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Interpreting WAV Data

Tags:

python

audio

pcm

I'm trying to write a program to display PCM data. I've been very frustrated trying to find a library with the right level of abstraction, but I've found the python wave library and have been using that. However, I'm not sure how to interpret the data.

The wave.getparams function returns (2 channels, 2 bytes, 44100 Hz, 96333 frames, No compression, No compression). This all seems cheery, but then I tried printing a single frame:'\xc0\xff\xd0\xff' which is 4 bytes. I suppose it's possible that a frame is 2 samples, but the ambiguities do not end there.

96333 frames * 2 samples/frame * (1/44.1k sec/sample) = 4.3688 seconds

However, iTunes reports the time as closer to 2 seconds and calculations based on file size and bitrate are in the ballpark of 2.7 seconds. What's going on here?

Additionally, how am I to know if the bytes are signed or unsigned?

Many thanks!

like image 341
SapphireSun Avatar asked Feb 09 '10 05:02

SapphireSun


1 Answers

Thank you for your help! I got it working and I'll post the solution here for everyone to use in case some other poor soul needs it:

import wave
import struct

def pcm_channels(wave_file):
    """Given a file-like object or file path representing a wave file,
    decompose it into its constituent PCM data streams.

    Input: A file like object or file path
    Output: A list of lists of integers representing the PCM coded data stream channels
        and the sample rate of the channels (mixed rate channels not supported)
    """
    stream = wave.open(wave_file,"rb")

    num_channels = stream.getnchannels()
    sample_rate = stream.getframerate()
    sample_width = stream.getsampwidth()
    num_frames = stream.getnframes()

    raw_data = stream.readframes( num_frames ) # Returns byte data
    stream.close()

    total_samples = num_frames * num_channels

    if sample_width == 1: 
        fmt = "%iB" % total_samples # read unsigned chars
    elif sample_width == 2:
        fmt = "%ih" % total_samples # read signed 2 byte shorts
    else:
        raise ValueError("Only supports 8 and 16 bit audio formats.")

    integer_data = struct.unpack(fmt, raw_data)
    del raw_data # Keep memory tidy (who knows how big it might be)

    channels = [ [] for time in range(num_channels) ]

    for index, value in enumerate(integer_data):
        bucket = index % num_channels
        channels[bucket].append(value)

    return channels, sample_rate
like image 110
SapphireSun Avatar answered Nov 03 '22 06:11

SapphireSun