pysox | Python wrapper around sox | Wrapper library

If you look at the examples at https://pythonhosted.org/pysox/intro.html#simple-examples, you need to import pysox, not sox:

I think I have solved the problem,

entered: (referred form this link)

The reason is that raw audio files do not contain information about the audio format in the file, so you need to provide these. Sample rate is just one such indicator, so you'll need to do this also for a few other parameters.

Quoted from sox.sourceforge.net:

SoX can work with ‘self-describing’ and ‘raw’ audio files. ‘self-describing’ formats (e.g. WAV, FLAC, MP3) have a header that completely describes the signal and encoding attributes of the audio data that follows. ‘raw’ or ‘headerless’ formats do not contain this information, so the audio characteristics of these must be described on the SoX command line or inferred from those of the input file.

The following four characteristics are used to describe the format of audio data such that it can be processed with SoX:

• sample rate

  The sample rate in samples per second (‘Hertz’ or ‘Hz’). Digital telephony traditionally uses a sample rate of 8000 Hz (8 kHz), though these days, 16 and even 32 kHz are becoming more common. Audio Compact Discs use 44100 Hz (44.1 kHz). Digital Audio Tape and many computer systems use 48 kHz. Professional audio systems often use 96 kHz.

• sample size [...]

• data encoding [...]
• channels [...]

The pysox documentation describes the set_input_format method:

set_input_format(file_type=None, rate=None, bits=None, channels=None, encoding=None, ignore_length=False)

Sets input file format arguments. This is primarily useful when dealing with audio files without a file extension. Overwrites any previously set input file arguments.

If this function is not explicitly called the input format is inferred from the file extension or the file’s header.

Parameters:

• file_type : str or None, default=None

  The file type of the input audio file. Should be the same as what the file extension would be, for ex. ‘mp3’ or ‘wav’.

• rate : float or None, default=None

  The sample rate of the input audio file. If None the sample rate is inferred.

• [...]

So, you should set the rate as follows:

1. The pysox module does not use libsox but creates a subprocess for each audio file to process (source code here). It certainly works, but if your server handles more than, say, 100 QPS, then you should take the overhead of creating and destroying processes into consideration.

2. As the doc goes, you should write

   tfm.build('path/to/input_audio', 'path/to/output_audio')

   after applying effects (tfm.trim in your case).

   Keep in mind that you should install sox (apt/yum install sox) on your server, and it does not support writting to MP3/M4A/OGG etc., so you'll probably need an additional call to lame/ffmpeg to encode the output for downloading.