sonusai 0.18.2__py3-none-any.whl → 0.18.4__py3-none-any.whl

sonusai/__init__.py

@@ -10,6 +10,7 @@ commands_doc = """
    calc_metric_spenh            Run speech enhancement and analysis
    doc                          Documentation
    genft                        Generate feature and truth data
+   genmetrics                   Generate mixture metrics data
    genmix                       Generate mixture and truth data
    genmixdb                     Generate a mixture database
    gentcst                      Generate target configuration from a subdirectory tree

sonusai/audiofe.py

@@ -142,7 +142,7 @@ def main() -> None:
         if hparams is None:
             logger.error(f'Error: ONNX model does not have required SonusAI hyperparameters, cannot proceed.')
             raise SystemExit(1)
-        feature_mode = hparams.feature
+        feature_mode = hparams['feature']
         in0name = sess_inputs[0].name
         in0type = sess_inputs[0].type
         out_names = [n.name for n in session.get_outputs()]

sonusai/calc_metric_spenh.py

@@ -24,7 +24,7 @@ For whisper ASR methods, the possible models used in local processing (ASR = whi
     {tiny.en,tiny,base.en,base,small.en,small,medium.en,medium,large-v1,large-v2,large}
 but note most are very computationally demanding and can overwhelm/hang a local system.
 
-Outputs the following to PLOC (where id is mixd number 0:num_mixtures):
+Outputs the following to PLOC (where id is mixid number 0:num_mixtures):
     <id>_metric_spenh.txt
 
     If --plot:
@@ -74,6 +74,9 @@ from sonusai.mixture import Feature
 from sonusai.mixture import MixtureDatabase
 from sonusai.mixture import Predict
 
+DB_99 = np.power(10, 99 / 10)
+DB_N99 = np.power(10, -99 / 10)
+
 
 def signal_handler(_sig, _frame):
     import sys
@@ -122,298 +125,6 @@ def power_uncompress(spec):
     return real_uncompress + 1j * imag_uncompress
 
 
-def snr(clean_speech, processed_speech, sample_rate):
-    # Check the length of the clean and processed speech. Must be the same.
-    clean_length = len(clean_speech)
-    processed_length = len(processed_speech)
-    if clean_length != processed_length:
-        raise ValueError('Both Speech Files must be same length.')
-
-    overall_snr = 10 * np.log10(np.sum(np.square(clean_speech)) / np.sum(np.square(clean_speech - processed_speech)))
-
-    # Global Variables
-    win_length = round(30 * sample_rate / 1000)  # window length in samples
-    skip_rate = int(np.floor(win_length / 4))  # window skip in samples
-    min_snr = -10  # minimum SNR in dB
-    max_snr = 35  # maximum SNR in dB
-
-    # For each frame of input speech, calculate the Segmental SNR
-    num_frames = int(clean_length / skip_rate - (win_length / skip_rate))  # number of frames
-    start = 0  # starting sample
-    window = 0.5 * (1 - np.cos(2 * np.pi * np.arange(1, win_length + 1) / (win_length + 1)))
-
-    segmental_snr = np.empty(num_frames)
-    eps = np.spacing(1)
-    for frame_count in range(num_frames):
-        # (1) Get the Frames for the test and reference speech. Multiply by Hanning Window.
-        clean_frame = clean_speech[start:start + win_length]
-        processed_frame = processed_speech[start:start + win_length]
-        clean_frame = np.multiply(clean_frame, window)
-        processed_frame = np.multiply(processed_frame, window)
-
-        # (2) Compute the Segmental SNR
-        signal_energy = np.sum(np.square(clean_frame))
-        noise_energy = np.sum(np.square(clean_frame - processed_frame))
-        segmental_snr[frame_count] = 10 * np.log10(signal_energy / (noise_energy + eps) + eps)
-        segmental_snr[frame_count] = max(segmental_snr[frame_count], min_snr)
-        segmental_snr[frame_count] = min(segmental_snr[frame_count], max_snr)
-
-        start = start + skip_rate
-
-    return overall_snr, segmental_snr
-
-
-def lp_coefficients(speech_frame, model_order):
-    # (1) Compute Autocorrelation Lags
-    win_length = np.size(speech_frame)
-    autocorrelation = np.empty(model_order + 1)
-    e = np.empty(model_order + 1)
-    for k in range(model_order + 1):
-        autocorrelation[k] = np.dot(speech_frame[0:win_length - k], speech_frame[k: win_length])
-
-    # (2) Levinson-Durbin
-    a = np.ones(model_order)
-    a_past = np.empty(model_order)
-    ref_coefficients = np.empty(model_order)
-    e[0] = autocorrelation[0]
-    for i in range(model_order):
-        a_past[0: i] = a[0: i]
-        sum_term = np.dot(a_past[0: i], autocorrelation[i:0:-1])
-        ref_coefficients[i] = (autocorrelation[i + 1] - sum_term) / e[i]
-        a[i] = ref_coefficients[i]
-        if i == 0:
-            a[0: i] = a_past[0: i] - np.multiply(a_past[i - 1:-1:-1], ref_coefficients[i])
-        else:
-            a[0: i] = a_past[0: i] - np.multiply(a_past[i - 1::-1], ref_coefficients[i])
-        e[i + 1] = (1 - ref_coefficients[i] * ref_coefficients[i]) * e[i]
-    lp_params = np.concatenate((np.array([1]), -a))
-    return autocorrelation, ref_coefficients, lp_params
-
-
-def llr(clean_speech, processed_speech, sample_rate):
-    from scipy.linalg import toeplitz
-
-    # Check the length of the clean and processed speech.  Must be the same.
-    clean_length = np.size(clean_speech)
-    processed_length = np.size(processed_speech)
-    if clean_length != processed_length:
-        raise ValueError('Both speech files must be same length.')
-
-    # Global Variables
-    win_length = (np.round(30 * sample_rate / 1000)).astype(int)  # window length in samples
-    skip_rate = (np.floor(win_length / 4)).astype(int)  # window skip in samples
-    if sample_rate < 10000:
-        p = 10  # LPC Analysis Order
-    else:
-        p = 16  # this could vary depending on sampling frequency.
-
-    # For each frame of input speech, calculate the Log Likelihood Ratio
-    num_frames = int((clean_length - win_length) / skip_rate)  # number of frames
-    start = 0  # starting sample
-    window = 0.5 * (1 - np.cos(2 * np.pi * np.arange(1, win_length + 1) / (win_length + 1)))
-
-    distortion = np.empty(num_frames)
-    for frame_count in range(num_frames):
-        # (1) Get the Frames for the test and reference speech. Multiply by Hanning Window.
-        clean_frame = clean_speech[start: start + win_length]
-        processed_frame = processed_speech[start: start + win_length]
-        clean_frame = np.multiply(clean_frame, window)
-        processed_frame = np.multiply(processed_frame, window)
-
-        # (2) Get the autocorrelation lags and LPC parameters used to compute the LLR measure.
-        r_clean, ref_clean, a_clean = lp_coefficients(clean_frame, p)
-        r_processed, ref_processed, a_processed = lp_coefficients(processed_frame, p)
-
-        # (3) Compute the LLR measure
-        numerator = np.dot(np.matmul(a_processed, toeplitz(r_clean)), a_processed)
-        denominator = np.dot(np.matmul(a_clean, toeplitz(r_clean)), a_clean)
-        distortion[frame_count] = np.log(numerator / denominator)
-        start = start + skip_rate
-    return distortion
-
-
-def wss(clean_speech, processed_speech, sample_rate):
-    from scipy.fftpack import fft
-
-    # Check the length of the clean and processed speech, which must be the same.
-    clean_length = np.size(clean_speech)
-    processed_length = np.size(processed_speech)
-    if clean_length != processed_length:
-        raise ValueError('Files must have same length.')
-
-    # Global variables
-    win_length = (np.round(30 * sample_rate / 1000)).astype(int)  # window length in samples
-    skip_rate = (np.floor(np.divide(win_length, 4))).astype(int)  # window skip in samples
-    max_freq = (np.divide(sample_rate, 2)).astype(int)  # maximum bandwidth
-    num_crit = 25  # number of critical bands
-
-    n_fft = (np.power(2, np.ceil(np.log2(2 * win_length)))).astype(int)
-    n_fft_by_2 = (np.multiply(0.5, n_fft)).astype(int)  # FFT size/2
-    k_max = 20.0  # value suggested by Klatt, pg 1280
-    k_loc_max = 1.0  # value suggested by Klatt, pg 1280
-
-    # Critical Band Filter Definitions (Center Frequency and Bandwidths in Hz)
-    cent_freq = np.array([50.0000, 120.000, 190.000, 260.000, 330.000, 400.000, 470.000,
-                          540.000, 617.372, 703.378, 798.717, 904.128, 1020.38, 1148.30,
-                          1288.72, 1442.54, 1610.70, 1794.16, 1993.93, 2211.08, 2446.71,
-                          2701.97, 2978.04, 3276.17, 3597.63])
-    bandwidth = np.array([70.0000, 70.0000, 70.0000, 70.0000, 70.0000, 70.0000, 70.0000,
-                          77.3724, 86.0056, 95.3398, 105.411, 116.256, 127.914, 140.423,
-                          153.823, 168.154, 183.457, 199.776, 217.153, 235.631, 255.255,
-                          276.072, 298.126, 321.465, 346.136])
-
-    bw_min = bandwidth[0]  # minimum critical bandwidth
-
-    # Set up the critical band filters.
-    # Note here that Gaussian-ly shaped filters are used.
-    # Also, the sum of the filter weights are equivalent for each critical band filter.
-    # Filter less than -30 dB and set to zero.
-    min_factor = np.exp(-30.0 / (2.0 * 2.303))  # -30 dB point of filter
-    crit_filter = np.empty((num_crit, n_fft_by_2))
-    for i in range(num_crit):
-        f0 = (cent_freq[i] / max_freq) * n_fft_by_2
-        bw = (bandwidth[i] / max_freq) * n_fft_by_2
-        norm_factor = np.log(bw_min) - np.log(bandwidth[i])
-        j = np.arange(n_fft_by_2)
-        crit_filter[i, :] = np.exp(-11 * np.square(np.divide(j - np.floor(f0), bw)) + norm_factor)
-        cond = np.greater(crit_filter[i, :], min_factor)
-        crit_filter[i, :] = np.where(cond, crit_filter[i, :], 0)
-    # For each frame of input speech, calculate the Weighted Spectral Slope Measure
-    num_frames = int(clean_length / skip_rate - (win_length / skip_rate))  # number of frames
-    start = 0  # starting sample
-    window = 0.5 * (1 - np.cos(2 * np.pi * np.arange(1, win_length + 1) / (win_length + 1)))
-
-    distortion = np.empty(num_frames)
-    for frame_count in range(num_frames):
-        # (1) Get the Frames for the test and reference speech. Multiply by Hanning Window.
-        clean_frame = clean_speech[start: start + win_length] / 32768
-        processed_frame = processed_speech[start: start + win_length] / 32768
-        clean_frame = np.multiply(clean_frame, window)
-        processed_frame = np.multiply(processed_frame, window)
-        # (2) Compute the Power Spectrum of Clean and Processed
-        clean_spec = np.square(np.abs(fft(clean_frame, n_fft)))
-        processed_spec = np.square(np.abs(fft(processed_frame, n_fft)))
-
-        # (3) Compute Filterbank Output Energies (in dB scale)
-        clean_energy = np.matmul(crit_filter, clean_spec[0:n_fft_by_2])
-        processed_energy = np.matmul(crit_filter, processed_spec[0:n_fft_by_2])
-
-        clean_energy = 10 * np.log10(np.maximum(clean_energy, 1E-10))
-        processed_energy = 10 * np.log10(np.maximum(processed_energy, 1E-10))
-
-        # (4) Compute Spectral Slope (dB[i+1]-dB[i])
-        clean_slope = clean_energy[1:num_crit] - clean_energy[0: num_crit - 1]
-        processed_slope = processed_energy[1:num_crit] - processed_energy[0: num_crit - 1]
-
-        # (5) Find the nearest peak locations in the spectra to each critical band.
-        #     If the slope is negative, we search to the left. If positive, we search to the right.
-        clean_loc_peak = np.empty(num_crit - 1)
-        processed_loc_peak = np.empty(num_crit - 1)
-
-        for i in range(num_crit - 1):
-            # find the peaks in the clean speech signal
-            if clean_slope[i] > 0:  # search to the right
-                n = i
-                while (n < num_crit - 1) and (clean_slope[n] > 0):
-                    n = n + 1
-                clean_loc_peak[i] = clean_energy[n - 1]
-            else:  # search to the left
-                n = i
-                while (n >= 0) and (clean_slope[n] <= 0):
-                    n = n - 1
-                clean_loc_peak[i] = clean_energy[n + 1]
-
-            # find the peaks in the processed speech signal
-            if processed_slope[i] > 0:  # search to the right
-                n = i
-                while (n < num_crit - 1) and (processed_slope[n] > 0):
-                    n = n + 1
-                processed_loc_peak[i] = processed_energy[n - 1]
-            else:  # search to the left
-                n = i
-                while (n >= 0) and (processed_slope[n] <= 0):
-                    n = n - 1
-                processed_loc_peak[i] = processed_energy[n + 1]
-
-        # (6) Compute the WSS Measure for this frame. This includes determination of the weighting function.
-        db_max_clean = np.max(clean_energy)
-        db_max_processed = np.max(processed_energy)
-        '''
-        The weights are calculated by averaging individual weighting factors from the clean and processed frame.
-        These weights w_clean and w_processed should range from 0 to 1 and place more emphasis on spectral peaks
-        and less emphasis on slope differences in spectral valleys.
-        This procedure is described on page 1280 of Klatt's 1982 ICASSP paper.
-        '''
-        w_max_clean = np.divide(k_max, k_max + db_max_clean - clean_energy[0: num_crit - 1])
-        w_loc_max_clean = np.divide(k_loc_max, k_loc_max + clean_loc_peak - clean_energy[0: num_crit - 1])
-        w_clean = np.multiply(w_max_clean, w_loc_max_clean)
-
-        w_max_processed = np.divide(k_max, k_max + db_max_processed - processed_energy[0: num_crit - 1])
-        w_loc_max_processed = np.divide(k_loc_max, k_loc_max + processed_loc_peak - processed_energy[0: num_crit - 1])
-        w_processed = np.multiply(w_max_processed, w_loc_max_processed)
-
-        w = np.divide(np.add(w_clean, w_processed), 2.0)
-        slope_diff = np.subtract(clean_slope, processed_slope)[0: num_crit - 1]
-        distortion[frame_count] = np.dot(w, np.square(slope_diff)) / np.sum(w)
-        # This normalization is not part of Klatt's paper, but helps to normalize the measure.
-        # Here we scale the measure by the sum of the weights.
-        start = start + skip_rate
-    return distortion
-
-
-def calc_speech_metrics(hypothesis: np.ndarray,
-                        reference: np.ndarray) -> tuple[float, int, int, int, float]:
-    """
-    Calculate speech metrics pesq_mos, c_sig, c_bak, c_ovl, seg_snr.  These are all related and thus included
-    in one function. Reference: matlab script "compute_metrics.m".
-
-    Usage:
-        pesq, c_sig, c_bak, c_ovl, ssnr = compute_metrics(hypothesis, reference, fs, path)
-        reference: clean audio as array
-        hypothesis: enhanced audio as array
-        Audio must have sampling rate = 16000 Hz.
-
-    Example call:
-        pesq_output, csig_output, cbak_output, covl_output, ssnr_output = \
-                calc_speech_metrics(predicted_audio, target_audio)
-    """
-    from sonusai.metrics import calc_pesq
-
-    fs = 16000
-
-    # compute the WSS measure
-    wss_dist_vec = wss(reference, hypothesis, fs)
-    wss_dist_vec = np.sort(wss_dist_vec)
-    alpha = 0.95  # value from CMGAN ref implementation
-    wss_dist = np.mean(wss_dist_vec[0: round(np.size(wss_dist_vec) * alpha)])
-
-    # compute the LLR measure
-    llr_dist = llr(reference, hypothesis, fs)
-    ll_rs = np.sort(llr_dist)
-    llr_len = round(np.size(llr_dist) * alpha)
-    llr_mean = np.mean(ll_rs[0: llr_len])
-
-    # compute the SNRseg
-    snr_dist, segsnr_dist = snr(reference, hypothesis, fs)
-    seg_snr = np.mean(segsnr_dist)
-
-    # compute the pesq (use Sonusai wrapper, only fs=16k, mode=wb support)
-    pesq_mos = calc_pesq(hypothesis=hypothesis, reference=reference)
-
-    # now compute the composite measures
-    c_sig = 3.093 - 1.029 * llr_mean + 0.603 * pesq_mos - 0.009 * wss_dist
-    c_sig = max(1, c_sig)
-    c_sig = min(5, c_sig)  # limit values to [1, 5]
-    c_bak = 1.634 + 0.478 * pesq_mos - 0.007 * wss_dist + 0.063 * seg_snr
-    c_bak = max(1, c_bak)
-    c_bak = min(5, c_bak)  # limit values to [1, 5]
-    c_ovl = 1.594 + 0.805 * pesq_mos - 0.512 * llr_mean - 0.007 * wss_dist
-    c_ovl = max(1, c_ovl)
-    c_ovl = min(5, c_ovl)  # limit values to [1, 5]
-
-    return pesq_mos, c_sig, c_bak, c_ovl, seg_snr
-
-
 def mean_square_error(hypothesis: np.ndarray,
                       reference: np.ndarray,
                       squared: bool = False) -> tuple[np.ndarray, np.ndarray, np.ndarray]:
@@ -494,48 +205,6 @@ def log_error(reference: np.ndarray, hypothesis: np.ndarray) -> tuple[np.ndarray
     return err, err_b, err_f
 
 
-def phase_distance(reference: np.ndarray,
-                   hypothesis: np.ndarray,
-                   eps: float = 1e-9) -> tuple[np.ndarray, np.ndarray, np.ndarray]:
-    """Calculate weighted phase distance error (weight normalization over bins per frame)
-
-    :param reference: complex [frames, bins]
-    :param hypothesis: complex [frames, bins]
-    :param eps: epsilon value
-    :return: mean, mean per bin, mean per frame
-    """
-    ang_diff = np.angle(reference) - np.angle(hypothesis)
-    phd_mod = (ang_diff + np.pi) % (2 * np.pi) - np.pi
-    rh_angle_diff = phd_mod * 180 / np.pi  # angle diff in deg
-
-    # Use complex divide to intrinsically keep angle diff +/-180 deg, but avoid div by zero (real hyp)
-    # hyp_real = np.real(hypothesis)
-    # near_zeros = np.real(hyp_real) < eps
-    # hyp_real = hyp_real * (np.logical_not(near_zeros))
-    # hyp_real = hyp_real + (near_zeros * eps)
-    # hypothesis = hyp_real + 1j*np.imag(hypothesis)
-    # rh_angle_diff = np.angle(reference / hypothesis) * 180 / np.pi  # angle diff +/-180
-
-    # weighted mean over all (scalar)
-    reference_mag = np.abs(reference)
-    ref_weight = reference_mag / (np.sum(reference_mag) + eps)  # frames x bins
-    err = np.around(np.sum(ref_weight * rh_angle_diff), 3)
-
-    # weighted mean over frames (value per bin)
-    err_b = np.zeros(reference.shape[1])
-    for bi in range(reference.shape[1]):
-        ref_weight = reference_mag[:, bi] / (np.sum(reference_mag[:, bi], axis=0) + eps)
-        err_b[bi] = np.around(np.sum(ref_weight * rh_angle_diff[:, bi]), 3)
-
-    # weighted mean over bins (value per frame)
-    err_f = np.zeros(reference.shape[0])
-    for fi in range(reference.shape[0]):
-        ref_weight = reference_mag[fi, :] / (np.sum(reference_mag[fi, :]) + eps)
-        err_f[fi] = np.around(np.sum(ref_weight * rh_angle_diff[fi, :]), 3)
-
-    return err, err_b, err_f
-
-
 def plot_mixpred(mixture: AudioT,
                  mixture_f: AudioF,
                  target: Optional[AudioT] = None,
@@ -543,7 +212,6 @@ def plot_mixpred(mixture: AudioT,
                  predict: Optional[Predict] = None,
                  tp_title: str = '') -> plt.Figure:
     from sonusai.mixture import SAMPLE_RATE
-
     num_plots = 2
     if feature is not None:
         num_plots += 1
@@ -706,12 +374,13 @@ def _process_mixture(mixid: int) -> tuple[pd.DataFrame, pd.DataFrame]:
     import h5py
     import mgzip
     from matplotlib.backends.backend_pdf import PdfPages
-    from numpy import inf
     from pystoi import stoi
 
     from sonusai import SonusAIError
     from sonusai import logger
     from sonusai.metrics import calc_pcm
+    from sonusai.metrics import calc_phase_distance
+    from sonusai.metrics import calc_speech
     from sonusai.metrics import calc_wer
     from sonusai.metrics import calc_wsdr
     from sonusai.mixture import forward_transform
@@ -771,24 +440,28 @@ def _process_mixture(mixid: int) -> tuple[pd.DataFrame, pd.DataFrame]:
     # noise_wo_dist_f = mixdb.mixture_noise_f(mixid, noise=noise_wo_dist)
     noise = mixture - target  # has time-domain distortion (ir,etc.) but does not have specaugment
     # noise_f = mixdb.mixture_noise_f(mixid, noise=noise)
-    segsnr_f = mixdb.mixture_segsnr(mixid, target=target, noise=noise)  # note: uses pre-IR, pre-specaug audio
+    # note: uses pre-IR, pre-specaug audio
+    segsnr_f: np.ndarray = mixdb.mixture_metrics(mixid, ['ssnr'])[0]  # type: ignore
     mixture_f = mixdb.mixture_mixture_f(mixid, mixture=mixture)
     noise_f = mixture_f - target_f  # true noise in freq domain includes specaugment and time-domain ir,distortions
     # segsnr_f = mixdb.mixture_segsnr(mixid, target=target, noise=noise)
-    segsnr_f[segsnr_f == inf] = 7.944e8  # 99db
-    segsnr_f[segsnr_f == -inf] = 1.258e-10  # -99db
+    segsnr_f[segsnr_f == np.inf] = DB_99
+    # segsnr_f should never be -np.inf
+    segsnr_f[segsnr_f == -np.inf] = DB_N99
     # need to use inv-tf to match #samples & latency shift properties of predict inv tf
     target_fi = inverse_transform(target_f, mixdb.it_config)
     noise_fi = inverse_transform(noise_f, mixdb.it_config)
     # mixture_fi = mixdb.inverse_transform(mixture_f)
 
     # gen feature, truth - note feature only used for plots
-    # TBD parse truth_f for different formats and also multi-truth
-    feature, truth_f = mixdb.mixture_ft(mixid, mixture=mixture)
-    truth_type = mixdb.target_file(mixdb.mixture(mixid).targets[0].file_id).truth_settings[0].function
-    if truth_type == 'target_mixture_f':
-        half = truth_f.shape[-1] // 2
-        truth_f = truth_f[..., :half]  # extract target_f only
+    # TODO: parse truth_f for different formats
+    feature, truth_f = mixdb.mixture_ft(mixid, mixture_f=mixture_f)
+    # ignore mixup
+    for truth_setting in mixdb.target_file(mixdb.mixture(mixid).targets[0].file_id).truth_settings:
+        if truth_setting.function == 'target_mixture_f':
+            half = truth_f.shape[-1] // 2
+            # extract target_f only
+            truth_f = truth_f[..., :half]
 
     if not truth_est_mode:
         if predict.shape[0] < target_f.shape[0]:  # target_f, truth_f, mixture_f, etc. same size
@@ -843,12 +516,12 @@ def _process_mixture(mixid: int) -> tuple[pd.DataFrame, pd.DataFrame]:
     pcm, pcm_bin, pcm_frame = calc_pcm(hypothesis=ypred_f, reference=ytrue_f, with_log=True)
 
     # Phase distance
-    phd, phd_bin, phd_frame = phase_distance(hypothesis=predict_complex, reference=truth_f_complex)
+    phd, phd_bin, phd_frame = calc_phase_distance(hypothesis=predict_complex, reference=truth_f_complex)
 
     # Noise td logerr
     # lerr_nt, lerr_nt_bin, lerr_nt_frame = log_error(noise_fi, noise_truth_est_audio)
 
-    # # SA-SDR (time-domain source-aggragated SDR)
+    # # SA-SDR (time-domain source-aggregated SDR)
     ytrue = np.concatenate((target_fi[:, np.newaxis], noise_fi[:, np.newaxis]), axis=1)
     ypred = np.concatenate((target_est_wav[:, np.newaxis], noise_est_wav[:, np.newaxis]), axis=1)
     # # note: w/o scale is more pessimistic number
@@ -863,8 +536,8 @@ def _process_mixture(mixid: int) -> tuple[pd.DataFrame, pd.DataFrame]:
     # Speech intelligibility measure - PESQ
     if int(mixdb.mixture(mixid).snr) > -99:
         # len = target_est_wav.shape[0]
-        pesq_speech, csig_tg, cbak_tg, covl_tg, sgsnr_tg = calc_speech_metrics(target_est_wav, target_fi)
-        pesq_mixture, csig_mx, cbak_mx, covl_mx, sgsnr_mx = calc_speech_metrics(mixture, target)
+        pesq_speech, csig_tg, cbak_tg, covl_tg = calc_speech(target_est_wav, target_fi)
+        pesq_mixture, csig_mx, cbak_mx, covl_mx = mixdb.mixture_metrics(mixid, ['mxpesq', 'mxcsig', 'mxcbak', 'mxcovl'])
         # pesq_speech_tst = calc_pesq(hypothesis=target_est_wav, reference=target)
         # pesq_mixture_tst = calc_pesq(hypothesis=mixture, reference=target)
         # pesq improvement
@@ -886,20 +559,15 @@ def _process_mixture(mixid: int) -> tuple[pd.DataFrame, pd.DataFrame]:
     asr_tt = None
     asr_mx = None
     asr_tge = None
-    if asr_method == 'none' or mixdb.mixture(mixid).snr == -99:  # noise only, ignore/reset target asr
-        wer_mx = float('nan')
-        wer_tge = float('nan')
-        wer_pi = float('nan')
-    else:
+    asr_engines = list(mixdb.asr_configs.keys())
+    if len(asr_engines) > 0 and mixdb.mixture(mixid).snr >= -96:  # noise only, ignore/reset target asr
+        wer_mx = float(mixdb.mixture_metrics(mixid, [f'mxwer.{asr_engines[0]}'])[0]) * 100
         asr_tt = MP_GLOBAL.mixdb.mixture_speech_metadata(mixid, 'text')[0]  # ignore mixup
         if asr_tt is None:
             asr_tt = calc_asr(target, engine=asr_method, whisper_model_name=asr_model_name).text  # target truth
 
         if asr_tt:
-            asr_mx = calc_asr(mixture, engine=asr_method, whisper_model_name=asr_model_name).text
             asr_tge = calc_asr(target_est_wav, engine=asr_method, whisper_model_name=asr_model_name).text
-
-            wer_mx = calc_wer(asr_mx, asr_tt).wer * 100  # mixture wer
             wer_tge = calc_wer(asr_tge, asr_tt).wer * 100  # target estimate wer
             if wer_mx == 0.0:
                 if wer_tge == 0.0:
@@ -913,6 +581,10 @@ def _process_mixture(mixid: int) -> tuple[pd.DataFrame, pd.DataFrame]:
             wer_mx = float(0)
             wer_tge = float(0)
             wer_pi = float(0)
+    else:
+        wer_mx = float('nan')
+        wer_tge = float('nan')
+        wer_pi = float('nan')
 
     # 5) Save per mixture metric results
     # Single row in table of scalar metrics per mixture
@@ -1088,7 +760,6 @@ def main():
     from os.path import basename
     from os.path import isdir
     from os.path import join
-    from os.path import split
 
     import psutil
     from tqdm import tqdm
@@ -1097,7 +768,7 @@ def main():
     from sonusai import initial_log_messages
     from sonusai import logger
     from sonusai import update_console_handler
-    from sonusai.mixture import DEFAULT_NOISE
+    from sonusai.mixture import DEFAULT_SPEECH
     from sonusai.mixture import MixtureDatabase
     from sonusai.mixture import read_audio
     from sonusai.utils import calc_asr
@@ -1173,8 +844,7 @@ def main():
         return
 
     if enable_asr_warmup:
-        default_speech = split(DEFAULT_NOISE)[0] + '/speech_ma01_01.wav'
-        audio = read_audio(default_speech)
+        audio = read_audio(DEFAULT_SPEECH)
         logger.info(f'Warming up asr method, note for cloud service this could take up to a few min ...')
         asr_chk = calc_asr(audio, engine=asr_method, whisper_model_name=asr_model_name)
         logger.info(f'Warmup completed, results {asr_chk}')

sonusai/data/genmixdb.yml

@@ -62,3 +62,5 @@ spectral_masks:
     t_max_width: 100
     t_num: 0
     t_max_percent: 100
+
+asr_configs: [ ]

sonusai/doc/doc.py

@@ -199,7 +199,6 @@ def get_truth_functions() -> str:
 def doc_truth_settings() -> str:
     import yaml
 
-    from sonusai.mixture import get_default_config
     default = f"\nDefault value:\n\n{yaml.dump(get_default_config()['truth_settings'])}"
     return """
 'truth_settings' is a mixture database configuration parameter that sets the truth
@@ -375,7 +374,6 @@ This rule expands to 6 unique augmentations being applied to each target
 def doc_target_augmentations() -> str:
     import yaml
 
-    from sonusai.mixture import get_default_config
     default = f"\nDefault value:\n\n{yaml.dump(get_default_config()['target_augmentations'])}"
     return """
 'target_augmentations' is a mixture database configuration parameter that
@@ -388,7 +386,6 @@ See 'augmentations' for details on augmentation rules.
 def doc_class_balancing_augmentation() -> str:
     import yaml
 
-    from sonusai.mixture import get_default_config
     default = f"\nDefault value:\n\n{yaml.dump(get_default_config()['class_balancing_augmentation'])}"
     return """
 'class_balancing_augmentation' is a mixture database configuration parameter
@@ -436,7 +433,6 @@ Required field:
 def doc_noise_augmentations() -> str:
     import yaml
 
-    from sonusai.mixture import get_default_config
     default = f"\nDefault value:\n\n{yaml.dump(get_default_config()['noise_augmentations'])}"
     return """
 'noise_augmentations' is a mixture database configuration parameter that
@@ -536,3 +532,48 @@ def doc_config() -> str:
     for c in VALID_CONFIGS:
         text += f' {c}\n'
     return text
+
+
+def doc_asr_configs() -> str:
+    from sonusai.utils import get_available_engines
+
+    default = f"\nDefault value: {get_default_config()['asr_configs']}"
+    engines = get_available_engines()
+    text = """
+'asr_configs' is a mixture database configuration parameter that sets the list of
+ASR engine(s) to use.
+
+Required fields:
+
+ 'name'         Unique identifier for the ASR engine.
+ 'engine'       ASR engine to use. Available engines:
+"""
+    text += f'                {", ".join(engines)}\n'
+    text += """
+Optional fields:
+
+ 'model'        Some ASR engines allow the specification of a model, but note most are
+                very computationally demanding and can overwhelm/hang a local system.
+                Available whisper ASR engines:
+                    tiny.en, tiny, base.en, base, small.en, small, medium.en, medium, large-v1, large-v2, large
+ 'device'       Some ASR engines allow the specification of a device, either 'cpu' or 'cuda'.
+ 'cpu_threads'  Some ASR engines allow the specification of the number of CPU threads to use.
+ 'compute_type' Some ASR engines allow the specification of a compute type, e.g. 'int8'.
+ 'beam_size'    Some ASR engines allow the specification of a beam size.
+ <other>        Other parameters can be injected into the ASR engine as needed; all
+                fields in each config are forwarded to the given engine.
+
+Example:
+
+asr_configs:
+  - name: faster_tiny_cuda
+    engine: faster_whisper
+    model: tiny
+    device: cuda
+    beam_size: 5
+  - name: google
+    engine: google
+
+Creates two ASR engines for use named faster_tiny_cuda and google.
+"""
+    return text + default