python - PyAudio 输入溢出

Question

我正在尝试在 python 中制作实时绘图声音。我需要从我的麦克风中获取块。

使用 PyAudio，尝试使用

import pyaudio
import wave
import sys

chunk = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 44100
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = "output.wav"

p = pyaudio.PyAudio()

stream = p.open(format = FORMAT,
                channels = CHANNELS,
                rate = RATE,
                input = True,
                frames_per_buffer = chunk)

print "* recording"
all = []
for i in range(0, RATE / chunk * RECORD_SECONDS):
    data = stream.read(chunk)
    all.append(data)
print "* done recording"

stream.close()
p.terminate()

之后，我收到以下错误：

* recording
Traceback (most recent call last):
  File "gg.py", line 23, in <module>
    data = stream.read(chunk)
  File "/usr/lib64/python2.7/site-packages/pyaudio.py", line 564, in read
    return pa.read_stream(self._stream, num_frames)
IOError: [Errno Input overflowed] -9981

我无法理解这个缓冲区。我想使用阻塞 IO 模式，所以如果块不可用，我想等待这些块。但是当我创建除段或睡眠（0.1）之外的尝试时，我听到咔嗒声，所以这不是我想要的。

请为我的问题提出最佳解决方案？

score 32 · Accepted Answer

pyaudio.Stream.read()有一个关键字参数exception_on_overflow，将其设置为 False。

对于您的示例代码，如下所示：

import pyaudio
import wave
import sys

chunk = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 44100
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = "output.wav"

p = pyaudio.PyAudio()

stream = p.open(format = FORMAT,
                channels = CHANNELS,
                rate = RATE,
                input = True,
                frames_per_buffer = chunk)

print "* recording"
all = []
for i in range(0, RATE / chunk * RECORD_SECONDS):
    data = stream.read(chunk, exception_on_overflow = False)
    all.append(data)
print "* done recording"

stream.close()
p.terminate()

有关更多详细信息，请参阅PyAudio 文档。

score 14 · Accepted Answer

好像很多人都遇到过这个问题。我对其进行了一些研究，我认为这意味着在上一次调用stream.read()和当前调用之间，流中的数据丢失了（即缓冲区填满的速度比你清除它的速度快）。

来自（最终调用Pa_ReadStream()的 PortAudio 函数）的文档：stream.read()

@return On success PaNoError will be returned, or PaInputOverflowed if
input data was discarded by PortAudio after the previous call and
before this call.

（PaInputOverflowed然后导致IOErrorpyaudio 包装器中的一个）。

如果您可以不捕获每一帧，那么您可以忽略此错误。如果拥有每一帧对您来说绝对至关重要，那么您需要找到一种方法来提高应用程序的优先级。我对 Python 不够熟悉，不知道用 Python 的方式来做这件事，但值得尝试一个简单的nice命令，或者将调度策略更改为 SCHED_DEADLINE。

编辑：

现在的一个问题是，当抛出 IOError 时，您会丢失该调用中收集的所有帧。要忽略溢出并只返回我们拥有的内容，您可以应用下面的补丁，这将导致 stream.read() 忽略来自 PortAudio 的输出欠载和输入溢出错误（但如果发生不同的错误仍然会抛出一些东西）。更好的方法是根据您的需要使这种行为（投掷/不投掷）可定制。

diff --git a/src/_portaudiomodule.c b/src/_portaudiomodule.c
index a8f053d..0878e74 100644
--- a/src/_portaudiomodule.c
+++ b/src/_portaudiomodule.c
@@ -2484,15 +2484,15 @@ pa_read_stream(PyObject *self, PyObject *args)
     } else {
       /* clean up */
       _cleanup_Stream_object(streamObject);
+
+      /* free the string buffer */
+      Py_XDECREF(rv);
+
+      PyErr_SetObject(PyExc_IOError,
+                       Py_BuildValue("(s,i)",
+                                     Pa_GetErrorText(err), err));
+      return NULL;
     }
-
-    /* free the string buffer */
-    Py_XDECREF(rv);
-
-    PyErr_SetObject(PyExc_IOError,
-                   Py_BuildValue("(s,i)",
-                                 Pa_GetErrorText(err), err));
-    return NULL;
   }

   return rv;

score 11 · Accepted Answer

当我运行你的代码时，我得到了同样的错误。我查看了我的默认音频设备的默认采样率，我的 macbook 的内置麦克风，它是 48000Hz 而不是 44100Hz。

p.get_device_info_by_index(0)['defaultSampleRate']
Out[12]: 48000.0

当我将 RATE 更改为此值时，它起作用了。

score 7 · Accepted Answer

我在 OS X 10.10 上工作，尝试从 SYBA USB 卡（C 媒体芯片组）中的麦克风获取音频时遇到相同的错误，并使用 fft 等实时处理它：

IOError: [Errno Input overflowed] -9981

当使用回调模式而不是阻塞模式时，溢出完全解决了，如 libbkmz 所写。（https://www.python.org/dev/peps/pep-0263/）

基于此，工作代码的一部分如下所示：

"""
Creating the audio stream from our mic
"""
rate=48000
self.chunk=2**12
width = 2

p = pyaudio.PyAudio()

# callback function to stream audio, another thread.
def callback(in_data,frame_count, time_info, status):
    self.audio = numpy.fromstring(in_data,dtype=numpy.int16)
    return (self.audio, pyaudio.paContinue)

#create a pyaudio object
self.inStream = p.open(format = p.get_format_from_width(width, unsigned=False),
                       channels=1,
                       rate=rate,
                       input=True,
                       frames_per_buffer=self.chunk,
                       stream_callback = callback)

"""
Setting up the array that will handle the timeseries of audio data from our input
"""
self.audio = numpy.empty((self.buffersize),dtype="int16")

    self.inStream.start_stream()

while True:
  try:
    self.ANY_FUNCTION() #any function to run parallel to the audio thread, running forever, until ctrl+C is pressed. 

  except KeyboardInterrupt:

    self.inStream.stop_stream()
    self.inStream.close()
    p.terminate()
    print("* Killed Process")
    quit()

此代码将创建一个回调函数，然后创建一个流对象，启动它，然后在任何函数中循环。一个单独的线程流式传输音频，并且当主循环停止时该流关闭。self.audio 用于任何功能。如果没有终止，我也遇到了线程永远运行的问题。

由于 Pyaudio 在单独的线程中运行此流，这使音频流稳定，因此阻塞模式可能已经饱和，具体取决于脚本中其余进程的速度或时间。

请注意，块大小为 2^12，但较小的块也可以。我考虑并使用了其他参数以确保它们都有意义：

块大小更大或更小（无影响）
缓冲区中字的位数和格式，在本例中为有符号 16 位。
变量的符号性（尝试使用无符号并获得饱和模式）
麦克风输入的性质，系统默认选择，增益等。

希望对某人有用！

score 4 · Accepted Answer

我的另一个答案在大多数情况下解决了这个问题。但是，有时错误仍然会发生。

这就是我放弃 pyaudio 并改用 pyalsaaudio 的原因。我的 Raspy 现在可以流畅地录制任何声音。

import alsaaudio   
import numpy as np
import array

# constants
CHANNELS    = 1
INFORMAT    = alsaaudio.PCM_FORMAT_FLOAT_LE
RATE        = 44100
FRAMESIZE   = 1024

# set up audio input
recorder=alsaaudio.PCM(type=alsaaudio.PCM_CAPTURE)
recorder.setchannels(CHANNELS)
recorder.setrate(RATE)
recorder.setformat(INFORMAT)
recorder.setperiodsize(FRAMESIZE)


buffer = array.array('f')
while <some condition>:
    buffer.fromstring(recorder.read()[1])

data = np.array(buffer, dtype='f')

score 3 · Accepted Answer

FORMAT = pyaudio.paInt16

确保设置正确的格式，我的内置麦克风设置为 24 位（请参阅 Audio-Midi-Setup 应用程序）。

score 2 · Accepted Answer

我在非常慢的树莓派上遇到了同样的问题，但我能够（在大多数情况下）通过使用更快的array模块来存储数据来解决它。

import array
import pyaudio 

FORMAT = pyaudio.paInt16
CHANNELS = 1
INPUT_CHANNEL=2
RATE = 48000
CHUNK = 512

p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
                channels=CHANNELS,
                rate=RATE,
                input=INPUT_CHANNEL,
                frames_per_buffer =CHUNK)

print("* recording")


try:
    data = array.array('h')
    for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
        data.fromstring(stream.read(CHUNK))
finally:
    stream.stop_stream()
    stream.close()
    p.terminate()

print("* done recording")

之后的内容data是相当二进制的。但是您可以使用它numpy.array(data, dtype='i')来获取一个 numpy 整数数组。

score 0 · Accepted Answer

对我来说，这有帮助：https ://stackoverflow.com/a/46787874/5047984

我使用多处理来并行写入文件以录制音频。这是我的代码：

记录AudioSamples.py

import pyaudio
import wave
import datetime
import signal
import ftplib
import sys
import os

# configuration for assos_listen
import config

# run the audio capture and send sound sample processes
# in parallel
from multiprocessing import Process

# CONFIG
CHUNK = config.chunkSize
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = config.samplingRate
RECORD_SECONDS = config.sampleLength

# HELPER FUNCTIONS

# write to ftp
def uploadFile(filename):

    print("start uploading file: " + filename)
    # connect to container
    ftp = ftplib.FTP(config.ftp_server_ip, config.username, config.password)

    # write file
    ftp.storbinary('STOR '+filename, open(filename, 'rb'))
    # close connection
    ftp.quit()
    print("finished uploading: " +filename)

# write to sd-card
def storeFile(filename,frames):

    print("start writing file: " + filename)
    wf = wave.open(filename, 'wb')
    wf.setnchannels(CHANNELS)
    wf.setsampwidth(p.get_sample_size(FORMAT))
    wf.setframerate(RATE)
    wf.writeframes(b''.join(frames))
    wf.close()
    print(filename + " written")

# abort the sampling process
def signal_handler(signal, frame):
    print('You pressed Ctrl+C!')

    # close stream and pyAudio
    stream.stop_stream()
    stream.close()
    p.terminate()

    sys.exit(0)

# MAIN FUNCTION
def recordAudio(p, stream):

    sampleNumber = 0
    while (True):
        print("*  recording")
        sampleNumber = sampleNumber +1

        frames = []
        startDateTimeStr = datetime.datetime.now().strftime("%Y_%m_%d_%I_%M_%S_%f")
        for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
            data = stream.read(CHUNK)
            frames.append(data)

        fileName =  str(config.sensorID) + "_" + startDateTimeStr + ".wav"

        # create a store process to write the file in parallel
        storeProcess = Process(target=storeFile, args=(fileName,frames))
        storeProcess.start()

        if (config.upload == True):
            # since waiting for the upload to finish will take some time
            # and we do not want to have gaps in our sample
            # we start the upload process in parallel
            print("start uploading...")
            uploadProcess = Process(target=uploadFile, args=(fileName,))
            uploadProcess.start()

# ENTRYPOINT FROM CONSOLE
if __name__ == '__main__':

    p = pyaudio.PyAudio()
    stream = p.open(format=FORMAT,
                    channels=CHANNELS,
                    rate=RATE,
                    input=True,
                    frames_per_buffer=CHUNK)

    # directory to write and read files from
    os.chdir(config.storagePath)

    # abort by pressing C
    signal.signal(signal.SIGINT, signal_handler)
    print('\n\n--------------------------\npress Ctrl+C to stop the recording')

    # start recording
    recordAudio(p, stream)

配置文件

### configuration file for assos_listen
# upload
upload = False

# config for this sensor
sensorID = "al_01"

# sampling rate & chunk size
chunkSize = 8192
samplingRate = 44100 # 44100 needed for Aves sampling
# choices=[4000, 8000, 16000, 32000, 44100] :: default 16000

# sample length in seconds
sampleLength = 10

# configuration for assos_store container
ftp_server_ip = "192.168.0.157"
username = "sensor"
password = "sensor"

# storage on assos_listen device
storagePath = "/home/pi/assos_listen_pi/storage/"

score 0 · Accepted Answer

这对我很有帮助：

input_ = stream.read(chunk, exception_on_overflow=False)
exception_on_overflow = False

python - PyAudio 输入溢出

9 回答 9

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