Files
Brandon Thomas 4a222ad815 realtime-voice-conversion based on last year's work
This is based on the work I did in (July - October)ish 2019 to get real
time voice conversion working. I've removed it from the `voder` library
and made it into an isolable program. I'll work on extending the
capabilities.
2020-09-11 09:06:48 -04:00

279 lines
11 KiB
Python
Executable File

#!/usr/bin/env python3
"""
Starts a demo HTTP server to capture and transform audio
as a live demonstration of the trained model.
Brandon Thomas 2019-07-29 <bt@brand.io> <echelon@gmail.com>
"""
import argparse
import falcon
import io
import librosa
import numpy as np
import os
import scipy
import soundfile
import tensorflow as tf
import subprocess
import tempfile
from falcon_multipart.middleware import MultipartMiddleware
from model import CycleGAN
from preprocess import *
from wsgiref import simple_server
print("TensorFlow version: {}".format(tf.version.VERSION))
INDEX_HTML = '''
<!doctype html>
<html>
<body>
<h1>Voice Demo</h1>
<script src="./script/recorder.js" type="application/javascript"></script>
<script type="application/javascript">
var audio_context;
var recorder;
// NB: Capture code originates from 'recorder.js' demo.
function startUserMedia(stream) {
var input = audio_context.createMediaStreamSource(stream);
window.recorder = new Recorder(input);
}
function createDownloadLink() {
window.recorder && window.recorder.exportWAV(function(blob) {
var xhr = new XMLHttpRequest();
xhr.responseType = 'blob';
xhr.onload=function(e) {
console.log('response received');
var objectUrl = window.URL.createObjectURL(this.response);
console.log('audio URL', objectUrl);
var audio = document.getElementById('audio');
audio.src = objectUrl;
audio.play();
};
var fd = new FormData();
fd.append("audio_data", blob, "audio_file.wav");
console.log(blob);
console.log('sending...');
xhr.open("POST","/upload",true);
xhr.send(fd);
console.log('sent');
});
}
function startRecording(button) {
window.recorder && window.recorder.record();
document.getElementById('start').disabled = true;
document.getElementById('stop').disabled = false;
}
function stopRecording(button) {
window.recorder && window.recorder.stop();
document.getElementById('start').disabled = false;
document.getElementById('stop').disabled = true;
createDownloadLink();
window.recorder.clear();
}
window.onload = function init() {
try {
audio_context = new AudioContext();
} catch (e) {
alert('No web audio support in this browser!');
}
var media = navigator.mediaDevices.getUserMedia({ audio: true, video: false });
window.media = media;
console.log('media', media);
media.then(startUserMedia);
};
</script>
<button id="start" onclick="startRecording(this);">record</button>
<br>
<br>
<button id="stop" onclick="stopRecording(this);" disabled>stop</button>
<br>
<br>
<audio controls id="audio">
<source id="source" src="" type="audio/wav">
Your browser does not support the audio element.
</audio>
</body>
</html>
'''
class Converter():
def __init__(self, model_dir, model_name):
self.num_features = 24
self.sampling_rate = 16000
self.frame_period = 5.0
self.model = CycleGAN(num_features = self.num_features, mode = 'test')
self.model.load(filepath = os.path.join(model_dir, model_name))
"""
# NB: Save the graph
definition = self.model.sess.graph_def
directory = 'saved_model_2'
tf.train.write_graph(definition, directory, 'saved_model_2.pb', as_text=True)
# https://github.com/tensorflow/models/issues/3530#issuecomment-395968881
output_dir = './saved_model/'
builder = tf.saved_model.builder.SavedModelBuilder(output_dir)
builder.add_meta_graph_and_variables(
self.model.sess,
[tf.saved_model.tag_constants.SERVING],
main_op=tf.tables_initializer(),
)
builder.save()
"""
"""
builder.add_meta_graph_and_variables(
self.model.sess,
[tf.saved_model.tag_constants.SERVING],
signature_def_map={
'predict_images':
prediction_signature,
signature_constants.DEFAULT_SERVING_SIGNATURE_DEF_KEY:
classification_signature,
},
main_op=tf.tables_initializer())
"""
self.mcep_normalization_params = np.load(os.path.join(model_dir, 'mcep_normalization.npz'))
self.mcep_mean_A = self.mcep_normalization_params['mean_A']
self.mcep_std_A = self.mcep_normalization_params['std_A']
self.mcep_mean_B = self.mcep_normalization_params['mean_B']
self.mcep_std_B = self.mcep_normalization_params['std_B']
self.logf0s_normalization_params = np.load(os.path.join(model_dir, 'logf0s_normalization.npz'))
self.logf0s_mean_A = self.logf0s_normalization_params['mean_A']
self.logf0s_std_A = self.logf0s_normalization_params['std_A']
self.logf0s_mean_B = self.logf0s_normalization_params['mean_B']
self.logf0s_std_B = self.logf0s_normalization_params['std_B']
def convert(self, wav, conversion_direction='A2B'):
wav = wav_padding(wav = wav, sr = self.sampling_rate, frame_period = self.frame_period, multiple = 4)
f0, timeaxis, sp, ap = world_decompose(wav = wav, fs = self.sampling_rate, frame_period = self.frame_period)
coded_sp = world_encode_spectral_envelop(sp = sp, fs = self.sampling_rate, dim = self.num_features)
coded_sp_transposed = coded_sp.T
if conversion_direction == 'A2B':
f0_converted = pitch_conversion(f0 = f0, mean_log_src = self.logf0s_mean_A, std_log_src = self.logf0s_std_A, mean_log_target = self.logf0s_mean_B, std_log_target = self.logf0s_std_B)
coded_sp_norm = (coded_sp_transposed - self.mcep_mean_A) / self.mcep_std_A
coded_sp_converted_norm = self.model.test(inputs = np.array([coded_sp_norm]), direction = conversion_direction)[0]
coded_sp_converted = coded_sp_converted_norm * self.mcep_std_B + self.mcep_mean_B
else:
f0_converted = pitch_conversion(f0 = f0, mean_log_src = self.logf0s_mean_B, std_log_src = self.logf0s_std_B, mean_log_target = self.logf0s_mean_A, std_log_target = self.logf0s_std_A)
coded_sp_norm = (coded_sp_transposed - self.mcep_mean_B) / self.mcep_std_B
coded_sp_converted_norm = self.model.test(inputs = np.array([coded_sp_norm]), direction = conversion_direction)[0]
coded_sp_converted = coded_sp_converted_norm * self.mcep_std_A + self.mcep_mean_A
coded_sp_converted = coded_sp_converted.T
coded_sp_converted = np.ascontiguousarray(coded_sp_converted)
decoded_sp_converted = world_decode_spectral_envelop(coded_sp = coded_sp_converted, fs = self.sampling_rate)
wav_transformed = world_speech_synthesis(f0 = f0_converted, decoded_sp = decoded_sp_converted, ap = ap, fs = self.sampling_rate, frame_period = self.frame_period)
# For debugging model output, uncomment the following line:
# librosa.output.write_wav('model_output.wav', wav_transformed, self.sampling_rate)
# TODO: Perhaps ditch this. It's probably unnecessary work.
upsampled = librosa.resample(wav_transformed, self.sampling_rate, 48000)
pcm_data = upsampled.astype(np.float64)
stereo_pcm_data = np.tile(pcm_data, (2,1)).T
buf = io.BytesIO()
scipy.io.wavfile.write(buf, 48000, stereo_pcm_data.astype(np.float32))
return buf
# Set up model
# This should live long in memory, so we do it up front.
model_dir_default = './model/sf1_tm1'
model_name_default = 'sf1_tm1.ckpt'
converter = Converter(model_dir_default, model_name_default)
class IndexHandler():
def on_get(self, request, response):
response.content_type = 'text/html'
response.body = INDEX_HTML
class ApiHandler():
def on_post(self, request, response):
# NB: uses middleware to pull out data.
form_data = request.params['audio_data'].file
data, samplerate = soundfile.read(form_data)
print('samplerate', samplerate)
print('data', data)
print('data.shape', data.shape)
print('data.dtype', data.dtype)
# For debugging browser input, uncomment the following line:
# scipy.io.wavfile.write('browser_input_audio.wav', samplerate, data)
# NB: Convert the input stereo signal into mono.
# In the future the frontend should be responsible for sampling details.
mono = data[:, 0]
print('mono', mono)
print('mono.shape', mono.shape)
print('mono.dtype', mono.dtype)
# NB: We must downsample to the rate that the network is trained on.
downsampled = librosa.resample(mono, samplerate, 16000)
print('downsampled', downsampled)
print('downsampled.shape', downsampled.shape)
print('downsampled.dtype', downsampled.dtype)
# Evaluate the model
print(">>> Converting...")
results = converter.convert(downsampled, conversion_direction = 'A2B')
print('type(results)', type(results))
temp_dir = tempfile.TemporaryDirectory(prefix='tmp_ml_audio')
temp_file = tempfile.NamedTemporaryFile(suffix='.wav')
temp_file.write(results.read())
out_file = temp_dir.name + '/output.ogg'
# NB: Browsers have a great deal of trouble decoding WAV files unless they are in the
# narrow slice of the WAV spec expected. None of the {librosa, scipy, soundfile} python
# tools do a good job of this, so here we shell out to ffmpeg and generate OGG.
# It's lazy and messy, but it works for now.
# See https://github.com/librosa/librosa/issues/361 for a survey of the library landscape
# See https://bugzilla.mozilla.org/show_bug.cgi?id=523837 for one of dozens of browser codec bugs
_stdout = subprocess.check_output(['ffmpeg', '-i', temp_file.name, '-acodec', 'libvorbis', out_file])
response.content_type = 'audio/ogg'
with open(out_file, mode='rb') as f:
response.data = f.read()
def main():
parser = argparse.ArgumentParser()
parser.add_argument('--port', type=int, default=8000)
args = parser.parse_args()
api = falcon.API(middleware=[MultipartMiddleware()])
api.add_route('/', IndexHandler())
api.add_route('/upload', ApiHandler())
api.add_static_route('/script', os.path.abspath('./script'))
api.add_static_route('/sound', os.path.abspath('./sound'))
print('Serving on 0.0.0.0:%d' % args.port)
simple_server.make_server('0.0.0.0', args.port, api).serve_forever()
if __name__ == '__main__':
main()