When I store the data in a .wav file into a byte array, what do these values mean?
I've read that they are in two-byte representations, but what exactly is contained in these two-byte values?
You will have heard, that audio signals are represented by some kind of wave. If you have ever seen this wave diagrams with a line going up and down -- that's basically what's inside those files. Take a look at this file picture from http://en.wikipedia.org/wiki/Sampling_rate
You see your audio wave (the gray line). The current value of that wave is repeatedly measured and given as a number. That's the numbers in those bytes. There are two different things that can be adjusted with this: The number of measurements you take per second (that's the sampling rate, given in Hz -- that's how many per second you grab). The other adjustment is how exact you measure. In the 2-byte case, you take two bytes for one measurement (that's values from -32768 to 32767 normally). So with those numbers given there, you can recreate the original wave (up to a limited quality, of course, but that's always so when storing stuff digitally). And recreating the original wave is what your speaker is trying to do on playback.
There are some more things you need to know. First, since it's two bytes, you need to know the byte order (big endian, little endian) to recreate the numbers correctly. Second, you need to know how many channels you have, and how they are stored. Typically you would have mono (one channel) or stereo (two), but more is possible. If you have more than one channel, you need to know, how they are stored. Often you would have them interleaved, that means you get one value for each channel for every point in time, and after that all values for the next point in time.
To illustrate: If you have data of 8 bytes for two channels and 16-bit number:
abcdefgh
Here a and b would make up the first 16bit number that's the first value for channel 1, c and d would be the first number for channel 2. e and f are the second value of channel 1, g and h the second value for channel 2. You wouldn't hear much there because that would not come close to a second of data...
If you take together all that information you have, you can calculate the bit rate you have, that's how many bits of information is generated by the recorder per second. In our example, you generate 2 bytes per channel on every sample. With two channels, that would be 4 bytes. You need about 44000 samples per second to represent the sounds a human beeing can normally hear. So you'll end up with 176000 bytes per second, which is 1408000 bits per second.
And of course, it is not 2-bit values, but two 2 byte values there, or you would have a really bad quality.
The first 44 bytes are commonly a standard RIFF header, as described here:
http://tiny.systems/software/soundProgrammer/WavFormatDocs.pdf
and here: http://www.topherlee.com/software/pcm-tut-wavformat.html
Apple/OSX/macOS/iOS created .wav files might add an 'FLLR' padding chunk to the header and thus increase the size of the initial header RIFF from 44 bytes to 4k bytes (perhaps for better disk or storage block alignment of the raw sample data).
The rest is very often 16-bit linear PCM in signed 2's-complement little-endian format, representing arbitrarily scaled samples at a rate of 44100 Hz.
The WAVE (.wav) file contain a header, which indicates the formatting information of the audio file's data. Following the header is the actual audio raw data. You can check their exact meaning below.
Positions Typical Value Description
1 - 4 "RIFF" Marks the file as a RIFF multimedia file.
Characters are each 1 byte long.
5 - 8 (integer) The overall file size in bytes (32-bit integer)
minus 8 bytes. Typically, you'd fill this in after
file creation is complete.
9 - 12 "WAVE" RIFF file format header. For our purposes, it
always equals "WAVE".
13-16 "fmt " Format sub-chunk marker. Includes trailing null.
17-20 16 Length of the rest of the format sub-chunk below.
21-22 1 Audio format code, a 2 byte (16 bit) integer.
1 = PCM (pulse code modulation).
23-24 2 Number of channels as a 2 byte (16 bit) integer.
1 = mono, 2 = stereo, etc.
25-28 44100 Sample rate as a 4 byte (32 bit) integer. Common
values are 44100 (CD), 48000 (DAT). Sample rate =
number of samples per second, or Hertz.
29-32 176400 (SampleRate * BitsPerSample * Channels) / 8
This is the Byte rate.
33-34 4 (BitsPerSample * Channels) / 8
1 = 8 bit mono, 2 = 8 bit stereo or 16 bit mono, 4
= 16 bit stereo.
35-36 16 Bits per sample.
37-40 "data" Data sub-chunk header. Marks the beginning of the
raw data section.
41-44 (integer) The number of bytes of the data section below this
point. Also equal to (#ofSamples * #ofChannels *
BitsPerSample) / 8
45+ The raw audio data.
I copied all of these from http://www.topherlee.com/software/pcm-tut-wavformat.html here
As others have pointed out, there's metadata in the wav file, but I think your question may be, specifically, what do the bytes (of data, not metadata) mean? If that's true, the bytes represent the value of the signal that was recorded.
What does that mean? Well, if you extract the two bytes (say) that represent each sample (assume a mono recording, meaning only one channel of sound was recorded), then you've got a 16-bit value. In WAV, 16-bit is (always?) signed and little-endian (AIFF, Mac OS's answer to WAV, is big-endian, by the way). So if you take the value of that 16-bit sample and divide it by 2^16 (or 2^15, I guess, if it's signed data), you'll end up with a sample that is normalized to be within the range -1 to 1. Do this for all samples and plot them versus time (and time is determined by how many samples/second is in the recording; e.g. 44.1KHz means 44.1 samples/millisecond, so the first sample value will be plotted at t=0, the 44th at t=1ms, etc) and you've got a signal that roughly represents what was originally recorded.
I suppose your question is "What do the bytes in data block of .wav file represent?" Let us know everything systematically.
Prelude:
Let us say we play a 5KHz sine wave using some device and record it in a file called 'sine.wav', and recording is done on a single channel (mono). Now you already know what the header in that file represents.
Let us go through some important definitions:
Sample: A sample of any signal means the amplitude of that signal at the point where sample is taken.
Sampling rate: Many such samples can be taken within a given interval of time. Suppose we take 10 samples of our sine wave within 1 second. Each sample is spaced by 0.1 second. So we have 10 samples per second, thus the sampling rate is 10Hz. Bytes 25th to 28th in the header denote sampling rate.
Now coming to the answer of your question:
It is not possible practically to write the whole sine wave to the file because there are infinite points on a sine wave. Instead, we fix a sampling rate and start sampling the wave at those intervals and record the amplitudes. (The sampling rate is chosen such that the signal can be reconstructed with minimal distortion, using the samples we are going to take. The distortion in the reconstructed signal because of the insufficient number of samples is called 'aliasing'.)
To avoid aliasing, the sampling rate is chosen to be more than twice the frequency of our sine wave (5kHz)(This is called 'sampling theorem' and the rate twice the frequency is called 'nyquist rate'). Thus we decide to go with sampling rate of 12kHz which means we will sample our sine wave, 12000 times in one second.
Once we start recording, if we record the signal, which is sine wave of 5kHz frequency, we will have 12000*5 samples(values). We take these 60000 values and put it in an array. Then we create the proper header to reflect our metadata and then we convert these samples, which we have noted in decimal, to their hexadecimal equivalents. These values are then written in the data bytes of our .wav files.
Plot plotted on : http://fooplot.com
Two bit audio wouldn't sound very good :) Most commonly, they represent sample values as 16-bit signed numbers that represent the audio waveform sampled at a frequency such as 44.1kHz.
Related
I'm trying to better understand how samples are aligned in the audio file.
Let's say we have a 2s audio file with sampling rate = 3.
I think there are three possible ways to align those samples. Looking at the picture below, can you tell me which one is correct?
Also, is this a standard for all audio files or does different formats have different rules?
Cheers!
Sampling rate in audio typically tells you how many samples are in one second, a unit called Hertz. Strictly speaking, the correct answer would be (1), as you have 3 samples within one second. Assuming there's no latency, PCM and other formats dictate that audio starts at 0. Next "cycle" (next second) also starts at zero, same principle like with a clock.
To get total length of the audio (following question in the comment), you should simply take number of samples / rate. Example from a 30s WAV using soxi, one of canonical tools used in the community for sound manipulation:
Input File : 'book_00396_chp_0024_reader_11416_5_door_Freesound_validated_380721_0-door_Freesound_validated_381380_0-9IfN8dUgGaQ_snr10_fileid_1138.wav'
Channels : 1
Sample Rate : 16000
Precision : 16-bit
Duration : 00:00:30.00 = 480000 samples ~ 2250 CDDA sectors
File Size : 960k
Bit Rate : 256k
Sample Encoding: 16-bit Signed Integer PCM
480000 samples / (16000 samples / seconds) = 30 seconds exactly. Citing manual, duration is "Equivalent to number of samples divided by the sample-rate."
Sorry if this question has been asked before but I did not find any matching what I'm looking for. I have a certain amount of time that I want for an audio file to be silent, so I can put that into some pcm data. But I've no idea how times I need to add b'\x00\x00' for it to equal a certain amount of time. I have this line of code:
pcm += struct.pack('<h', 0)*some_number
so how would I calculate "some_number" based on a certain amount of time?
Thanks in advance for any help
16-bits means 2 bytes per PCM value. Stereo means two channels (two PCM values per frame). So that totals 4 bytes per frame. 48000 fps is frames per second, so multiply the number of seconds by 4.
This is kind of a basic question which might sound too obvious to many of you , but I am getting confused so bad.
Here is what a Quora user says. Now It is clear to me what a Sampling rate is - The number of samples you take of a sound signal (in one second) is it's sampling rate.
Now my doubt here is - This rate should have nothing to do with the quantisation, right?
About bit-depth, Is the quantisation dependant on bit-depth? As in 32-bit (2^32 levels) and 64-bit (2^64 levels). Or is it something else?
and the bit-rate, is number of bits transferred in one second? If I an audio file says 320 kbps what does that really mean?
I assume the readers have got some sense on how I am panicking on where does the bit rate, and bit depth have significance?
EDIT: Also find this question if you have worked with linux OS and gstreamer framework.
Now my doubt here is - This rate should have nothing to do with the
quantisation, right?
Wrong. Sampling is a process that results in quantisation. Sampling, as the name implies, means taking samples (amplitudes) of a (usually) continuous signal (e.g audio) at regular time intervals and converting them to a different represantation thereof. In digital signal processing, this represantation is discrete (not continuous). An example of this process is a wave file (e.g recording your own voice and saving it as a wav).
About bit-depth, Is the quantisation dependant on bit-depth? As in
32-bit (2^32 levels) and 64-bit (2^64 levels). Or is it something
else?
Yes. The CD format, for example, has a bit depth of 16 (16 bits per sample). Bit depth is a part of the format of a sound (wave) file (along with the number of channels and sampling rate).
Since sound (think of a pure sine tone) has both positive and negative parts, I'd argue that you can represent (2^16 / 2) amplitude levels using 16 bits.
and the bit-rate, is number of bits transferred in one second? If I an
audio file says 320 kbps what does that really mean?
Yes. Bit rates are usually meaningful in the context of network transfers. 320 kbps == 320 000 bits per second. (for kilobit you multiply by 1000, rather than 1024)
Let's take a worked example 'Red-book' CD audio
The Bit depth is 16-bit. This is the number of bits used to represent each sample. This is intimately coupled with quantisation.
The Smaple-rate is 44.1kHz
The Frame-rate is 44.1kHz (two audio channels make up a stereo pair)
The Bit-rate is therefore 16 * 44100 * 2 = 1411200 bits/sec
There are a few twists with compressed audio streams such such as MP3 or AAC. In these, there is a non-linear relationship between bit-rate, sample-rate and bit-depth. The bit-rate is generally the maximum rate per-second and the efficiency of the codec is content dependant.
In anime, does frame means number of scene per second? Each scene can consist of several layer background, hero, object, etc. I think this is the reason why I am confused.
In wave (raw audio) file,
Does one frame contain data for one side (left or right) only?
Does bit sampling precision refer to a single side/channel?
With audio, do frames represent changes in loudness?
One frame can consist of left and right?
I.e. stereo 8 bit sampling depth => 1 frame => 2 bytes?
I do not know whether a formal definition of a frame exists, but when referring to an audio frame we usually mean a single time sample of a number of channels. So 2 audio channels # 8 bits per channel results in 2 bytes per frame. 4 channels # 16 bit per sample is 8 bytes.
I am developing a directshow audio decoder filter, to decode AC3 audio.
the filter is used in a live graph, decoding TS multicast.
the demuxer (mainconcept) provides me with the audio data demuxed, but does not provide timestamps for the sample.
how can I get/compute the correct timestamp of the audio?
I found this forum post:
http://www.ureader.com/msg/14712447.aspx
In it, a member gives the following formula for calculating the timestamps for audio, given it's format (sample rate, number of channels, bits per sample):
With PCM audio, duration_in_secs = 8 * buffer_size / wBitsPerSample /
nChannels / nSamplesPerSec or duration_in_secs = buffer_size /
nAvgBytesPerSec (since, for PCM audio, nAvgBytesPerSec =
wBitsPerSample * nChannels * nSamplesPerSec / 8).
The only thing you need to add is a tracking variable that tells you what sample number in the stream that you are at, so you can use it to offset the start time and end time by the duration (duration_in_secs) when doing linear streaming. For seek operations you would of course need to know or calculate the sample number into the stream.
Don't forget that the units for timestamps in DirectShow are typed as REFERENCE_TIME, a long integer or Int64. Each unit is equal to 100 nanoseconds. That is why you see in video filters the value 10,000,000 being divided by the relevant number of frames per second (FPS) to calculate timestamps for each frame because 10,000,000 equals 1 second in a REFERENCE_TIME variable.
Each AC-3 frame embeds data for 6 * 256 samples. Sampling rate can be 32 kHz, 44.1 kHz or 48 kHz (as defined by AC-3 specification Digital Audio Compression Standard (AC-3, E-AC-3)). The frames themselves do not carry timestamps, so you needs to assume continuous stream and increment time stamps respectively. As you mentioned the source is live, you might need to re-adjust time stamps on data starvation.
Each AC-3 frame is of fixed length (which you can identify from bitstream header), so you might also be checking if demultiplexer is giving you a single AC-3 frame or a few in a batch.