|
|
|
"""Server-end for the ASR demo."""
|
|
|
|
import os
|
|
|
|
import time
|
|
|
|
import random
|
|
|
|
import argparse
|
|
|
|
import distutils.util
|
|
|
|
from time import gmtime, strftime
|
|
|
|
import SocketServer
|
|
|
|
import struct
|
|
|
|
import wave
|
|
|
|
import pyaudio
|
|
|
|
import paddle.v2 as paddle
|
|
|
|
from utils import print_arguments
|
|
|
|
from data_utils.data import DataGenerator
|
|
|
|
from model import DeepSpeech2Model
|
|
|
|
from data_utils.utils import read_manifest
|
|
|
|
|
|
|
|
parser = argparse.ArgumentParser(description=__doc__)
|
|
|
|
parser.add_argument(
|
|
|
|
"--host_ip",
|
|
|
|
default="localhost",
|
|
|
|
type=str,
|
|
|
|
help="Server IP address. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--host_port",
|
|
|
|
default=8086,
|
|
|
|
type=int,
|
|
|
|
help="Server Port. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--speech_save_dir",
|
|
|
|
default="demo_cache",
|
|
|
|
type=str,
|
|
|
|
help="Directory for saving demo speech. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--vocab_filepath",
|
|
|
|
default='datasets/vocab/eng_vocab.txt',
|
|
|
|
type=str,
|
|
|
|
help="Vocabulary filepath. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--mean_std_filepath",
|
|
|
|
default='mean_std.npz',
|
|
|
|
type=str,
|
|
|
|
help="Manifest path for normalizer. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--warmup_manifest_path",
|
|
|
|
default='datasets/manifest.test',
|
|
|
|
type=str,
|
|
|
|
help="Manifest path for warmup test. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--specgram_type",
|
|
|
|
default='linear',
|
|
|
|
type=str,
|
|
|
|
help="Feature type of audio data: 'linear' (power spectrum)"
|
|
|
|
" or 'mfcc'. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--num_conv_layers",
|
|
|
|
default=2,
|
|
|
|
type=int,
|
|
|
|
help="Convolution layer number. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--num_rnn_layers",
|
|
|
|
default=3,
|
|
|
|
type=int,
|
|
|
|
help="RNN layer number. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--rnn_layer_size",
|
|
|
|
default=512,
|
|
|
|
type=int,
|
|
|
|
help="RNN layer cell number. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--use_gpu",
|
|
|
|
default=True,
|
|
|
|
type=distutils.util.strtobool,
|
|
|
|
help="Use gpu or not. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--model_filepath",
|
|
|
|
default='checkpoints/params.latest.tar.gz',
|
|
|
|
type=str,
|
|
|
|
help="Model filepath. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--decode_method",
|
|
|
|
default='beam_search',
|
|
|
|
type=str,
|
|
|
|
help="Method for ctc decoding: best_path or beam_search. "
|
|
|
|
"(default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--beam_size",
|
|
|
|
default=100,
|
|
|
|
type=int,
|
|
|
|
help="Width for beam search decoding. (default: %(default)d)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--language_model_path",
|
|
|
|
default="lm/data/common_crawl_00.prune01111.trie.klm",
|
|
|
|
type=str,
|
|
|
|
help="Path for language model. (default: %(default)s)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--alpha",
|
|
|
|
default=0.36,
|
|
|
|
type=float,
|
|
|
|
help="Parameter associated with language model. (default: %(default)f)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--beta",
|
|
|
|
default=0.25,
|
|
|
|
type=float,
|
|
|
|
help="Parameter associated with word count. (default: %(default)f)")
|
|
|
|
parser.add_argument(
|
|
|
|
"--cutoff_prob",
|
|
|
|
default=0.99,
|
|
|
|
type=float,
|
|
|
|
help="The cutoff probability of pruning"
|
|
|
|
"in beam search. (default: %(default)f)")
|
|
|
|
args = parser.parse_args()
|
|
|
|
|
|
|
|
|
|
|
|
class AsrTCPServer(SocketServer.TCPServer):
|
|
|
|
"""The ASR TCP Server."""
|
|
|
|
|
|
|
|
def __init__(self,
|
|
|
|
server_address,
|
|
|
|
RequestHandlerClass,
|
|
|
|
speech_save_dir,
|
|
|
|
audio_process_handler,
|
|
|
|
bind_and_activate=True):
|
|
|
|
self.speech_save_dir = speech_save_dir
|
|
|
|
self.audio_process_handler = audio_process_handler
|
|
|
|
SocketServer.TCPServer.__init__(
|
|
|
|
self, server_address, RequestHandlerClass, bind_and_activate=True)
|
|
|
|
|
|
|
|
|
|
|
|
class AsrRequestHandler(SocketServer.BaseRequestHandler):
|
|
|
|
"""The ASR request handler."""
|
|
|
|
|
|
|
|
def handle(self):
|
|
|
|
# receive data through TCP socket
|
|
|
|
chunk = self.request.recv(1024)
|
|
|
|
target_len = struct.unpack('>i', chunk[:4])[0]
|
|
|
|
data = chunk[4:]
|
|
|
|
while len(data) < target_len:
|
|
|
|
chunk = self.request.recv(1024)
|
|
|
|
data += chunk
|
|
|
|
# write to file
|
|
|
|
filename = self._write_to_file(data)
|
|
|
|
|
|
|
|
print("Received utterance[length=%d] from %s, saved to %s." %
|
|
|
|
(len(data), self.client_address[0], filename))
|
|
|
|
start_time = time.time()
|
|
|
|
transcript = self.server.audio_process_handler(filename)
|
|
|
|
finish_time = time.time()
|
|
|
|
print("Response Time: %f, Transcript: %s" %
|
|
|
|
(finish_time - start_time, transcript))
|
|
|
|
self.request.sendall(transcript)
|
|
|
|
|
|
|
|
def _write_to_file(self, data):
|
|
|
|
# prepare save dir and filename
|
|
|
|
if not os.path.exists(self.server.speech_save_dir):
|
|
|
|
os.mkdir(self.server.speech_save_dir)
|
|
|
|
timestamp = strftime("%Y%m%d%H%M%S", gmtime())
|
|
|
|
out_filename = os.path.join(
|
|
|
|
self.server.speech_save_dir,
|
|
|
|
timestamp + "_" + self.client_address[0] + ".wav")
|
|
|
|
# write to wav file
|
|
|
|
file = wave.open(out_filename, 'wb')
|
|
|
|
file.setnchannels(1)
|
|
|
|
file.setsampwidth(4)
|
|
|
|
file.setframerate(16000)
|
|
|
|
file.writeframes(data)
|
|
|
|
file.close()
|
|
|
|
return out_filename
|
|
|
|
|
|
|
|
|
|
|
|
def warm_up_test(audio_process_handler,
|
|
|
|
manifest_path,
|
|
|
|
num_test_cases,
|
|
|
|
random_seed=0):
|
|
|
|
"""Warming-up test."""
|
|
|
|
manifest = read_manifest(manifest_path)
|
|
|
|
rng = random.Random(random_seed)
|
|
|
|
samples = rng.sample(manifest, num_test_cases)
|
|
|
|
for idx, sample in enumerate(samples):
|
|
|
|
print("Warm-up Test Case %d: %s", idx, sample['audio_filepath'])
|
|
|
|
start_time = time.time()
|
|
|
|
transcript = audio_process_handler(sample['audio_filepath'])
|
|
|
|
finish_time = time.time()
|
|
|
|
print("Response Time: %f, Transcript: %s" %
|
|
|
|
(finish_time - start_time, transcript))
|
|
|
|
|
|
|
|
|
|
|
|
def start_server():
|
|
|
|
"""Start the ASR server"""
|
|
|
|
# prepare data generator
|
|
|
|
data_generator = DataGenerator(
|
|
|
|
vocab_filepath=args.vocab_filepath,
|
|
|
|
mean_std_filepath=args.mean_std_filepath,
|
|
|
|
augmentation_config='{}',
|
|
|
|
specgram_type=args.specgram_type,
|
|
|
|
num_threads=1)
|
|
|
|
# prepare ASR model
|
|
|
|
ds2_model = DeepSpeech2Model(
|
|
|
|
vocab_size=data_generator.vocab_size,
|
|
|
|
num_conv_layers=args.num_conv_layers,
|
|
|
|
num_rnn_layers=args.num_rnn_layers,
|
|
|
|
rnn_layer_size=args.rnn_layer_size,
|
|
|
|
pretrained_model_path=args.model_filepath)
|
|
|
|
|
|
|
|
# prepare ASR inference handler
|
|
|
|
def file_to_transcript(filename):
|
|
|
|
feature = data_generator.process_utterance(filename, "")
|
|
|
|
result_transcript = ds2_model.infer_batch(
|
|
|
|
infer_data=[feature],
|
|
|
|
decode_method=args.decode_method,
|
|
|
|
beam_alpha=args.alpha,
|
|
|
|
beam_beta=args.beta,
|
|
|
|
beam_size=args.beam_size,
|
|
|
|
cutoff_prob=args.cutoff_prob,
|
|
|
|
vocab_list=data_generator.vocab_list,
|
|
|
|
language_model_path=args.language_model_path,
|
|
|
|
num_processes=1)
|
|
|
|
return result_transcript[0]
|
|
|
|
|
|
|
|
# warming up with utterrances sampled from Librispeech
|
|
|
|
print('-----------------------------------------------------------')
|
|
|
|
print('Warming up ...')
|
|
|
|
warm_up_test(
|
|
|
|
audio_process_handler=file_to_transcript,
|
|
|
|
manifest_path=args.warmup_manifest_path,
|
|
|
|
num_test_cases=3)
|
|
|
|
print('-----------------------------------------------------------')
|
|
|
|
|
|
|
|
# start the server
|
|
|
|
server = AsrTCPServer(
|
|
|
|
server_address=(args.host_ip, args.host_port),
|
|
|
|
RequestHandlerClass=AsrRequestHandler,
|
|
|
|
speech_save_dir=args.speech_save_dir,
|
|
|
|
audio_process_handler=file_to_transcript)
|
|
|
|
print("ASR Server Started.")
|
|
|
|
server.serve_forever()
|
|
|
|
|
|
|
|
|
|
|
|
def main():
|
|
|
|
print_arguments(args)
|
|
|
|
paddle.init(use_gpu=args.use_gpu, trainer_count=1)
|
|
|
|
start_server()
|
|
|
|
|
|
|
|
|
|
|
|
if __name__ == "__main__":
|
|
|
|
main()
|