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  • Algorithm for digit summing?

    - by Joe
    I'm searching for an algorithm for Digit summing. Let me outline the basic principle: Say you have a number: 18268. 1 + 8 + 2 + 6 + 8 = 25 2 + 5 = 7 And 7 is our final number. It's basically adding each number of the whole number until we get down to a single (also known as a 'core') digit. It's often used by numerologists. I'm searching for an algorithm (doesn't have to be language in-specific) for this. I have searched Google for the last hour with terms such as digit sum algorithm and whatnot but got no suitable results. Any help would be great, thanks.

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  • Kohonen SOM Maps: Normalizing the input with unknown range

    - by S.N
    According to "Introduction to Neural Networks with Java By Jeff Heaton", the input to the Kohonen neural network must be the values between -1 and 1. It is possible to normalize inputs where the range is known beforehand: For instance RGB (125, 125, 125) where the range is know as values 0 and 255: 1. Divide by 255: (125/255) = 0.49 (0.49,0.49,0.49) 2. Multiply by two and subtract one: ((0.49*2)-1)=-0.02 (-0.02,-0.02,-0.02) The question is how can we normalize the input where the range is unknown like our height or weight. Also, some other papers mention that the input must be normalized to the values between 0 and 1. Which is the proper way, "-1 and 1" or "0 and 1"?

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  • "'data(...).options' is null or not an object" in jquery-ui

    - by ripper234
    I'm using jquery-ui 1.8, and getting this error in Internet Explorer: Webpage error details User Agent: Mozilla/4.0 (compatible; MSIE 8.0; Windows NT 6.1; WOW64; Trident/4.0; SLCC2; .NET CLR 2.0.50727; .NET CLR 3.5.30729; .NET CLR 3.0.30729; Media Center PC 6.0; .NET4.0C; .NET4.0E) Timestamp: Mon, 10 May 2010 06:26:48 UTC Message: 'data(...).options' is null or not an object Line: 75 Char: 13074 Code: 0 URI: http://localhost:58365/Scripts/Lib/jquery-ui-1.8.custom.min.js Is this a known bug? Is there a workaround? The error happens when I use droppable/draggable.

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  • Nested flowlayout panel not wrapping

    - by SnOrfus
    I've got a FlowLayoutPanel with properties: Dock = Fill (in a usercontrol) FlowDirection = TopDown WrapContents = false I do it this way so that each item added to the panel gets added to the bottom. The items that I add to this panel are usercontrols which themselves have FlowLayoutPanels on them, however they have the standard behaviour (LeftToRight, WrapContents = true). The problem that I'm having is that the interior usercontrol's FlowLayoutPanel isn't resizing to fill the outer control, but when I set autosizing to true on these controls, then the panel won't wrap its contents - which is a known problem apparently. If it helps visualize what I'm trying to do, it looks like this: ______________________________ | __________________________ | Outer box = exterior flowlayout | |Text____________________| | (TopDown, NoWrap) | | # # # # # # # # # # # #| | | | # # # # | | Interior boxes = usercontrols with text and a | |________________________| | flowlayoutpanel on them | __________________________ | (LeftToRight, Wrap) | |Text____________________| | | | # # # # # # # # # # # #| | # = pictures | | # # | | | |________________________| | |____________________________|

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  • Send custom headers with UIWebView loadRequest

    - by Thomas Clayson
    I want to be able to send some extra headers with my UIWebView loadRequest method. I have tried: NSMutableURLRequest *req = [NSMutableURLRequest requestWithURL:[NSURL URLWithString:@"http://www.reliply.org/tools/requestheaders.php"]]; [req addValue:@"hello" forHTTPHeaderField:@"aHeader"]; [self.theWebView loadRequest:req]; I have also tried subclassing the UIWebView and intercepting the - (BOOL)webView:(UIWebView *)webView shouldStartLoadWithRequest:(NSURLRequest *)request navigationType:(UIWebViewNavigationType)navigationType method. In that method I had a block of code which looked like this: NSMutableURLRequest *newRequest = [request mutableCopy]; for(NSString *key in [customHeaders allKeys]) { [newRequest setValue:[customHeaders valueForKey:key] forHTTPHeaderField:key]; } [self loadRequest:newRequest]; But for some unknown reason it was causing the web view to not load anything (blank frame) and the error message NSURLErrorCancelled (-999) comes up (all known fixes don't fix it for me). So I am at a loss as to what to do. How can I send a custom header along with a UIWebView request? Many thanks!

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  • What version numbering scheme to use?

    - by deamon
    I'm looking for a version numbering scheme that expresses the extent of change, especially compatiblity. Apache APR, for example, use the well known version numbering scheme <major>.<minor>.<patch> example: 4.5.11 Maven suggests a similar but more detailed schema: <major>.<minor>.<patch>-<qualifier>-<build number> example: 4.5.11-RC1-3732 Where is the Maven versioning scheme defined? Are there conventions for qualifier and build number? Probably it is a bad idea to use maven but not to follow the Maven version scheme ... What other version numbering schemes do you know? What scheme would you prefer and why?

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  • Why is J2ee/EJBs a dying Trend?

    - by Taranfx
    I might be wrong on this, but I no longer see heavy Business and Web services being hosted using J2ee technologies (Especially EJBs). Having known the power and scalability of J2ee applications, what is keeping developers/decision makers to restrict themselves to Core Java (POJOs) or even other web technologies like PHP, python. Is it the development time? Is it the Ease of configuration? (I feel this should not be a strong reason with Java EE 6, things are simplified) Of course scripting languages are faster to develop, we cannot ignore the fact that they are inherently not-as-scalable as Java Applications are (using App servers)

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  • Detect block size for quota in Linux

    - by Chen Levy
    The limit placed on disk quota in Linux is counted in blocks. However, I found no reliable way to determine the block size. Tutorials I found refer to block size as 512 bytes, and sometimes as 1024 bytes. I got confused reading a post on LinuxForum.org for what a block size really means. So I tried to find that meaning in the context of quota. I found a "Determine the block size on hard disk filesystem for disk quota" tip on NixCraft, that suggested the command: dumpe2fs /dev/sdXN | grep -i 'Block size' or blockdev --getbsz /dev/sdXN But on my system those commands returned 4096, and when I checked the real quota block size on the same system, I got a block size of 1024 bytes. Is there a scriptable way to determine the quota block size on a device, short of creating a known sized file, and checking it's quota usage?

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  • Which Secure Software Development Practices do you Employ?

    - by Michael Howard-MSFT
    I work on a project known as the Security Development Lifecycle (SDL) project at Microsoft (http://microsoft.com/sdl) - in short it's a set of practices that must be used by product groups before they ship products to help improve security. Over the last couple of years, we have published a great deal of SDL documentation, as customers ask for more information about what we're doing. But what I'd like to know is: 1) What are you doing within your organization to help improve the security of your product? 2) What works? What doesn't work? 3) How did you get management to agree to this work? Thanks.

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  • Can an arbitrary email adress be used in workflow send email activity

    - by Greg McGuffey
    I'm wondering if there is any way to be able to include an arbitrary email address as the To:, From:, CC: or BCC: fields of a send email activity? It appears that they must be contacts in the CRM. I ask this because I have a requirement to cc a known group email (no actual user associated with the email...something like [email protected] it's not a queue at all). I'm concerned that if I create a CRM user for this email, that when I move to production, I'll have to change all the workflows using this email to point to the CRM entity on the production box (assuming GUID is saved with activity). If an arbitrary email isn't possible, any other suggestions? Thanks!

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  • What are the CS fundamentals behind package/dependency management?

    - by Frep D-Oronge
    Often I hear about situations where companies are developing extensable in house software (the dreaded enterprise 'framework') which is supposed to support multiple 'plugins' from diffirent teams. Usually this ends up being a half baked solution that does not really work due to compatibility prolems between addins, or between addins and the framework itself. Usually this means QA have to 'rubber stamp' a global set of versions accross all plugins, or more usually plugins are released and stuff breaks in nasty ways. This problem has been solved before however, for example the package management systems like apt for debian linux. I suspect that the reason it works is that it is built from the start on a known 'Computer Science-y' concept. My question is what is it?

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  • WPF RichTextBox with no width set

    - by Mick
    I have the following XAML code: <Window x:Class="RichText_Wrapping.Window1" xmlns="http://schemas.microsoft.com/winfx/2006/xaml/presentation" xmlns:x="http://schemas.microsoft.com/winfx/2006/xaml" Title="Window1"> <Grid> <RichTextBox Height="100" Margin="2" Name="richTextBox1"> <FlowDocument> <Paragraph> This is a RichTextBox - if you don't specify a width, the text appears in a single column </Paragraph> </FlowDocument> </RichTextBox> </Grid> ... If you create this window in XAML, you can see that when you don't specify a width for the window, it wraps the text in a single column, one letter at a time. Is there something I'm missing? If it's a known deficiency in the control, is there any workaround? Thanks, Michael

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  • gethostname() returns accurate hostname, bind() doesn't like it

    - by user2072848
    Doing a python socket tutorial, entire codebase is as follows import socket as so s = so.socket() host = so.gethostname() port = 12345 s.bind((host, port)) s.listen(5) while True: c, addr = s.accept() print 'Got connection from', addr c.send('Thank you for connecting') c.close() and error message: Traceback (most recent call last): File "server.py", line 13, in <module> s.bind((host, port)) File "/Users/solid*name*/anaconda/lib/python2.7/socket.py", line 224, in meth return getattr(self._sock,name)(*args) socket.gaierror: [Errno 8] nodename nor servname provided, or not known Printing hostname gives me super*name* Which is, in fact, my computer's hostname.

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  • Skip sanitization for videos in html5lib

    - by pug
    I am using a wmd-editor in django, much like this one in which I am typing. I would like to allow the users to embed videos in it. For that I am using the Markdown video extension here. The problem is that I am also sanitizing user input using html5lib sanitization and it doesn't allow object tags which are required to embed the videos. One solution could be to check the input for urls of well-known video sites and skip the sanitization in those cases. Is there a better solution?

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  • What happened to Perl?

    - by llasa
    I will try to keep this as objective as possible. I've been dealing with PHP since 3 years know, I have always known of Perl but never really "dived" into it. So I took a look at some Perl code examples and I thought: Wow, It's like PHP just failed at cloning it. My questions are: What is bad about Perl? What are the disadvantages that made it so extremely unpopular so that it is actually dying right know? Why could PHP take over? What does PHP have (or what did it have in the times of PHP4) that made it rise in popularity compared to Perl? I'm rather young and the questions above are a bit subjective and I think you can only really answer them when you have experienced the rise of PHP along with the fall of Perl. Unless my question before I hope that this one here can be more or less completely answered. There have to be definite disadvantages Perl has compared to PHP that made it fall.

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  • Delphi 2009 RAD Studio - Desktop Selector not Working

    - by Andreas Rejbrand
    In the Delphi 2009 RAD Studio, there is a drop-down combo box where you can select the desktop configuration, i.e. "Default Layout", "Debug Layout, "Classic Undocked", etc. This has stopped working for me, i.e. when I select a desktop, nothing happens. I think I once "customized" the toolbar, by removing the drop-down combo box, and then I added it again, and ever since it has been dead. Is this a known bug? Is there a cure?

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  • 3D World to Local transformation

    - by Bill Kotsias
    Hello. I am having a real headache trying to set a node's local position to match a given world position. I was given a solution but, AFAICS, it only takes into account orientation and position but NOT scaling : node_new_local_position = node_parent.derivedOrientation().Inverse() * ( world_position_to_match - node_parent.derivedPosition() ); The node in question is a child of node_parent; node_parent local and derived properties (orientation, position and scaling) are known, as well as its full matrix transform. All the positions are 3d vectors; the orientation is a quaternion; the full transform is a 4x4 matrix. Could someone please help me to modify the solution to support scaling in the node hierarchy? Many thanks in advance, Bill

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  • Visual Studio msbuild

    - by user62958
    I have a question regarding the commandline options of msbuild. I am currently using msbuild to build projects using the existing solution files. These solution files have references to external dll which have different paths on each machine. I am currently writing a build script and passing the specific path to the project file via the /p: switch of msbuild. My current build line is: msbuild test.sln /p:ReferencePath="c:\abc" /p:ReferencePath="c:\rca" What i have noticed that Reference Path now contains only c:\rca and not c:\abc. this is causing problems for me since, the external dlls lie in two different directorys. I am allowed to keep multiple reference paths via visual studio, but not via the commandline. Is there any known way by which i can do this

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  • synchronizing audio over a network

    - by sharkin
    I'm in startup of designing a client/server audio system which can stream audio arbitrarily over a network. One central server pumps out an audio stream and x number of clients receives the audio data and plays it. So far no magic needed and I have even got this scenario to work with VLC media player out of the box. However, the tricky part seems to be synchronizing the audio playback so that all clients are in audible synch (actual latency can be allowed as long as it is perceived to be in sync by a human listener). My question is if there's any known method or algorithm to use for these types of synchronization problems (video is probably solved the same way). My own initial thoughts centers around synchronizing clocks between physical machines and thereby creating a virtual "main timer" and somehow aligning audio data packets against it. Some products already solving the problem: http://www.sonos.com http://netchorus.com/ Any pointers are most welcome. Thanks. PS: This related question seem to have died long ago.

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  • Hidden features of x86 assembly

    - by Earlz
    I am still a fan of x86 assembly(sorta) and know a lot of developers still using x86 assembly, although by far there are very few features available in assembly, let us list out the most useful and not so well known ones. Of course the question is on the lines of the Hidden Features questions listed below.: Hidden Features of JavaScript Hidden Features of CSS Hidden Features of C# Hidden Features of VB.NET Hidden Features of Java Hidden Features of ASP.NET Hidden Features of Python Hidden Features of TextPad Hidden Features of Eclipse Hidden Features of Classic ASP Please specify one feature per answer. Also, you can specify all bits of the x86 such as 16bit(real mode), 32bit, and 64bit. Please keep it neutral of assembler though. Both Intel and AT&T syntax is welcome but please don't for example demonstrate a useful macro feature for yasm.

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  • Can not clone git repo to server

    - by Classified
    I'm running the same command on 2 different servers. One works, the other doesn't. I'm running git clone https://blah.com:8443/blah.git On server A, it works fine. I get the objects, files, etc. no problems. On server B, I get the following message. git clone https://blah.com:8443/blah.git Cloning into 'blah'... error: Peer certificate cannot be authenticated with known CA certificates while accessing https://blah.com:8443/blah.git/info/refs?service=git-upload-pack fatal: HTTP request failed Does anyone know what this means or what I need to do to get this to work? Thanks in advance for any help you can give me.

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  • SSHing into EC2 instance fails - -v details below!

    - by ming yeow
    Hi folks! I created a new ec2 instance, but i am unable to ssh in with the key i normally use with my other instances. The -v details are below. Thanks! debug1: Host 'dbl01' is known and matches the RSA host key. debug1: Found key in /Users/mingyeow/.ssh/known_hosts:26 debug1: ssh_rsa_verify: signature correct debug1: SSH2_MSG_NEWKEYS sent debug1: expecting SSH2_MSG_NEWKEYS debug1: SSH2_MSG_NEWKEYS received debug1: SSH2_MSG_SERVICE_REQUEST sent debug1: SSH2_MSG_SERVICE_ACCEPT received debug1: Authentications that can continue: publickey debug1: Next authentication method: publickey debug1: Offering public key: /Users/mingyeow/.ssh/id_rsa debug1: Authentications that can continue: publickey debug1: Trying private key: /Users/mingyeow/.ssh/identity debug1: Trying private key: /Users/mingyeow/.ssh/id_dsa debug1: No more authentication methods to try. Permission denied (publickey).

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  • Uploadify refuses to upload WMV, FLV and MP4 files

    - by Jon Winstanley
    The uploadify plugin for JQuery seems very good and works for most file types. However, it allows me to upload all file types apart from the ones I need! Namely .WMV, .FLV and .MP4 I have googled the issue and not found anyonw having such difficulties. I have already tried changing the fileExt parameter and also tried removing it altogether. I have testing in Google Chrome, IE7 and Firefox and none work for thes efile types. Is there a known reason for this behaviour?

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  • Conceal packet loss in PCM stream

    - by ZeroDefect
    I am looking to use 'Packet Loss Concealment' to conceal lost PCM frames in an audio stream. Unfortunately, I cannot find a library that is accessible without all the licensing restrictions and code bloat (...up for some suggestions though). I have located some GPL code written by Steve Underwood for the Asterisk project which implements PLC. There are several limitations; although, as Steve suggests in his code, his algorithm can be applied to different streams with a bit of work. Currently, the code works with 8kHz 16-bit signed mono streams. Variations of the code can be found through a simple search of Google Code Search. My hope is that I can adapt the code to work with other streams. Initially, the goal is to adjust the algorithm for 8+ kHz, 16-bit signed, multichannel audio (all in a C++ environment). Eventually, I'm looking to make the code available under the GPL license in hopes that it could be of benefit to others... Attached is the code below with my efforts. The code includes a main function that will "drop" a number of frames with a given probability. Unfortunately, the code does not quite work as expected. I'm receiving EXC_BAD_ACCESS when running in gdb, but I don't get a trace from gdb when using 'bt' command. Clearly, I'm trampimg on memory some where but not sure exactly where. When I comment out the *amdf_pitch* function, the code runs without crashing... int main (int argc, char *argv[]) { std::ifstream fin("C:\\cc32kHz.pcm"); if(!fin.is_open()) { std::cout << "Failed to open input file" << std::endl; return 1; } std::ofstream fout_repaired("C:\\cc32kHz_repaired.pcm"); if(!fout_repaired.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } std::ofstream fout_lossy("C:\\cc32kHz_lossy.pcm"); if(!fout_lossy.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } audio::PcmConcealer Concealer; Concealer.Init(1, 16, 32000); //Generate random numbers; srand( time(NULL) ); int value = 0; int probability = 5; while(!fin.eof()) { char arr[2]; fin.read(arr, 2); //Generate's random number; value = rand() % 100 + 1; if(value <= probability) { char blank[2] = {0x00, 0x00}; fout_lossy.write(blank, 2); //Fill in data; Concealer.Fill((int16_t *)blank, 1); fout_repaired.write(blank, 2); } else { //Write data to file; fout_repaired.write(arr, 2); fout_lossy.write(arr, 2); Concealer.Receive((int16_t *)arr, 1); } } fin.close(); fout_repaired.close(); fout_lossy.close(); return 0; } PcmConcealer.hpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #ifndef __PCMCONCEALER_HPP__ #define __PCMCONCEALER_HPP__ /** 1. What does it do? The packet loss concealment module provides a suitable synthetic fill-in signal, to minimise the audible effect of lost packets in VoIP applications. It is not tied to any particular codec, and could be used with almost any codec which does not specify its own procedure for packet loss concealment. Where a codec specific concealment procedure exists, the algorithm is usually built around knowledge of the characteristics of the particular codec. It will, therefore, generally give better results for that particular codec than this generic concealer will. 2. How does it work? While good packets are being received, the plc_rx() routine keeps a record of the trailing section of the known speech signal. If a packet is missed, plc_fillin() is called to produce a synthetic replacement for the real speech signal. The average mean difference function (AMDF) is applied to the last known good signal, to determine its effective pitch. Based on this, the last pitch period of signal is saved. Essentially, this cycle of speech will be repeated over and over until the real speech resumes. However, several refinements are needed to obtain smooth pleasant sounding results. - The two ends of the stored cycle of speech will not always fit together smoothly. This can cause roughness, or even clicks, at the joins between cycles. To soften this, the 1/4 pitch period of real speech preceeding the cycle to be repeated is blended with the last 1/4 pitch period of the cycle to be repeated, using an overlap-add (OLA) technique (i.e. in total, the last 5/4 pitch periods of real speech are used). - The start of the synthetic speech will not always fit together smoothly with the tail of real speech passed on before the erasure was identified. Ideally, we would like to modify the last 1/4 pitch period of the real speech, to blend it into the synthetic speech. However, it is too late for that. We could have delayed the real speech a little, but that would require more buffer manipulation, and hurt the efficiency of the no-lost-packets case (which we hope is the dominant case). Instead we use a degenerate form of OLA to modify the start of the synthetic data. The last 1/4 pitch period of real speech is time reversed, and OLA is used to blend it with the first 1/4 pitch period of synthetic speech. The result seems quite acceptable. - As we progress into the erasure, the chances of the synthetic signal being anything like correct steadily fall. Therefore, the volume of the synthesized signal is made to decay linearly, such that after 50ms of missing audio it is reduced to silence. - When real speech resumes, an extra 1/4 pitch period of sythetic speech is blended with the start of the real speech. If the erasure is small, this smoothes the transition. If the erasure is long, and the synthetic signal has faded to zero, the blending softens the start up of the real signal, avoiding a kind of "click" or "pop" effect that might occur with a sudden onset. 3. How do I use it? Before audio is processed, call plc_init() to create an instance of the packet loss concealer. For each received audio packet that is acceptable (i.e. not including those being dropped for being too late) call plc_rx() to record the content of the packet. Note this may modify the packet a little after a period of packet loss, to blend real synthetic data smoothly. When a real packet is not available in time, call plc_fillin() to create a sythetic substitute. That's it! */ /*! Minimum allowed pitch (66 Hz) */ #define PLC_PITCH_MIN(SAMPLE_RATE) ((double)(SAMPLE_RATE) / 66.6) /*! Maximum allowed pitch (200 Hz) */ #define PLC_PITCH_MAX(SAMPLE_RATE) ((SAMPLE_RATE) / 200) /*! Maximum pitch OLA window */ //#define PLC_PITCH_OVERLAP_MAX(SAMPLE_RATE) ((PLC_PITCH_MIN(SAMPLE_RATE)) >> 2) /*! The length over which the AMDF function looks for similarity (20 ms) */ #define CORRELATION_SPAN(SAMPLE_RATE) ((20 * (SAMPLE_RATE)) / 1000) /*! History buffer length. The buffer must also be at leat 1.25 times PLC_PITCH_MIN, but that is much smaller than the buffer needs to be for the pitch assessment. */ //#define PLC_HISTORY_LEN(SAMPLE_RATE) ((CORRELATION_SPAN(SAMPLE_RATE)) + (PLC_PITCH_MIN(SAMPLE_RATE))) namespace audio { typedef struct { /*! Consecutive erased samples */ int missing_samples; /*! Current offset into pitch period */ int pitch_offset; /*! Pitch estimate */ int pitch; /*! Buffer for a cycle of speech */ float *pitchbuf;//[PLC_PITCH_MIN]; /*! History buffer */ short *history;//[PLC_HISTORY_LEN]; /*! Current pointer into the history buffer */ int buf_ptr; } plc_state_t; class PcmConcealer { public: PcmConcealer(); ~PcmConcealer(); void Init(int channels, int bit_depth, int sample_rate); //Process a block of received audio samples. int Receive(short amp[], int frames); //Fill-in a block of missing audio samples. int Fill(short amp[], int frames); void Destroy(); private: int amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames); void save_history(plc_state_t *s, short *buf, int channel_index, int frames); void normalise_history(plc_state_t *s); /** Holds the states of each of the channels **/ std::vector< plc_state_t * > ChannelStates; int plc_pitch_min; int plc_pitch_max; int plc_pitch_overlap_max; int correlation_span; int plc_history_len; int channel_count; int sample_rate; bool Initialized; }; } #endif PcmConcealer.cpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #include "audio/PcmConcealer.hpp" /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */ #define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */ #if !defined(INT16_MAX) #define INT16_MAX (32767) #define INT16_MIN (-32767-1) #endif #ifdef WIN32 inline double rint(double x) { return floor(x + 0.5); } #endif inline short fsaturate(double damp) { if (damp > 32767.0) return INT16_MAX; if (damp < -32768.0) return INT16_MIN; return (short)rint(damp); } namespace audio { PcmConcealer::PcmConcealer() : Initialized(false) { } PcmConcealer::~PcmConcealer() { Destroy(); } void PcmConcealer::Init(int channels, int bit_depth, int sample_rate) { if(Initialized) return; if(channels <= 0 || bit_depth != 16) return; Initialized = true; channel_count = channels; this->sample_rate = sample_rate; ////////////// double min = PLC_PITCH_MIN(sample_rate); int imin = (int)min; double max = PLC_PITCH_MAX(sample_rate); int imax = (int)max; plc_pitch_min = imin; plc_pitch_max = imax; plc_pitch_overlap_max = (plc_pitch_min >> 2); correlation_span = CORRELATION_SPAN(sample_rate); plc_history_len = correlation_span + plc_pitch_min; ////////////// for(int i = 0; i < channel_count; i ++) { plc_state_t *t = new plc_state_t; memset(t, 0, sizeof(plc_state_t)); t->pitchbuf = new float[plc_pitch_min]; t->history = new short[plc_history_len]; ChannelStates.push_back(t); } } void PcmConcealer::Destroy() { if(!Initialized) return; while(ChannelStates.size()) { plc_state_t *s = ChannelStates.at(0); if(s) { if(s->history) delete s->history; if(s->pitchbuf) delete s->pitchbuf; memset(s, 0, sizeof(plc_state_t)); delete s; } ChannelStates.erase(ChannelStates.begin()); } ChannelStates.clear(); Initialized = false; } //Process a block of received audio samples. int PcmConcealer::Receive(short amp[], int frames) { if(!Initialized) return 0; int j = 0; for(int k = 0; k < ChannelStates.size(); k++) { int i; int overlap_len; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples) { /* Although we have a real signal, we need to smooth it to fit well with the synthetic signal we used for the previous block */ /* The start of the real data is overlapped with the next 1/4 cycle of the synthetic data. */ pitch_overlap = s->pitch >> 2; if (pitch_overlap > frames) pitch_overlap = frames; gain = 1.0 - s->missing_samples * ATTENUATION_INCREMENT; if (gain < 0.0) gain = 0.0; new_step = 1.0/pitch_overlap; old_step = new_step*gain; new_weight = new_step; old_weight = (1.0 - new_step)*gain; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[index]); if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->missing_samples = 0; } save_history(s, amp, j, frames); j++; } return frames; } //Fill-in a block of missing audio samples. int PcmConcealer::Fill(short amp[], int frames) { if(!Initialized) return 0; int j =0; for(int k = 0; k < ChannelStates.size(); k++) { short *tmp = new short[plc_pitch_overlap_max]; int i; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; short *orig_amp; int orig_len; orig_amp = amp; orig_len = frames; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples == 0) { // As the gap in real speech starts we need to assess the last known pitch, //and prepare the synthetic data we will use for fill-in normalise_history(s); s->pitch = amdf_pitch(plc_pitch_min, plc_pitch_max, s->history + plc_history_len - correlation_span - plc_pitch_min, j, correlation_span); // We overlap a 1/4 wavelength pitch_overlap = s->pitch >> 2; // Cook up a single cycle of pitch, using a single of the real signal with 1/4 //cycle OLA'ed to make the ends join up nicely // The first 3/4 of the cycle is a simple copy for (i = 0; i < s->pitch - pitch_overlap; i++) s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]; // The last 1/4 of the cycle is overlapped with the end of the previous cycle new_step = 1.0/pitch_overlap; new_weight = new_step; for ( ; i < s->pitch; i++) { s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]*(1.0 - new_weight) + s->history[plc_history_len - 2*s->pitch + i]*new_weight; new_weight += new_step; } // We should now be ready to fill in the gap with repeated, decaying cycles // of what is in pitchbuf // We need to OLA the first 1/4 wavelength of the synthetic data, to smooth // it into the previous real data. To avoid the need to introduce a delay // in the stream, reverse the last 1/4 wavelength, and OLA with that. gain = 1.0; new_step = 1.0/pitch_overlap; old_step = new_step; new_weight = new_step; old_weight = 1.0 - new_step; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->history[plc_history_len - 1 - i] + new_weight * s->pitchbuf[i]); new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->pitch_offset = i; } else { gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT; i = 0; } for ( ; gain > 0.0 && i < frames; i++) { int index = (i * channel_count) + j; amp[index] = s->pitchbuf[s->pitch_offset]*gain; gain -= ATTENUATION_INCREMENT; if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; } for ( ; i < frames; i++) { int index = (i * channel_count) + j; amp[i] = 0; } s->missing_samples += orig_len; save_history(s, amp, j, frames); delete [] tmp; j++; } return frames; } void PcmConcealer::save_history(plc_state_t *s, short *buf, int channel_index, int frames) { if (frames >= plc_history_len) { /* Just keep the last part of the new data, starting at the beginning of the buffer */ //memcpy(s->history, buf + len - plc_history_len, sizeof(short)*plc_history_len); int frames_to_copy = plc_history_len; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + frames - plc_history_len)) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = 0; return; } if (s->buf_ptr + frames > plc_history_len) { /* Wraps around - must break into two sections */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*(plc_history_len - s->buf_ptr)); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = plc_history_len - s->buf_ptr; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } frames -= (plc_history_len - s->buf_ptr); //memcpy(s->history, buf + (plc_history_len - s->buf_ptr), sizeof(short)*len); frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + (plc_history_len - s->buf_ptr))) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = frames; return; } /* Can use just one section */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*len); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } s->buf_ptr += frames; } void PcmConcealer::normalise_history(plc_state_t *s) { short *tmp = new short[plc_history_len]; if (s->buf_ptr == 0) return; memcpy(tmp, s->history, sizeof(short)*s->buf_ptr); memcpy(s->history, s->history + s->buf_ptr, sizeof(short)*(plc_history_len - s->buf_ptr)); memcpy(s->history + plc_history_len - s->buf_ptr, tmp, sizeof(short)*s->buf_ptr); s->buf_ptr = 0; delete [] tmp; } int PcmConcealer::amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames) { int i; int j; int acc; int min_acc; int pitch; pitch = min_pitch; min_acc = INT_MAX; for (i = max_pitch; i <= min_pitch; i++) { acc = 0; for (j = 0; j < frames; j++) { int index1 = (channel_count * (i+j)) + channel_index; int index2 = (channel_count * j) + channel_index; //std::cout << "Index 1: " << index1 << ", Index 2: " << index2 << std::endl; acc += abs(amp[index1] - amp[index2]); } if (acc < min_acc) { min_acc = acc; pitch = i; } } std::cout << "Pitch: " << pitch << std::endl; return pitch; } } P.S. - I must confess that digital audio is not my forte...

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  • Multilangual Unicode rendering in opengl

    - by sum1stolemyname
    Hi Folks, I have to extend an OpenGL-Rendering System to support international characters (especially Hebrew, Arabic and cyrillic). Development Platform is Windows(XP|Vista|7), Alas using Embercardero Delphi 2010. I currently use wglOutLineFont(...) to build my font's display list and glCallLists(length(m_Text), UNSIGNED_SHORT, PWchar(m_Text) ) to render my strings. While this is feasable for Latin-1 Characters, building the full unicode character set in advanced is pretty time-consuming (about 8.5 minutes on my machine), so i am looking for a more efficient solution. I thought about limiting the range from u+0020 - u+077f (latin, greek, cyrillic, arbaic and hebrew) to include just the glyphs i need, but that would just be a solution for my current needs, and will become insufficent once other encoding is needed. On the upside, i do not have to worry about left-to right or right-to left direction as our application can handle this already. I would expect this to be a well-known problem, so i would like to ask if there is any reference material on this on the web, or if you could share some insight on this?

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