Xinghai Sun
8 years ago
parent
a3807d9cb5
commit
0ebf36b98f
@ -0,0 +1,75 @@
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from pynput import keyboard
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import struct
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import socket
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import sys
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import pyaudio
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HOST, PORT = "10.104.18.14", 8086
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is_recording = False
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enable_trigger_record = True
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def on_press(key):
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global is_recording, enable_trigger_record
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if key == keyboard.Key.space:
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if (not is_recording) and enable_trigger_record:
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sys.stdout.write("Start Recording ... ")
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sys.stdout.flush()
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is_recording = True
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def on_release(key):
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global is_recording, enable_trigger_record
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if key == keyboard.Key.esc:
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return False
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elif key == keyboard.Key.space:
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if is_recording == True:
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is_recording = False
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data_list = []
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def callback(in_data, frame_count, time_info, status):
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global data_list, is_recording, enable_trigger_record
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if is_recording:
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data_list.append(in_data)
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enable_trigger_record = False
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elif len(data_list) > 0:
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# Connect to server and send data
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sock = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
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sock.connect((HOST, PORT))
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sent = ''.join(data_list)
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sock.sendall(struct.pack('>i', len(sent)) + sent)
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print('Speech[length=%d] Sent.' % len(sent))
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# Receive data from the server and shut down
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received = sock.recv(1024)
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print "Recognition Results: {}".format(received)
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sock.close()
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data_list = []
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enable_trigger_record = True
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return (in_data, pyaudio.paContinue)
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def main():
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p = pyaudio.PyAudio()
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stream = p.open(
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format=pyaudio.paInt32,
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channels=1,
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rate=16000,
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input=True,
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stream_callback=callback)
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stream.start_stream()
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with keyboard.Listener(
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on_press=on_press, on_release=on_release) as listener:
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listener.join()
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stream.stop_stream()
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stream.close()
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p.terminate()
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if __name__ == "__main__":
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main()
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@ -0,0 +1,208 @@
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import os
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import time
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import argparse
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import distutils.util
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from time import gmtime, strftime
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import SocketServer
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import struct
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import wave
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import pyaudio
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import paddle.v2 as paddle
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from data_utils.data import DataGenerator
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from model import DeepSpeech2Model
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import utils
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parser = argparse.ArgumentParser(description=__doc__)
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parser.add_argument(
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"--host_ip",
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default="10.104.18.14",
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type=str,
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help="Server IP address. (default: %(default)s)")
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parser.add_argument(
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"--host_port",
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default=8086,
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type=int,
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help="Server Port. (default: %(default)s)")
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parser.add_argument(
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"--speech_save_dir",
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default="demo_cache",
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type=str,
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help="Directory for saving demo speech. (default: %(default)s)")
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parser.add_argument(
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"--vocab_filepath",
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default='datasets/vocab/eng_vocab.txt',
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type=str,
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help="Vocabulary filepath. (default: %(default)s)")
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parser.add_argument(
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"--mean_std_filepath",
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default='mean_std.npz',
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type=str,
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help="Manifest path for normalizer. (default: %(default)s)")
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parser.add_argument(
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"--specgram_type",
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default='linear',
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type=str,
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help="Feature type of audio data: 'linear' (power spectrum)"
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" or 'mfcc'. (default: %(default)s)")
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parser.add_argument(
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"--num_conv_layers",
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default=2,
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type=int,
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help="Convolution layer number. (default: %(default)s)")
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parser.add_argument(
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"--num_rnn_layers",
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default=3,
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type=int,
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help="RNN layer number. (default: %(default)s)")
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parser.add_argument(
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"--rnn_layer_size",
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default=512,
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type=int,
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help="RNN layer cell number. (default: %(default)s)")
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parser.add_argument(
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"--use_gpu",
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default=True,
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type=distutils.util.strtobool,
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help="Use gpu or not. (default: %(default)s)")
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parser.add_argument(
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"--model_filepath",
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default='checkpoints/params.latest.tar.gz',
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type=str,
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help="Model filepath. (default: %(default)s)")
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parser.add_argument(
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"--decode_method",
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default='beam_search',
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type=str,
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help="Method for ctc decoding: best_path or beam_search. "
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"(default: %(default)s)")
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parser.add_argument(
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"--beam_size",
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default=500,
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type=int,
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help="Width for beam search decoding. (default: %(default)d)")
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parser.add_argument(
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"--language_model_path",
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default="lm/data/common_crawl_00.prune01111.trie.klm",
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type=str,
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help="Path for language model. (default: %(default)s)")
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parser.add_argument(
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"--alpha",
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default=0.36,
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type=float,
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help="Parameter associated with language model. (default: %(default)f)")
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parser.add_argument(
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"--beta",
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default=0.25,
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type=float,
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help="Parameter associated with word count. (default: %(default)f)")
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parser.add_argument(
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"--cutoff_prob",
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default=0.99,
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type=float,
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help="The cutoff probability of pruning"
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"in beam search. (default: %(default)f)")
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args = parser.parse_args()
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class AsrTCPServer(SocketServer.TCPServer):
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def __init__(self,
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server_address,
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RequestHandlerClass,
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speech_save_dir,
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audio_process_handler,
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bind_and_activate=True):
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self.speech_save_dir = speech_save_dir
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self.audio_process_handler = audio_process_handler
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SocketServer.TCPServer.__init__(
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self, server_address, RequestHandlerClass, bind_and_activate=True)
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class AsrRequestHandler(SocketServer.BaseRequestHandler):
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"""The ASR request handler.
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"""
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def handle(self):
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# receive data through TCP socket
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chunk = self.request.recv(1024)
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target_len = struct.unpack('>i', chunk[:4])[0]
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data = chunk[4:]
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while len(data) < target_len:
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chunk = self.request.recv(1024)
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data += chunk
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# write to file
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filename = self._write_to_file(data)
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print("Received utterance[length=%d] from %s, saved to %s." %
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(len(data), self.client_address[0], filename))
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#filename = "/home/work/.cache/paddle/dataset/speech/Libri/train-other-500/LibriSpeech/train-other-500/811/130143/811-130143-0025.flac"
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start_time = time.time()
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transcript = self.server.audio_process_handler(filename)
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finish_time = time.time()
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print("Response Time: %f, Transcript: %s" %
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(finish_time - start_time, transcript))
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self.request.sendall(transcript)
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def _write_to_file(self, data):
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# prepare save dir and filename
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if not os.path.exists(self.server.speech_save_dir):
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os.mkdir(self.server.speech_save_dir)
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timestamp = strftime("%Y%m%d%H%M%S", gmtime())
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out_filename = os.path.join(
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self.server.speech_save_dir,
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timestamp + "_" + self.client_address[0] + "_" + ".wav")
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# write to wav file
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file = wave.open(out_filename, 'wb')
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file.setnchannels(1)
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file.setsampwidth(4)
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file.setframerate(16000)
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file.writeframes(data)
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file.close()
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return out_filename
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def start_server():
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data_generator = DataGenerator(
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vocab_filepath=args.vocab_filepath,
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mean_std_filepath=args.mean_std_filepath,
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augmentation_config='{}',
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specgram_type=args.specgram_type,
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num_threads=1)
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ds2_model = DeepSpeech2Model(
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vocab_size=data_generator.vocab_size,
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num_conv_layers=args.num_conv_layers,
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num_rnn_layers=args.num_rnn_layers,
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rnn_layer_size=args.rnn_layer_size,
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pretrained_model_path=args.model_filepath)
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def file_to_transcript(filename):
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feature = data_generator.process_utterance(filename, "")
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result_transcript = ds2_model.infer_batch(
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infer_data=[feature],
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decode_method=args.decode_method,
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beam_alpha=args.alpha,
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beam_beta=args.beta,
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beam_size=args.beam_size,
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cutoff_prob=args.cutoff_prob,
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vocab_list=data_generator.vocab_list,
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language_model_path=args.language_model_path,
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num_processes=1)
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return result_transcript[0]
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server = AsrTCPServer(
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server_address=(args.host_ip, args.host_port),
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RequestHandlerClass=AsrRequestHandler,
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speech_save_dir=args.speech_save_dir,
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audio_process_handler=file_to_transcript)
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print("ASR Server Started.")
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server.serve_forever()
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def main():
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utils.print_arguments(args)
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paddle.init(use_gpu=args.use_gpu, trainer_count=1)
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start_server()
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if __name__ == "__main__":
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main()
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