Python3实现语音识别和语音合成
本文将分享如何使用Python3实现语音识别和语音合成的过程,主要使用的是Google Speech API和Google Text-to-Speech API。
安装依赖
在开始之前需要安装以下库:
pip install google-cloud-speech google-cloud-texttospeech pyaudio
同时需要安装Google的API,我们需要创建一个Google Cloud Platform帐户并为它启用Google Cloud Speech-to-Text API和Google Cloud Text-to-Speech API。获取授权文件后将其放入项目目录中。
实现语音识别
import io
import os
# 导入语音识别库
from google.cloud import speech
from google.cloud.speech import enums
from google.cloud.speech import types
# 启用授权
os.environ["GOOGLE_APPLICATION_CREDENTIALS"] = "your_auth_file.json"
# 初始化语音识别客户端
client = speech.SpeechClient()
# 从音频文件获取语音内容并进行识别
def transcribe_file(speech_file):
with io.open(speech_file, 'rb') as f:
content = f.read()
audio = types.RecognitionAudio(content=content)
config = types.RecognitionConfig(
encoding=enums.RecognitionConfig.AudioEncoding.LINEAR16,
sample_rate_hertz=16000, # 采样率需要与音频文件的采样率匹配
language_code='zh-CN') # 语言设置为中文
response = client.recognize(config, audio)
for result in response.results:
return result.alternatives[0].transcript # 返回最佳识别结果
# 按下回车后录制音频并识别
def recognize_speech():
input("Press Enter to start recording...")
CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 16000
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = "test.wav"
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
frames = []
print("Recording...")
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
print("Finished recording.")
stream.stop_stream()
stream.close()
p.terminate()
wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
wf.setnchannels(CHANNELS)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(RATE)
wf.writeframes(b''.join(frames))
wf.close()
print("Transcribing...")
text = transcribe_file(WAVE_OUTPUT_FILENAME)
print("Transcription:", text)
# 调用录音函数
recognize_speech()
实现了一个简单的语音识别程序,同时可以录制音频输入(通过按下回车键开始录制,录制5秒钟后自动停止),并输出识别的文字结果。
实现语音合成
from google.cloud import texttospeech
# 启用授权
os.environ["GOOGLE_APPLICATION_CREDENTIALS"] = "your_auth_file.json"
# 初始化语音合成客户端
client = texttospeech.TextToSpeechClient()
# 保存TTS合成后的语音
def save_audio(synthesis_input, voice, audio_config, output_file):
response = client.synthesize_speech(synthesis_input, voice, audio_config)
with open(output_file, 'wb') as out:
out.write(response.audio_content)
print('Audio content saved to file {output_file}')
# 合成指定文字并保存
def synthesize_text(text, output_file):
input_text = texttospeech.SynthesisInput(text=text)
voice = texttospeech.VoiceSelectionParams(
language_code='zh-CN', # 语言设置为中文
name='zh-CN-Wavenet-D') # 选择语音类型为Wavenet
audio_config = texttospeech.AudioConfig(
audio_encoding=texttospeech.AudioEncoding.MP3)
save_audio(input_text, voice, audio_config, output_file)
# 调用语音合成函数
synthesize_text("你好,很高兴认识你", "output.mp3")
这是一个简单的语音合成程序,将输入的文字转为语音并输出为MP3文件。
以上两个示例程序都是使用Google的API,但是其他厂商如阿里云、腾讯云、百度云等也提供了类似的API,开发者可以根据自己的需要进行选择。
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