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."
I am new to audio programming,
But I am wondering formula of bitRate,
According to wiki https://en.wikipedia.org/wiki/Bit_rate#Audio,
bit rate = sample rate X bit depth X channels
and
sample rate is the number of samples (or snapshots taken) per second obtained by a digital audio device.
bit depth is the number of bits of information in each sample.
So why bit rate = sample rate X bit depth X channels?
From my perspective, if bitDepth = 2 bit, sample rate = 3 HZ
then I can transfer 6 bit data in 1 second
For example:
Sample data = 00 //at 1/3 second.
Sample data = 01 //at 2/3 second.
Sample data = 10 //at 3/3 second.
So I transfer 000110 in 1 second, is that correct logic?
Bit-rate is the expected amount of bits per interval (eg: per second).
Sound cycles are measured in hertz, where 1 hertz == 1 second. So to get full sound data that represents that 1 second of audio, you calculate how many bits are needed to be sent (or for media players, they check the bit-rate in a file-format's settings so they can read & playback correctly).
Why is channels involved (isn't sample rate X bit-depth enough)?
In digital audio the samples are sent for each "ear" (L/R channel). There will always be double the amount of samples in a stereo sound versus if it was mono sound. Usually there is a "flag" to specify if sound is stereo or mono.
Logic Example: (without bit-depth, and assuming 1-bit per sample)...
There is speech "Hello" recorded at 200 samples/sec at bitrate of 100/sec. What happens?
If stereo flag, each ear gets 100 samples per sec (correct total of 200 played)
If mono, audio speech will sound slow by half (since only 100 samples played at expected bit-rate of 100, but remember, a full second was recorded at 200 sample/sec. You get half of "hello" in one second and the other at next second to (== slowed speech).
Taking the above example, you will find these audio gives slow/double speed adventures in your "new to audio programming" experience. The fix will be either setting channels amount or setting bit-rate correctly. Good luck.
The 'sample rate' is the rate at which each channel is sampled.
So 'sample rate X bit depth' will give you the bit rate for a single channel.
You then need to multiply that by the number of channels to get the total bit rate flowing through the system.
For example the CD standard has a sample rate of 44100 samples per second and a bit depth of 16 giving a bit rate of 705600 per channel and a total bit rate of 1411200 bits per seconds for stereo.
I have recently read that uncompressed CD-quality audio has a bandwidth of 1.411 Mbps in case of stereo, does it mean a CD can be played to output audio at the rate of 1.411 Mbps, i mean does it play 1.411 Mbits of stereo audio every second..?
Two channels, each with 44,100 16-bit samples per second. That is 2 x 44100 x 16 = 1,411,200bps. That is 1.411Mbps. (176400 bytes per second)
Each second requires 1.411Mb. If you reduced the sample rate by half, you would double the number of seconds that can be recorded on a CD. Same if you dropped it to one channel, or 8-bit.
To imagine the impact of reducing the sample rate, lets suppose a technology that sampled every 1 second. This would be like pressing mute over and over, you would only catch parts.
Reducing the channel to one is easy to imagine, that's monaural.
Reducing to 8-bit is harder to describe. Imagine we reduced it to 1-bit. That would essentially mean the speaker has two states, fully centered and fully driven. That is not much variation. 16 bits gives 65536 positions.
I am trying to understand the meaning of "rate" as it applies to ALSA. It is always reported in units of Hz, and is often expanded in text as "sample rate". However, usage seems to indicate that it is actually frame rate or, possibly, byte rate of an audio stream.
The confusion may arise from what exactly is referred to by "sample". If each channel is sampling at a particular frequency, then that is the frame rate of the overall stream.
So, for example, if I have a rate of 44100 Hz on a 3-channel, 16-bit audio stream, am I processing 44,100 bytes per second, 88,200 bytes per second (44,100 samples per second), or 264,600 bytes per second (44,100 frames per second)?
Question rather related to [1] and [2], and was probably the motive behind [3].
Elaboration of ALSA's meaning of "frame" and "sample" at Introduction to Sound Programming with ALSA.
In ALSA, the rate is the frame rate.
Historically, this value is called "sample rate" because it is the rate at which samples arrive at each DAC. This view is correct only if each channel has its own DAC. Nowadays, most DAC chips have at least two channels, so the actual sample rate does not really occur anywhere in the system.
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.