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  • Reading and writing C++ vector to a file

    - by JB
    For some graphics work I need to read in a large amount of data as quickly as possible and would ideally like to directly read and write the data structures to disk. Basically I have a load of 3d models in various file formats which take too long to load so I want to write them out in their "prepared" format as a cache that will load much faster on subsequent runs of the program. Is it safe to do it like this? My worries are around directly reading into the data of the vector? I've removed error checking, hard coded 4 as the size of the int and so on so that i can give a short working example, I know it's bad code, my question really is if it is safe in c++ to read a whole array of structures directly into a vector like this? I believe it to be so, but c++ has so many traps and undefined behavour when you start going low level and dealing directly with raw memory like this. I realise that number formats and sizes may change across platforms and compilers but this will only even be read and written by the same compiler program to cache data that may be needed on a later run of the same program. #include <fstream> #include <vector> using namespace std; struct Vertex { float x, y, z; }; typedef vector<Vertex> VertexList; int main() { // Create a list for testing VertexList list; Vertex v1 = {1.0f, 2.0f, 3.0f}; list.push_back(v1); Vertex v2 = {2.0f, 100.0f, 3.0f}; list.push_back(v2); Vertex v3 = {3.0f, 200.0f, 3.0f}; list.push_back(v3); Vertex v4 = {4.0f, 300.0f, 3.0f}; list.push_back(v4); // Write out a list to a disk file ofstream os ("data.dat", ios::binary); int size1 = list.size(); os.write((const char*)&size1, 4); os.write((const char*)&list[0], size1 * sizeof(Vertex)); os.close(); // Read it back in VertexList list2; ifstream is("data.dat", ios::binary); int size2; is.read((char*)&size2, 4); list2.resize(size2); // Is it safe to read a whole array of structures directly into the vector? is.read((char*)&list2[0], size2 * sizeof(Vertex)); }

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  • Function Point Analysis -- a seriously overestimating technique?

    - by kizzx2
    I know questions about FPA has been asked numerous times before, but this time I'm taking a more analytical angle at it, backed up with data. 1. First, some data This question is based on a tutorial. He had a "Sample Count" section where he demonstrated it step by step. You can see some screenshots of his sample application here. In the end, he calculated the unadjusted FP to be 99. There is another article on InformIT with industry data on typical hour/FP. It ranges from 2 hours/FP to 27.4 hours/FP. Let's try to stick with 2 for the moment (since SO readers are probably the more efficient crowd :p). 2. Reality check!? Now just check out the screenshots again. Do a little math here 99 * 2 = 198 hours 198 hours / 40 hours per week = 5 weeks Seriously? That sample application is going to take 5 weeks to implement? Is it just my feeling that it wouldn't take any decent programmer longer than one week (I"m not even saying weekend) to have it completed? Now let's try estimating the cost of the project. We'll use New York's minimum wage at the moment (Wikipedia), which is $7.25 198 * 7.25 = $1435.5 From what I could see from the screenshots, this application is a small excel-improvement app. I could have bought MS Office Pro for 200 bucks which gives me greater interoperability (.xls files) and flexibility (spreadsheets). (For the record, that same Web site has another article discussing productivity. It seems like they typically use 4.2 hours/FP, which gives us even more shocking stats: 99 * 4.2 = 415 hours = 10 weeks = almost 3 whopping months! 415 hours * $7.25 = $3000 zomg (That's even assuming that all our poor coders get the minimum wage!) 3. Am I missing something here? Right now, I could come up with several possible explanation: FPA is really only suited for bigger projects (1000+ FPs) so it becomes extremely inaccurate at smaller scale. The hours/FP metric fluctuates abruptly from team to team, project to project. For a small project like this, we could have used something like 0.5 hour/FP or something. (Now this kind of makes the whole estimation thing pointless, unless my firm does the same type of projects for several years with the same team, not really common.) From my experience with several software metrics, Function Point is really not a lightweight metric. If the hour/FP thing fluctuates so much, then what's the point, maybe I could have gone with User Story Points which is a lot faster to get and arguably almost as uncertain. What would be the FP experts' answers to this?

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  • Calculate pixels within a polygon

    - by DoomStone
    In an assignment for school do we need to do some image recognizing, where we have to find a path for a robot. So far have we been able to find all the polygons in the image, but now we need to generate a pixel map, that be used for an astar algorithm later. We have found a way to do this, show below, but the problem is that is very slow, as we go though each pixel and test if it is inside the polygon. So my question is, are there a way that we can generate this pixel map faster? We have a list of cordinates for the polygon private List<IntPoint> hull; The fuction "getMap" is called to get the pixel map public Point[] getMap() { List<Point> points = new List<Point>(); lock (hull) { Rectangle rect = getRectangle(); for (int x = rect.X; x <= rect.X + rect.Width; x++) { for (int y = rect.Y; y <= rect.Y + rect.Height; y++) { if (inPoly(x, y)) points.Add(new Point(x, y)); } } } return points.ToArray(); } Get Rectangle is used to limit the search, se we don't have to go thoug the whole image public Rectangle getRectangle() { int x = -1, y = -1, width = -1, height = -1; foreach (IntPoint item in hull) { if (item.X < x || x == -1) x = item.X; if (item.Y < y || y == -1) y = item.Y; if (item.X > width || width == -1) width = item.X; if (item.Y > height || height == -1) height = item.Y; } return new Rectangle(x, y, width-x, height-y); } And atlast this is how we check to see if a pixel is inside the polygon public bool inPoly(int x, int y) { int i, j = hull.Count - 1; bool oddNodes = false; for (i = 0; i < hull.Count; i++) { if (hull[i].Y < y && hull[j].Y >= y || hull[j].Y < y && hull[i].Y >= y) { try { if (hull[i].X + (y - hull[i].X) / (hull[j].X - hull[i].X) * (hull[j].X - hull[i].X) < x) { oddNodes = !oddNodes; } } catch (DivideByZeroException e) { if (0 < x) { oddNodes = !oddNodes; } } } j = i; } return oddNodes; }

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  • Estimating the boundary of arbitrarily distributed data

    - by Dave
    I have two dimensional discrete spatial data. I would like to make an approximation of the spatial boundaries of this data so that I can produce a plot with another dataset on top of it. Ideally, this would be an ordered set of (x,y) points that matplotlib can plot with the plt.Polygon() patch. My initial attempt is very inelegant: I place a fine grid over the data, and where data is found in a cell, a square matplotlib patch is created of that cell. The resolution of the boundary thus depends on the sampling frequency of the grid. Here is an example, where the grey region are the cells containing data, black where no data exists. OK, problem solved - why am I still here? Well.... I'd like a more "elegant" solution, or at least one that is faster (ie. I don't want to get on with "real" work, I'd like to have some fun with this!). The best way I can think of is a ray-tracing approach - eg: from xmin to xmax, at y=ymin, check if data boundary crossed in intervals dx y=ymin+dy, do 1 do 1-2, but now sample in y An alternative is defining a centre, and sampling in r-theta space - ie radial spokes in dtheta increments. Both would produce a set of (x,y) points, but then how do I order/link neighbouring points them to create the boundary? A nearest neighbour approach is not appropriate as, for example (to borrow from Geography), an isthmus (think of Panama connecting N&S America) could then close off and isolate regions. This also might not deal very well with the holes seen in the data, which I would like to represent as a different plt.Polygon. The solution perhaps comes from solving an area maximisation problem. For a set of points defining the data limits, what is the maximum contiguous area contained within those points To form the enclosed area, what are the neighbouring points for the nth point? How will the holes be treated in this scheme - is this erring into topology now? Apologies, much of this is me thinking out loud. I'd be grateful for some hints, suggestions or solutions. I suspect this is an oft-studied problem with many solution techniques, but I'm looking for something simple to code and quick to run... I guess everyone is, really! Cheers, David

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  • how to develop a program to minimize errors in human transcription of hand written surveys

    - by Alex. S.
    I need to develop custom software to do surveys. Questions may be of multiple choice, or free text in a very few cases. I was asked to design a subsystem to check if there is any error in the manual data entry for the multiple choices part. We're trying to speed up the user data entry process and to minimize human input differences between digital forms and the original questionnaires. The surveys are filled with handwritten marks and text by human interviewers, so it's possible to find hard to read marks, or also the user could accidentally select a different value in some question, and we would like to avoid that. The software must include some automatic control to detect possible typing differences. Each answer of the multiple choice questions has the same probability of being selected. This question has two parts: The GUI. The most simple thing I have in mind is to implement the most usable design of the questions display: use of large and readable fonts and space generously the choices. Is there something else? For faster input, I would like to use drop down lists (favoring keyboard over mouse). Given the questions are grouped in sections, I would like to show the answers selected for the questions of that section, but this could slow down the process. Any other ideas? The error checking subsystem. What else can I do to minimize or to check human typos in the multiple choice questions? Is this a solvable problem? is there some statistical methodology to check values that were entered by the users are the same from the hand filled forms? For example, let's suppose the survey has 5 questions, and each has 4 options. Let's say I have n survey forms filled in paper by interviewers, and they're ready to be entered in the software, then how to minimize the accidental differences that can have the manual transcription of the n surveys, without having to double check everything in the 5 questions of the n surveys? My first suggestion is that at the end of the processing of all the hand filled forms, the software could choose some forms randomly to make a double check of the responses in a few instances, but on what criteria can I make this selection? This validation would be enough to cover everything in a significant way? The actual survey is nation level and it has 56 pages with over 200 questions in total, so it will be a lot of hand written pages by many people, and the intention is to reduce the likelihood of errors and to optimize speed in the data entry process. The surveys must filled in paper first, given the complications of taking laptops or handhelds with the interviewers.

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  • approximating log10[x^k0 + k1]

    - by Yale Zhang
    Greetings. I'm trying to approximate the function Log10[x^k0 + k1], where .21 < k0 < 21, 0 < k1 < ~2000, and x is integer < 2^14. k0 & k1 are constant. For practical purposes, you can assume k0 = 2.12, k1 = 2660. The desired accuracy is 5*10^-4 relative error. This function is virtually identical to Log[x], except near 0, where it differs a lot. I already have came up with a SIMD implementation that is ~1.15x faster than a simple lookup table, but would like to improve it if possible, which I think is very hard due to lack of efficient instructions. My SIMD implementation uses 16bit fixed point arithmetic to evaluate a 3rd degree polynomial (I use least squares fit). The polynomial uses different coefficients for different input ranges. There are 8 ranges, and range i spans (64)2^i to (64)2^(i + 1). The rational behind this is the derivatives of Log[x] drop rapidly with x, meaning a polynomial will fit it more accurately since polynomials are an exact fit for functions that have a derivative of 0 beyond a certain order. SIMD table lookups are done very efficiently with a single _mm_shuffle_epi8(). I use SSE's float to int conversion to get the exponent and significand used for the fixed point approximation. I also software pipelined the loop to get ~1.25x speedup, so further code optimizations are probably unlikely. What I'm asking is if there's a more efficient approximation at a higher level? For example: Can this function be decomposed into functions with a limited domain like log2((2^x) * significand) = x + log2(significand) hence eliminating the need to deal with different ranges (table lookups). The main problem I think is adding the k1 term kills all those nice log properties that we know and love, making it not possible. Or is it? Iterative method? don't think so because the Newton method for log[x] is already a complicated expression Exploiting locality of neighboring pixels? - if the range of the 8 inputs fall in the same approximation range, then I can look up a single coefficient, instead of looking up separate coefficients for each element. Thus, I can use this as a fast common case, and use a slower, general code path when it isn't. But for my data, the range needs to be ~2000 before this property hold 70% of the time, which doesn't seem to make this method competitive. Please, give me some opinion, especially if you're an applied mathematician, even if you say it can't be done. Thanks.

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  • web service filling gridview awfully slow, as is paging/sorting

    - by nat
    Hi I am making a page which calls a web service to fill a gridview this is returning alot of data, and is horribly slow. i ran the svcutil.exe on the wsdl page and it generated me the class and config so i have a load of strongly typed objects coming back from each request to the many service functions. i am then using LINQ to loop around the objects grabbing the necessary information as i go, but for each row in the grid i need to loop around an object, and grab another list of objects (from the same request) and loop around each of them.. 1 to many parent object child one.. all of this then gets dropped into a custom datatable a row at a time.. hope that makes sense.... im not sure there is any way to speed up the initial load. but surely i should be able to page/sort alot faster than it is doing. as at the moment, it appears to be taking as long to page/sort as it is to load initially. i thought if when i first loaded i put the datasource of the grid in the session, that i could whip it out of the session to deal with paging/sorting and the like. basically it is doing the below protected void Page_Load(object sender, EventArgs e) { //init the datatable //grab the filter vars (if there are any) WebServiceObj WS = WSClient.Method(args); //fill the datatable (around and around we go) foreach (ParentObject po in WS.ReturnedObj) { var COs = from ChildObject c in WS.AnotherReturnedObj where c.whatever.equals(...) ...etc foreach(ChildObject c in COs){ myDataTable.Rows.Add(tlo.this, tlo.that, c.thisthing, c.thatthing, etc......); } } grdListing.DataSource = myDataTable; Session["dt"] = myDataTable; grdListing.DataBind(); } protected void Listing_PageIndexChanging(object sender, GridViewPageEventArgs e) { grdListing.PageIndex = e.NewPageIndex; grdListing.DataSource = Session["dt"] as DataTable; grdListing.DataBind(); } protected void Listing_Sorting(object sender, GridViewSortEventArgs e) { DataTable dt = Session["dt"] as DataTable; DataView dv = new DataView(dt); string sortDirection = " ASC"; if (e.SortDirection == SortDirection.Descending) sortDirection = " DESC"; dv.Sort = e.SortExpression + sortDirection; grdListing.DataSource = dv.ToTable(); grdListing.DataBind(); } am i doing this totally wrongly? or is the slowness just coming from the amount of data being bound in/return from the Web Service.. there are maybe 15 columns(ish) and a whole load of rows.. with more being added to the data the webservice is querying from all the time any suggestions / tips happily received thanks

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  • B-trees, databases, sequential inputs, and speed.

    - by IanC
    I know from experience that b-trees have awful performance when data is added to them sequentially (regardless of the direction). However, when data is added randomly, best performance is obtained. This is easy to demonstrate with the likes of an RB-Tree. Sequential writes cause a maximum number of tree balances to be performed. I know very few databases use binary trees, but rather used n-order balanced trees. I logically assume they suffer a similar fate to binary trees when it comes to sequential inputs. This sparked my curiosity. If this is so, then one could deduce that writing sequential IDs (such as in IDENTITY(1,1)) would cause multiple re-balances of the tree to occur. I have seen many posts argue against GUIDs as "these will cause random writes". I never use GUIDs, but it struck me that this "bad" point was in fact a good point. So I decided to test it. Here is my code: SET ANSI_NULLS ON GO SET QUOTED_IDENTIFIER ON GO CREATE TABLE [dbo].[T1]( [ID] [int] NOT NULL CONSTRAINT [T1_1] PRIMARY KEY CLUSTERED ([ID] ASC) ) GO CREATE TABLE [dbo].[T2]( [ID] [uniqueidentifier] NOT NULL CONSTRAINT [T2_1] PRIMARY KEY CLUSTERED ([ID] ASC) ) GO declare @i int, @t1 datetime, @t2 datetime, @t3 datetime, @c char(300) set @t1 = GETDATE() set @i = 1 while @i < 2000 begin insert into T2 values (NEWID(), @c) set @i = @i + 1 end set @t2 = GETDATE() WAITFOR delay '0:0:10' set @t3 = GETDATE() set @i = 1 while @i < 2000 begin insert into T1 values (@i, @c) set @i = @i + 1 end select DATEDIFF(ms, @t1, @t2) AS [Int], DATEDIFF(ms, @t3, getdate()) AS [GUID] drop table T1 drop table T2 Note that I am not subtracting any time for the creation of the GUID nor for the considerably extra size of the row. The results on my machine were as follows: Int: 17,340 ms GUID: 6,746 ms This means that in this test, random inserts of 16 bytes was almost 3 times faster than sequential inserts of 4 bytes. Would anyone like to comment on this? Ps. I get that this isn't a question. It's an invite to discussion, and that is relevant to learning optimum programming.

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  • How to get a physics engine like Nape working?

    - by Glacius
    Introduction: I think Nape is a relatively new engine so some of you may not know it. It's supposedly faster than box2d and I like that there is decent documentation. Here's the site: http://code.google.com/p/nape/ I'm relatively new to programming. I am decent at AS3's basic functionality, but every time I try to implement some kind of engine or framework I can't even seem to get it to work. With Nape I feel I got a little further than before but I still got stuck. My problem: I'm using Adobe CS5, I managed to import the SWC file like described here. Next I tried to copy the source of one of the demo's like this one and get it to work but I keep getting errors. I made a new class file, copied the demo source to it, and tried to add it to the stage. My stage code basically looks like this: import flash.Boot; // these 2 lines are as described in the tutorial new Boot(); var demo = new Main(); // these 2 are me guessing what I'm supposed to do addChild(demo); Well, it seems the source code is not even being recognized by flash as a valid class file. I tried editing it, but even if I get it recognized (give a package name and add curly brackets) but I still get a bunch of errors. Is it psuedo code or something? What is going on? My goal: I can imagine I'm going about this the wrong way. So let me explain what I'm trying to achieve. I basically want to learn how to use the engine by starting from a simple basic example that I can edit and mess around with. If I can't even get a working example then I'm unable to learn anything. Preferably I don't want to start using something like FlashDevelop (as I'd have to learn how to use the program) but if it can't be helped then I can give it a try. Thank you.

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  • How to cache queries in EJB and return result efficient (performance POV)

    - by Maxym
    I use JBoss EJB 3.0 implementation (JBoss 4.2.3 server) At the beginning I created native query all the time using construction like Query query = entityManager.createNativeQuery("select * from _table_"); Of couse it is not that efficient, I performed some tests and found out that it really takes a lot of time... Then I found a better way to deal with it, to use annotation to define native queries: @NamedNativeQuery( name = "fetchData", value = "select * from _table_", resultClass=Entity.class ) and then just use it Query query = entityManager.createNamedQuery("fetchData"); the performance of code line above is two times better than where I started from, but still not that good as I expected... then I found that I can switch to Hibernate annotation for NamedNativeQuery (anyway, JBoss's implementation of EJB is based on Hibernate), and add one more thing: @NamedNativeQuery( name = "fetchData2", value = "select * from _table_", resultClass=Entity.class, readOnly=true) readOnly - marks whether the results are fetched in read-only mode or not. It sounds good, because at least in this case of mine I don't need to update data, I wanna just fetch it for report. When I started server to measure performance I noticed that query without readOnly=true (by default it is false) returns result with each iteration better and better, and at the same time another one (fetchData2) works like "stable" and with time difference between them is shorter and shorter, and after 5 iterations speed of both was almost the same... The questions are: 1) is there any other way to speed query using up? Seems that named queries should be prepared once, but I can't say it... In fact if to create query once and then just use it it would be better from performance point of view, but it is problematic to cache this object, because after creating query I can set parameters (when I use ":variable" in query), and it changes query object (isn't it?). well, is here any way to cache them? Or named query is the best option I can use? 2) any other approaches how to make results retrieveng faster. I mean, for instance I don't need those Entities to be attached, I won't update them, all I need is just fetch collection of data. Maybe readOnly is the only available way, so I can't speed it up, but who knows :) P.S. I don't ask about DB performance, all I need now is how not to create query all the time, so use it efficient, and to "allow" EJB to do less job with the same result concerning data returning.

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  • Fixed strptime exception with thread lock, but slows down the program

    - by eWizardII
    I have the following code, which when is running inside of a thread (the full code is here - https://github.com/eWizardII/homobabel/blob/master/lovebird.py) for null in range(0,1): while True: try: with open('C:/Twitter/tweets/user_0_' + str(self.id) + '.json', mode='w') as f: f.write('[') threadLock.acquire() for i, seed in enumerate(Cursor(api.user_timeline,screen_name=self.ip).items(200)): if i>0: f.write(", ") f.write("%s" % (json.dumps(dict(sc=seed.author.statuses_count)))) j = j + 1 threadLock.release() f.write("]") except tweepy.TweepError, e: with open('C:/Twitter/tweets/user_0_' + str(self.id) + '.json', mode='a') as f: f.write("]") print "ERROR on " + str(self.ip) + " Reason: ", e with open('C:/Twitter/errors_0.txt', mode='a') as a_file: new_ii = "ERROR on " + str(self.ip) + " Reason: " + str(e) + "\n" a_file.write(new_ii) break Now without the thread lock I generate the following error: Exception in thread Thread-117: Traceback (most recent call last): File "C:\Python27\lib\threading.py", line 530, in __bootstrap_inner self.run() File "C:/Twitter/homobabel/lovebird.py", line 62, in run for i, seed in enumerate(Cursor(api.user_timeline,screen_name=self.ip).items(200)): File "build\bdist.win-amd64\egg\tweepy\cursor.py", line 110, in next self.current_page = self.page_iterator.next() File "build\bdist.win-amd64\egg\tweepy\cursor.py", line 85, in next items = self.method(page=self.current_page, *self.args, **self.kargs) File "build\bdist.win-amd64\egg\tweepy\binder.py", line 196, in _call return method.execute() File "build\bdist.win-amd64\egg\tweepy\binder.py", line 182, in execute result = self.api.parser.parse(self, resp.read()) File "build\bdist.win-amd64\egg\tweepy\parsers.py", line 75, in parse result = model.parse_list(method.api, json) File "build\bdist.win-amd64\egg\tweepy\models.py", line 38, in parse_list results.append(cls.parse(api, obj)) File "build\bdist.win-amd64\egg\tweepy\models.py", line 49, in parse user = User.parse(api, v) File "build\bdist.win-amd64\egg\tweepy\models.py", line 86, in parse setattr(user, k, parse_datetime(v)) File "build\bdist.win-amd64\egg\tweepy\utils.py", line 17, in parse_datetime date = datetime(*(time.strptime(string, '%a %b %d %H:%M:%S +0000 %Y')[0:6])) File "C:\Python27\lib\_strptime.py", line 454, in _strptime_time return _strptime(data_string, format)[0] File "C:\Python27\lib\_strptime.py", line 300, in _strptime _TimeRE_cache = TimeRE() File "C:\Python27\lib\_strptime.py", line 188, in __init__ self.locale_time = LocaleTime() File "C:\Python27\lib\_strptime.py", line 77, in __init__ raise ValueError("locale changed during initialization") ValueError: locale changed during initialization The problem is with thread lock on, each thread runs itself serially basically, and it takes way to long for each loop to run for there to be any advantage to having a thread anymore. So if there isn't a way to get rid of the thread lock, is there a way to have it run the for loop inside of the try statement faster?

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  • compressed archive with quick access to individual file

    - by eric.frederich
    I need to come up with a file format for new application I am writing. This file will need to hold a bunch other text files which are mostly text but can be other formats as well. Naturally, a compressed tar file seems to fit the bill. The problem is that I want to be able to retrieve some data from the file very quickly and getting just a particular file from a tar.gz file seems to take longer than it should. I am assumeing that this is because it has to decompress the entire file even though I just want one. When I have just a regular uncompressed tar file I can get that data real quick. Lets say the file I need quickly is called data.dat For example the command... tar -x data.dat -zf myfile.tar.gz ... is what takes a lot longer than I'd like. MP3 files have id3 data and jpeg files have exif data that can be read in quickly without opening the entire file. I would like my data.dat file to be available in a similar way. I was thinking that I could leave it uncompressed and seperate from the rest of the files in myfile.tar.gz I could then create a tar file of data.dat and myfile.tar.gz and then hopefully that data would be able to be retrieved faster because it is at the head of outer tar file and is uncompressed. Does this sound right?... putting a compressed tar inside of a tar file? Basically, my need is to have an archive type of file with quick access to one particular file. Tar does this just fine, but I'd also like to have that data compressed and as soon as I do that, I no longer have quick access. Are there other archive formats that will give me that quick access I need? As a side note, this application will be written in Python. If the solution calls for a re-invention of the wheel with my own binary format I am familiar with C and would have no problem writing the Python module in C. Idealy I'd just use tar, dd, cat, gzip, etc though. Thanks, ~Eric

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  • Modify audio pitch of recorded clip (m4v)

    - by devcube
    I'm writing an app in which I'm trying to change the pitch of the audio when I'm recording a movie (.m4v). Or by modifying the audio pitch of the movie afterwards. I want the end result to be a movie (.m4v) that has the original length (i.e. same visual as original) but with modified sound pitch, e.g. a "chipmunk voice". A realtime conversion is to prefer if possible. I've read alot about changing audio pitch in iOS but most examples focus on playback, i.e. playing the sound with a different pitch. In my app I'm recording a movie (.m4v / AVFileTypeQuickTimeMovie) and saving it using standard AVAssetWriter. When saving the movie I have access to the following elements where I've tried to manipulate the audio (e.g. modify the pitch): audio buffer (CMSampleBufferRef) audio input writer (AVAssetWriterAudioInput) audio input writer options (e.g. AVNumberOfChannelsKey, AVSampleRateKey, AVChannelLayoutKey) asset writer (AVAssetWriter) I've tried to hook into the above objects to modify the audio pitch, but without success. I've also tried with Dirac as described here: Real Time Pitch Change In iPhone Using Dirac And OpenAL with AL_PITCH as described here: Piping output from OpenAL into a buffer And the "BASS" library from un4seen: Change Pitch/Tempo In Realtime I haven't found success with any of the above libs, most likely because I don't really know how to use them, and where to hook them into the audio saving code. There seems to be alot of librarys that have similar effects but focuses on playback or custom recording code. I want to manipulate the audio stream I've already got (AVAssetWriterAudioInput) or modify the saved movie clip (.m4v). I want the video to be unmodifed visually, i.e. played at the same speed. But I want the audio to go faster (like a chipmunk) or slower (like a ... monster? :)). Do you have any suggestions how I can modify the pitch in either real time (when recording the movie) or afterwards by converting the entire movie (.m4v file)? Should I look further into Dirac, OpenAL, SoundTouch, BASS or some other library? I want to be able to share the movie to others with modified audio, that's the reason I can't rely on modifying the pitch for playback only. Any help is appreciated, thanks!

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  • thread management in nbody code of cuda-sdk

    - by xnov
    When I read the nbody code in Cuda-SDK, I went through some lines in the code and I found that it is a little bit different than their paper in GPUGems3 "Fast N-Body Simulation with CUDA". My questions are: First, why the blockIdx.x is still involved in loading memory from global to share memory as written in the following code? for (int tile = blockIdx.y; tile < numTiles + blockIdx.y; tile++) { sharedPos[threadIdx.x+blockDim.x*threadIdx.y] = multithreadBodies ? positions[WRAP(blockIdx.x + q * tile + threadIdx.y, gridDim.x) * p + threadIdx.x] : //this line positions[WRAP(blockIdx.x + tile, gridDim.x) * p + threadIdx.x]; //this line __syncthreads(); // This is the "tile_calculation" function from the GPUG3 article. acc = gravitation(bodyPos, acc); __syncthreads(); } isn't it supposed to be like this according to paper? I wonder why sharedPos[threadIdx.x+blockDim.x*threadIdx.y] = multithreadBodies ? positions[WRAP(q * tile + threadIdx.y, gridDim.x) * p + threadIdx.x] : positions[WRAP(tile, gridDim.x) * p + threadIdx.x]; Second, in the multiple threads per body why the threadIdx.x is still involved? Isn't it supposed to be a fix value or not involving at all because the sum only due to threadIdx.y if (multithreadBodies) { SX_SUM(threadIdx.x, threadIdx.y).x = acc.x; //this line SX_SUM(threadIdx.x, threadIdx.y).y = acc.y; //this line SX_SUM(threadIdx.x, threadIdx.y).z = acc.z; //this line __syncthreads(); // Save the result in global memory for the integration step if (threadIdx.y == 0) { for (int i = 1; i < blockDim.y; i++) { acc.x += SX_SUM(threadIdx.x,i).x; //this line acc.y += SX_SUM(threadIdx.x,i).y; //this line acc.z += SX_SUM(threadIdx.x,i).z; //this line } } } Can anyone explain this to me? Is it some kind of optimization for faster code?

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  • Hidden divs for "lazy javascript" loading? Possible security/other issues?

    - by xyld
    I'm curious about people's opinion's and thoughts about this situation. The reason I'd like to lazy load javascript is because of performance. Loading javascript at the end of the body reduces the browser blocking and ends up with much faster page loads. But there is some automation I'm using to generate the html (django specifically). This automation has the convenience of allowing forms to be built with "Widgets" that output content it needs to render the entire widget (extra javascript, css, ...). The problem is that the widget wants to output javascript immediately into the middle of the document, but I want to ensure all javascript loads at the end of the body. When the following widget is added to a form, you can see it renders some <script>...</script> tags: class AutoCompleteTagInput(forms.TextInput): class Media: css = { 'all': ('css/jquery.autocomplete.css', ) } js = ( 'js/jquery.bgiframe.js', 'js/jquery.ajaxQueue.js', 'js/jquery.autocomplete.js', ) def render(self, name, value, attrs=None): output = super(AutoCompleteTagInput, self).render(name, value, attrs) page_tags = Tag.objects.usage_for_model(DataSet) tag_list = simplejson.dumps([tag.name for tag in page_tags], ensure_ascii=False) return mark_safe(u'''<script type="text/javascript"> jQuery("#id_%s").autocomplete(%s, { width: 150, max: 10, highlight: false, scroll: true, scrollHeight: 100, matchContains: true, autoFill: true }); </script>''' % (name, tag_list,)) + output What I'm proposing is that if someone uses a <div class=".lazy-js">...</div> with some css (.lazy-js { display: none; }) and some javascript (jQuery('.lazy-js').each(function(index) { eval(jQuery(this).text()); }), you can effectively force all javascript to load at the end of page load: class AutoCompleteTagInput(forms.TextInput): class Media: css = { 'all': ('css/jquery.autocomplete.css', ) } js = ( 'js/jquery.bgiframe.js', 'js/jquery.ajaxQueue.js', 'js/jquery.autocomplete.js', ) def render(self, name, value, attrs=None): output = super(AutoCompleteTagInput, self).render(name, value, attrs) page_tags = Tag.objects.usage_for_model(DataSet) tag_list = simplejson.dumps([tag.name for tag in page_tags], ensure_ascii=False) return mark_safe(u'''<div class="lazy-js"> jQuery("#id_%s").autocomplete(%s, { width: 150, max: 10, highlight: false, scroll: true, scrollHeight: 100, matchContains: true, autoFill: true }); </div>''' % (name, tag_list,)) + output Nevermind all the details of my specific implementation (the specific media involved), I'm looking for a consensus on whether the method of using lazy-loaded javascript through hidden a hidden tags can pose issues whether security or other related? One of the most convenient parts about this is that it follows the DRY principle rather well IMO because you don't need to hack up a specific lazy-load for each instance in the page. It just "works". UPDATE: I'm not sure if django has the ability to queue things (via fancy template inheritance or something?) to be output just before the end of the </body>?

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  • WPF: Improving Performance for Running on Older PCs

    - by Phil Sandler
    So, I'm building a WPF app and did a test deployment today, and found that it performed pretty poorly. I was surprised, as we are really not doing much in the way of visual effects or animations. I deployed on two machines: the fastest and the slowest that will need to run the application (the slowest PC has an Intel Celeron 1.80GHz with 2GB RAM). The application ran pretty well on the faster machine, but was choppy on the slower machine. And when I say "choppy", I mean the cursor jumped even just passing it over any open window of the app that had focus. I opened the Task Manager Performance window, and could see that the CPU usage jumped whenever the app had focus and the cursor was moving over it. If I gave focus to another (e.g. Excel), the CPU usage went back down after a second. This happened on both machines, but the choppiness was only noticeable on the slower machine. I had very limited time to tinker on the deployment machines, so didn't do a lot of detailed testing. The app runs fine on my development machine, but I also see the CPU spiking up to 10% there, just running the cursor over the window. I downloaded the WPF performance tool from MS and have been tinkering with it (on my dev machine). The docs say this about the "Frame Rate" metric in the Perforator tool: For applications without animation, this value should be near 0. The app is not doing any heavy animation, but the frame rate stays near 50 when the cursor is over any window. The screens I tested on have column headers in a grid that "highlight" and buttons that change color and appearance when scrolled over. Even moving the mouse on blank areas of the windows cause the same Frame rate and CPU usage (doesn't seem to be related to these minor animations). (Also, I am unable to figure out how to get anything but the two default tools--Perforator and Visual Profiler--installed into the WPF performance tool. That is probably a separate question). I also have Redgate's profiling tool, but I'm not sure if that can shed any light on rendering performance. So, I realize this is not an easy thing to troubleshoot without specifics or sample code (which I can't post). My questions are: What are some general things to look for (or avoid) in the code to improve performance? What steps can I take using the WPF performance tool to narrow down the problem? Is the PC spec listed above (Intel Celeron 1.80GHz with 2GB RAM) too slow to be running even vanilla WPF applications?

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  • CSS selectors : should I minimise my use of the class attribute in the HTML or optimise the speed

    - by Laurent Bourgault-Roy
    As I was working on a small website, I decided to use the PageSpeed extension to check if their was some improvement I could do to make the site load faster. However I was quite surprise when it told me that my use of CSS selector was "inefficient". I was always told that you should keep the usage of the class attribute in the HTML to a minimum, but if I understand correctly what PageSpeed tell me, it's much more efficient for the browser to match directly against a class name. It make sense to me, but it also mean that I need to put more CSS classes in my HTML. It also make my .css file a little harder to read. I usually tend to mark my CSS like this : #mainContent p.productDescription em.priceTag { ... } Which make it easy to read : I know this will affect the main content and that it affect something in a paragraph tag (so I wont start to put all sort of layout code in it) that describe a product and its something that need emphasis. However it seem I should rewrite it as .priceTag { ... } Which remove all context information about the style. And if I want to use differently formatted price tag (for example, one in a list on the sidebar and one in a paragraph), I need to use something like that .paragraphPriceTag { ... } .listPriceTag { ... } Which really annoy me since I seem to duplicate the semantic of the HTML in my classes. And that mean I can't put common style in an unqualified .priceTag { ... } and thus I need to replicate the style in both CSS rule, making it harder to make change. (Altough for that I could use multiple class selector, but IE6 dont support them) I believe making code harder to read for the sake of speed has never been really considered a very good practice . Except where it is critical, of course. This is why people use PHP/Ruby/C# etc. instead of C/assembly to code their site. It's easier to write and debug. So I was wondering if I should stick with few CSS classes and complex selector or if I should go the optimisation route and remove my fancy CSS selectors for the sake of speed? Does PageSpeed make over the top recommandation? On most modern computer, will it even make a difference?

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  • how to deal with the position in a c# stream

    - by CapsicumDreams
    The (entire) documentation for the position property on a stream says: When overridden in a derived class, gets or sets the position within the current stream. The Position property does not keep track of the number of bytes from the stream that have been consumed, skipped, or both. That's it. OK, so we're fairly clear on what it doesn't tell us, but I'd really like to know what it in fact does stand for. What is 'the position' for? Why would we want to alter or read it? If we change it - what happens? In a pratical example, I have a a stream that periodically gets written to, and I have a thread that attempts to read from it (ideally ASAP). From reading many SO issues, I reset the position field to zero to start my reading. Once this is done: Does this affect where the writer to this stream is going to attempt to put the data? Do I need to keep track of the last write position myself? (ie if I set the position to zero to read, does the writer begin to overwrite everything from the first byte?) If so, do I need a semaphore/lock around this 'position' field (subclassing, perhaps?) due to my two threads accessing it? If I don't handle this property, does the writer just overflow the buffer? Perhaps I don't understand the Stream itself - I'm regarding it as a FIFO pipe: shove data in at one end, and suck it out at the other. If it's not like this, then do I have to keep copying the data past my last read (ie from position 0x84 on) back to the start of my buffer? I've seriously tried to research all of this for quite some time - but I'm new to .NET. Perhaps the Streams have a long, proud (undocumented) history that everyone else implicitly understands. But for a newcomer, it's like reading the manual to your car, and finding out: The accelerator pedal affects the volume of fuel and air sent to the fuel injectors. It does not affect the volume of the entertainment system, or the air pressure in any of the tires, if fitted. Technically true, but seriously, what we want to know is that if we mash it to the floor you go faster..

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  • How to handle very frequent updates to a Lucene index

    - by fsm
    I am trying to prototype an indexing/search application which uses very volatile indexing data sources (forums, social networks etc), here are some of the performance requirements, Very fast turn-around time (by this I mean that any new data (such as a new message on a forum) should be available in the search results very soon (less than a minute)) I need to discard old documents on a fairly regular basis to ensure that the search results are not dated. Last but not least, the search application needs to be responsive. (latency on the order of 100 milliseconds, and should support at least 10 qps) All of the requirements I have currently can be met w/o using Lucene (and that would let me satisfy all 1,2 and 3), but I am anticipating other requirements in the future (like search relevance etc) which Lucene makes easier to implement. However, since Lucene is designed for use cases far more complex than the one I'm currently working on, I'm having a hard time satisfying my performance requirements. Here are some questions, a. I read that the optimize() method in the IndexWriter class is expensive, and should not be used by applications that do frequent updates, what are the alternatives? b. In order to do incremental updates, I need to keep committing new data, and also keep refreshing the index reader to make sure it has the new data available. These are going to affect 1 and 3 above. Should I try duplicate indices? What are some common approaches to solving this problem? c. I know that Lucene provides a delete method, which lets you delete all documents that match a certain query, in my case, I need to delete all documents which are older than a certain age, now one option is to add a date field to every document and use that to delete documents later. Is it possible to do range queries on document ids (I can create my own id field since I think that the one created by lucene keeps changing) to delete documents? Is it any faster than comparing dates represented as strings? I know these are very open questions, so I am not looking for a detailed answer, I will try to treat all of your answers as suggestions and use them to inform my design. Thanks! Please let me know if you need any other information.

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  • What are the pros and cons of using manual list iteration vs recursion through fail

    - by magus
    I come up against this all the time, and I'm never sure which way to attack it. Below are two methods for processing some season facts. What I'm trying to work out is whether to use method 1 or 2, and what are the pros and cons of each, especially large amounts of facts. methodone seems wasteful since the facts are available, why bother building a list of them (especially a large list). This must have memory implications too if the list is large enough ? And it doesn't take advantage of Prolog's natural backtracking feature. methodtwo takes advantage of backtracking to do the recursion for me, and I would guess would be much more memory efficient, but is it good programming practice generally to do this? It's arguably uglier to follow, and might there be any other side effects? One problem I can see is that each time fail is called, we lose the ability to pass anything back to the calling predicate, eg. if it was methodtwo(SeasonResults), since we continually fail the predicate on purpose. So methodtwo would need to assert facts to store state. Presumably(?) method 2 would be faster as it has no (large) list processing to do? I could imagine that if I had a list, then methodone would be the way to go.. or is that always true? Might it make sense in any conditions to assert the list to facts using methodone then process them using method two? Complete madness? But then again, I read that asserting facts is a very 'expensive' business, so list handling might be the way to go, even for large lists? Any thoughts? Or is it sometimes better to use one and not the other, depending on (what) situation? eg. for memory optimisation, use method 2, including asserting facts and, for speed use method 1? season(spring). season(summer). season(autumn). season(winter). % Season handling showseason(Season) :- atom_length(Season, LenSeason), write('Season Length is '), write(LenSeason), nl. % ------------------------------------------------------------- % Method 1 - Findall facts/iterate through the list and process each %-------------------------------------------------------------- % Iterate manually through a season list lenseason([]). lenseason([Season|MoreSeasons]) :- showseason(Season), lenseason(MoreSeasons). % Findall to build a list then iterate until all done methodone :- findall(Season, season(Season), AllSeasons), lenseason(AllSeasons), write('Done'). % ------------------------------------------------------------- % Method 2 - Use fail to force recursion %-------------------------------------------------------------- methodtwo :- % Get one season and show it season(Season), showseason(Season), % Force prolog to backtrack to find another season fail. % No more seasons, we have finished methodtwo :- write('Done').

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  • Generics vs Object performance

    - by Risho
    I'm doing practice problems from MCTS Exam 70-536 Microsft .Net Framework Application Dev Foundation, and one of the problems is to create two classes, one generic, one object type that both perform the same thing; in which a loop uses the class and iterated over thousand times. And using the timer, time the performance of both. There was another post at C# generics question that seeks the same questoion but nonone replied. Basically if in my code I run the generic class first it takes loger to process. If I run the object class first than the object class takes longer to process. The whole idea was to prove that generics perform faster. I used the original users code to save me some time. I didn't particularly see anything wrong with the code and was puzzled by the outcome. Can some one explain why the unusual results? Thanks, Risho Here is the code: class Program { class Object_Sample { public Object_Sample() { Console.WriteLine("Object_Sample Class"); } public long getTicks() { return DateTime.Now.Ticks; } public void display(Object a) { Console.WriteLine("{0}", a); } } class Generics_Samle<T> { public Generics_Samle() { Console.WriteLine("Generics_Sample Class"); } public long getTicks() { return DateTime.Now.Ticks; } public void display(T a) { Console.WriteLine("{0}", a); } } static void Main(string[] args) { long ticks_initial, ticks_final, diff_generics, diff_object; Object_Sample OS = new Object_Sample(); Generics_Samle<int> GS = new Generics_Samle<int>(); //Generic Sample ticks_initial = 0; ticks_final = 0; ticks_initial = GS.getTicks(); for (int i = 0; i < 50000; i++) { GS.display(i); } ticks_final = GS.getTicks(); diff_generics = ticks_final - ticks_initial; //Object Sample ticks_initial = 0; ticks_final = 0; ticks_initial = OS.getTicks(); for (int j = 0; j < 50000; j++) { OS.display(j); } ticks_final = OS.getTicks(); diff_object = ticks_final - ticks_initial; Console.WriteLine("\nPerformance of Generics {0}", diff_generics); Console.WriteLine("Performance of Object {0}", diff_object); Console.ReadKey(); } }

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  • casting doubles to integers in order to gain speed

    - by antirez
    Hello all, in Redis (http://code.google.com/p/redis) there are scores associated to elements, in order to take this elements sorted. This scores are doubles, even if many users actually sort by integers (for instance unix times). When the database is saved we need to write this doubles ok disk. This is what is used currently: snprintf((char*)buf+1,sizeof(buf)-1,"%.17g",val); Additionally infinity and not-a-number conditions are checked in order to also represent this in the final database file. Unfortunately converting a double into the string representation is pretty slow. While we have a function in Redis that converts an integer into a string representation in a much faster way. So my idea was to check if a double could be casted into an integer without lost of data, and then using the function to turn the integer into a string if this is true. For this to provide a good speedup of course the test for integer "equivalence" must be fast. So I used a trick that is probably undefined behavior but that worked very well in practice. Something like that: double x = ... some value ... if (x == (double)((long long)x)) use_the_fast_integer_function((long long)x); else use_the_slow_snprintf(x); In my reasoning the double casting above converts the double into a long, and then back into an integer. If the range fits, and there is no decimal part, the number will survive the conversion and will be exactly the same as the initial number. As I wanted to make sure this will not break things in some system, I joined #c on freenode and I got a lot of insults ;) So I'm now trying here. Is there a standard way to do what I'm trying to do without going outside ANSI C? Otherwise, is the above code supposed to work in all the Posix systems that currently Redis targets? That is, archs where Linux / Mac OS X / *BSD / Solaris are running nowaday? What I can add in order to make the code saner is an explicit check for the range of the double before trying the cast at all. Thank you for any help.

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  • Finding open contiguous blocks of time for every day of a month, fast

    - by Chris
    I am working on a booking availability system for a group of several venues, and am having a hard time generating the availability of time blocks for days in a given month. This is happening server-side in PHP, but the concept itself is language agnostic -- I could be doing this in JS or anything else. Given a venue_id, month, and year (6/2012 for example), I have a list of all events occurring in that range at that venue, represented as unix timestamps start and end. This data comes from the database. I need to establish what, if any, contiguous block of time of a minimum length (different per venue) exist on each day. For example, on 6/1 I have an event between 2:00pm and 7:00pm. The minimum time is 5 hours, so there's a block open there from 9am - 2pm and another between 7pm and 12pm. This would continue for the 2nd, 3rd, etc... every day of June. Some (most) of the days have nothing happening at all, some have 1 - 3 events. The solution I came up with works, but it also takes waaaay too long to generate the data. Basically, I loop every day of the month and create an array of timestamps for each 15 minutes of that day. Then, I loop the time spans of events from that day by 15 minutes, marking any "taken" timeslot as false. Remaining, I have an array that contains timestamp of free time vs. taken time: //one day's array after processing through loops (not real timestamps) array( 12345678=>12345678, // <--- avail 12345878=>12345878, 12346078=>12346078, 12346278=>false, // <--- not avail 12346478=>false, 12346678=>false, 12346878=>false, 12347078=>12347078, // <--- avail 12347278=>12347278 ) Now I would need to loop THIS array to find continuous time blocks, then check to see if they are long enough (each venue has a minimum), and if so then establish the descriptive text for their start and end (i.e. 9am - 2pm). WHEW! By the time all this looping is done, the user has grown bored and wandered off to Youtube to watch videos of puppies; it takes ages to so examine 30 or so days. Is there a faster way to solve this issue? To summarize the problem, given time ranges t1 and t2 on day d, how can I determine the remaining time left in d that is longer than the minimum time block m. This data is assembled on demand via AJAX as the user moves between calendar months. Results are cached per-page-load, so if the user goes to July a second time, the data that was generated the first time would be reused. Any other details that would help, let me know. Edit Per request, the database structure (or the part that is relevant here) *events* id (bigint) title (varchar) *event_times* id (bigint) event_id (bigint) venue_id (bigint) start (bigint) end (bigint) *venues* id (bigint) name (varchar) min_block (int) min_start (varchar) max_start (varchar)

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  • How to setup Lucene/Solr for a B2B web app?

    - by Bill Paetzke
    Given: 1 database per client (business customer) 5000 clients Clients have between 2 to 2000 users (avg is ~100 users/client) 100k to 10 million records per database Users need to search those records often (it's the best way to navigate their data) Possibly relevant info: Several new clients each week (any time during business hours) Multiple web servers and database servers (users can login via any web server) Let's stay agnostic of language or sql brand, since Lucene (and Solr) have a breadth of support For Example: Joel Spolsky said in Podcast #11 that his hosted web app product, FogBugz On-Demand, uses Lucene. He has thousands of on-demand clients. And each client gets their own database. They use an index per client and store it in the client's database. I'm not sure on the details. And I'm not sure if this is a serious mod to Lucene. The Question: How would you setup Lucene search so that each client can only search within its database? How would you setup the index(es)? Where do you store the index(es)? Would you need to add a filter to all search queries? If a client cancelled, how would you delete their (part of the) index? (this may be trivial--not sure yet) Possible Solutions: Make an index for each client (database) Pro: Search is faster (than one-index-for-all method). Indices are relative to the size of the client's data. Con: I'm not sure what this entails, nor do I know if this is beyond Lucene's scope. Have a single, gigantic index with a database_name field. Always include database_name as a filter. Pro: Not sure. Maybe good for tech support or billing dept to search all databases for info. Con: Search is slower (than index-per-client method). Flawed security if query filter removed. One last thing: I would also accept an answer that uses Solr (the extension of Lucene). Perhaps it's better suited for this problem. Not sure.

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  • c++ std::ostringstream vs std::string::append

    - by NickSoft
    In all examples that use some kind of buffering I see they use stream instead of string. How is std::ostringstream and << operator different than using string.append. Which one is faster and which one uses less resourses (memory). One difference I know is that you can output different types into output stream (like integer) rather than the limited types that string::append accepts. Here is an example: std::ostringstream os; os << "Content-Type: " << contentType << ";charset=" << charset << "\r\n"; std::string header = os.str(); vs std::string header("Content-Type: "); header.append(contentType); header.append(";charset="); header.append(charset); header.append("\r\n"); Obviously using stream is shorter, but I think append returns reference to the string so it can be written like this: std::string header("Content-Type: "); header.append(contentType) .append(";charset=") .append(charset) .append("\r\n"); And with output stream you can do: std::string content; ... os << "Content-Length: " << content.length() << "\r\n"; But what about memory usage and speed? Especially when used in a big loop. Update: To be more clear the question is: Which one should I use and why? Is there situations when one is preferred or the other? For performance and memory ... well I think benchmark is the only way since every implementation could be different. Update 2: Well I don't get clear idea what should I use from the answers which means that any of them will do the job, plus vector. Cubbi did nice benchmark with the addition of Dietmar Kühl that the biggest difference is construction of those objects. If you are looking for an answer you should check that too. I'll wait a bit more for other answers (look previous update) and if I don't get one I think I'll accept Tolga's answer because his suggestion to use vector is already done before which means vector should be less resource hungry.

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