_interval
- type: peak
name: LZpeak_1min
id: LZpeak_1min
unit_of_measurement: dBZ

- type: eq
name: LAeq_1min
id: LAeq_1min
unit_of_measurement: dBA

# The dsp_filters field is a list of filter IDs defined
# in the main component's corresponding section.
# If you're only using a single filter, you can omit the brackets.
# Alternatively, instead of referencing an existing filter,
# you can define a filter directly here in the same format
# as in the main component. However, this filter won't be reusable
# across other sensors, so it’s best to use this approach only when
# each sensor requires its own unique filters that don't overlap with others.
dsp_filters: [f_inmp441, f_a]
- type: max
name: LAmax_1s_1min
id: LAmax_1s_1min
window_size: 1s
unit_of_measurement: dBA
dsp_filters: [f_inmp441, f_a]
- type: min
name: LAmin_1s_1min
id: LAmin_1s_1min
window_size: 1s
unit_of_measurement: dBA
dsp_filters: [f_inmp441, f_a]
- type: peak
name: LApeak_1min
id: LApeak_1min
unit_of_measurement: dBA
dsp_filters: [f_inmp441, f_a]

- type: eq
name: LCeq_1min
id: LCeq_1min
unit_of_measurement: dBC
dsp_filters: [f_inmp441, f_c]
- type: max
name: LCmax_1s_1min
id: LCmax_1s_1min
window_size: 1s
unit_of_measurement: dBC
dsp_filters: [f_inmp441, f_c]
- type: min
name: LCmin_1s_1min
id: LCmin_1s_1min
window_size: 1s
unit_of_measurement: dBC
dsp_filters: [f_inmp441, f_c]
- type: peak
name: LCpeak_1min
id: LCpeak_1min
unit_of_measurement: dBC
dsp_filters: [f_inmp441, f_c]

# automation
# available actions:
Expand All @@ -211,6 +208,7 @@ sound_level_meter:
switch:
- platform: template
name: "Sound Level Meter Switch"
icon: mdi:power
# if you want is_on property on component to have effect, then set
# restore_mode to DISABLED, or alternatively you can use other modes
# (more on them in esphome docs), then is_on property on the component will
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