Reputation: 1929
I am reading bytes from wav audio downloaded from a URL. I would like to "reconstruct" these bytes into a .wav file. I have attempted the code below, but the resulting file is pretty much static. For example, when I download audio of myself speaking, the .wav file produced is static only, but I can hear slight alterations/distortions when I know the audio should be playing my voice. What am I doing wrong?
from pprint import pprint
import scipy.io.wavfile
import numpy
#download a wav audio recording from a url
>>>response = client.get_recording(r"someurl.com")
>>>pprint(response)
(b'RIFFv\xfc\x03\x00WAVEfmt \x10\x00\x00\x00\x01\x00\x01\x00\x80>\x00\x00'
...
b'\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff'
...
b'\xea\xff\xfd\xff\x10\x00\x0c\x00\xf0\xff\x06\x00\x10\x00\x06\x00'
...)
>>>a=bytearray(response)
>>>pprint(a)
bytearray(b'RIFFv\xfc\x03\x00WAVEfmt \x10\x00\x00\x00\x01\x00\x01\x00'
b'\x80>\x00\x00\x00}\x00\x00\x02\x00\x10\x00LISTJ\x00\x00\x00INFOINAM'
b'0\x00\x00\x00Conference d95ac842-08b7-4380-83ec-85ac6428cc41\x00'
b'IART\x06\x00\x00\x00Nexmo\x00data\x00\xfc\x03\x00\xff\xff'
b'\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff'
...
b'\x12\x00\xf6\xff\t\x00\xed\xff\xf6\xff\xfc\xff\xea\xff\xfd\xff'
...)
>>>b = numpy.array(a, dtype=numpy.int16)
>>>pprint(b)
array([ 82, 73, 70, ..., 255, 248, 255], dtype=int16)
>>>scipy.io.wavfile.write(r"C:\Users\somefolder\newwavfile.wav",
16000, b)
Upvotes: 8
Views: 42527
Reputation: 11
You can do this with the wave
module:
def save_audio_wf(audio_data, filename="./output-wf.wav"):
with wave.open(filename, 'wb') as wf:
wf.setnchannels(1) # mono
wf.setsampwidth(2) # 2 bytes per sample
wf.setframerate(16000) # 16kHz sample rate
wf.writeframes(audio_data)
For context, I used it to store streamed data like so:
@sock.route('/stream')
def stream(ws):
try:
audio_buffer = b''
while True:
message = ws.receive()
packet = json.loads(message)
if packet['event'] == 'start':
print('Starting Stream')
elif packet['event'] == 'stop':
print('Stopping Stream')
save_audio_wf(audio_buffer)
audio_buffer = b''
elif packet['event'] == 'media':
audio = base64.b64decode(packet['media']['payload'])
audio = audioop.ulaw2lin(audio, 2)
audio = audioop.ratecv(audio, 2, 1, 8000, 16000, None)[0]
audio_buffer += audio
except Exception as e:
raise e
I first created a buffer for the data, then appended incoming packets to the buffer (after a little bit of "preprocessing") as shown in the last elif
block.
At the end of the stream, I saved it all to a .wav
file using the save_audio_wf
function.
Upvotes: 1
Reputation: 2614
I faced the same problem while streaming and I used the answers above to write a complete function. In my case, the byte array was coming from streaming an audio file (the frontend) and the backend needs to process it as a ndarray.
This function simulates how the front-ends sends the audio file as chunks that are accumulated into a byte array:
audio_file_path = 'offline_input/zoom283.wav'
chunk = 1024
wf = wave.open(audio_file_path, 'rb')
audio_input = b''
d = wf.readframes(chunk)
while len(d) > 0:
d = wf.readframes(chunk)
audio_input = audio_input + d
some import libraries:
import io
import wave
import numpy as np
import scipy.io.wavfile
import soundfile as sf
from scipy.io.wavfile import write
Finally, the backend will take a byte array and convert it to ndarray:
def convert_bytearray_to_wav_ndarray(input_bytearray: bytes, sampling_rate=16000):
bytes_wav = bytes()
byte_io = io.BytesIO(bytes_wav)
write(byte_io, sampling_rate, np.frombuffer(input_bytearray, dtype=np.int16))
output_wav = byte_io.read()
output, samplerate = sf.read(io.BytesIO(output_wav))
return output
output = convert_bytearray_to_wav_ndarray(input_bytearray=audio_input)
The output represents the audio file to be processed by the backend:
To check that the file has been received correctly, we write it to the desk:
scipy.io.wavfile.write("output1.wav", 16000, output)
Upvotes: 5
Reputation: 85
To add wave file header to raw audio bytes (extracted from wave library):
import struct
def write_header(_bytes, _nchannels, _sampwidth, _framerate):
WAVE_FORMAT_PCM = 0x0001
initlength = len(_bytes)
bytes_to_add = b'RIFF'
_nframes = initlength // (_nchannels * _sampwidth)
_datalength = _nframes * _nchannels * _sampwidth
bytes_to_add += struct.pack('<L4s4sLHHLLHH4s',
36 + _datalength, b'WAVE', b'fmt ', 16,
WAVE_FORMAT_PCM, _nchannels, _framerate,
_nchannels * _framerate * _sampwidth,
_nchannels * _sampwidth,
_sampwidth * 8, b'data')
bytes_to_add += struct.pack('<L', _datalength)
return bytes_to_add + _bytes
Upvotes: 3
Reputation: 143
AudioSegment.from_raw() also will work while you have a continues stream of bytes:
import io
from pydub import AudioSegment
current_data is defined as the stream of bytes that you receive
s = io.BytesIO(current_data)
audio = AudioSegment.from_raw(s, sample_width, frame_rate, channels).export(filename, format='wav')
Upvotes: 4
Reputation: 4884
You can simply write the data in response
to a file:
with open('myfile.wav', mode='bx') as f:
f.write(response)
If you want to access the audio data as a NumPy array without writing it to a file first, you can do this with the soundfile module like this:
import io
import soundfile as sf
data, samplerate = sf.read(io.BytesIO(response))
See also this example: https://pysoundfile.readthedocs.io/en/0.9.0/#virtual-io
Upvotes: 12