├── .gitignore
├── LICENSE
├── README.md
├── __init__.py
├── audio
├── __init__.py
├── helpers.py
├── pitch.py
└── visuals.py
├── bin
├── __init__.py
└── compute_metrics.py
├── config
├── __init__.py
└── global_config.py
├── metrics
├── DTW.py
├── FFE.py
├── GPE.py
├── MCD.py
├── MSD.py
├── VDE.py
├── __init__.py
├── dists.py
├── helpers.py
└── moments.py
└── requirements.txt
/.gitignore:
--------------------------------------------------------------------------------
1 | __pycache__
2 | *.json
3 | .env
--------------------------------------------------------------------------------
/LICENSE:
--------------------------------------------------------------------------------
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--------------------------------------------------------------------------------
/README.md:
--------------------------------------------------------------------------------
1 | # TTS Objective Metrics 🎯
2 |
3 | This repository comprises a compilation of the objective metrics used in several text-to-speech (TTS) papers.
4 |
5 | ## Available Metrics
6 | | Metric | Used In |
7 | | ------ | ------ |
8 | | Voicing Decision Error (VDE) | [E2E-Prosody](https://arxiv.org/pdf/1803.09047.pdf), [Mellotron](https://arxiv.org/abs/1910.11997)|
9 | | Gross Pitch Error (GPE) | [E2E-Prosody](https://arxiv.org/pdf/1803.09047.pdf), [Mellotron](https://arxiv.org/abs/1910.11997)|
10 | | F0 Frame Error (FFE) | [E2E-Prosody](https://arxiv.org/pdf/1803.09047.pdf), [Mellotron](https://arxiv.org/abs/1910.11997)|
11 | | Dynamic Time Warping (DTW) | [FastSpeech2](https://arxiv.org/abs/2006.04558) |
12 | | Mel Spectral Distortion (MSD) | [Wave-Tacotron](https://arxiv.org/abs/2011.03568) |
13 | | Mel Cepstral Distortion (MCD) | [E2E-Prosody](https://arxiv.org/pdf/1803.09047.pdf), [Wave-Tacotron](https://arxiv.org/abs/2011.03568) |
14 | | Statistical Moments (STD, SKEW, KURT) | [FastSpeech2](https://arxiv.org/abs/2006.04558) |
15 |
16 | ## Available Pitch Computation
17 | | Alogrithm | Proposed In |
18 | | ------ | ------ |
19 | | YIN | [(Cheveigné and Kawahara, 2002)](http://audition.ens.fr/adc/pdf/2002_JASA_YIN.pdf) |
20 | | DIO | [(Morise, Kawahara, and Katayose, 2009)](https://www.aes.org/e-lib/browse.cfm?elib=15165)|
21 | | PYIN (Testing) | [(Mauch and Dixon, 2014)](https://ieeexplore.ieee.org/document/6853678) |
22 |
23 | ## How to Run
24 | First, clone and enter the repo:
25 | ```sh
26 | git clone https://github.com/AI-Unicamp/TTS-Objective-Metrics
27 | cd TTS-Objective-Metrics
28 | ```
29 |
30 | Install dependencies:
31 | ```
32 | pip install -r requirements.txt
33 | ```
34 |
35 | Then, configure the global parameters as you wish in config-> global.py. If not done, all statistics will be computed with the default parameters already set.
36 |
37 | The main usage of the repo is to calculate all available metrics for a batch of (ground truth, synthesized) audio pairs (test/eval set). For this make sure you have the (ground_truth, synthesized) audio pairs names **matching** and in **numbered order, each with the same number of digits of the greatest file** (eg. if there are 100 files, you shall start with 000.wav, if there are [10,99] you shall start with 00.wav). as in:
38 |
39 | 📂My Audio\
40 | ┣ 📂Ground Truths\
41 | ┃ ┣ 📜00.wav\
42 | ┃ ┣ 📜01.wav\
43 | ┃ ┣ ...\
44 | ┣ 📂Synthesizeds\
45 | ┃ ┣ 📜00.wav\
46 | ┃ ┣ 📜01.wav\
47 | ┃ ┣ ...
48 |
49 | Then, choose one pitch computing algorithm and run the following command:
50 | ```sh
51 | python -m bin.compute_metrics --gt_folder_path 'My Audio\Ground Truths' --synth_folder_path 'My Audio\Synthesizeds' --pitch_algorithm 'yin'
52 | ```
53 | The result will be saved in a file named metrics.json in the main repo folder: 'TTS Objective Metrics/metrics.json'.
54 |
55 | Alternatively, it is possible to calculate a single metric for a pair of (ground truth, synthesized) audio, by choosing an available metric and pitch computation method (yin or dio) with one of the following commands:
56 | ```sh
57 | # For DTW, FFE, GPE, VDE, moments
58 | python -m metrics.DTW --gt_path 'ground_truth_audio.wav' --synth_path 'synthesized_audio.wav' --pitch_algorithm 'yin'
59 | ```
60 | ```
61 | # For MSD or MCD
62 | python -m metrics.MSD --gt_path 'path_to_ground_truth_audio.wav' --synth_path 'path_to_synthesized_audio.wav'
63 | ```
64 | The result will be displayed in the terminal.
65 |
66 | ## Repo Organization
67 | 📦TTS Objective Metrics\
68 | ┣ 📂audio\
69 | ┃ ┣ 📜helpers.py\
70 | ┃ ┣ 📜pitch.py\
71 | ┃ ┣ 📜visuals.py\
72 | ┣ 📂bin\
73 | ┃ ┣ 📜compute_metrics.py\
74 | ┣ 📂config\
75 | ┃ ┣ 📜global_config.py\
76 | ┣ 📂metrics\
77 | ┃ ┣ 📜dists.py\
78 | ┃ ┣ 📜DTW.py\
79 | ┃ ┣ 📜FFE.py\
80 | ┃ ┣ 📜GPE.py\
81 | ┃ ┣ 📜helpers.py\
82 | ┃ ┣ 📜MCD.py\
83 | ┃ ┣ 📜moments.py\
84 | ┃ ┣ 📜MSD.py\
85 | ┃ ┣ 📜VDE.py\
86 | ┣ 📜README.md
87 |
88 | ## How to Contribute
89 | As the repo is still in its infancy, feel free to either open an issue, discussion or send a pull request, or even contact us by e-mail.
90 |
91 | ## Authors
92 | - Leonardo B. de M. M. Marques (leonardoboulitreau@gmail.com)
93 | - Lucas Hideki Ueda (lucashueda@gmail.com)
94 |
95 | ## Github references
96 | - [Coqui-AI](https://github.com/coqui-ai/TTS)
97 | - [Facebook Fairseq](https://github.com/pytorch/fairseq)
98 | - [NVIDIA Deep Learning Examples](https://github.com/NVIDIA/DeepLearningExamples)
99 | - [NVIDIA Mellotron](https://github.com/NVIDIA/mellotron/tree/d5362ccae23984f323e3cb024a01ec1de0493aff)
100 | - [MAPS](https://github.com/bastibe/MAPS-Scripts)
101 | - [Yin](https://github.com/patriceguyot/Yin)
102 |
103 | All references are listened on top of the used code itself.
104 |
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/__init__.py:
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https://raw.githubusercontent.com/AI-Unicamp/TTS-Objective-Metrics/e183fd0f4ec484173b13b86ae315c862af9db888/__init__.py
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/audio/__init__.py:
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https://raw.githubusercontent.com/AI-Unicamp/TTS-Objective-Metrics/e183fd0f4ec484173b13b86ae315c862af9db888/audio/__init__.py
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/audio/helpers.py:
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1 | import os
2 | import glob
3 | import librosa
4 |
5 | # Load Audios
6 | def read_folder(path):
7 | out = list()
8 | for filename in sorted(glob.glob(os.path.join(path, '*.wav'))):
9 | x, sr = librosa.load(filename)
10 | out.append(x)
11 | return out, sr
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/audio/pitch.py:
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1 | import librosa
2 | import numpy as np
3 | import torch
4 | import pyworld as pw
5 |
6 | # https://github.com/pytorch/fairseq/blob/fcca32258c8e8bcc9f9890bf4714fa2f96b6b3e1/examples/speech_synthesis/evaluation/eval_f0.py#L87
7 | def yin(sig, config):
8 |
9 | '''Compute the Yin Algorithm. Return fundamental frequency and harmonic rate.
10 | An adaptation of https://github.com/NVIDIA/mellotron which is an adaption of
11 | https://github.com/patriceguyot/Yin
12 |
13 | Args:
14 | sig (list of floats): Audio Signal
15 | sr (int): Sample Rate
16 | win_len (int): Size of the analysis window (samples)
17 | hop_length (int): Size of the lag between two consecutives windows (samples)
18 | f0_min (int): Minimum fundamental frequency that can be detected (hertz)
19 | f0_max (int): Maximum fundamental frequency that can be detected (hertz)
20 | harmo_thresh (float): Threshold of detection. The algorithm returns the
21 | first minimum of the CMND function below this threshold.
22 |
23 | Returns:
24 | (dict) with:
25 | 'pitches' (1-D np.array): fundamental frequencies,
26 | 'harmonic_rates' (1-D np.array): list of harmonic rate values for each
27 | fundamental frequency value (= confidence value)
28 | 'argmins' (1-D np.array): minimums of the Cumulative Mean Normalized
29 | DifferenceFunction
30 | 'times' (1-D np.array): list of time of each estimation
31 | '''
32 | sr = config.sr
33 | win_length = config.win_length
34 | hop_length = config.hop_length
35 | f0_min = config.f0_min_pitch
36 | f0_max = config.f0_max_pitch
37 | harmo_thresh = config.harmo_thresh
38 |
39 | tau_min = int(sr / f0_max)
40 | tau_max = int(sr / f0_min)
41 |
42 | # Time values for each analysis window
43 | time_scale = range(0, len(sig) - win_length, hop_length)
44 | times = [t/float(sr) for t in time_scale]
45 | frames = [sig[t:t + win_length] for t in time_scale]
46 |
47 | pitches = [0.0] * len(time_scale)
48 | harmonic_rates = [0.0] * len(time_scale)
49 | argmins = [0.0] * len(time_scale)
50 |
51 | for i, frame in enumerate(frames):
52 | # Compute YIN
53 | df = _difference_function(frame, win_length, tau_max)
54 | cm_df = _cumulative_mean_normalized_difference_function(df, tau_max)
55 | p = _get_pitch(cm_df, tau_min, tau_max, harmo_thresh)
56 |
57 | # Get results
58 | if np.argmin(cm_df) > tau_min:
59 | argmins[i] = float(sr / np.argmin(cm_df))
60 | if p != 0: # A pitch was found
61 | pitches[i] = float(sr / p)
62 | harmonic_rates[i] = cm_df[p]
63 | else: # No pitch, but we compute a value of the harmonic rate
64 | harmonic_rates[i] = min(cm_df)
65 |
66 | return {'pitches' : np.array(pitches), 'harmonic_rates' : np.array(harmonic_rates), 'argmins' : np.array(argmins), 'times' : np.array(times)}
67 |
68 | def _difference_function(x, n, tau_max):
69 |
70 | '''Compute difference function of data x. This solution is implemented directly
71 | with Numpy fft.
72 |
73 | Args:
74 | x (list): Audio data frame
75 | n (int): Length of data
76 | tau_max (int): Integration window size
77 |
78 | Returns:
79 | (list) of difference function
80 | '''
81 | x = np.array(x, np.float64)
82 | w = x.size
83 | tau_max = min(tau_max, w)
84 | x_cumsum = np.concatenate((np.array([0.]), (x * x).cumsum()))
85 | size = w + tau_max
86 | p2 = (size // 32).bit_length()
87 | nice_numbers = (16, 18, 20, 24, 25, 27, 30, 32)
88 | size_pad = min(x * 2 ** p2 for x in nice_numbers if x * 2 ** p2 >= size)
89 | fc = np.fft.rfft(x, size_pad)
90 | conv = np.fft.irfft(fc * fc.conjugate())[:tau_max]
91 | return x_cumsum[w:w - tau_max:-1] + x_cumsum[w] - x_cumsum[:tau_max] - \
92 | 2 * conv
93 |
94 | def _cumulative_mean_normalized_difference_function(df, n):
95 |
96 | '''Compute cumulative mean normalized difference function (CMND).
97 |
98 | Args:
99 | df (list): Difference function
100 | n (int): Length of data
101 |
102 | Returns:
103 | (list) of cumulative mean normalized difference function
104 | '''
105 |
106 | # Scipy method
107 | cmn_df = df[1:] * range(1, n) / np.cumsum(df[1:]).astype(float)
108 | return np.insert(cmn_df, 0, 1)
109 |
110 | def _get_pitch(cmdf, tau_min, tau_max, harmo_th=0.1):
111 |
112 | '''Return fundamental period of a frame based on CMND function.
113 |
114 | Args:
115 | cmdf (list): Cumulative Mean Normalized Difference function
116 | tau_min (int): Minimum period for speech
117 | tau_max (int): Maximum period for speech
118 | harmo_th (float): harmonicity threshold to determine if
119 | it is necessary to compute pitch frequency
120 |
121 | Returns:
122 | (float) fundamental period if there is values under threshold,
123 | 0 otherwise
124 | '''
125 |
126 | tau = tau_min
127 | while tau < tau_max:
128 | if cmdf[tau] < harmo_th:
129 | while tau + 1 < tau_max and cmdf[tau + 1] < cmdf[tau]:
130 | tau += 1
131 | return tau
132 | tau += 1
133 |
134 | return 0 # if unvoiced
135 |
136 | # https://github.com/coqui-ai/TTS/blob/main/TTS/utils/audio.py#L710
137 | def dio(sig, config):
138 | '''Compute pitch (f0) of a waveform using the same parameters used for
139 | computing melspectrogram.
140 |
141 | Args:
142 | sig (list of floats): Audio Signal
143 | sr (int): Sample Rate
144 | hop_length (int): Number of frames between STFT columns
145 | f0_min (int): Minimum fundamental frequency that can be detected (hertz)
146 | f0_max (int): Maximum fundamental frequency that can be detected (hertz)
147 |
148 | Returns:
149 | (dict) with:
150 | 'pitches' (1-D np.array): fundamental frequencies,
151 | 'times' (1-D np.array): list of time of each estimation
152 | '''
153 | sr = config.sr
154 | hop_length = config.hop_length
155 | f0_min = config.f0_min_pitch
156 | f0_max = config.f0_max_pitch
157 |
158 |
159 | # Align F0 length to the spectrogram length
160 | if len(sig) % hop_length == 0:
161 | sig = np.pad(sig, (0, hop_length // 2), mode="reflect")
162 |
163 | f0, t = pw.dio(
164 | sig.astype(np.double),
165 | fs=sr,
166 | f0_ceil=f0_max,
167 | f0_floor = f0_min,
168 | frame_period=1000 * hop_length / sr,
169 | )
170 | f0 = pw.stonemask(sig.astype(np.double), f0, t, sr)
171 | return {'pitches':f0, 'times':t}
172 |
173 | # NOT TESTED YET
174 | # https://github.com/NVIDIA/DeepLearningExamples/blob/a43ffd01cb002b23a98c97c3c5a231e24a57fa71/PyTorch/SpeechSynthesis/FastPitch/fastpitch/data_function.py#L81
175 | def compute_pyin(wav, mel_len, method='pyin', normalize_mean=None,normalize_std=None, n_formants=1):
176 |
177 | if type(normalize_mean) is float or type(normalize_mean) is list:
178 | normalize_mean = torch.tensor(normalize_mean)
179 |
180 | if type(normalize_std) is float or type(normalize_std) is list:
181 | normalize_std = torch.tensor(normalize_std)
182 |
183 | if method == 'pyin':
184 |
185 | snd, sr = librosa.load(wav)
186 | pitch_mel, voiced_flag, voiced_probs = librosa.pyin(
187 | snd, fmin=librosa.note_to_hz('C2'),
188 | fmax=librosa.note_to_hz('C7'), frame_length=1024)
189 | assert np.abs(mel_len - pitch_mel.shape[0]) <= 1.0
190 |
191 | pitch_mel = np.where(np.isnan(pitch_mel), 0.0, pitch_mel)
192 | pitch_mel = torch.from_numpy(pitch_mel).unsqueeze(0)
193 | pitch_mel = torch.nn.functional.pad(pitch_mel, (0, mel_len - pitch_mel.size(1)))
194 |
195 | if n_formants > 1:
196 | raise NotImplementedError
197 |
198 | else:
199 | raise ValueError
200 |
201 | pitch_mel = pitch_mel.float()
202 |
203 | if normalize_mean is not None:
204 | assert normalize_std is not None
205 | pitch_mel = _normalize_pitch(pitch_mel, normalize_mean, normalize_std)
206 |
207 | return pitch_mel
208 |
209 | def _normalize_pitch(pitch, mean, std):
210 | zeros = (pitch == 0.0)
211 | pitch -= mean[:, None]
212 | pitch /= std[:, None]
213 | pitch[zeros] = 0.0
214 | return pitch
--------------------------------------------------------------------------------
/audio/visuals.py:
--------------------------------------------------------------------------------
1 | import matplotlib.pyplot as plt
2 | import numpy as np
3 | import torch
4 |
5 | # https://github.com/coqui-ai/TTS/blob/0592a5805ca7fb877c8fc8df56a4eacac13f2657/TTS/tts/utils/visual.py#L35
6 | def plot_spectrogram(spectrogram, fig_size):
7 | '''Plot spectrogram.
8 |
9 | Args:
10 | spectrogram (np.array): Spectrogram values.
11 |
12 | Returns:
13 | plt.figure
14 | '''
15 | if isinstance(spectrogram, torch.Tensor):
16 | spectrogram_ = spectrogram.detach().cpu().numpy().squeeze().T
17 | else:
18 | spectrogram_ = spectrogram.T
19 | spectrogram_ = spectrogram_.astype(np.float32) if spectrogram_.dtype == np.float16 else spectrogram_
20 | fig = plt.figure(figsize=fig_size)
21 | plt.imshow(spectrogram_, aspect="auto", origin="lower")
22 | plt.colorbar()
23 | plt.tight_layout()
24 | plt.close()
25 | return fig
26 |
27 | # https://github.com/coqui-ai/TTS/blob/0592a5805ca7fb877c8fc8df56a4eacac13f2657/TTS/tts/utils/visual.py#L52
28 | def plot_pitch(pitch, spectrogram, fig_size):
29 | '''Plot pitch curves on top of the spectrogram.
30 |
31 | Args:
32 | pitch ( (T,)np.array): Pitch values.
33 | spectrogram ( (C, T) np.array): Spectrogram values.
34 |
35 | Returns:
36 | plt.figure
37 | '''
38 |
39 | if isinstance(spectrogram, torch.Tensor):
40 | spectrogram_ = spectrogram.detach().cpu().numpy().squeeze().T
41 | else:
42 | spectrogram_ = spectrogram.T
43 | spectrogram_ = spectrogram_.astype(np.float32) if spectrogram_.dtype == np.float16 else spectrogram_
44 |
45 | old_fig_size = plt.rcParams["figure.figsize"]
46 | if fig_size is not None:
47 | plt.rcParams["figure.figsize"] = fig_size
48 |
49 | fig, ax = plt.subplots()
50 |
51 | ax.imshow(spectrogram_, aspect="auto", origin="lower")
52 | ax.set_xlabel("time")
53 | ax.set_ylabel("spec_freq")
54 |
55 | ax2 = ax.twinx()
56 | ax2.plot(pitch, linewidth=5.0, color="red")
57 | ax2.set_ylabel("F0")
58 |
59 | plt.rcParams["figure.figsize"] = old_fig_size
60 | plt.close()
61 | return fig
62 |
63 |
64 | # Testing Plotting DTW Alignment between pitchs/spectrograms
65 | #def plot_aligned_pitch(x, y, path, sample_init=0, sample_end=400):
66 | #
67 | # x = x[sample_init:sample_end]
68 | # y = y[sample_init:sample_end]
69 | # path = path[sample_init:sample_end]
70 |
71 | # plt.figure(figsize=(10, 8))
72 | # dis_y = 150
73 | # dis_x = 0
74 | # plt.plot(np.arange(x.shape[0]) + dis_x, x + dis_y, "-", c="C3", linewidth = 3)
75 | # plt.plot(np.arange(y.shape[0]) - dis_x, y - dis_y, "-", c="C0", linewidth = 3)
76 | # for x_i, y_j in path:
77 | # plt.plot([x_i + dis_x, y_j - dis_x], [x[x_i] + dis_y, y[y_j] - dis_y], "-", c="C7", linewidth = 0.5)
78 | # plt.axis("off")
79 | # plt.savefig("mygraph.png")
80 |
81 | # a = []
82 | # b = []
83 | # a.append(torch.Tensor(refs_yin[0][0]).unsqueeze_(1))
84 | # b.append(torch.Tensor(synths_yin[0][0]).unsqueeze_(1))
85 | # x,y,z,k = batch_dynamic_time_warping(a,b, _compute_rms_dist)
86 | # plot_aligned_pitch(np.array(refs_yin[0][0]), np.array(synths_yin[0][0]), k)
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/bin/__init__.py:
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/bin/compute_metrics.py:
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1 | # Basic Imports
2 | import argparse
3 | import json
4 | import numpy as np
5 | import torch
6 |
7 | # Import Audio Loader
8 | from audio.helpers import read_folder
9 |
10 | # Import Pitch Computation
11 | from audio.pitch import dio, yin
12 |
13 | # Import Config
14 | from config.global_config import GlobalConfig
15 |
16 | # Import Metrics
17 | from metrics.VDE import voicing_decision_error
18 | from metrics.GPE import gross_pitch_error
19 | from metrics.FFE import f0_frame_error
20 | from metrics.DTW import batch_dynamic_time_warping
21 | from metrics.MSD import batch_mel_spectral_distortion
22 | from metrics.MCD import batch_mel_cepstral_distortion
23 | from metrics.moments import estimate_moments
24 |
25 | # Import Basic Stats
26 | from metrics.helpers import add_basic_stats
27 |
28 | # Get Input Args
29 | parser = argparse.ArgumentParser(description='Script to calculate the all repo metrics between the ground truth and the synthesized audios.')
30 | parser.add_argument("--gt_folder_path", required = True, type=str, help = 'Path to the folder containing all ground truth audio in .wav format')
31 | parser.add_argument("--synth_folder_path", required = True, type=str, help = 'Path to the folder containing all respective synthesized audio in .wav format')
32 | parser.add_argument("--pitch_algorithm", required = True, type=str, choices = ["dio", "yin"], help = 'Choose method of computing pitch')
33 | args = parser.parse_args()
34 | gt_folder_path = args.gt_folder_path
35 | synth_folder_path = args.synth_folder_path
36 | pitch_algorithm = args.pitch_algorithm
37 |
38 | # Import Audios
39 | refs, _ = read_folder(gt_folder_path)
40 | synths, _ = read_folder(synth_folder_path)
41 | refs_tensor = [torch.Tensor(item) for item in refs]
42 | synths_tensor = [torch.Tensor(item) for item in synths]
43 |
44 | # Define Configurations
45 | config = GlobalConfig()
46 |
47 | # Compute Pitch
48 | refs_pitch = [eval(pitch_algorithm)(item, config) for item in refs]
49 | synths_pitch = [eval(pitch_algorithm)(item, config) for item in synths]
50 | refs_pitch_tensor = [torch.Tensor(item['pitches']).unsqueeze_(1) for item in refs_pitch]
51 | synths_pitch_tensor = [torch.Tensor(item['pitches']).unsqueeze_(1) for item in synths_pitch]
52 |
53 | # Pitch DistributionS
54 | refs_dist = [item for sublist in refs_pitch for item in sublist['pitches']]
55 | synths_dist = [item for sublist in synths_pitch for item in sublist['pitches']]
56 |
57 | # Pairwise Metrics
58 | VDE = {}
59 | GPE = {}
60 | FFE = {}
61 | for i in range(len(refs)):
62 | VDE[i] = voicing_decision_error(refs_pitch[i]['times'], refs_pitch[i]['pitches'], synths_pitch[i]['times'], synths_pitch[i]['pitches'])
63 | GPE[i] = gross_pitch_error(refs_pitch[i]['times'], refs_pitch[i]['pitches'], synths_pitch[i]['times'], synths_pitch[i]['pitches'])
64 | FFE[i] = f0_frame_error(refs_pitch[i]['times'], refs_pitch[i]['pitches'], synths_pitch[i]['times'], synths_pitch[i]['pitches'])
65 |
66 | # Batched Metrics
67 | DTW = {v:k for v,k in enumerate(batch_dynamic_time_warping(refs_pitch_tensor, synths_pitch_tensor, config.dist_fn, config.norm_align_type)['norm_align_costs'])}
68 | MSD = {v:k for v,k in enumerate(batch_mel_spectral_distortion(refs_tensor, synths_tensor, config))}
69 | MCD = {v:k for v,k in enumerate(batch_mel_cepstral_distortion(refs_tensor, synths_tensor, config))}
70 |
71 | # Distribution Metrics
72 | MOMENTS = estimate_moments(refs_dist, synths_dist)
73 |
74 | # Compute Basic Stats and Store
75 | stats = {}
76 | stats['VDE'] = add_basic_stats(VDE)
77 | stats['GPE'] = add_basic_stats(GPE)
78 | stats['FFE'] = add_basic_stats(FFE)
79 | stats['DTW'] = add_basic_stats(DTW)
80 | stats['MSD'] = add_basic_stats(MSD)
81 | stats['MCD'] = add_basic_stats(MCD)
82 | stats['Moments'] = MOMENTS
83 |
84 | # Create Output File
85 | tts_objective_metrics = {'tts_objecive_metrics': stats}
86 | json_object = json.dumps(tts_objective_metrics, indent = 4)
87 | with open("metrics.json", "w") as outfile:
88 | outfile.write(json_object)
89 |
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/config/__init__.py:
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/config/global_config.py:
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1 | import torch
2 | import torchaudio
3 |
4 | class GlobalConfig():
5 | def __init__(self):
6 |
7 | self.sr = 22050
8 |
9 | self.f0_min_pitch: int = 100
10 | self.f0_max_pitch: int = 500
11 |
12 | self.n_fft = 1024
13 | self.win_length: int = 1024
14 | self.hop_length: int = 256
15 | self.f0_min_mel: int = 0
16 | self.f0_max_mel = None
17 | self.harmo_thresh: float = 0.1
18 | self.window_fn = torch.hann_window
19 | self.log_mels = True
20 | self.n_mels = 80
21 | self.n_mfcc = 13
22 | self.power = 2
23 | self.fig_size: tuple = (16,10)
24 | self.dist_fn = 'compute_rms_dist'
25 | self.norm_align_type = 'path'
26 |
27 | self.mel_fn = torchaudio.transforms.MelSpectrogram(
28 | sample_rate = self.sr, n_fft=self.n_fft, win_length=self.win_length,
29 | hop_length=self.hop_length, f_min=self.f0_min_mel, f_max = self.f0_max_mel,
30 | n_mels=self.n_mels, window_fn=self.window_fn, power = self.power
31 | )
32 |
33 | self.melkwargs = {
34 | "n_fft": self.n_fft,
35 | "win_length": self.win_length,
36 | "hop_length": self.hop_length,
37 | "f_min": self.f0_min_mel,
38 | "f_max": self.f0_max_mel,
39 | "window_fn": self.window_fn,
40 | "n_mels": self.n_mels,
41 | "power": self.power
42 | }
43 |
44 | self.mfcc_fn = torchaudio.transforms.MFCC(
45 | sample_rate = self.sr, n_mfcc=self.n_mfcc, log_mels=self.log_mels, melkwargs=self.melkwargs
46 | )
47 |
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/metrics/DTW.py:
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1 | import argparse
2 | import librosa
3 | import numpy as np
4 | import torch
5 | from config.global_config import GlobalConfig
6 | from audio.pitch import dio, yin
7 | from metrics.dists import compute_rms_dist
8 |
9 | # https://github.com/pytorch/fairseq/blob/fcca32258c8e8bcc9f9890bf4714fa2f96b6b3e1/fairseq/tasks/text_to_speech.py#L304
10 | def batch_dynamic_time_warping(x1, x2, dist_fn, norm_align_type):
11 | """full batched DTW without any constraints
12 | x1: list of tensors of the references
13 | x2: list of tensors of the synthesizeds
14 | dist_fn: distance metric used to compute
15 | outputs: list of floats/ints or list of numpy arrays indexed by batch
16 | """
17 |
18 | distance, shapes = [], []
19 | for cur_x1, cur_x2 in zip(x1, x2):
20 | # Compute Distance between the Ref/Synths
21 | cur_d = eval(dist_fn)(cur_x1, cur_x2)
22 | # cur_d = dist_fn(cur_x1, cur_x2)
23 | distance.append(cur_d)
24 | shapes.append(distance[-1].size())
25 |
26 | # Get Max Spec Size of References
27 | max_m = max(ss[0] for ss in shapes)
28 | # Get Max Spec Size of Synths
29 | max_n = max(ss[1] for ss in shapes)
30 |
31 | # Pad all distance matrices with to be [Max_Synth, Max_Ref]
32 | distance = torch.stack(
33 | [torch.nn.functional.pad(dd, (0, max_n - dd.size(1), 0, max_m - dd.size(0))) for dd in distance]
34 | )
35 |
36 | # All Spec Sizes (List of [REF_SPEC_SIZE, SYNTH_SPEC_SIZE])
37 | shapes = torch.LongTensor(shapes).to(distance.device)
38 |
39 | # ptr: 0=left, 1=up-left, 2=up
40 | ptr2dij = {0: (0, -1), 1: (-1, -1), 2: (-1, 0)}
41 |
42 | bsz, m, n = distance.size()
43 | cumdist = torch.zeros_like(distance)
44 | backptr = torch.zeros_like(distance).type(torch.int32) - 1
45 |
46 | # initialize
47 | cumdist[:, 0, :] = distance[:, 0, :].cumsum(dim=-1)
48 | cumdist[:, :, 0] = distance[:, :, 0].cumsum(dim=-1)
49 | backptr[:, 0, :] = 0
50 | backptr[:, :, 0] = 2
51 |
52 | # DP with optimized anti-diagonal parallelization, O(M+N) steps
53 | for offset in range(2, m + n - 1):
54 | ind = _antidiag_indices(offset, 1, m, 1, n)
55 | c = torch.stack(
56 | [
57 | cumdist[:, ind[0], ind[1] - 1],
58 | cumdist[:, ind[0] - 1, ind[1] - 1],
59 | cumdist[:, ind[0] - 1, ind[1]],
60 | ],
61 | dim=2,
62 | )
63 | v, b = c.min(axis=-1)
64 |
65 | # Best Backpaths from each starting point
66 | backptr[:, ind[0], ind[1]] = b.int()
67 |
68 | # D(i,j) = d(x_i,y_i) + min(D(i-1,j-1), D(i-1,j), D(i,j-1))
69 | cumdist[:, ind[0], ind[1]] = v + distance[:, ind[0], ind[1]]
70 |
71 | # Evaluate the Optimal Backpath
72 | pathmap = torch.zeros_like(backptr)
73 | for b in range(bsz):
74 | i = m - 1 if shapes is None else (shapes[b][0] - 1).item()
75 | j = n - 1 if shapes is None else (shapes[b][1] - 1).item()
76 | dtwpath = [(i, j)]
77 | while (i != 0 or j != 0) and len(dtwpath) < 10000:
78 | assert i >= 0 and j >= 0
79 | di, dj = ptr2dij[backptr[b, i, j].item()]
80 | i, j = i + di, j + dj
81 | dtwpath.append((i, j))
82 | dtwpath = dtwpath[::-1]
83 | indices = torch.from_numpy(np.array(dtwpath)).long()
84 | pathmap[b, indices[:, 0], indices[:, 1]] = 1
85 |
86 | norm_align_costs = []
87 | cumdists = []
88 | backptrs = []
89 | pathmaps = []
90 | idxs = []
91 | itr = zip(shapes, cumdist, backptr, pathmap)
92 |
93 | # Get distortion for each item in the batch
94 | for (m, n), cumdist, backptr, pathmap in itr:
95 | cumdist = cumdist[:m, :n]
96 | backptr = backptr[:m, :n]
97 | pathmap = pathmap[:m, :n]
98 | divisor = _get_divisor(pathmap, norm_align_type)
99 | norm_align_cost = cumdist[-1, -1] / divisor
100 |
101 | # Get Indices
102 | p = pathmap.squeeze(0).cpu().detach().numpy()
103 | idx = np.transpose(np.nonzero(p))
104 |
105 | norm_align_costs.append(float(norm_align_cost.item()))
106 | cumdists.append(cumdist.numpy())
107 | backptrs.append(backptr.numpy())
108 | pathmaps.append(pathmap.numpy())
109 | idxs.append(idx)
110 |
111 | return {'norm_align_costs' : norm_align_costs, 'cumdists' : cumdists, 'backptrs' : backptrs, 'pathmaps' : pathmaps, 'idxs': idxs}
112 |
113 | def _antidiag_indices(offset, min_i=0, max_i=None, min_j=0, max_j=None):
114 | """
115 | for a (3, 4) matrix with min_i=1, max_i=3, min_j=1, max_j=4, outputs
116 | offset=2 (1, 1),
117 | offset=3 (2, 1), (1, 2)
118 | offset=4 (2, 2), (1, 3)
119 | offset=5 (2, 3)
120 | constraints:
121 | i + j = offset
122 | min_j <= j < max_j
123 | min_i <= offset - j < max_i
124 | """
125 | if max_i is None:
126 | max_i = offset + 1
127 | if max_j is None:
128 | max_j = offset + 1
129 | min_j = max(min_j, offset - max_i + 1, 0)
130 | max_j = min(max_j, offset - min_i + 1, offset + 1)
131 | j = torch.arange(min_j, max_j)
132 | i = offset - j
133 | return torch.stack([i, j])
134 |
135 | def _get_divisor(pathmap, normalize_type):
136 | if normalize_type is None:
137 | return 1
138 | elif normalize_type == "len1":
139 | return pathmap.size(0)
140 | elif normalize_type == "len2":
141 | return pathmap.size(1)
142 | elif normalize_type == "path":
143 | return pathmap.sum().item()
144 | else:
145 | raise ValueError(f"normalize_type {normalize_type} not supported")
146 |
147 | def main(gt_path, synth_path, pitch_algorithm):
148 |
149 | # Load Audio
150 | x_gt, _ = librosa.load(gt_path)
151 | x_synth, _ = librosa.load(synth_path)
152 |
153 | # Load Config
154 | config = GlobalConfig()
155 |
156 | # Compute Pitch
157 | pitch_gt = eval(pitch_algorithm)(x_gt, config)
158 | pitch_synth = eval(pitch_algorithm)(x_synth, config)
159 |
160 | # Initiate Lists of Tensors for Batched DTW
161 | x1 = []
162 | x2 = []
163 |
164 | # Append Each Tensor
165 | x1.append(torch.Tensor(pitch_gt['pitches']).unsqueeze_(1))
166 | x2.append(torch.Tensor(pitch_synth['pitches']).unsqueeze_(1))
167 |
168 | return print(batch_dynamic_time_warping(x1, x2, config.dist_fn, config.norm_align_type)['norm_align_costs'])
169 |
170 | if __name__ == "__main__":
171 |
172 | parser = argparse.ArgumentParser(description='Script to calculate the Dynamic Time Warping between the ground truth and the synthesized audios.')
173 | parser.add_argument("--gt_path", required = True, type=str, help = 'Path to corresponding ground truth audio in .wav format')
174 | parser.add_argument("--synth_path", required = True, type=str, help = 'Path to corresponding synthesized audio in .wav format')
175 | parser.add_argument("--pitch_algorithm", required = True, type=str, choices = ["dio", "yin"], help = 'Choose method of computing pitch')
176 | args = parser.parse_args()
177 |
178 | gt_path = args.gt_path
179 | synth_path = args.synth_path
180 | pitch_algorithm = args.pitch_algorithm
181 |
182 | main(gt_path, synth_path, pitch_algorithm)
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/metrics/FFE.py:
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1 | import argparse
2 | import librosa
3 | import numpy as np
4 | from audio.pitch import yin, dio
5 | from config.global_config import GlobalConfig
6 | from metrics.helpers import same_t_in_true_and_est, gross_pitch_error_frames, voicing_decision_error_frames
7 |
8 | # https://github.com/bastibe/MAPS-Scripts/blob/master/helper.py
9 | @same_t_in_true_and_est
10 | def f0_frame_error(true_t, true_f, est_t, est_f):
11 | """Measurs the percentage of frames that either are outside a threshold
12 | around the true pitch or that contain a voicing decision error.
13 | """
14 | gpe_frames = gross_pitch_error_frames(
15 | true_t, true_f, est_t, est_f
16 | )
17 | vde_frames = voicing_decision_error_frames(
18 | true_t, true_f, est_t, est_f
19 | )
20 | return (np.sum(gpe_frames) +
21 | np.sum(vde_frames)) / (len(true_t))
22 |
23 | def main(gt_path, synth_path, pitch_algorithm):
24 |
25 | # Load Audio
26 | x_gt, _ = librosa.load(gt_path)
27 | x_synth, _ = librosa.load(synth_path)
28 |
29 | # Load Config
30 | config = GlobalConfig()
31 |
32 | # Compute Pitch
33 | pitch_gt = eval(pitch_algorithm)(x_gt, config)
34 | pitch_synth = eval(pitch_algorithm)(x_synth, config)
35 |
36 | return print(f0_frame_error(np.array(pitch_gt['times']), np.array(pitch_gt['pitches']), np.array(pitch_synth['times']), np.array(pitch_synth['pitches'])))
37 |
38 | if __name__ == "__main__":
39 |
40 | parser = argparse.ArgumentParser(description='Script to calculate the F0 Frame Error between the ground truth and the synthesized audios.')
41 | parser.add_argument("--gt_path", required = True, type=str, help = 'Path to corresponding ground truth audio in .wav format')
42 | parser.add_argument("--synth_path", required = True, type=str, help = 'Path to corresponding synthesized audio in .wav format')
43 | parser.add_argument("--pitch_algorithm", required = True, type=str, choices = ["dio", "yin", "pyin"], help = 'Choose method of computing pitch')
44 |
45 | args = parser.parse_args()
46 |
47 | gt_path = args.gt_path
48 | synth_path = args.synth_path
49 | pitch_algorithm = args.pitch_algorithm
50 |
51 | main(gt_path, synth_path, pitch_algorithm)
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/metrics/GPE.py:
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1 | import argparse
2 | import librosa
3 | import numpy as np
4 | from audio.pitch import yin, dio
5 | from config.global_config import GlobalConfig
6 | from metrics.helpers import true_voiced_frames, gross_pitch_error_frames, same_t_in_true_and_est
7 |
8 | # https://github.com/bastibe/MAPS-Scripts/blob/master/helper.py
9 | @same_t_in_true_and_est
10 | def gross_pitch_error(true_t, true_f, est_t, est_f):
11 | """The relative frequency in percent of pitch estimates that are
12 | outside a threshold around the true pitch. Only frames that are
13 | considered pitched by both the ground truth and the estimator (if
14 | applicable) are considered.
15 | """
16 | correct_frames = true_voiced_frames(true_t, true_f, est_t, est_f)
17 | gpe_frames = gross_pitch_error_frames(true_t, true_f, est_t, est_f)
18 | return np.sum(gpe_frames) / np.sum(correct_frames)
19 |
20 | def main(gt_path, synth_path, pitch_algorithm):
21 |
22 | # Load Audio
23 | x_gt, _ = librosa.load(gt_path)
24 | x_synth, _ = librosa.load(synth_path)
25 |
26 | # Load Config
27 | config = GlobalConfig()
28 |
29 | # Compute Pitch
30 | pitch_gt = eval(pitch_algorithm)(x_gt, config)
31 | pitch_synth = eval(pitch_algorithm)(x_synth, config)
32 |
33 | return print(gross_pitch_error(np.array(pitch_gt['times']), np.array(pitch_gt['pitches']), np.array(pitch_synth['times']), np.array(pitch_synth['pitches'])))
34 |
35 | if __name__ == "__main__":
36 |
37 | parser = argparse.ArgumentParser(description='Script to calculate the Gross Pitch Error between the ground truth and the synthesized audios.')
38 | parser.add_argument("--gt_path", required = True, type=str, help = 'Path to corresponding ground truth audio in .wav format')
39 | parser.add_argument("--synth_path", required = True, type=str, help = 'Path to corresponding synthesized audio in .wav format')
40 | parser.add_argument("--pitch_algorithm", required = True, type=str, choices = ["dio", "yin"], help = 'Choose method of computing pitch')
41 |
42 | args = parser.parse_args()
43 |
44 | gt_path = args.gt_path
45 | synth_path = args.synth_path
46 | pitch_algorithm = args.pitch_algorithm
47 |
48 | main(gt_path, synth_path, pitch_algorithm)
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/metrics/MCD.py:
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1 | import argparse
2 | from config.global_config import GlobalConfig
3 | import librosa
4 | import torch
5 | import torchaudio
6 | from metrics.helpers import batch_compute_distortion
7 |
8 | # https://github.com/pytorch/fairseq/blob/fcca32258c8e8bcc9f9890bf4714fa2f96b6b3e1/examples/speech_synthesis/utils.py
9 | def batch_mel_cepstral_distortion(y1, y2, config):
10 | """
11 | https://arxiv.org/pdf/2011.03568.pdf
12 | The root mean squared error computed on 13-dimensional MFCC using DTW for
13 | alignment. MFCC features are computed from an 80-channel log-mel
14 | spectrogram using a 50ms Hann window and hop of 12.5ms.
15 | y1: list of arrays of waveforms of type double
16 | y2: list of arrays of waveforms of type double
17 | sr: sampling rate
18 | """
19 |
20 | return batch_compute_distortion(
21 | y1,
22 | y2,
23 | config.sr,
24 | lambda y: config.mfcc_fn.to(y1[0].device)(y).transpose(-1, -2),
25 | 'compute_rms_dist',
26 | config.norm_align_type,
27 | )
28 |
29 | def main(gt_path, synth_path):
30 |
31 | # Load Audio
32 | x_gt, sr_gt = librosa.load(gt_path)
33 | x_synth, sr_synth = librosa.load(synth_path)
34 |
35 | # Load Config
36 | config = GlobalConfig()
37 |
38 | # Initiate Lists of Tensors for Batched DTW
39 | x1 = []
40 | x2 = []
41 |
42 | # Append Each Tensor
43 | x1.append(torch.Tensor(x_gt))
44 | x2.append(torch.Tensor(x_synth))
45 |
46 | return print(batch_mel_cepstral_distortion(x1, x2, config))
47 |
48 | if __name__ == "__main__":
49 |
50 | parser = argparse.ArgumentParser(description='Script to calculate the Mel Cepstral Distortion between the ground truth and the synthesized audios.')
51 | parser.add_argument("--gt_path", required = True, type=str, help = 'Path to corresponding ground truth audio in .wav format')
52 | parser.add_argument("--synth_path", required = True, type=str, help = 'Path to corresponding synthesized audio in .wav format')
53 |
54 | args = parser.parse_args()
55 |
56 | gt_path = args.gt_path
57 | synth_path = args.synth_path
58 |
59 | main(gt_path, synth_path)
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/metrics/MSD.py:
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1 | import argparse
2 | from config.global_config import GlobalConfig
3 | import librosa
4 | import torchaudio
5 | import torch
6 | from metrics.helpers import batch_compute_distortion
7 |
8 | # https://github.com/pytorch/fairseq/blob/fcca32258c8e8bcc9f9890bf4714fa2f96b6b3e1/examples/speech_synthesis/utils.py
9 | def batch_mel_spectral_distortion(
10 | y1, y2, config
11 | ):
12 | """
13 | https://arxiv.org/pdf/2011.03568.pdf
14 | Same as Mel Cepstral Distortion, but computed on log-mel spectrograms.
15 | """
16 | offset = 1e-6
17 | return batch_compute_distortion(
18 | y1, y2, config.sr, lambda y: torch.log(config.mel_fn.to(y1[0].device)(y) + offset).transpose(-1, -2),
19 | 'compute_rms_dist', config.norm_align_type
20 | )
21 |
22 | def main(gt_path, synth_path):
23 |
24 | # Load Audio
25 | x_gt, _ = librosa.load(gt_path)
26 | x_synth, _ = librosa.load(synth_path)
27 |
28 | # Load Config
29 | config = GlobalConfig()
30 |
31 | # Initiate Lists of Tensors for Batched DTW
32 | x1 = []
33 | x2 = []
34 |
35 | # Append Each Tensor
36 | x1.append(torch.Tensor(x_gt))
37 | x2.append(torch.Tensor(x_synth))
38 |
39 | return print(batch_mel_spectral_distortion(x1, x2, config))
40 |
41 | if __name__ == "__main__":
42 |
43 | parser = argparse.ArgumentParser(description='Script to calculate the Mel Spectral Distortion between the ground truth and the synthesized audios.')
44 | parser.add_argument("--gt_path", required = True, type=str, help = 'Path to corresponding ground truth audio in .wav format')
45 | parser.add_argument("--synth_path", required = True, type=str, help = 'Path to corresponding synthesized audio in .wav format')
46 |
47 | args = parser.parse_args()
48 |
49 | gt_path = args.gt_path
50 | synth_path = args.synth_path
51 |
52 | main(gt_path, synth_path)
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/metrics/VDE.py:
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1 | import argparse
2 | from distutils.command.config import config
3 | import librosa
4 | import numpy as np
5 | from audio.pitch import yin, dio
6 | from config.global_config import GlobalConfig
7 | from metrics.helpers import same_t_in_true_and_est, voicing_decision_error_frames
8 |
9 | # https://github.com/bastibe/MAPS-Scripts/blob/master/helper.py
10 | @same_t_in_true_and_est
11 | def voicing_decision_error(true_t, true_f, est_t, est_f):
12 | """Percentages of frames that contain error in the voicing
13 | decision, according to the reference.
14 | """
15 | vde_frames = voicing_decision_error_frames(
16 | true_t, true_f, est_t, est_f
17 | )
18 | return np.sum(vde_frames) / (len(true_t))
19 |
20 | def main(gt_path, synth_path, pitch_algorithm):
21 |
22 | # Load Audio
23 | x_gt, _ = librosa.load(gt_path)
24 | x_synth, _ = librosa.load(synth_path)
25 |
26 | # Load Config
27 | config = GlobalConfig()
28 |
29 | # Compute Pitch
30 | pitch_gt = eval(pitch_algorithm)(x_gt, config)
31 | pitch_synth = eval(pitch_algorithm)(x_synth, config)
32 |
33 | return print(voicing_decision_error(np.array(pitch_gt['times']), np.array(pitch_gt['pitches']), np.array(pitch_synth['times']), np.array(pitch_synth['pitches'])))
34 |
35 | if __name__ == "__main__":
36 |
37 | parser = argparse.ArgumentParser(description='Script to calculate the Voicing Decision Error between the ground truth and the synthesized audios.')
38 | parser.add_argument("--gt_path", required = True, type=str, help = 'Path to corresponding ground truth audio in .wav format')
39 | parser.add_argument("--synth_path", required = True, type=str, help = 'Path to corresponding synthesized audio in .wav format')
40 | parser.add_argument("--pitch_algorithm", required = True, type=str, choices = ["dio", "yin", "pyin"], help = 'Choose method of computing pitch')
41 |
42 | args = parser.parse_args()
43 |
44 | gt_path = args.gt_path
45 | synth_path = args.synth_path
46 | pitch_algorithm = args.pitch_algorithm
47 |
48 | main(gt_path, synth_path, pitch_algorithm)
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/metrics/__init__.py:
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https://raw.githubusercontent.com/AI-Unicamp/TTS-Objective-Metrics/e183fd0f4ec484173b13b86ae315c862af9db888/metrics/__init__.py
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/metrics/dists.py:
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1 | import torch
2 |
3 | # try to make the distance function more general (only between two vectors and the expand it?)
4 | def compute_lp_dist(x1, x2, p):
5 | """compute an (m, n) Lp distance matrix from (m, d) and (n, d) matrices
6 | with all combination of distances between m and n"""
7 | return torch.cdist(x1.unsqueeze(0), x2.unsqueeze(0), p=p).squeeze(0)
8 |
9 | def compute_rms_dist(x1, x2):
10 | l2_dist = compute_lp_dist(x1, x2, 2)
11 | return (l2_dist.pow(2) / x1.size(1)).pow(0.5)
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/metrics/helpers.py:
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1 | # https://github.com/bastibe/MAPS-Scripts/blob/master/helper.py
2 | # GPL 3.0 LICENSE
3 | import numpy as np
4 | import torch
5 | from metrics.DTW import batch_dynamic_time_warping
6 | from scipy.interpolate import interp1d
7 |
8 | def same_t_in_true_and_est(func):
9 | def new_func(true_t, true_f, est_t, est_f):
10 | assert type(true_t) is np.ndarray
11 | assert type(true_f) is np.ndarray
12 | assert type(est_t) is np.ndarray
13 | assert type(est_f) is np.ndarray
14 |
15 | interpolated_f = interp1d(
16 | est_t, est_f, bounds_error=False, kind='nearest', fill_value=0
17 | )(true_t)
18 | return func(true_t, true_f, true_t, interpolated_f)
19 | return new_func
20 |
21 | def gross_pitch_error_frames(true_t, true_f, est_t, est_f, eps=1e-8):
22 | voiced_frames = true_voiced_frames(true_t, true_f, est_t, est_f)
23 | true_f_p_eps = [x + eps for x in true_f]
24 | pitch_error_frames = np.abs(est_f / true_f_p_eps - 1) > 0.2
25 | return voiced_frames & pitch_error_frames
26 |
27 | def voicing_decision_error_frames(true_t, true_f, est_t, est_f):
28 | return (est_f != 0) != (true_f != 0)
29 |
30 | def true_voiced_frames(true_t, true_f, est_t, est_f):
31 | return (est_f != 0) & (true_f != 0)
32 |
33 |
34 | def batch_compute_distortion(y1, y2, sr, feat_fn, dist_fn, normalize_align_type):
35 | # Distances, Sizes, Spectrograms of Ref, Spectrograms of Synths
36 | s, x1, x2 = [], [], []
37 |
38 | # Loop throug each (Ref, Synth) pair
39 | for cur_y1, cur_y2 in zip(y1, y2):
40 | assert cur_y1.ndim == 1 and cur_y2.ndim == 1
41 |
42 | # Compute Mel-Specs for MSD or MFCC for MCD
43 | cur_x1 = feat_fn(cur_y1)
44 | cur_x2 = feat_fn(cur_y2)
45 | x1.append(cur_x1)
46 | x2.append(cur_x2)
47 |
48 | # Save Sizes for UnPadding
49 | size = torch.empty(cur_x1.size(0), cur_x2.size(0))
50 | s.append(size.size())
51 | s = torch.LongTensor(s).to(cur_y1.device)
52 |
53 | # Get DTW D matrix, Minimal Cost Paths, Optimal Path
54 | return batch_dynamic_time_warping(x1,x2,dist_fn, normalize_align_type)['norm_align_costs']
55 |
56 |
57 | def add_basic_stats(dic):
58 | mean = np.mean(list(dic.values()))
59 | std = np.std(list(dic.values()))
60 | max = np.max(list(dic.values()))
61 | argmax = float(np.argmax(list(dic.values())))
62 | min = np.min(list(dic.values()))
63 | argmin = float(np.argmin(list(dic.values())))
64 | dic['stats'] = {'mean':mean, 'std':std, 'max':max, 'argmax':argmax, 'min':min, 'argmin':argmin}
65 | return dic
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/metrics/moments.py:
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1 | import argparse
2 | import librosa
3 | import numpy as np
4 | from audio.pitch import yin, dio
5 | from config.global_config import GlobalConfig
6 | from scipy.stats import tstd, skew, kurtosis
7 |
8 | def estimate_moments(pitch_gt, pitch_skew):
9 | gt_moments = {'std':tstd(pitch_gt), 'skew':skew(pitch_gt), 'kurt':kurtosis(pitch_gt)}
10 | synth_moments = {'std':tstd(pitch_skew), 'skew':skew(pitch_skew), 'kurt':kurtosis(pitch_skew)}
11 | return {'GT': gt_moments, 'SYNTH':synth_moments}
12 |
13 | def main(gt_path, synth_path, pitch_algorithm):
14 |
15 | # Load Audio
16 | x_gt, _ = librosa.load(gt_path)
17 | x_synth, _ = librosa.load(synth_path)
18 |
19 | # Load Config
20 | config = GlobalConfig()
21 |
22 | # Compute Pitch
23 | pitch_gt = eval(pitch_algorithm)(x_gt, config)['pitches']
24 | pitch_synth = eval(pitch_algorithm)(x_synth, config)['pitches']
25 |
26 | # Compute Moments
27 | return print(estimate_moments(pitch_gt, pitch_synth))
28 |
29 | if __name__ == "__main__":
30 |
31 | parser = argparse.ArgumentParser(description='Script to estimate the Statistical Moments of the ground truth and the synthesized audios.')
32 | parser.add_argument("--gt_path", required = True, type=str, help = 'Path to corresponding ground truth audio in .wav format')
33 | parser.add_argument("--synth_path", required = True, type=str, help = 'Path to corresponding synthesized audio in .wav format')
34 | parser.add_argument("--pitch_algorithm", required = True, type=str, choices = ["dio", "yin"], help = 'Choose method of computing pitch')
35 |
36 | args = parser.parse_args()
37 |
38 | gt_path = args.gt_path
39 | synth_path = args.synth_path
40 | pitch_algorithm = args.pitch_algorithm
41 |
42 | main(gt_path, synth_path, pitch_algorithm)
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/requirements.txt:
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https://raw.githubusercontent.com/AI-Unicamp/TTS-Objective-Metrics/e183fd0f4ec484173b13b86ae315c862af9db888/requirements.txt
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