I want to convert a sound from Mic to binary and match it from the database(a type of voice identification program but don't getting idea how to get sound from Mic directly so that i can convert it to binary?Also it is possible or not. Please guide me )
See this:
http://www.dotnetspider.com/resources/4967-How-record-voice-from-microphone.aspx
You're not going to be able to identify voices by doing a binary comparison on sound data. The binary of a particular sound will not be identical to an imitation of that sound unless it is literally the same file because of minor variations in just about everything. You'll need to do some signals processing to do a fuzzy comparison of the data. You can read about signal processing on wikipedia.
You will probably find it easier to use a third party library to process the sound for you. Something like this might be a good start.
You're looking at two very distinct problems here.
The first is pretty technical: Getting sound from the microphone into a digital waveform. How you do this exactly depends on the OS and API you're using (on Windows, you're probably looking at DirectX audio or, if available, ASIO). Typically, this is how you'd proceed:
Set up a recording buffer for the microphone, with suitable parameters (number of channels, physical input on the sound card, sample rate, bit depth, buffer size)
Start the recording. This usually involves pointing the sound library to a callback function to process the recorded buffer.
In the callback, read the buffer, convert it to a suitable format, and append it to the audio file of your choice. (You could also record to RAM only, but longer recordings may exceed available storage).
Store the recorded audio in a suitable database field (some kind of binary blob)
This is the easy part though; the harder part is matching a chunk of audio data against other chunks. A naïve approach would be to try and find exact matches, but that won't help you much, because the chance that you find one is practically zero - recording equipment, even the best, introduces a bit of random noise, and recording setups vary slightly whether you want to or not, so even if you'd have someone say something twice, perfectly identical, you'd still see differences in the recorded audio.
What you need to do, then, is find certain typical characteristics of the waveform. Things you could look for are:
Overall amplitude shape
Base frequencies
Selected harmonics (formants)
Extracting these is non-trivial and involves pretty severe math; and then you'll have to condense them into some sort of fingerprint, and find a way to compare them with some fuzziness (so that a near-match is good enough, rather than requiring exact matches). Finding the right parameters and comparison algorithms isn't easy, and it takes a lot of tweaking and testing; your best bet is to go find a library that does this for you.
Related
I am trying to disassemble a code from a old radio containing a 68xx (68hc12 like) microcontroller. The problem is, I dont have the access to the interrupt vector of the micro in the top of the ROM, so I don't know where start to look. I only have the code below the top. There is some suggestion of where or how can I find meaningful routines in the code data?
You can't really disassemble reliably without knowing where the reset vector points. What you can do, however, is try to narrow down the possible reset addresses by eliminating all those other addresses that cannot possibly be a starting point.
So, given that any address in the memory map that contains a valid opcode is a potential reset point, you need to either eliminate it, or keep it for further analysis.
For the 68HC11 case, you could try to guess somewhat the entry point by looking for LDS instructions with legitimate operand value (i.e., pointing at or near the top of available RAM -- if multiple RAM banks, then to any of them).
It may help a bit if you know the device's full memory map, i.e., if external memory is used, its mapping and possible mapped peripherals (e.g., LCD). Do you also know CONFIG register contents?
The LDS instruction is usually either the very first instruction, or close thereafter (so look back a few instructions when you feel you have finally singled out your reset address). The problem here is some data may, by chance, appear as LDS instructions so you could end up with multiple potentially valid entry points. Only one of them is valid, of course.
You can eliminate further by disassembling a few instructions starting from each of these LDS instructions until you either hit an illegal opcode (i.e. obviously not a valid code sequence but an accidental data arrangement that looks like opcodes), or you see a series of instructions that are commonly used in 68HC11 initialization. These involve (usually) initialization of any one or more of the registers BPROT, OPTION, SCI, INIT ($103D in most parts, but for some $3D), etc.
You could write a relatively small script (e.g., in Lua) to do the basic scanning of the memory map and produce a (hopefully small) set of potential reset points to be examined further with a true disassembler for hints like the ones I mentioned.
Now, once you have the reset vector figured out the job becomes somewhat easier but you still need to figure out where any interrupt handlers are located. For this your hint is an RTI instruction and whatever preceding code that normally should acknowledge the specific interrupt it handles.
Hope this helps.
Let's say I have some numbers, like
5,10,7,8,9,6,2,4,8,5,3,9,78,5,6
I need to send this to another computer, but as the least number of possible bytes. I know what there is a way to do that, I just forgot what it's called and how it works, but generally doing some math with those numbers, getting a big number that, from this number, I'll be able to export the data and get this numbers from this number. Thanks in advance.
EDIT
OK so I need to send this text in UDP but I need it as less bits as possible. I'm sending some options, like firstcolor-secondcolor, let's say I have 15 colors. Every color is just number, from 1 to 199, but maybe there is a better way to send this data? thanks.
No one can say which compression scheme is the best for you. We don't have any information about the numbers. But as a first try, you could just write them into a file and use gzip compression on it. Or bzip2, or 7zip.
And only if all these don't help, you should think about doing the compression yourself.
You also didn't tell us your operating systems (source computer, destination computer) and from where you get the data.
[Update, based on the edit in the question:] So basically you want to send some numbers in the range of 1 to 199. This is pretty close to what a single byte can hold.
If it is ok that you use 8 bits per number (meaning you waste 0.4 bits per number), this is trivial but highly depends on the programming language. Here is how it might look like in Java syntax:
ByteBuffer buf = new ByteBuffer();
buf.add(1);
buf.add(199);
buf.add(78);
buf.add(7);
udpSocket.send(buf.toArray());
Get a compression library (like zlib, for example) and feed your numbers in (as an array of integers, for example). This is compressing your data. That same library should allow you to reverse the process and decompress the data at the other end to get your values back out.
If you want to improve your algorithmic knowledge and your requirements are simple and non-critical I'd recommend having a go at writing your own compression/decompression code. If not, grab some code off the shelf - there are loads of good libraries around.
Would it be possible to transfer large files using only a system of checksums, and then reconstruct the original file by calculations?
Say that you transfer the MD5 checksum of a file and the size of the file. By making a "virtual file" and calculating it's checksum, trying every single bit combination, you should eventually "reach" the original file. But on the way you would also get a lot of "collisions" where the checksum also match.
So we change the first byte of the original file to some specified value, calculate the checksum again, and send this too. If we make the same substitution in the virtual file we can test each "collision" to see if it still matches. This should narrow it down a bit, and we can do this several times.
Of course, the computing power to do this would be enormous. But is it theoretically possible, and how many checksums would you need to transfer something (say 1mb)? Or would perhaps the amount of data needed to transfer the checksums almost as large as the file, making it pointless?
The amount of data you need to transfer would most certainly be the same size as the file. Consider: If you could communicate a n byte file with n-1 bytes of data, that means you've got 256^(n-1) possible patterns of data you may have sent, but are selecting from a space of size 256^n. This means that one out of every 256 files won't be expressible using this method - this is often referred to as the pidegonhole principle.
Now, even if that wasn't a problem, there's no guarentee that you won't have a collision after any given amount of checksumming. Checksum algorithms are designed to avoid collisions, but for most checksum/hash algorithms there's no strong proof that after X hashes you can guarantee no collisions in a N-byte space.
Finally, hash algorithms, at least, are designed to be hard to reverse, so even if it were possible it would take an impossible huge amount of CPU power to do so.
That said, for a similar approach, you might be interested in reading about Forward Error Correction codes - they're not at all hash algorithms, but I think you may find them interesting.
What you have here is a problem of information. A checksum is not necessarily unique to a particular set of data, in fact to be so it would effectively need to have a many bits of information as the source. What it can indicate is that the data received is not the exact data that the checksum was generated from but in most cases it can't prove it.
In short "no".
To take a hypothetical example, consider a 24 bpp photo with 6 pixels -- there are 2^(24 * 6) (2^144) possible combinations of intensities for each colour channel on those six pixels, so you can gaurantee that if you were to evaluate every possibility, you are guaranteed an MD5 collision (as MD5 is a 128 bit number).
Short answer: Not in any meaningfull form.
Long answer:
Let us assume an arbitrary file file.bin with a 1000-byte size. There are 2^(8*1000) different combinations that could be its actual contents. By sending e.g. a 1000-bit checksum,
you still have about 2^(7*1000) colliding alternatives.
By sending a single additional bit, you might be able cut those down by half... and you still have 2^6999 collisions. By the time you eliminate the colisions, you will have sent at least 8000 bits i.e. an amount equal or greater to the file size.
The only way for this to be theoretically possible (Note: I did not say "feasible", let alone "practical") would be if the file did not really contain random data and you could use that knowledge to prune alternatives. In that case you'd be better off using compression, ayway. Content-aware compression algorithms (e.g. FLAC for audio) use a-priori knowledge on the properties of the input data to improve the compression ratio.
I think what you are thinking of is in fact an interesting topic, but you haven't hit upon the right method. If I can try and rephrase your question, you are asking if there is a way to apply a function to some data, transmit the result of the function, and then reconstruct the original data from the terser function result. For a single MD5 checksum the answer is no, but with other functions, provided you are willingly to send several function results, it is possible. In general this area of research is called compressed sensing. Sometimes exact reconstruction is possible, but more often it is used as a lossy compression scheme for images and other visual or sound data.
I am working on a game engine which is loosely descended from Quake 2, adding some things like scripted effects (allowing the server to specify special effects in detail to a client, instead of having only a limited number of hardcoded effects which the client is capable of.) This is a tradeoff of network efficiency for flexibility.
I've hit an interesting barrier. See, the maximum packet size is 2800 bytes, and only one can go out per client per frame.
Here is the script to do a "sparks" effect (could be good for bullet impact sparks, electrical shocks, etc.)
http://pastebin.com/m7acdf519 (If you don't understand it, don't sweat it; it's a custom syntax I made and not relevant to the question I am asking.)
I have done everything possible to shrink the size of that script. I've even reduced the variable names to single letters. But the result is exactly 405 bytes. Meaning you can fit at most 6 of these per frame. I also have in mind a few server-side changes which could shave it down another 12, and a protocol change which might save another 6. Although the savings would vary depending on what script you are working with.
However, of those 387 bytes, I estimate that only 41 would be unique between multiple usages of the effect. In other words, this is a prime candidate for compression.
It just so happens that R1Q2 (a backward-compatible Quake 2 engine with an extended network protocol) has Zlib compression code. I could lift this code, or at least follow it closely as a reference.
But is Zlib necessarily the best choice here? I can think of at least one alternative, LZMA, and there could easily be more.
The requirements:
Must be very fast (must have very small performance hit if run over 100 times a second.)
Must cram as much data as possible into 2800 bytes
Small metadata footprint
GPL compatible
Zlib is looking good, but is there anything better? Keep in mind, none of this code is being merged yet, so there's plenty of room for experimentation.
Thanks,
-Max
EDIT: Thanks to those who have suggested compiling the scripts into bytecode. I should have made this clear-- yes, I am doing this. If you like you can browse the relevant source code on my website, although it's still not "prettied up."
This is the server-side code:
Lua component: http://meliaserlow.dyndns.tv:8000/alienarena/lua_source/lua/scriptedfx.lua
C component: http://meliaserlow.dyndns.tv:8000/alienarena/lua_source/game/g_scriptedfx.c
For the specific example script I posted, this gets a 1172 byte source down to 405 bytes-- still not small enough. (Keep in mind I want to fit as many of these as possible into 2800 bytes!)
EDIT2: There is no guarantee that any given packet will arrive. Each packet is supposed to contain "the state of the world," without relying on info communicated in previous packets. Generally, these scripts will be used to communicate "eye candy." If there's no room for one, it gets dropped from the packet and that's no big deal. But if too many get dropped, things start to look strange visually and this is undesirable.
LZO might be a good candidate for this.
FINAL UPDATE: The two libraries seem about equivalent. Zlib gives about 20% better compression, while LZO's decoding speed is about twice as fast, but the performance hit for either is very small, nearly negligible. That is my final answer. Thanks for all other answers and comments!
UPDATE: After implementing LZO compression and seeing only sightly better performance, it is clear that my own code is to blame for the performance hit (massively increased number of scripted effects possible per packet, thus my effect "interpreter" is getting exercised a lot more.) I would like to humbly apologize for scrambling to shift blame, and I hope there are no hard feelings. I will do some profiling and then maybe I will be able to get some numbers which will be more useful to someone else.
ORIGINAL POST:
OK, I finally got around to writing some code for this. I started out with Zlib, here are the first of my findings.
Zlib's compression is insanely great. It is reliably reducing packets of, say, 8.5 kib down to, say, 750 bytes or less, even when compressing with Z_BEST_SPEED (instead of Z_DEFAULT_COMPRESSION.) The compression time is also pretty good.
However, I had no idea the decompression speed of anything could even possibly be this bad. I don't have actual numbers, but it must be taking 1/8 second per packet at least! (Core2Duo T550 # 1.83 Ghz.) Totally unacceptable.
From what I've heard, LZMA is a tradeoff of worse performance vs. better compression when compared to Zlib. Since Zlib's compression is already overkill and its performance is already incredibly bad, LZMA is off the table sight unseen for now.
If LZO's decompression time is as good as it's claimed to be, then that is what I will be using. I think in the end the server will still be able to send Zlib packets in extreme cases, but clients can be configured to ignore them and that will be the default.
zlib might be a good candidate - license is very good, works fast and its authors say it has very little overhead and overhead is the thing that makes use for small amounts of data problematic.
you should look at OpenTNL and adapt some of the techniques they use there, like the concept of Network Strings
I would be inclinded to use the most significant bit of each character, which is currently wasted, by shifting groups of 9 bytes leftwards, you will fit into 8 bytes.
You could go further and map the characters into a small space - can you get them down to 6 bits (i.e. only having 64 valid characters) by, for example, not allowing capital letters and subtracting 0x20 from each character ( so that space becomes value 0 )
You could go further by mapping the frequency of each character and make a Huffman type compression to reduce the avarage number bits of each character.
I suspect that there are no algorithms that will save data any better that, in the general case, as there is essentially no redundancy in the message after the changes that you have alrady made.
How about sending a binary representation of your script?
So I'm thinking in the lines of a Abstract Syntax Tree with each procedure having a identifier.
This means preformance gain on the clients due to the one time parsing, and decrease of size due to removing the method names.
Say you have a conference room and meetings take place at arbitrary impromptu times. You would like to keep an audio record of all meetings. In order to make it as easy to use as possible, no action would be required on the part of meeting attenders, they just know that when they have a meeting in a specific room they will have a record of it.
Obviously just recording nonstop would be inefficient as it would be a waste of data storage and a pain to sift through.
I figure there are two basic ways to go about it.
Recording simply starts and stops according to sound level thresholds.
Recording is continuous, but split into X minute blocks. Blocks found to contain no content are discarded.
I like the second way better because I feel there is less risk for losing data because of late starts, or triggers failing.
I would like to implement in Python, and on Windows if possible.
Implementation suggestions?
Bonus considerations that probably deserve their own questions:
best audio format and compression for this purpose
any way of determining how many speakers are present, assuming identification is unrealistic
This is one of those projects where the path is going to be defined more about what's on hand for ready reuse.
You'll probably find it easier to continuously record and saving the data off in chunks (for example, hour long pieces).
Format is going to be dependent on what you in the form of recording tools and audio processing library. You may even find that you use two. One format, like PCM encoded WAV for recording and processing, but compressed MP3 for storage.
Once you have an audio stream, you'll need to access it in a PCM form (list of amplitude values). A simple averaging approach will probably be good enough to detect when there is a conversation. Typical tuning attributes:
* Average energy level to trigger
* Amount of time you need to be at the energy level or below to identify stop and start (I recommend two different values)
* Size of analysis window for averaging
As for number of participants, unless you find a library that does this, I don't see an easy solution. I've used speech recognition engines before and also done a reasonable amount of audio processing and I haven't seen any 'easy' ways to do this. If you were to look, search out universities doing speech analysis research. You may find some prototypes you can modify to give your software some clues.
I think you'll have difficulty doing this entirely in Python. You're talking about doing frequency/amplitude analysis of MP3 files. You would have to open up the file and look for a volume threshold, then cut out the portions that go below that threshold. Figuring out how many speakers are present would require very advanced signal processing.
A cursory Google search turned up nothing for me. You might have better luck looking for an off-the-shelf solution.
As an aside- there may be legal complications to having a recorder running 24/7 without letting people know.