Source code for devices.xylo.syns61201.afe_sim_empirical

"""
Simulation of an analog audio filtering front-end

Defines the :py:class:`.AFESim` module, with an empirically-validated simulation of the audio front-end on XyloAudio 2 (SYNS61201).

See Also:
    For example usage of the :py:class:`.AFESim` Module, see :ref:`/devices/analog-frontend-example.ipynb`
"""

# - Rockpool imports
from rockpool.nn.modules.module import Module
from rockpool.nn.modules.native.filter_bank import ButterFilter
from rockpool.timeseries import TSEvent, TSContinuous
from rockpool.parameters import State, SimulationParameter

# - Other imports
import numpy as np
from scipy.signal import butter, lfilter
from scipy import signal, fftpack
import enum

from typing import Tuple

from rockpool.typehints import P_int, P_float, P_bool, P_ndarray

from .afe_spike_generation import _encode_spikes

# Define exports
__all__ = ["AFESim"]


[docs]class AFESim(Module): """ A :py:class:`.Module` that simulates analog hardware for preprocessing audio and converting into spike features. This module simulates the XyloAudio 2 front-end stage. This is a signal-to-event core that consists of a number of band-pass filters, followed by rectifying event production simulating a spiking LIF neuron. The event rate in each channel is roughly correlated to the energy in each filter band. .. warning:: - The AFE contains frequency tripling internally. **For accurate simulation, the sampling frequency must be at least 6 times higher than the highest frequency component in the filtering chain.** This would be the centre frequency of the highest filter, plus half the BW of that signal. - **To prevent signal aliasing, you should apply a low-pass filter to restrict the bandwidth of the input, to ensure you don't exceed this target highest frequency.** - Input to the module is in Volts. **Input amplitude should be scaled to a maximum of 112mV RMS.** .. note:: - By default, the module simulates HW mismatch in the analog encoding block. This is controlled on instantiation with the ``add_noise``, ``add_offset`` and ``add_mismatch`` arguments on initialisation and the corresponding simulation parameters. - Mismatch can be re-sampled by calling the :py:meth:`~.AFESim.generate_mismatch` method. See Also: For example usage of the :py:class:`.AFESim` Module, see :ref:`/devices/analog-frontend-example.ipynb` """
[docs] def __init__( self, fs: int, raster_period: float = 0.01, max_spike_per_raster_period: int = 15, add_noise: bool = True, add_offset: bool = True, add_mismatch: bool = True, seed: int = np.random.randint(2**32 - 1, dtype=np.uint32), num_workers: int = 1, *args, **kwargs, ): """ Parameters ---------- fs: int The sampling frequency of the input, in Hz NOTE: (i) AFE contains frequency tripling in LNA and also in microphone. So the maximum representable frequency is ``fs/6``. (ii) If this frequency is different than the sampling frequency of the audio, the frequencies are proportionally shifted and extracted spikes features will be wrong. raster_period (float): this is the period to which the spikes are rastered and counted. Default 10ms. max_spike_per_raster_period (int): maximum number of spikes in each rastering period. This is equal to 15 spikes in Xylo-A2. add_noise: bool Enables / disables the simulated noise generated by the AFE. Default: ``True``, include noise add_offset: bool If ``True`` (default), add mismatch offset to each filter add_mismatch: bool If ``True`` (default), add simualted mismatch to each filter seed: int The AFE is subject to mismatch, this can be seeded by providing an integer seed. Default: random seed. Provide ``None`` to prevent seeding. num_workers: int Number of cpu units used to speed up filter computation. Default: 1. """ ###### Check shape argument and Initialize the superclass ###### shape = (1, 16) super().__init__(shape=shape, spiking_output=True, *args, **kwargs) ## Provide pRNG seed self.seed: P_int = SimulationParameter(seed, shape=(), init_func=lambda _: None) if self.seed is not None: np.random.seed(self.seed) ##### Set the rastering parameters for the produced spike ##### self.raster_period = raster_period self.max_spike_per_raster_period = max_spike_per_raster_period ###### Power supply features and maximum voltage of the chip ###### # Maximum bipolar amplitude that can be supported by the the chip without being clipped # NOTE: in the chip, we have a voltage supply in the range [0, 1.1] volts where the refernce voltages of LNA and filters is # shifted to the middle value of 0.55 volts. # So, in terms of signal processing, we can always assume that the input signal is a bipolar one in the range [-0.55, 0.55] with clipping # if the amplitude goes beyond this range self.VCC: P_float = SimulationParameter(1.1) # in volts self.INPUT_MAX_AMPLITUDE: P_float = SimulationParameter( self.VCC / 2.0 ) # in volts ###### microphone fetaures ###### # NOTE: microphone has severe THD for sound level above 120 dBL SPL, which is 20 Pa pressure. # Microphone has a sensitivity of around -40 dB in units of Volt/Pa which will be 10 mV/Pa. # For maximum 20 Pa pressure, this yields a maximum amplitude of 200 mV. self.MIC_DISTORTION: P_float = SimulationParameter(0.01) self.INPUT_MIC_MAX_AMPLITUDE: P_float = SimulationParameter(200e-3) # 200mV """ float: Maximum amplitude of sound that can be produced with microphone without having sever THD (above 1%)""" ###### from microphone to LNA ###### self.MAX_INPUT_OFFSET: P_float = SimulationParameter(0.0) # from microphone """ float: Maxmimum input offset from microphone (Default 0.) """ # Corner frequency of the AC coupling between the microphone and LNA self.F_CORNER_HIGHPASS: P_float = SimulationParameter(20) """ float: High pass corner frequency due to AC Coupling from BPF to FWR in Hz. (Default 20 Hz)""" ###### LNA features ###### # the linear regime of LNA with a given nonlinearity threshold # NOTE: in Xylo-A2 the microphone has a very low sensitivity and the maximum voltage it can produce is 200 mV where beyond that # THD is very large. # For this reson, the LNA distortion would be almost negiligible compared with that of the microphone. self.LNA_DISTORTION: P_float = SimulationParameter(0.01) self.INPUT_LNA_MAX_AMPLITUDE: P_float = SimulationParameter( self.INPUT_MAX_AMPLITUDE * 0.8 ) """ float: LNA Distortion parameter when the amplitude goes beyond its linear regime. Default 0.01 """ # LNA gain # possible gains in LNA: it is possible to adjust LNA by to have an amplification of order 2 or 4 class LNA_GAIN(enum.Enum): G0dB = 0.0 # in dB G6dB = 6.0 # in dB G12dB = 12.0 # in dB self.lna_gain_db: P_float = SimulationParameter(LNA_GAIN.G0dB.value) # in dB """ float: Low-noise amplifer gain in dB (Default 0.) """ self.MAX_LNA_OFFSET: P_float = SimulationParameter(5.0e-3) # +/-5mV random """ float: Maxmimum low-noise amplifier offset in mV (Default 5mV) """ ###### filterbank features ###### # - Parameters for BPF self.Q: P_int = SimulationParameter(4) """ int: Quality parameter for band-pass filters """ self.Qs: np.ndarray = self.Q * np.ones(self.size_out) """ np.ndarray: Q factors for each filter """ # - sampling frequency of the incoming audio signal self.Fs: P_float = SimulationParameter(fs) """ float: Sample frequency of input data """ # - nominal/design values of the filters center frequencies and bandwidths self.design_fcs: P_ndarray = SimulationParameter( np.asarray( [ 40, 54, 77, 137, 203, 290, 428, 674, 1177, 1700, 2226, 3418, 5154, 7884, 11630, 16940, ] ) ) """ np.ndarray: Centre frequency of each band-pass filter in Hz """ self.fcs: np.ndarray = self.design_fcs """ np.ndarray: Actual centre frequencies of each band-pass filter in Hz """ self.bws: np.ndarray = self.fcs / self.Qs """ np.ndarray: Actual bandwidth for each filter """ # - Check the filters w.r.t the sampling frequency if self.Fs < (6 * np.max(self.design_fcs)): raise ValueError( f"""Sampling frequency ({self.Fs}) must be at least 6 times the highest BPF centre freq. (i.e. >{6 * np.max(self.design_fcs)} Hz) The main reason is that the microphone produces THD (third-order distortion) which may fallback into the wrong frequency if the sampling frequency is not large enough. """ ) # - nominal/design Bandwidths of the filters self.design_bws: P_ndarray = SimulationParameter(self.design_fcs / self.Q) """ np.ndarray: Bandwidths of each filter in Hz """ # - order of the Butterworth BPF used in the filterbank self.ORDER_BPF: P_int = SimulationParameter(2) """ int: Band-pass filter order (Default 2)""" self.MAX_BPF_OFFSET: P_float = SimulationParameter(5.0e-3) # +/-5mV random """ float: Maxmum band-pass filter offset in mV (Default 5mV)""" self.BPF_FC_SHIFT: P_float = SimulationParameter( -5e-2 ) # 5 for +5% -5 for -5% NOTE: 16 channels center freq shift in the same direction """ float: Centre frequency band-pass filter shift in % (Default -5%) """ self.Q_MIS_MATCH: P_float = SimulationParameter(10e-2) # +/-10% random """ float: Mismatch in Q in % (Default 10%) """ self.FC_MIS_MATCH: P_float = SimulationParameter(5e-2) # +/-5% random """ float: Mismatch in centre freq. in % (Default 5%)""" ###### spike generation features ###### # capacitor in LIF circuit used for integration and spike generation self.C_IAF: P_float = SimulationParameter(5e-12) # 5 pF """ float: Integrator Capacitance for IAF (Default 5e-12)""" # V2I module gain: how the input voltage is converted to current for integration and spike generation # this is set to be I=2e-8 for a max voltage of V=60mv self.V2I_GIAN: P_float = SimulationParameter(0.333e-6) self.LEAKAGE: P_float = SimulationParameter(1e-9) """ float: Leakage conductance for LIF neuron producing the spikes. Default: 1.0e-9 : 1.0 nA for a cpacitor at volatage 1.0V """ self.DIGITAL_COUNTER: P_int = SimulationParameter(4) """ int: Digital counter factor to reduce output spikes by. Default 4 (by a factor 4) """ # Threshold for spike generation using IAF (essentially LIF) neuron self.THR_UP: P_float = SimulationParameter( 0.5 ) # 0.1-0.9 V to be in the linear regime of system. """ float: Threshold for delta modulation in V (0.1--0.9) (Default 0.5V)""" ##### Other settings for simulation ##### self.add_noise: P_bool = SimulationParameter(add_noise) """ bool: Flag indicating that noise should be simulated during operation. Default `True` """ self.add_offset: P_bool = SimulationParameter(add_offset) """ bool: Flag indicating that offset should be simulated during operation. Default `True` """ self.add_mismatch: P_bool = SimulationParameter(add_mismatch) """ bool: Flag indicating that mismatch in the parameters should be simulated during operation. Default `True` """ self.num_workers: P_int = SimulationParameter(num_workers) """ int: number of independent CPU units used for simulating the filters in the filterbank. Default 1""" ### Macro definitions related to noise ### self.VRMS_SQHZ_LNA: P_float = SimulationParameter(70e-9) self.F_KNEE_LNA: P_float = SimulationParameter(70e3) self.F_ALPHA_LNA: P_float = SimulationParameter(1) self.VRMS_SQHZ_BPF: P_float = SimulationParameter(1e-9) self.F_KNEE_BPF: P_float = SimulationParameter(100e3) self.F_ALPHA_BPF: P_float = SimulationParameter(1) self.VRMS_SQHZ_FWR: P_float = SimulationParameter(700e-9) self.F_KNEE_FWR: P_float = SimulationParameter(158) self.F_ALPHA_FWR: P_float = SimulationParameter(1) ### Initialise mismatch parameters self.input_offset: P_float = SimulationParameter(0.0) """ float: Mismatch offset in the signal comming from microphone -- typically 0 due to AC coupling """ self.lna_offset: P_float = SimulationParameter(0.0) """ float: Mismatch offset in low-noise amplifier """ self.bpf_offset: P_ndarray = SimulationParameter(np.zeros(self.size_out)) """ float: Mismatch offset in band-pass filters """ self.Q_mismatch: P_ndarray = SimulationParameter(np.zeros(self.size_out)) """ float: Mismatch in Q over band-pass filters """ self.fc_mismatch: P_ndarray = SimulationParameter(np.zeros(self.size_out)) """ float: Mismatch in centre frequency for band-pass filters """ self.bpf_fc_shift: P_float = SimulationParameter(0.0) """ float: Common shift in center frequencies due to temperature, etc. """ # - Event generation state self.lif_state: P_ndarray = State(np.zeros(self.size_out)) """ (np.ndarray) Internal state of the LIF neurons used to generate events """ # Initialize chip parameters, generate mismatch if required self.generate_mismatch()
[docs] def generate_mismatch(self): """ This function generates mismatch for the parameters of AFESim, based on analog non-idealities. It may be called: (i) once for all inputs simulated if we are interested in a single chip for all simulations. (ii) once for each input if we would like some sort of augmentation w.r.t. chip non-idealities. """ self.input_offset = ( self.MAX_INPUT_OFFSET * (2 * np.random.rand(1).item() - 1.0) if self.add_offset else 0.0 ) self.lna_offset = ( self.MAX_LNA_OFFSET * (2 * np.random.rand(1).item() - 1.0) if self.add_offset else 0.0 ) self.bpf_offset = ( self.MAX_BPF_OFFSET * (2 * np.random.rand(self.size_out) - 1.0) if self.add_offset else np.zeros(self.size_out) ) self.Q_mismatch = ( self.Q_MIS_MATCH * (2 * np.random.rand(self.size_out) - 1.0) if self.add_mismatch else np.zeros(self.size_out) ) self.fc_mismatch = ( self.FC_MIS_MATCH * (2 * np.random.rand(self.size_out) - 1.0) if self.add_mismatch else np.zeros(self.size_out) ) self.bpf_fc_shift = ( self.BPF_FC_SHIFT * (2 * np.random.rand(1).item() - 1.0) if self.add_mismatch else 0.0 ) # Shift center frequencies and bandwidths based on mismatch parameters self.fcs = np.asarray( [ freq * (1.0 + mismatch) for (freq, mismatch) in zip(self.design_fcs, self.fc_mismatch) ] ) # Shift the center frequencies together by mismatch factor self.fcs = np.asarray([freq * (1.0 + self.bpf_fc_shift) for freq in self.fcs]) # Produce Q for the filters self.Qs = np.asarray( [self.Q * (1.0 + mismatch) for mismatch in self.Q_mismatch] ) # Produce bandwidths # self.bws = np.asarray([freq / Q for (freq, Q) in zip(self.fcs, self.Qs)]) self.bws = self.fcs / self.Qs # - Re-generate the filterbank module self._butter_filterbank = ButterFilter( frequency=self.fcs, bandwidth=self.bws, fs=self.Fs, order=self.ORDER_BPF, num_workers=self.num_workers, use_lowpass=False, ) # - High-pass filter parameters (for AC coupling between microphone and LNA) self._HP_filt = self._butter_highpass(self.F_CORNER_HIGHPASS, self.Fs, order=1) """ High-pass filter on input """ # - Reset internal neuron state self.lif_state: P_ndarray = State(np.zeros(self.size_out)) """ (np.ndarray) Internal state of the LIF neurons used to generate events """
##### Utility functions: filters #####
[docs] def _butter_bandpass( self, lowcut: float, highcut: float, fs: float, order: int = 2 ) -> Tuple[float, float]: """ Build a Butterworth bandpass filter from specification Args: lowcut (float): Low-cut frequency in Hz highcut (float): High-cut frequency in Hz fs (float): Sampling frequecy in Hz order (int): Order of the filter Returns: (float, float): b, a Parameters for the bandpass filter """ nyq = 0.5 * fs low = lowcut / nyq high = highcut / nyq b, a = butter(order, [low, high], btype="band", output="ba") return b, a
[docs] def _butter_bandpass_filter( self, data: np.ndarray, lowcut: float, highcut: float, fs: float, order: int = 2 ) -> np.ndarray: """ Filter data with a bandpass Butterworth filter, according to specifications Args: data (np.ndarray): Input data with shape ``(T, N)`` lowcut (float): Low-cut frequency in Hz highcut (float): High-cut frequency in Hz fs (float): Sampling frequency in Hz order (int): Order of the filter Returns: np.ndarray: Filtered data with shape ``(T, N)`` """ b, a = self._butter_bandpass(lowcut, highcut, fs, order=order) y = lfilter(b, a, data) return y
[docs] def _butter_highpass( self, cutoff: float, fs: float, order: int = 1 ) -> Tuple[float, float]: """ Build a Butterworth high-pass filter from specifications Args: cutoff (float): High-pass cutoff frequency in Hz fs (float): Sampling rate in Hz order (int): Order of the filter Returns: (float, float): b, a Parameters for the high-pass filter """ nyq = 0.5 * fs normal_cutoff = cutoff / nyq b, a = butter(order, normal_cutoff, btype="high", analog=False) return b, a
[docs] def _butter_highpass_filter( self, data: np.ndarray, cutoff: float, fs: float, order: int = 1 ) -> np.ndarray: """ Filter some data with a Butterworth high-pass filter from specifications Args: data (np.ndarray): Array of input data to filter, with shape ``(T, N)`` cutoff (float): Cutoff frequency of the high-pass filter, in Hz fs (float): Sampling frequency of ``data``, in Hz order (int): Order of the Butterwoth filter Returns: np.ndarray: Filtered output data with shape ``(T, N)`` """ b, a = self._butter_highpass(cutoff, fs, order=order) y = signal.filtfilt(b, a, data) return y
#### Utility functions: noise ####
[docs] def _generateNoise( self, T, Fs: float = 16e3, VRMS_SQHZ: float = 1e-6, F_KNEE: float = 1e3, F_ALPHA: float = 1.4, ) -> np.ndarray: """ Generate band-limited noise, for use in simulating the AFE architecture Args: x (np.ndarray): Input signal defining desired shape of noise ``(T,)`` Fs (float): Sampling frequency in Hz VRMS_SQHZ (float): F_KNEE (float): F_ALPHA (float): Returns: np.ndarray: Generated noise with shape ``(T,)`` """ def one_over_f(f: np.ndarray, knee: float, alpha: float) -> np.ndarray: d = np.ones_like(f) f = np.clip(f, 1e-12, np.inf) d[f < knee] = np.abs(((knee / f[f < knee]) ** (alpha))) d[0] = 1 return d W_NOISE_SIGMA = VRMS_SQHZ * np.sqrt(Fs / 2) # Noise in the bandwidth 0 - Fs/2 wn = np.random.normal(0, W_NOISE_SIGMA, T) s = fftpack.rfft(wn) f = fftpack.rfftfreq(len(s)) * Fs ff = s * one_over_f(f, F_KNEE, F_ALPHA) x_t = fftpack.irfft(ff) return x_t
#### Utility functions: spike generation #####
[docs] def _sampling_spikes(self, spikes: np.ndarray, count: int) -> np.ndarray: """ Down-sample events in a signal, by passing one in every ``N`` events Args: spikes (np.ndarray): Raster ``(T, N)`` of events count (int): Number of events to ignore before passing one event Returns: np.ndarray: Raster ``(T, N)`` of down-sampled events """ return (np.cumsum(spikes, axis=0) % count * spikes) == (count - 1)
#### Utility functions: modelling the distortion #####
[docs] def _MIC_evolve( self, sig_in: np.ndarray, v_corner: float, THD_level, max_output: float ): """this function incorporates the effect of third-order distortion in the input audio signal. Args: sig_in (np.ndarray): 1-dim input signal. v_corner (float): voltage level at which THD is equal to THD_level. THD_level (float, optional): level of third-order distortion. max_output (float): maximum signal amplitude in the whole chip. """ # we use the simple formula sin(3 th) = 3 sin(th) - 4 sin^3(th) for simulating THD distortion = ( THD_level * v_corner * (3 * sig_in / v_corner - 4 * (sig_in / v_corner) ** 3) ) sig_out = sig_in + distortion sig_out[sig_out > max_output] = max_output sig_out[sig_out < -max_output] = -max_output return sig_out
[docs] def _LNA_evolve( self, sig_in: np.ndarray, v_corner: float, lna_distortion_level: float, max_output: float, ): """this function takes the nonlinearity due to LNA into account. Args: sig_in (np.ndarray): input signal received from the microphone. v_corner (float): the maximum linear range of the LNA. lna_distortion_level (float): the nonlinear distortion level added by LNA. max_output (float): maximum signal amplitude in the whole chip. """ # input signal is amplified by LNA lna_gain = 2 ** (self.lna_gain_db / 6.0) sig_out = lna_gain * (sig_in + self.lna_offset) # for a symmetric LNA, the main contribution of distortion is due to 3rd order nonlinearity lna_out = v_corner * ( sig_out / v_corner - lna_distortion_level * (sig_out / v_corner) ** 3 ) # truncate the amplitude when the LNA goes into the saturation regime lna_out[lna_out > max_output] = max_output lna_out[lna_out < -max_output] = -max_output return lna_out
#### Utility functions: state representation ##### @property def dt(self) -> float: """ Simulation time-step in seconds Returns: float: Simulation time-step """ return 1 / self.Fs
[docs] def _wrap_recorded_state(self, state_dict: dict, t_start: float = 0.0) -> dict: args = {"dt": self.dt, "t_start": t_start} return { "LNA_out": TSContinuous.from_clocked( state_dict["LNA_out"], name="LNA", **args ), "BPF": TSContinuous.from_clocked(state_dict["BPF"], name="BPF", **args), "rect": TSContinuous.from_clocked(state_dict["rect"], name="Rect", **args), "spks_out": TSEvent.from_raster( state_dict["spks_out"], name="Spikes", num_channels=self.size_out, **args, ), }
[docs] def evolve( self, input: np.ndarray, record: bool = False, *args, **kwargs, ) -> Tuple[np.ndarray, dict, dict]: """Evolve AFESim and return the generated spikes Args: input (np.ndarray, optional): input audio signal. record (bool, optional): Record the internal state of AFESim during evolution. Defaults to ``False``. Raises: ValueError: If the input data is not 1D Returns: np.ndarray: Output events (T, 16), where each bin contains the count of events in that bin. dict: Internal state of the module at the end of evolution dict: Recording dictionary if requested, otherwise an empty dictionary """ input, _ = self._auto_batch(input) Nb, Nt, Nc = input.shape # - Make sure input is 1D if Nb > 1 or Nc > 1: raise ValueError("the input signal should be 1-dim.") input = input[0, :, 0] # -- Revert and repeat the input signal in the beginning to avoid boundary effects l = np.shape(input)[0] __input_rev = np.flip(input, axis=0) input = np.concatenate((__input_rev, input), axis=0) #### Microphone model #### mic_out = self._MIC_evolve( sig_in=input, v_corner=self.INPUT_MIC_MAX_AMPLITUDE, THD_level=self.MIC_DISTORTION, max_output=self.INPUT_MAX_AMPLITUDE, ) if self.add_offset: mic_out += self.input_offset #### LNA - Gain #### lna_out = self._LNA_evolve( sig_in=mic_out, v_corner=self.INPUT_LNA_MAX_AMPLITUDE, lna_distortion_level=self.LNA_DISTORTION, max_output=self.INPUT_MAX_AMPLITUDE, ) if self.add_offset: lna_out += self.lna_offset #### Add Noise ##### if self.add_noise: noise = self._generateNoise( input.shape[0], self.Fs, self.VRMS_SQHZ_LNA, self.F_KNEE_LNA, self.F_ALPHA_LNA, ) lna_out += noise #### filterbank processing #### # - Expand lna_output dimensions and add offset filter_in = np.tile(np.atleast_2d(lna_out).T, (1, self.size_out)) if self.add_offset: filter_in += self.bpf_offset # - Perform the filtering filtered, _, _ = self._butter_filterbank(filter_in) # add noise if self.add_noise: for i in range(self.size_out): filtered[:, i] += self._generateNoise( input.shape[0], self.Fs, self.VRMS_SQHZ_BPF, self.F_KNEE_BPF, self.F_ALPHA_BPF, ) # - HP filt, additional noise, rectify # NOTE: HP filter should be essentially at the input where the AC coupling between microphone and LNA lies. # However, we can also do it at the output together with rectifier rectified = np.zeros_like(filtered) for i in range(self.size_out): rectified[:, i] = abs( signal.filtfilt(*self._HP_filt, filtered[:, i]) # rectified[:, i] + self._generateNoise( input.shape[0], self.Fs, self.VRMS_SQHZ_FWR, self.F_KNEE_FWR, self.F_ALPHA_FWR, ) ) # removing the part of the signal coresponding to __input_rev (which was added to avoid sharo boundary effects) rectified = rectified[l:, :] filtered = filtered[l:, :] # Encoding to spike by integrating the FWR output for positive going(UP) spikes, new_state = _encode_spikes( initial_state=self.lif_state, dt=self.dt, data=rectified, v2i_gain=self.V2I_GIAN, c_iaf=self.C_IAF, leakage=self.LEAKAGE, thr_up=self.THR_UP, vcc=self.VCC, ) # - Keep a record of the LIF neuron states self.lif_state = new_state if self.DIGITAL_COUNTER > 1: spikes = self._sampling_spikes(spikes, self.DIGITAL_COUNTER) recording = ( { "LNA_out": lna_out, "BPF": filtered, "rect": rectified, "spks_out": spikes, } if record else {} ) return spikes, self.state(), recording
[docs] def raster(self, spikes: np.ndarray): """ Rasterise the produced spikes within the rastering period Args: spikes (np.ndarray): input spikes of dimension `T x F` where `F`: number of filters. """ # convert spikes into numpy: this is due to having jax.numpy array when jax is active spikes = np.asarray(spikes) # number of clocks within a rastering period num_clk_per_period = self.raster_period / self.dt # faster method if num_clk_per_period == int(num_clk_per_period): # sum of spikes during several clocks spike_sum = np.cumsum(spikes, axis=0)[:: int(num_clk_per_period), :] # number of spikes colected in rastering periods spike_sum[1:, :] -= spike_sum[:-1, :] # truncate the number of spikes spike_sum[ spike_sum > self.max_spike_per_raster_period ] = self.max_spike_per_raster_period return spike_sum # use rockpool rastering: slower spike_sum = TSEvent.from_raster(spikes, dt=self.dt).raster( dt=self.raster_period, add_events=True ) spike_sum[ spike_sum > self.max_spike_per_raster_period ] = self.max_spike_per_raster_period return spike_sum