使用 python 减慢语音音频（不是来自 mp3/wav）

Question

我需要减慢通过麦克风捕获的短时间的语音突发，然后在 python 脚本中实时播放。我可以使用 PyAudio 的输入和输出流在不改变速度的情况下捕获和播放音频，但我不知道如何减慢它的速度。

我看到 this post 使用 pydub 对来自文件的音频做了类似的事情，但不知道如何为我的目的修改它。

只是为了强调问题标题中的关键点，“（不是来自 mp3/wav 或任何其他文件类型）”，因为我想在 实时 短脉冲，理想情况下 所以只想处理从 PyAudio 流中读取的数据。

有使用 pydub 经验的人知道它是否可以满足我的需要吗？

注意，我意识到输出会越来越落后，并且可能存在缓冲问题，但是我只是在长达 30 秒的短暂时间内这样做，并且只想将语音速度减慢约 10%。

Answer 1

所以事实证明它非常非常简单。

查看pydub 和pyaudio 代码库后，我意识到，与输入音频流（麦克风）相比，只需为输出音频流（扬声器）上的“rate”参数指定一个较低的值即可) stream.write() 函数会为我处理它。

我一直期望需要对原始数据进行物理操作才能将数据转换为更大的缓冲区。

这是一个简单的例子：

import pyaudio

FORMAT = pyaudio.paInt16
CHANNELS = 1
FRAME_RATE = 44100
CHUNK = 1024*4

# simply modify the value for the 'rate' parameter to change the playback speed
# <1 === slow down;  >1 === speed up
FRAMERATE_OFFSET = 0.8

audio = pyaudio.PyAudio()

#output stream
stream_out = audio.open(format=FORMAT,
                        channels=CHANNELS,
                        rate= int(FRAME_RATE * FRAMERATE_OFFSET),
                        output=True)

# open input steam to start recording mic audio
stream_in = audio.open(format=FORMAT, 
                       channels=CHANNELS,
                       rate=FRAME_RATE, 
                       input=True)

for i in range(1):
    # modify the chunk multiplier below to captyre longer time durations
    data = stream_in.read(CHUNK*25)
    stream_out.write(data)

stream_out.stop_stream()
stream_out.close()
audio.terminate()

为了实现这个操作，我需要设置一个共享内存数据缓冲区并设置一个子进程来处理输出，这样我就不会错过输入信号中的任何重要内容。

Answer 2

这就是我所做的。

import wave
channels = 1
swidth = 2
multiplier = 0.2

spf = wave.open('flute-a4.wav', 'rb')
fr=spf.getframerate() # frame rate
signal = spf.readframes(-1)

wf = wave.open('ship.wav', 'wb')
wf.setnchannels(channels)
wf.setsampwidth(swidth)
wf.setframerate(fr*multiplier)
wf.writeframes(signal)
wf.close()

我用这个repo的长笛。

Answer 3

如 cmets 中所述，通过简单地增加或减少采样频率/帧速率，您可以加速减速音频。虽然如果您打算通过麦克风实时进行录制，其中一个想法是在几秒钟内录制，播放放慢的音频，然后再次进行录制。

这是一个使用 sounddevice 的示例，这基本上是我的回答 here 的轻微修改。 我们循环录制音频 4 秒 3 次，并立即播放帧速率偏移（> 1 表示加速，

import sounddevice as sd
import numpy as np
import scipy.io.wavfile as wav
import time

fs=44100
duration = 4  # seconds
#fs_offset = 1.3 #speedup
fs_offset = 0.8 #speedup slow down

for count in range(1,4):
    myrecording = sd.rec(duration * fs, samplerate=fs, channels=2, dtype='float64')
    print "Recording Audio chunk {} for {} seconds".format(count, duration)
    sd.wait()
    print "Recording complete, Playing chunk {} with offset {} ".format(count, fs_offset)
    sd.play(myrecording, fs * fs_offset)
    sd.wait()
    print "Playing chunk {} Complete".format(count)
    time.sleep(1)

输出：

$python sdaudio.py
Recording Audio chunk 1 for 4 seconds
Recording complete, Playing chunk 1 with offset 0.8 
Playing chunk 1 Complete
Recording Audio chunk 2 for 4 seconds
Recording complete, Playing chunk 2 with offset 0.8 
Playing chunk 2 Complete
Recording Audio chunk 3 for 4 seconds
Recording complete, Playing chunk 3 with offset 0.8 
Playing chunk 3 Complete

这是一个使用 PyAudio 从麦克风录音和 pydub 进行回放的示例。尽管您也可以使用 pyaudio 阻塞线功能来修改传出音频。我使用了 pydub，因为您提到了基于 pydub 的解决方案。这是来自here的代码模组。

import pyaudio
import wave
from pydub import AudioSegment
from pydub.playback import play
import time

FORMAT = pyaudio.paInt16
CHANNELS = 2
RATE = 44100
CHUNK = 1024
RECORD_SECONDS = 4
#FRAMERATE_OFFSET = 1.4  #speedup
FRAMERATE_OFFSET = 0.7 #slowdown
WAVE_OUTPUT_FILENAME = "file.wav"

def get_audio():

    audio = pyaudio.PyAudio()

    # start Recording
    stream = audio.open(format=FORMAT, channels=CHANNELS,
                    rate=RATE, input=True,
                    frames_per_buffer=CHUNK)
    frames = []

    for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
        data = stream.read(CHUNK)
        frames.append(data)

    # stop Recording
    stream.stop_stream()
    stream.close()
    audio.terminate()

    #save to file
    waveFile = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
    waveFile.setnchannels(CHANNELS)
    waveFile.setsampwidth(audio.get_sample_size(FORMAT))
    waveFile.setframerate(RATE * FRAMERATE_OFFSET)
    waveFile.writeframes(b''.join(frames))
    waveFile.close()


for count in range(1,4):
    print "recording segment {} ....".format(count)
    frame_array = get_audio()
    print "Playing segment {} .... at offset {}".format(count, FRAMERATE_OFFSET)
    audio_chunk = AudioSegment.from_wav(WAVE_OUTPUT_FILENAME)
    print "Finished playing segment {} .... at offset {}".format(count, FRAMERATE_OFFSET)
    play(audio_chunk)
    time.sleep(1)

输出：

$python slowAudio.py 
recording segment 1 ....
Playing segment 1 .... at offset 0.7
Finished playing segment 1 .... at offset 0.7
recording segment 2 ....
Playing segment 2 .... at offset 0.7
Finished playing segment 2 .... at offset 0.7
recording segment 3 ....
Playing segment 3 .... at offset 0.7

Answer 4

这个问题已经回答here。

from pydub import AudioSegment
sound = AudioSegment.from_file(…)

def speed_change(sound, speed=1.0):
    # Manually override the frame_rate. This tells the computer how many
    # samples to play per second
    sound_with_altered_frame_rate = sound._spawn(sound.raw_data, overrides={
        "frame_rate": int(sound.frame_rate * speed)
    })

    # convert the sound with altered frame rate to a standard frame rate
    # so that regular playback programs will work right. They often only
    # know how to play audio at standard frame rate (like 44.1k)
    return sound_with_altered_frame_rate.set_frame_rate(sound.frame_rate)

slow_sound = speed_change(sound, 0.75)
fast_sound = speed_change(sound, 2.0)