in the example of this post
http://dracoater.blogspot.com/2008/11/comparing-2-audio-wav-files.html
bits per sample is 0010 or 16 (in DEC)
but why each sample contains 4 bytes ?
4 X 8 = 32 bits !?
Each of those "samples" labeled on the bottom line of your diagram is actually 2 samples - i.e. each circled pair of 2 bytes is a sample - The diagram is wrong.
If it's a stereo file, the samples will be in pairs (one for left, one for right) - that group is usually called a "frame". The sample channels are interleaved in the file e.g. LRLRLRLRLRLRLRLRLR
As title; for some compressed format such as EAC3, AC3 frame starts as a sync word.
So what's PCM (raw audio)'s sync word? How to identify the beginning of a PCM frame?
I met a problem where audio is concatenated by several audio segments and each of them has different frame size. I need to identify the start position.
Thanks in advance.
There is no such concept as a frame in PCM. The concept of a frame is to indicate points of random access. In PCM every single sample is a point of random access, hence start indicators are not required, and there are no standard frame size. It all up to you.
A PCM frame is different from the frames you're describing, in that a frame is just a single sample on all channels. That is, if I'm recording 16-bit stereo PCM audio, each frame is 4 bytes (32 bits) long.
There is no sync word, nor frame header in raw PCM. It's just a stream of data. You need to know the bit depth, channel count, and current offset if you want to sync to it. (Or, you need to do some simple heuristics. For example, apply several different formats and offsets to a small chunk of data and see which one has the least variance/randomness from sample to sample.)
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.
When I was studying Cocoa Audio Queue document, I met several terms in audio codec. There are defined in a structure named AudioStreamBasicDescription.
Here are the terms:
1. Sample rate
2. Packet
3. Frame
4. Channel
I known about sample rate and channel. How I was confused by the other two. What do the other two terms mean?
Also you can answer this question by example. For example, I have an dual-channel PCM-16 source with a sample rate 44.1kHz, which means there are 2*44100 = 88200 Bytes PCM data per second. But how about packet and frame?
Thank you at advance!
You are already familiar with the sample rate defintion.
The sampling frequency or sampling rate, fs, is defined as the number of samples obtained in one second (samples per second), thus fs = 1/T.
So for a sampling rate of 44100 Hz, you have 44100 samples per second (per audio channel).
The number of frames per second in video is a similar concept to the number of samples per second in audio. Frames for our eyes, samples for our ears. Additional infos here.
If you have 16 bits depth stereo PCM it means you have 16*44100*2 = 1411200 bits per second => ~ 172 kB per second => around 10 MB per minute.
To the definition in reworded terms from Apple:
Sample: a single number representing the value of one audio channel at one point in time.
Frame: a group of one or more samples, with one sample for each channel, representing the audio on all channels at a single point on time.
Packet: a group of one or more frames, representing the audio format's smallest encoding unit, and the audio for all channels across a short amount of time.
As you can see there is a subtle difference between audio and video frame notions. In one second you have for stereo audio at 44.1 kHz: 88200 samples and thus 44100 frames.
Compressed format like MP3 and AAC pack multiple frames in packets (these packets can then be written in MP4 file for example where they could be efficiently interleaved with video content). You understand that dealing with large packets helps to identify bits patterns for better coding efficiency.
MP3, for example, uses packets of 1152 frames, which are the basic atomic unit of an MP3 stream. PCM audio is just a series of samples, so it can be divided down to the individual frame, and it really has no packet size at all.
For AAC you can have 1024 (or 960) frames per packet. This is described in the Apple document you pointed at:
The number of frames in a packet of audio data. For uncompressed audio, the value is 1. For variable bit-rate formats, the value is a larger fixed number, such as 1024 for AAC. For formats with a variable number of frames per packet, such as Ogg Vorbis, set this field to 0.
In MPEG-based file format a packet is referred to as a data frame (not to be
mingled with the previous audio frame notion). See Brad comment for more information on the subject.
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.