I have a question about the basic wav file data chunk.
I know that each 2 bytes in the data chunk section represent mono sound and left and right channel is alternately stored, but I still have no idea how to understand the 2 byte value.
Is it high and low?
value 0000 => lowest?
value FFFF => highest?
value 8FFF => no sound?
If so, when I want to reduce volume 50%, is all I have to do just classfy low and high
and divive by 2?
WAV files can contain samples at different bit depths and encodings, but one of the most commonly used is 16 bit PCM. With 16 bit PCM data in a WAV file, each two bytes should be interpreted as a short (i.e. a signed, two-byte number). In WAV files, the samples are always "little-endian". If you tag your question with a language, someone might be able to provide some simple sample code. In C/C++, you would cast your array of data read from the file into a (short *), allowing you to easily access each sample.
To answer the second part of your question, yes you can reduce the volume by halving the value of each sample.
Your understanding seems fine. The only thing I can think to say is one byte will be a most significant byte, and one will be least significant. Other than that, you seem spot on.
And your question is fine (In my opinion anyway!) Although more useful answers might be gotten if you were to ask a specific programming question, but I can see how that might not be possible.
Related
I am posed with the task of mixing raw data from audio files. I am currently struggling to get a clean sound from mixing the data, I keep getting distortion or white noise.
Lets say that I have a two byte array of data from two AudioInputStream's. The AIS is used to stream a byte array from a given audio file. Here I can playback single audio files using SourceDataLine's write method. I want to play two audio files simultaneously, therefore I am aware that I need to perform some sort of PCM addition.
Can anyone recommend whether this addition should be done with float values or byte values? Also, when it comes to adding 3,4 or more audio files, I am guessing my problem will be even harder! Do I need to divide by a certain amount to avoid this overflow? Lets say I am adding two 16-bit audio files (min -32,768, max 32,767).
I admit, I have had some advice on this before but can't seem to get it working! I have code of what I have tried but not with me!
Any advice would be great.
Thanks
First off, I question whether you are actually working with fully decoded PCM data values. If you are directly adding bytes, that would only make sense if the sound was recorded at 8-bit resolution, which is done less and less. These days, audio is recorded more commonly as 16-bit values, or more. I think there are some situations that don't require as much frequency content, but with current systems, the cpu savings aren't as critical so people opt to keep at least "CD Quality" (16-bit resolution, stereo, 41000 fps).
So step one, you have to make sure that you are properly converting the byte streams to valid PCM. For example, if 16-bit encoding, the two bytes have to be appended in the correct order (may be either big-endian or little-endian), and the resulting value used.
Once that is properly handled, it is usually sufficient to simply add the values and maybe impose a min and max filter to ensure the signal doesn't go beyond the defined range. I can think of two reasons why this works: (a) audio is usually recorded at a low enough volume that summing will not cause overflow, (b) the signals are random enough, with both positive and negative values, that moments where all the contributors line up in either the positive or negative direction are rare and short-lived.
Using a min and max will "clip" the signals, and can introduce some audible distortion, but it is a much less horrible sound than overflow! If your sources are routinely hitting the min and max, you can simply multiply a volume factor (within the range 0 to 1) to one or more of the contributing signals as a whole, to bring the audio values down.
For 16-bit data, it works to perform operations directly on the signed integers that result from appending the two bytes together (-32768 to 32767). But it is a more common practice to "normalize" the values, i.e., convert the 16-bit integers to floats ranging from -1 to 1, perform operations at that level, and then convert back to integers in the range -32768 to 32767 and break those integers into byte pairs.
There is a free book on digital signal processing that is well worth reading: Steven Smith's "The Scientists and Engineers Guide to Digital Signal Processing." It will give much more detail and background.
I am trying to understand what the raw data from an audio file looks like and how to get that data. I want to take the data and analyze it and see if I am able to make a program that can recognize patterns in a song such as in a hip hop song, finding the same beat in a chorus. In my head I think this could be a doable task if the data is in an integer form.
I've looked up many tutorials for this but all the tutorials use other libraries or don't explain it in a way I understand(more than likely the source of my issue).
I am wondering if there is someone out there that can help me understand a few things.
1). In an MP3 file, what is actually being stored in the file. Is it an integer that tells the radio/amp/audioPlayer a frequency, another integer for amplitude, etc...(over simplified because I don't know what other data is stored in an audio file).
2). If it is stored in an integer format, is there a way to read the integers and analyze it. If it is not stored in an integer format, how is it stored, and is there a way to convert it to an integer format?
3). In visual representations of an audio files like this one, it seems more clear what is what. It seems like the frequency is where on the circle the audio is represented, and the amplitude is how high it jumps. Is this right? Or does it just appear that way and I am understanding it incorrectly.
4). Is this task harder than I think it is? Considering I haven't found any good explanations or tutorials on how to do so, I am skeptical on how easy this would be.
(Sorry if this was poorly phrased, first question on stack and I am just illiterate :^)
I've read through many questions on stack overflow which states that to mix the audios, you just have to add the byte frames together (and make sure to clip when necessary). But what should I do if I want to say mix an audio with an another with some offset. For example, I want to mix second audio into the first one when the first audio reaches 5th second.
Any help would be appreciated!
Typically when working with audio on a computer, you will be working with audio in the time domain, in the format of PCM samples. That is, many times per second, the pressure level at that point of time will be measured an quantified into a number. If you are working with CD-quality audio, 44,1000 samples per second is the sample rate. The number is often quantified into 16-bit integers. (-32,767 to 32,768). (Other sample rates, bit depths, and quantization are out there and often used, this is just an example.)
If you want to mix two audio streams of the same sample rate, it is possible to simply add the values of each sample together. If you think about it, if you were to hear sound from two sources, their pressure levels would affect each other in much the same way. Sometimes they will cancel each other out, sometimes they will add to each other. You mentioned clipping... you can do this, but you will be introducing distortion into the mix. When a sound is too loud to be quantified, it is clipped at the maximum and minimums of the quantifiable range, causing audible clicks, pops, and poor quality sound. If you want to avoid this problem, you can cut the level of each in half, guaranteeing that even with both streams at their maximum level, they will be within the appropriate range.
Now, your question is about mixing audio with offset. It's absolutely no different. If you want to start mixing 5 seconds in, then 5 * 44,100 = 220500, meaning align sample zero of one stream to sample 220500 of the other stream and mix.
i am wondering if there is any way to cycle through a .wav file to get the amplitude/DB of a specific point in the wav file. i am reading it into a byte array now but that has no help to me from what i can see.
i am using this in conjunction with some hardware i have developed that encodes light data into binary and outputs audio. i wont get into the details but i need to be able to do this in c# or c++. i cant find any info on this anywhere. i have never programmed anything relating to audio so excuse me if this is a very easy thing.
i dont have anything started since this is the starting point so if anybody can point me to some functions, libraries, or methods to being able to collect the amplitude of the wave at a specific time in the file, i would greatly appreciate it.
i hope this is enough info, and thank you in advance if you are kind enough to help.
It is possible and it is done in a straightforward way: the file with PCM audio contains one value for every channel, for every (1/sample-rate) of second.
The values however might vary: 8-bit, 16-bit, single precision floating point values. You certainly have to take this into account and this is the reason you cannot take the bytes from byte array directly.
The .WAV file also has a header preceding the actual payload.
I am wondering on how to estimate where I am currently in an audio with regards to time, by using the data.
For example, I read data by byte[8192] blocks. How can I know how much byte[8192] is equivalent to in time?
If this is some sort of raw-ish encoding, like PCM, this is simple. The length in time is a function of the sample rate, bit depth, and number of channels. 30 seconds of 16-bit audio at 44.1kHz in mono is 2.5MB. However, you also need to factor in headers and container format crapola. WAV files for example can have a lot of other stuff in them.
Compressed formats are much more tricky. You can never be sure where you are without playing through the file to get to where you are. Of course you can always guesstimate based on the percentage of the file length, if that is good enough for your case.
I think this is not what he was asking.
First you have to tell us what kind of data you are using. WAV? MP3? Usually without knowing where that block came from - so you know if you have some kind of frame information and where to find it - you are not able to determine that block's position.
If you have the full stream and this data then you can do a search