#!/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 """ 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 = '''

Voice Demo





''' 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()