Interpreting WAV Data

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开发者_JAVA百科 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!

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

"Two channels" means stereo, so it makes no sense to sum each channel's duration -- so you're off by a factor of two (2.18 seconds, not 4.37). As for signedness, as explained for example here, and I quote:

8-bit samples are stored as unsigned bytes, ranging from 0 to 255. 16-bit samples are stored as 2's-complement signed integers, ranging from -32768 to 32767.

This is part of the specs of the WAV format (actually of its superset RIFF) and thus not dependent on what library you're using to deal with a WAV file.

I know that an answer has already been accepted, but I did some things with audio a while ago and you have to unpack the wave doing something like this.

pcmdata = wave.struct.unpack("%dh"%(wavedatalength),wavedata)

Also, one package that I used was called PyAudio, though I still had to use the wave package with it.

Each sample is 16 bits and there 2 channels, so the frame takes 4 bytes

The duration is simply the number of frames divided by the number of frames per second. From your data this is: 96333 / 44100 = 2.18 seconds.

Building upon this answer, you can get a good performance boost by using numpy.fromstring or numpy.fromfile. Also see this answer.

Here is what I did:

def interpret_wav(raw_bytes, n_frames, n_channels, sample_width, interleaved = True):

    if sample_width == 1:
        dtype = np.uint8 # unsigned char
    elif sample_width == 2:
        dtype = np.int16 # signed 2-byte short
    else:
        raise ValueError("Only supports 8 and 16 bit audio formats.")

    channels = np.fromstring(raw_bytes, dtype=dtype)

    if interleaved:
        # channels are interleaved, i.e. sample N of channel M follows sample N of channel M-1 in raw data
        channels.shape = (n_frames, n_channels)
        channels = channels.T
    else:
        # channels are not interleaved. All samples from channel M occur before all samples from channel M-1
        channels.shape = (n_channels, n_frames)

    return channels

Assigning a new value to shape will throw an error if it requires data to be copied in memory. This is a good thing, since you want to use the data in place (using less time and memory overall). The ndarray.T function also does not copy (i.e. returns a view) if possible, but I'm not sure how you ensure that it does not copy.

Reading directly from the file with np.fromfile will be even better, but you would have to skip the header using a custom dtype. I haven't tried this yet.