I want to turn an InputStream to a seq of frames. If I correctly understand the code, gloss.io/lazy-decode-all eagerly consumes its second argument, which is not an option in my case.
I've found a way to do this with aleph/input-stream->channel, decode-channel and lamina.core.operators/channel->lazy-seq, but it looks a bit too much for such a common task. Also I'm a bit concerned with performance in this scenario: the app will be fed several gigabytes of data.
I could manually read the frames into ByteBuffers and decode them, but they have dynamic length, so I'll duplicate much of the logic in gloss.
Is there some concise way to lazily decode the stream?
Related
Here's the problem - I want to generate the delta of a binary file (> 1 MB in size) on a server and send the delta to a memory-constrained (low on RAM and no dynamic memory) embedded device over HTTP. Deltas are preferred (as opposed to sending the full binary file from the server) because of the high cost involved in transmitting data over the wire.
Trouble is, the embedded device cannot decode deltas and create the contents of the new file in memory. I have looked into various binary delta encoding/decoding algorithms like bsdiff, VCDiff etc. but was unable to find libraries that supported streaming.
Perhaps, rather than asking if there are suitable libraries out there, are there alternate approaches I can take that will still solve the original problem (send minimal data over the wire)? Although it would certainly help if there are suitable delta libraries out there that support streaming decode (written in C or C++ without using dynamic memory).
Maintain a copy on the server of the current file as held by the embedded device. When you want to send an update, XOR the new version of the file with the old version and compress the resultant stream with any sensible compressor. (Algorithms which allow high-cost encoding to allow low-cost decoding would be particularly helpful here.) Send the compressed stream to the embedded device, which reads the stream, decompresses it on the fly and XORs directly (a copy of) the target file.
If your updates are such that the file content changes little over time and retains a fixed structure, the XOR stream will be predominantly zeroes, and will compress extremely well: number of bytes transmitted will be small, effort to decompress will be low, memory requirements on the embedded device will be minimal. The further your model is from these assumptions, the less this approach will gain you.
Since you said the delta could be arbitrarily random (from zero delta to a completely different file), compression of the delta may be a lost cause. Lossless compression of random binary data is theoretically impossible. Also, since the embedded device has limited memory anyway, using a sophisticated -and therefore computationally expensive- library for compression/decompression of the occasional "simple" delta will probably be infeasible.
I would recommend simply sending the new file to the device in raw byte format, and overwriting the existing old file.
As Kevin mentioned, compressing random data should not be your goal. A few more comments about the type of data your working with would be helpful. Context is key in compression.
You used the term image which makes it sound like the classic video codec challenge. If you've ever seen weird video aliasing effects that impact the portion of the frame that has changed, and then suddenly everything clears up. You've likely witnessed the notion of a key frame along with a series of delta frames. Where the delta frames were not properly applied.
In this model, the server decides what's cheaper:
complete key frame
delta commands
The delta commands are communicated as a series of write instructions that can overlay the clients existing buffer.
Example Format:
[Address][Length][Repeat][Delta Payload]
[Address][Length][Repeat][Delta Payload]
[Address][Length][Repeat][Delta Payload]
There are likely a variety of methods for computing these delta commands. A brute force method would be:
Perform Smith Waterman between two images.
Compress the resulting transform into delta commands.
I have an int16_t[] buffer with PCM raw audio data and I want to apply some effects (like echo, reverb, gain...) into it.
I thought that SoX or similar can do the trick for me, but SoX only works with files and other similar libraries that supports adding sound effects seems to add the effects only when the sound is played. So my problem with this is that I want to apply the effect to the samples into my buffer without playing them.
I have never worked with audio, but reading about PCM data I have learned that I can apply gain multiplying each sample value, for example. But I'm looking for any library or relatively easy algorithms that I can use directly in my buffer to get the sound effects applied.
I'm sure there are a lot of solutions to my problem out there if you know what to look for, but it's my first time with audio "processing" and I'm lost, as you can see.
For everyone like me, interested in learning DSP related to audio processing with C++ I want to share my little research results and opinion, and perhaps save you some time :)
After trying several DSP libraries, finally I have found The Synthesis ToolKit in C++ (STK), an open-source library that offer easy and clear interfaces and easy to understand code that you can dive in to learn about various basic DSP algorithms.
So, I recommend to anyone who is starting out and have no previous experience to take a look at this library.
Your int16_t[] buffer contains a sequence of samples. They represent instantaneous amplitude levels. Think of them as the voltage to apply to the speaker at the corresponding instant in time. They are signed numbers with values in the range (-32767,32767]. A stream of constant zeros means silence. A stream of constant -32000 (for example) also means silence, but it will eventually burn your your speaker coil. The position in the array represents time, and the value of each sample represents voltage.
If you want to mix two sample streams together, for example to apply a chirp, you get yourself a sample stream with the chirp in it (record a bird or something). You then add the two sounds sample by sample.
You can do a super-cheesy reverb effect by taking your original sound buffer, lowering its volume (perhaps by dividing all the samples by a constant), and adding it back to your original stream, but shifting the samples by a tenth of a second's worth of array position.
Those are the basics of audio processing. Things get very sophisticated indeed. This field is known as "digital signal processing" and there are plenty of books on the subject.
You can do it either with hacking the audio buffer and trying to do some effects like gain and threshold with simple math operations or do it correct using proper DSP algorithms. If you wish to do it correct, I would recommend using the Speex Library. It's open source and and well tested. www (dot)speex (dot)org. The code should compile on MSVC or linux with minimal effort. This is the fastest way to get a good audio code working with proper DSP techniques. Your code would look like .. please read the AEC example.
st = speex_echo_state_init(NN, TAIL);
den = speex_preprocess_state_init(NN, sampleRate);
speex_echo_ctl(st, SPEEX_ECHO_SET_SAMPLING_RATE, &sampleRate);
speex_preprocess_ctl(den, SPEEX_PREPROCESS_SET_ECHO_STATE, st);
You need to setup the states, the code testecho includes these.
I've been wanting to compress a string in C++ and display it's compressed state to console. I've been looking around for something and can't find anything appropriate so far. The closest thing I've come to finding it this one:
How to simply compress a C++ string with LZMA
However, I can't find the lzma.h header which works with it anywhere.
Basically, I am looking for a function like this:
std::string compressString(std::string uncompressedString){
//Compression Code
return compressedString;
}
The compression algorithm choice doesn't really matter. Anybody can help me out finding something like this? Thank you in advance! :)
Based on the pointers in the article I'm fairly certain they are using XZ Utils, so download that project and make the produced library available in your project.
However, two caveats:
dumping a compressed string to the console isn't very useful, as that string will contain all possible byte values, most of which aren't displayable on a console
compressing short strings (actually any small quantity of data) isn't what most general-purpose compressors were designed for, in many cases the compressed result will be as big or even bigger than the input. However, I have no experience with LZMA on small data quantities, an extensive test with data representative for your use case will tell you whether it works as expected.
One algorithm I've been playing with that gives good compression on small amounts of data (tested on data chunks sized 300-500 bytes) is range encoding.
I have a set of mp3 files, some of which have extended periods of silence or periodic intervals of silence. How can I programmatically detect this?
I am looking for a library in C++, or preferably C#, that will allow me to examine the sound content of these files for the silences.
EDIT: I should elaborate what I am trying to achieve. I am capturing streaming sports commentary using VLC and saving it to mp3. When a game is delayed, or cancelled, the streaming commentary is replaced by a repetitive message saying commentary is not available. By looking for these periodic silences (or total silence), I can detect if there is no commentary and stop the streaming recording
For this reason I am reluctant to decompress the mp3 because if would mean my test for these silences would be very slow. Unless I can decode the last 5 minutes of the file?
Thanks
Andrew
I'm not aware of a library that will detect silence directly in the MP3 encoded data, since its not a trivial task to detect silence without first decompressing. Luckily, its easy to find libraries that decode MP3 files and access them as PCM data, and its trivial to detect silence in PCM Data. Here is one such Library for C# I found, but I'm sure there are tons: http://www.robburke.net/mle/mp3sharp/
Once you decode the data, you will have a list of PCM samples. In the most basic form, the algorithm you need to detect silence is simply to analyze a small chunks (could be as little as .25s or as much as several seconds), and make sure that the absolute value of each sample in the chunk is below a threshold. The threshold value you use determines how 'quiet' the sound has to be to be considered silence, and the chunk size determines how long the volume needs to be below that threshold to be considered silence (If you go with very short chunks, you will get lots of false positives due to samples near zero-crossings, but .25s or higher should be ok. There are improvements to the basic approach such as using historesis (which is basically using two thresholds, one for the transition to silence, and one for the transition from silence), and filtering.
Unfortunately, I don't know a library for C++ or C# that implements level detection off hand, and nothing immediately springs up on google, but at least for the simple version its pretty easy to code.
Edit: Also, this library seems interesting: http://naudio.codeplex.com/
Also, while not a true duplicate question, the answers here will be useful for you:
Detecting audio silence in WAV files using C#
I want to concat two or more gzip streams without recompressing them.
I mean I have A compressed to A.gz and B to B.gz, I want to compress them to single gzip (A+B).gz without compressing once again, using C or C++.
Several notes:
Even you can just concat two files and gunzip would know how to deal with them, most of programs would not be able to deal with two chunks.
I had seen once an example of code that does this just by decompression of the files and then manipulating original and this significantly faster then normal re-compression, but still requires O(n) CPU operation.
Unfortunaly I can't found this example I had found once (concatenation using decompression only), if someone can point it I would be greatful.
Note: it is not duplicate of this because proposed solution is not fits my needs.
Clearification edit:
I want to concate several compressed HTML pices and send them to browser as one page, as per request: "Accept-Encoding: gzip", with respnse "Content-Encoding: gzip"
If the stream is concated as simple as cat a.gz b.gz >ab.gz, Gecko (firefox) and KHTML web engines gets only first part (a); IE6 does not display anything and Google Chrome displays first part (a) correctly and the second part (b) as garbage (does not decompress at all).
Only Opera handles this well.
So I need to create a single gzip stream of several chunks and send them without re-compressing.
Update: I had found gzjoin.c in the examples of zlib, it does it using only decompression. The problem is that decompression is still slower them simple memcpy.
It is still faster 4 times then fastest gzip compression. But it is not enough.
What I need is to find the data I need to save together with gzip file in order to
not run decompression procedure, and how do I find this data during compression.
Look at the RFC1951 and RFC1952
The format is simply a suites of members, each composed of three parts, an header, data and a trailer. The data part is itself a set of chunks with each chunks having an header and data part.
To simulate the effect of gzipping the result of the concatenation of two (or more files), you simply have to adjust the headers (there is a last chunk flag for instance) and trailer correctly and copying the data parts.
There is a problem, the trailer has a CRC32 of the uncompressed data and I'm not sure if this one is easy to compute when you know the CRC of the parts.
Edit: the comments in the gzjoin.c file you found imply that, while it is possible to compute the CRC32 without decompressing the data, there are other things which need the decompression.
The gzip manual says that two gzip files can be concatenated as you attempted.
http://www.gnu.org/software/gzip/manual/gzip.html#Advanced-usage
So it appears that the other tools may be broken. As seen in this bug report.
http://connect.microsoft.com/VisualStudio/feedback/ViewFeedback.aspx?FeedbackID=97263
Apart from filing a bug report with each one of the browser makers, and hoping they comply, perhaps your program can cache the most common concatenations of the required data.
As others have mentioned you may be able to perform surgery:
http://www.gzip.org/zlib/rfc-gzip.html
And this requires a CRC-32 of the final uncompressed file. The required size of the uncompressed file can be easily calculated by adding the lengths of the individual sub-files.
In the bottom of the last link, there is code for calculating a running crc-32 named update_crc.
Calculating the crc on the uncompressed files each time your process is run, is probably cheaper than the gzip algorithm itself.
It seems that the original compression of the individual files is done by you. It also seems that the desired result (concatenation of several pieces) is small enough to be sent to a web browser in one page. In that case your efficiency concerns seem to be unwarranted.
Please note that (1) the gzjoin.c approach is highly likely to be the best answer that you could get to your question as stated (2) it is complicated microsurgery performed by one of the gzip originators and may not have been subject to extensive stress testing.
Please consider a boring understandable reliable approach: storing the original pieces UNcompressed, then select required pieces, and concatenate and compress them. Note that the compression ratio may be better than that obtained by glueing together small compressed pieces.
If taring them is not out of the question (since the linked cat solution isn't viable for you):
tar cf A_B.gz.tar A.gz B.gz
Then, to get them back:
tar xf A_B.gz.tar