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• Big Data – Is Big Data Relevant to me? – Big Data Questionnaires – Guest Post by Vinod Kumar

  - by Pinal Dave

  This guest post is by Vinod Kumar. Vinod Kumar has worked with SQL Server extensively since joining the industry over a decade ago. Working on various versions of SQL Server 7.0, Oracle 7.3 and other database technologies – he now works with the Microsoft Technology Center (MTC) as a Technology Architect. Let us read the blog post in Vinod’s own voice. I think the series from Pinal is a good one for anyone planning to start on Big Data journey from the basics. In my daily customer interactions this buzz of “Big Data” always comes up, I react generally saying – “Sir, do you really have a ‘Big Data’ problem or do you have a big Data problem?” Generally, there is a silence in the air when I ask this question. Data is everywhere in organizations – be it big data, small data, all data and for few it is bad data which is same as no data :). Wow, don’t discount me as someone who opposes “Big Data”, I am a big supporter as much as I am a critic of the abuse of this term by the people. In this post, I wanted to let my mind flow so that you can also think in the direction I want you to see these concepts. In any case, this is not an exhaustive dump of what is in my mind – but you will surely get the drift how I am going to question Big Data terms from customers!!! Is Big Data Relevant to me? Many of my customers talk to me like blank whiteboard with no idea – “why Big Data”. They want to jump into the bandwagon of technology and they want to decipher insights from their unexplored data a.k.a. unstructured data with structured data. So what are these industry scenario’s that come to mind? Here are some of them: Financials Fraud detection: Banks and Credit cards are monitoring your spending habits on real-time basis. Customer Segmentation: applies in every industry from Banking to Retail to Aviation to Utility and others where they deal with end customer who consume their products and services. Customer Sentiment Analysis: Responding to negative brand perception on social or amplify the positive perception. Sales and Marketing Campaign: Understand the impact and get closer to customer delight. Call Center Analysis: attempt to take unstructured voice recordings and analyze them for content and sentiment. Medical Reduce Re-admissions: How to build a proactive follow-up engagements with patients. Patient Monitoring: How to track Inpatient, Out-Patient, Emergency Visits, Intensive Care Units etc. Preventive Care: Disease identification and Risk stratification is a very crucial business function for medical. Claims fraud detection: There is no precise dollars that one can put here, but this is a big thing for the medical field. Retail Customer Sentiment Analysis, Customer Care Centers, Campaign Management. Supply Chain Analysis: Every sensors and RFID data can be tracked for warehouse space optimization. Location based marketing: Based on where a check-in happens retail stores can be optimize their marketing. Telecom Price optimization and Plans, Finding Customer churn, Customer loyalty programs Call Detail Record (CDR) Analysis, Network optimizations, User Location analysis Customer Behavior Analysis Insurance Fraud Detection & Analysis, Pricing based on customer Sentiment Analysis, Loyalty Management Agents Analysis, Customer Value Management This list can go on to other areas like Utility, Manufacturing, Travel, ITES etc. So as you can see, there are obviously interesting use cases for each of these industry verticals. These are just representative list. Where to start? A lot of times I try to quiz customers on a number of dimensions before starting a Big Data conversation. Are you getting the data you need the way you want it and in a timely manner? Can you get in and analyze the data you need? How quickly is IT to respond to your BI Requests? How easily can you get at the data that you need to run your business/department/project? How are you currently measuring your business? Can you get the data you need to react WITHIN THE QUARTER to impact behaviors to meet your numbers or is it always “rear-view mirror?” How are you measuring: The Brand Customer Sentiment Your Competition Your Pricing Your performance Supply Chain Efficiencies Predictive product / service positioning What are your key challenges of driving collaboration across your global business?  What the challenges in innovation? What challenges are you facing in getting more information out of your data? Note: Garbage-in is Garbage-out. Hold good for all reporting / analytics requirements Big Data POCs? A number of customers get into the realm of setting a small team to work on Big Data – well it is a great start from an understanding point of view, but I tend to ask a number of other questions to such customers. Some of these common questions are: To what degree is your advanced analytics (natural language processing, sentiment analysis, predictive analytics and classification) paired with your Big Data’s efforts? Do you have dedicated resources exploring the possibilities of advanced analytics in Big Data for your business line? Do you plan to employ machine learning technology while doing Advanced Analytics? How is Social Media being monitored in your organization? What is your ability to scale in terms of storage and processing power? Do you have a system in place to sort incoming data in near real time by potential value, data quality, and use frequency? Do you use event-driven architecture to manage incoming data? Do you have specialized data services that can accommodate different formats, security, and the management requirements of multiple data sources? Is your organization currently using or considering in-memory analytics? To what degree are you able to correlate data from your Big Data infrastructure with that from your enterprise data warehouse? Have you extended the role of Data Stewards to include ownership of big data components? Do you prioritize data quality based on the source system (that is Facebook/Twitter data has lower quality thresholds than radio frequency identification (RFID) for a tracking system)? Do your retention policies consider the different legal responsibilities for storing Big Data for a specific amount of time? Do Data Scientists work in close collaboration with Data Stewards to ensure data quality? How is access to attributes of Big Data being given out in the organization? Are roles related to Big Data (Advanced Analyst, Data Scientist) clearly defined? How involved is risk management in the Big Data governance process? Is there a set of documented policies regarding Big Data governance? Is there an enforcement mechanism or approach to ensure that policies are followed? Who is the key sponsor for your Big Data governance program? (The CIO is best) Do you have defined policies surrounding the use of social media data for potential employees and customers, as well as the use of customer Geo-location data? How accessible are complex analytic routines to your user base? What is the level of involvement with outside vendors and third parties in regard to the planning and execution of Big Data projects? What programming technologies are utilized by your data warehouse/BI staff when working with Big Data? These are some of the important questions I ask each customer who is actively evaluating Big Data trends for their organizations. These questions give you a sense of direction where to start, what to use, how to secure, how to analyze and more. Sign off Any Big data is analysis is incomplete without a compelling story. The best way to understand this is to watch Hans Rosling – Gapminder (2:17 to 6:06) videos about the third world myths. Don’t get overwhelmed with the Big Data buzz word, the destination to what your data speaks is important. In this blog post, we did not particularly look at any Big Data technologies. This is a set of questionnaire one needs to keep in mind as they embark their journey of Big Data. I did write some of the basics in my blog: Big Data – Big Hype yet Big Opportunity. Do let me know if these questions make sense?  Reference: Pinal Dave (http://blog.sqlauthority.com)Filed under: Big Data, PostADay, SQL, SQL Authority, SQL Query, SQL Server, SQL Tips and Tricks, T SQL

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• The Stub Proto: Not Just For Stub Objects Anymore

  - by user9154181

  One of the great pleasures of programming is to invent something for a narrow purpose, and then to realize that it is a general solution to a broader problem. In hindsight, these things seem perfectly natural and obvious. The stub proto area used to build the core Solaris consolidation has turned out to be one of those things. As discussed in an earlier article, the stub proto area was invented as part of the effort to use stub objects to build the core ON consolidation. Its purpose was merely as a place to hold stub objects. However, we keep finding other uses for it. It turns out that the stub proto should be more properly thought of as an auxiliary place to put things that we would like to put into the proto to help us build the product, but which we do not wish to package or deliver to the end user. Stub objects are one example, but private lint libraries, header files, archives, and relocatable objects, are all examples of things that might profitably go into the stub proto. Without a stub proto, these items were handled in a variety of ad hoc ways: If one part of the workspace needed private header files, libraries, or other such items, it might modify its Makefile to reach up and over to the place in the workspace where those things live and use them from there. There are several problems with this: Each component invents its own approach, meaning that programmers maintaining the system have to invest extra effort to understand what things mean. In the past, this has created makefile ghettos in which only the person who wrote the makefiles feels confident to modify them, while everyone else ignores them. This causes many difficulties and benefits no one. These interdependencies are not obvious to the make, utility, and can lead to races. They are not obvious to the human reader, who may therefore not realize that they exist, and break them. Our policy in ON is not to deliver files into the proto unless those files are intended to be packaged and delivered to the end user. However, sometimes non-shipping files were copied into the proto anyway, causing a different set of problems: It requires a long list of exceptions to silence our normal unused proto item error checking. In the past, we have accidentally shipped files that we did not intend to deliver to the end user. Mixing cruft with valuable items makes it hard to discern which is which. The stub proto area offers a convenient and robust solution. Files needed to build the workspace that are not delivered to the end user can instead be installed into the stub proto. No special exceptions or custom make rules are needed, and the intent is always clear. We are already accessing some private lint libraries and compilation symlinks in this manner. Ultimately, I'd like to see all of the files in the proto that have a packaging exception delivered to the stub proto instead, and for the elimination of all existing special case makefile rules. This would include shared objects, header files, and lint libraries. I don't expect this to happen overnight — it will be a long term case by case project, but the overall trend is clear. The Stub Proto, -z assert_deflib, And The End Of Accidental System Object Linking We recently used the stub proto to solve an annoying build issue that goes back to the earliest days of Solaris: How to ensure that we're linking to the OS bits we're building instead of to those from the running system. The Solaris product is made up of objects and files from a number of different consolidations, each of which is built separately from the others from an independent code base called a gate. The core Solaris OS consolidation is ON, which stands for "Operating System and Networking". You will frequently also see ON called the OSnet. There are consolidations for X11 graphics, the desktop environment, open source utilities, compilers and development tools, and many others. The collection of consolidations that make up Solaris is known as the "Wad Of Stuff", usually referred to simply as the WOS. None of these consolidations is self contained. Even the core ON consolidation has some dependencies on libraries that come from other consolidations. The build server used to build the OSnet must be running a relatively recent version of Solaris, which means that its objects will be very similar to the new ones being built. However, it is necessarily true that the build system objects will always be a little behind, and that incompatible differences may exist. The objects built by the OSnet link to other objects. Some of these dependencies come from the OSnet, while others come from other consolidations. The objects from other consolidations are provided by the standard library directories on the build system (/lib, /usr/lib). The objects from the OSnet itself are supposed to come from the proto areas in the workspace, and not from the build server. In order to achieve this, we make use of the -L command line option to the link-editor. The link-editor finds dependencies by looking in the directories specified by the caller using the -L command line option. If the desired dependency is not found in one of these locations, ld will then fall back to looking at the default locations (/lib, /usr/lib). In order to use OSnet objects from the workspace instead of the system, while still accessing non-OSnet objects from the system, our Makefiles set -L link-editor options that point at the workspace proto areas. In general, this works well and dependencies are found in the right places. However, there have always been failures: Building objects in the wrong order might mean that an OSnet dependency hasn't been built before an object that needs it. If so, the dependency will not be seen in the proto, and the link-editor will silently fall back to the one on the build server. Errors in the makefiles can wipe out the -L options that our top level makefiles establish to cause ld to look at the workspace proto first. In this case, all objects will be found on the build server. These failures were rarely if ever caught. As I mentioned earlier, the objects on the build server are generally quite close to the objects built in the workspace. If they offer compatible linking interfaces, then the objects that link to them will behave properly, and no issue will ever be seen. However, if they do not offer compatible linking interfaces, the failure modes can be puzzling and hard to pin down. Either way, there won't be a compile-time warning or error. The advent of the stub proto eliminated the first type of failure. With stub objects, there is no dependency ordering, and the necessary stub object dependency will always be in place for any OSnet object that needs it. However, makefile errors do still occur, and so, the second form of error was still possible. While working on the stub object project, we realized that the stub proto was also the key to solving the second form of failure caused by makefile errors: Due to the way we set the -L options to point at our workspace proto areas, any valid object from the OSnet should be found via a path specified by -L, and not from the default locations (/lib, /usr/lib). Any OSnet object found via the default locations means that we've linked to the build server, which is an error we'd like to catch. Non-OSnet objects don't exist in the proto areas, and so are found via the default paths. However, if we were to create a symlink in the stub proto pointing at each non-OSnet dependency that we require, then the non-OSnet objects would also be found via the paths specified by -L, and not from the link-editor defaults. Given the above, we should not find any dependency objects from the link-editor defaults. Any dependency found via the link-editor defaults means that we have a Makefile error, and that we are linking to the build server inappropriately. All we need to make use of this fact is a linker option to produce a warning when it happens. Although warnings are nice, we in the OSnet have a zero tolerance policy for build noise. The -z fatal-warnings option that was recently introduced with -z guidance can be used to turn the warnings into fatal build errors, forcing the programmer to fix them. This was too easy to resist. I integrated 7021198 ld option to warn when link accesses a library via default path PSARC/2011/068 ld -z assert-deflib option into snv_161 (February 2011), shortly after the stub proto was introduced into ON. This putback introduced the -z assert-deflib option to the link-editor: -z assert-deflib=[libname] Enables warning messages for libraries specified with the -l command line option that are found by examining the default search paths provided by the link-editor. If a libname value is provided, the default library warning feature is enabled, and the specified library is added to a list of libraries for which no warnings will be issued. Multiple -z assert-deflib options can be specified in order to specify multiple libraries for which warnings should not be issued. The libname value should be the name of the library file, as found by the link-editor, without any path components. For example, the following enables default library warnings, and excludes the standard C library. ld ... -z assert-deflib=libc.so ... -z assert-deflib is a specialized option, primarily of interest in build environments where multiple objects with the same name exist and tight control over the library used is required. If is not intended for general use. Note that the definition of -z assert-deflib allows for exceptions to be specified as arguments to the option. In general, the idea of using a symlink from the stub proto is superior because it does not clutter up the link command with a long list of objects. When building the OSnet, we usually use the plain from of -z deflib, and make symlinks for the non-OSnet dependencies. The exception to this are dependencies supplied by the compiler itself, which are usually found at whatever arbitrary location the compiler happens to be installed at. To handle these special cases, the command line version works better. Following the integration of the link-editor change, I made use of -z assert-deflib in OSnet builds with 7021896 Prevent OSnet from accidentally linking to build system which integrated into snv_162 (March 2011). Turning on -z assert-deflib exposed between 10 and 20 existing errors in our Makefiles, which were all fixed in the same putback. The errors we found in our Makefiles underscore how difficult they can be prevent without an automatic system in place to catch them. Conclusions The stub proto is proving to be a generally useful construct for ON builds that goes beyond serving as a place to hold stub objects. Although invented to hold stub objects, it has already allowed us to simplify a number of previously difficult situations in our makefiles and builds. I expect that we'll find uses for it beyond those described here as we go forward.

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• Actionscript 3: Monitoring the activity level for multiple Microphones doesn't seem to work.

  - by Dave

  For a project I want to show all available webcams and microphones, so that the user can easily select whichever webcam/microphone combination they prefer. I run into an issue with the microphones listing though. Each microphone is listed with an activity animation and it's name. I am able to list all Microphones just fine (using the Microphone.names Array), but it seems like I can only get the activity viewer to work for one microphone. The other microphones show up with '-1' activity, which (as far as I know) is Flex for 'present, but not in use'. When unplugging the microphone that does show activity, the next one (in my case, the mic-in line on my motherboard) shows up with '0' activity (it's not connected, so that makes sense). During my testing I have a total of 3 microphones available, the not-connected onboard mic-in port, and two connected microphones. For testing purposes I use a timer that traces the current microphone activity each 100ms and the graph is also shown. It does not seem to matter what default microphone I set via flash' settings panel. The code I've only attached the revelant code snippets below to make it easier for you to read through them. Please let me know if you prefer the entire code. Main application.mxml Note: cont is a VBox. i is defined before this code snippet. var mics:Array = Microphone.names; for(i=0; i < mics.length; i++){ var mic:settingsMicEntry = new assets.settingsMicEntry; mic.d = {name: mics[i], index: i}; cont.addChild(mic); } assets/settingsMicEntry.mxml timer is defined before this code snippet. the SoundTransform is added to silence local microphone playback. Excluding this code does not solve the problem, sadly (I've tried). display is an MXML Canvas object. mic = Microphone.getMicrophone(d.index); if(mic){ // Temporary: The Microphones' visualizer var bar:Box = new Box(); bar.y = 50; bar.height = 0; bar.width = 66; bar.setStyle("backgroundColor", 0x003300); display.addChild(bar); var tf:SoundTransform = new SoundTransform(0); mic.setLoopBack(true); mic.soundTransform = tf; timer = new Timer(100); timer.addEventListener(TimerEvent.TIMER, function(e:TimerEvent):void{ var h:int = Math.floor((display.height/100)*mic.activityLevel); bar.height = (h>-1) ? h : 0; bar.y = (h>-1) ? display.height-h : display.height; trace('TIMER: '+h+' from '+d.name); }); timer.start(); } I'm pulling my hear out here, so any help is much appreciated! Thanks, -Dave Ps.: Pardon the messiness of the code!

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• Why can't my program display this dialog box, while another program can?

  - by nonoitall

  I'm trying to write a wrapper for Winamp input plugins and have hit a bit of a snag. I'd like my wrapper to be able to display a plugin's configuration dialog, which is (or should be) achieved by calling the plugin's Config(HWND hwndParent) function. For most plugins, this works fine and my program is able to display the plugin's configuration dialog. However, 64th Note (a plugin for playing USF files) is giving me problems. Winamp can display its configuration dialog just fine, but whenever I try to display it from my wrapper, the dialog gets destroyed before it ever shows itself. Thankfully, 64th Note is open source, so I took a look at its innards to try and get an idea of what's going wrong. I've trimmed off the irrelevant bits and am left with this: Config function in the plugin (should show configuration dialog): void Config(HWND hwndParent) { DialogBox(slave, (const char *) IDD_CONFIG_WINDOW, NULL, configDlgProc); } (Slave is the plugin DLL's HINSTANCE handle.) The proc for the dialog is as follows (I have stripped out all the functionality, since it doesn't appear to have an influence on this problem): BOOL CALLBACK configDlgProc(HWND hDlg, UINT uMsg, WPARAM wParam, LPARAM lParam) { return 0; } The template for IDD_CONFIG_WINDOW is as follows: IDD_CONFIG_WINDOW DIALOGEX 0, 0, 269, 149 STYLE DS_SETFONT | DS_MODALFRAME | WS_POPUP | WS_CAPTION | WS_SYSMENU CAPTION "64th Note configuration" FONT 8, "MS Sans Serif", 0, 0, 0x0 BEGIN DEFPUSHBUTTON "OK",IDOK,212,38,50,14 CONTROL "Play Forever",IDC_NOLENGTH,"Button",BS_AUTORADIOBUTTON,7,7,55,8 CONTROL "Always Use Default Length",IDC_SETLEN,"Button",BS_AUTORADIOBUTTON,7,17,101,8 CONTROL "Default Length",IDC_DEFLEN,"Button",BS_AUTORADIOBUTTON,7,29,63,8 EDITTEXT IDC_DEFLENVAL,71,28,38,12,ES_AUTOHSCROLL EDITTEXT IDC_DEFFADEVAL,71,42,38,12,ES_AUTOHSCROLL CONTROL "Detect Silence",IDC_DETSIL,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,7,56,63,8 EDITTEXT IDC_DETSILVAL,71,56,38,12,ES_AUTOHSCROLL CONTROL "Slider2",IDC_PRISLIDER,"msctls_trackbar32",TBS_AUTOTICKS | WS_TABSTOP,74,90,108,11 EDITTEXT IDC_TITLEFMT,7,127,255,15,ES_AUTOHSCROLL CONTROL "Default to file name on missing field",IDC_FNONMISSINGTAG, "Button",BS_AUTOCHECKBOX | WS_TABSTOP,50,114,124,8 CONTROL "Use Recompiler CPU",IDC_RECOMPILER,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,145,7,83,8 CONTROL "Round Frequency",IDC_ROUNDFREQ,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,145,16,73,8 CONTROL "Seek Backwards",IDC_BACKWARDS,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,145,26,70,8 CONTROL "Fast Seek",IDC_FASTSEEK,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,145,35,48,8 CONTROL "RSP Sections",IDC_SECTIONS,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,145,45,60,8 CONTROL "Soft Amplify",IDC_SOFTAMP,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,145,54,53,8 CONTROL "Audio HLE",IDC_AUDIOHLE,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,145,63,50,8 CONTROL "Auto Audio HLE",IDC_AUTOAUDIOHLE,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,145,72,64,8 CONTROL "Display Errors",IDC_DISPERROR,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,145,81,58,8 EDITTEXT IDC_RELVOL,211,104,28,12,ES_AUTOHSCROLL PUSHBUTTON "Cancel",IDCANCEL,212,54,50,14 PUSHBUTTON "Help",IDHELPBUTTON,212,71,50,14 LTEXT "Title format:",IDC_STATIC,7,113,38,8 LTEXT "seconds",IDC_STATIC,112,29,28,8 LTEXT "Default Fade",IDC_STATIC,19,43,42,8 LTEXT "seconds",IDC_STATIC,112,43,28,8 LTEXT "seconds",IDC_STATIC,112,57,28,8 CTEXT "CPU Thread Priority",IDC_STATIC,7,91,63,8 CTEXT "Look ma, I'm data!",IDC_CPUPRI,75,104,108,8 LTEXT "Relative Volume",IDC_STATIC,199,94,52,8 LTEXT "Fade Type",IDC_STATIC,7,75,35,8 COMBOBOX IDC_FADETYPE,45,72,87,74,CBS_DROPDOWNLIST | WS_TABSTOP END Naturally, without any substance in the proc function, the dialog doesn't have any functionality, but it still displays in Winamp when the Config function is invoked. However, it does not appear when I invoke it from my wrapper program. When I monitored the messages sent to the dialog in its proc function, I saw that WM_DESTROY and WM_NCDESTROY were sent within the first few messages, though I have no clue as to why. If I change the Config function so that it displays the plugin's About dialog instead of its configuration dialog, both Winamp and my wrapper will display the About dialog, which suggests that there is something unique to the configuration dialog template that's causing the problem. The modified Config function reads like so: void Config(HWND hwndParent) { DialogBox(slave, (const char *) IDD_ABOUTBOX, NULL, configDlgProc); } The template for the About dialog is as follows: IDD_ABOUTBOX DIALOGEX 0, 0, 152, 151 STYLE DS_SETFONT | DS_MODALFRAME | WS_POPUP | WS_CAPTION | WS_SYSMENU CAPTION "About 64th Note" FONT 8, "MS Sans Serif", 0, 0, 0x1 BEGIN LTEXT "64th Note v1.2 beta 3\nBased on Project 64 1.6 by Zilmar and Jabo\nAudio HLE by Azimer\nPSF concept and tagging by Neill Corlett\nPlayer by hcs, Josh W, dr0\nhttp://hcs64.com/usf",IDC_STATIC,7,94,138,50 CONTROL 110,IDC_STATIC,"Static",SS_BITMAP,26,7,95,86,WS_EX_DLGMODALFRAME END Like I said, my wrapper displays the About dialog just fine, as does Winamp. Why can Winamp display the Config dialog, while my wrapper cannot?

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• Why do I get a segmentation fault while redirecting sys.stdout to Tkinter.Text widget in Python?

  - by Brent Nash

  I'm in the process of building a GUI-based application with Python/Tkinter that builds on top of the existing Python bdb module. In this application, I want to silence all stdout/stderr from the console and redirect it to my GUI. To accomplish this purpose, I've written a specialized Tkinter.Text object (code at the end of the post). The basic idea is that when something is written to sys.stdout, it shows up as a line in the "Text" with the color black. If something is written to sys.stderr, it shows up as a line in the "Text" with the color red. As soon as something is written, the Text always scrolls down to view the most recent line. I'm using Python 2.6.1 at the moment. On Mac OS X 10.5, this seems to work great. I have had zero problems with it. On RedHat Enterprise Linux 5, however, I pretty reliably get a segmentation fault during the run of a script. The segmentation fault doesn't always occur in the same place, but it pretty much always occurs. If I comment out the sys.stdout= and sys.stderr= lines from my code, the segmentation faults seem to go away. I'm sure there are other ways around this that I will probably have to resort to, but can anyone see anything I'm doing blatantly wrong here that could be causing these segmentation faults? It's driving me nuts. Thanks! PS - I realize redirecting sys.stderr to the GUI might not be a great idea, but I still get segmentation faults even when I only redirect sys.stdout and not sys.stderr. I also realize that I'm allowing the Text to grow indefinitely at the moment. class ConsoleText(tk.Text): '''A Tkinter Text widget that provides a scrolling display of console stderr and stdout.''' class IORedirector(object): '''A general class for redirecting I/O to this Text widget.''' def __init__(self,text_area): self.text_area = text_area class StdoutRedirector(IORedirector): '''A class for redirecting stdout to this Text widget.''' def write(self,str): self.text_area.write(str,False) class StderrRedirector(IORedirector): '''A class for redirecting stderr to this Text widget.''' def write(self,str): self.text_area.write(str,True) def __init__(self, master=None, cnf={}, **kw): '''See the __init__ for Tkinter.Text for most of this stuff.''' tk.Text.__init__(self, master, cnf, **kw) self.started = False self.write_lock = threading.Lock() self.tag_configure('STDOUT',background='white',foreground='black') self.tag_configure('STDERR',background='white',foreground='red') self.config(state=tk.DISABLED) def start(self): if self.started: return self.started = True self.original_stdout = sys.stdout self.original_stderr = sys.stderr stdout_redirector = ConsoleText.StdoutRedirector(self) stderr_redirector = ConsoleText.StderrRedirector(self) sys.stdout = stdout_redirector sys.stderr = stderr_redirector def stop(self): if not self.started: return self.started = False sys.stdout = self.original_stdout sys.stderr = self.original_stderr def write(self,val,is_stderr=False): #Fun Fact: The way Tkinter Text objects work is that if they're disabled, #you can't write into them AT ALL (via the GUI or programatically). Since we want them #disabled for the user, we have to set them to NORMAL (a.k.a. ENABLED), write to them, #then set their state back to DISABLED. self.write_lock.acquire() self.config(state=tk.NORMAL) self.insert('end',val,'STDERR' if is_stderr else 'STDOUT') self.see('end') self.config(state=tk.DISABLED) self.write_lock.release()

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• Java algorithm for normalizing audio

  - by Marty Pitt

  I'm trying to normalize an audio file of speech. Specifically, where an audio file contains peaks in volume, I'm trying to level it out, so the quiet sections are louder, and the peaks are quieter. I know very little about audio manipulation, beyond what I've learnt from working on this task. Also, my math is embarrassingly weak. I've done some research, and the Xuggle site provides a sample which shows reducing the volume using the following code: (full version here) @Override public void onAudioSamples(IAudioSamplesEvent event) { // get the raw audio byes and adjust it's value ShortBuffer buffer = event.getAudioSamples().getByteBuffer().asShortBuffer(); for (int i = 0; i < buffer.limit(); ++i) buffer.put(i, (short)(buffer.get(i) * mVolume)); super.onAudioSamples(event); } Here, they modify the bytes in getAudioSamples() by a constant of mVolume. Building on this approach, I've attempted a normalisation modifies the bytes in getAudioSamples() to a normalised value, considering the max/min in the file. (See below for details). I have a simple filter to leave "silence" alone (ie., anything below a value). I'm finding that the output file is very noisy (ie., the quality is seriously degraded). I assume that the error is either in my normalisation algorithim, or the way I manipulate the bytes. However, I'm unsure of where to go next. Here's an abridged version of what I'm currently doing. Step 1: Find peaks in file: Reads the full audio file, and finds this highest and lowest values of buffer.get() for all AudioSamples @Override public void onAudioSamples(IAudioSamplesEvent event) { IAudioSamples audioSamples = event.getAudioSamples(); ShortBuffer buffer = audioSamples.getByteBuffer().asShortBuffer(); short min = Short.MAX_VALUE; short max = Short.MIN_VALUE; for (int i = 0; i < buffer.limit(); ++i) { short value = buffer.get(i); min = (short) Math.min(min, value); max = (short) Math.max(max, value); } // assign of min/max ommitted for brevity. super.onAudioSamples(event); } Step 2: Normalize all values: In a loop similar to step1, replace the buffer with normalized values, calling: buffer.put(i, normalize(buffer.get(i)); public short normalize(short value) { if (isBackgroundNoise(value)) return value; short rawMin = // min from step1 short rawMax = // max from step1 short targetRangeMin = 1000; short targetRangeMax = 8000; int abs = Math.abs(value); double a = (abs - rawMin) * (targetRangeMax - targetRangeMin); double b = (rawMax - rawMin); double result = targetRangeMin + ( a/b ); // Copy the sign of value to result. result = Math.copySign(result,value); return (short) result; } Questions: Is this a valid approach for attempting to normalize an audio file? Is my math in normalize() valid? Why would this cause the file to become noisy, where a similar approach in the demo code doesn't?

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• OpenVPN Client timing out

  - by Austin

  I recently installed OpenVPN on my Ubuntu VPS. Whenenver I try to connect to it, I can establish a connection just fine. However, everything I try to connect to times out. If I try to ping something, it will resolve the IP, but will time out after resolving the IP. (So DNS Server seems to be working correctly) My server.conf has this relevant information (At least I think it's relevant. I'm not sure if you need more or not) # Which local IP address should OpenVPN # listen on? (optional) ;local a.b.c.d # Which TCP/UDP port should OpenVPN listen on? # If you want to run multiple OpenVPN instances # on the same machine, use a different port # number for each one. You will need to # open up this port on your firewall. port 1194 # TCP or UDP server? ;proto tcp proto udp # "dev tun" will create a routed IP tunnel, # "dev tap" will create an ethernet tunnel. # Use "dev tap0" if you are ethernet bridging # and have precreated a tap0 virtual interface # and bridged it with your ethernet interface. # If you want to control access policies # over the VPN, you must create firewall # rules for the the TUN/TAP interface. # On non-Windows systems, you can give # an explicit unit number, such as tun0. # On Windows, use "dev-node" for this. # On most systems, the VPN will not function # unless you partially or fully disable # the firewall for the TUN/TAP interface. ;dev tap dev tun # Windows needs the TAP-Win32 adapter name # from the Network Connections panel if you # have more than one. On XP SP2 or higher, # you may need to selectively disable the # Windows firewall for the TAP adapter. # Non-Windows systems usually don't need this. ;dev-node MyTap # SSL/TLS root certificate (ca), certificate # (cert), and private key (key). Each client # and the server must have their own cert and # key file. The server and all clients will # use the same ca file. # # See the "easy-rsa" directory for a series # of scripts for generating RSA certificates # and private keys. Remember to use # a unique Common Name for the server # and each of the client certificates. # # Any X509 key management system can be used. # OpenVPN can also use a PKCS #12 formatted key file # (see "pkcs12" directive in man page). ca ca.crt cert server.crt key server.key # This file should be kept secret # Diffie hellman parameters. # Generate your own with: # openssl dhparam -out dh1024.pem 1024 # Substitute 2048 for 1024 if you are using # 2048 bit keys. dh dh1024.pem # Configure server mode and supply a VPN subnet # for OpenVPN to draw client addresses from. # The server will take 10.8.0.1 for itself, # the rest will be made available to clients. # Each client will be able to reach the server # on 10.8.0.1. Comment this line out if you are # ethernet bridging. See the man page for more info. server 10.8.0.0 255.255.255.0 # Maintain a record of client <-> virtual IP address # associations in this file. If OpenVPN goes down or # is restarted, reconnecting clients can be assigned # the same virtual IP address from the pool that was # previously assigned. ifconfig-pool-persist ipp.txt # Configure server mode for ethernet bridging. # You must first use your OS's bridging capability # to bridge the TAP interface with the ethernet # NIC interface. Then you must manually set the # IP/netmask on the bridge interface, here we # assume 10.8.0.4/255.255.255.0. Finally we # must set aside an IP range in this subnet # (start=10.8.0.50 end=10.8.0.100) to allocate # to connecting clients. Leave this line commented # out unless you are ethernet bridging. ;server-bridge 10.8.0.4 255.255.255.0 10.8.0.50 10.8.0.100 # Configure server mode for ethernet bridging # using a DHCP-proxy, where clients talk # to the OpenVPN server-side DHCP server # to receive their IP address allocation # and DNS server addresses. You must first use # your OS's bridging capability to bridge the TAP # interface with the ethernet NIC interface. # Note: this mode only works on clients (such as # Windows), where the client-side TAP adapter is # bound to a DHCP client. ;server-bridge # Push routes to the client to allow it # to reach other private subnets behind # the server. Remember that these # private subnets will also need # to know to route the OpenVPN client # address pool (10.8.0.0/255.255.255.0) # back to the OpenVPN server. ;push "route 192.168.10.0 255.255.255.0" ;push "route 192.168.20.0 255.255.255.0" # To assign specific IP addresses to specific # clients or if a connecting client has a private # subnet behind it that should also have VPN access, # use the subdirectory "ccd" for client-specific # configuration files (see man page for more info). # EXAMPLE: Suppose the client # having the certificate common name "Thelonious" # also has a small subnet behind his connecting # machine, such as 192.168.40.128/255.255.255.248. # First, uncomment out these lines: ;client-config-dir ccd ;route 192.168.40.128 255.255.255.248 # Then create a file ccd/Thelonious with this line: # iroute 192.168.40.128 255.255.255.248 # This will allow Thelonious' private subnet to # access the VPN. This example will only work # if you are routing, not bridging, i.e. you are # using "dev tun" and "server" directives. # EXAMPLE: Suppose you want to give # Thelonious a fixed VPN IP address of 10.9.0.1. # First uncomment out these lines: ;client-config-dir ccd ;route 10.9.0.0 255.255.255.252 # Then add this line to ccd/Thelonious: # ifconfig-push 10.9.0.1 10.9.0.2 # Suppose that you want to enable different # firewall access policies for different groups # of clients. There are two methods: # (1) Run multiple OpenVPN daemons, one for each # group, and firewall the TUN/TAP interface # for each group/daemon appropriately. # (2) (Advanced) Create a script to dynamically # modify the firewall in response to access # from different clients. See man # page for more info on learn-address script. ;learn-address ./script # If enabled, this directive will configure # all clients to redirect their default # network gateway through the VPN, causing # all IP traffic such as web browsing and # and DNS lookups to go through the VPN # (The OpenVPN server machine may need to NAT # or bridge the TUN/TAP interface to the internet # in order for this to work properly). push "redirect-gateway def1 bypass-dhcp" push "dhcp-option DNS 8.8.8.8" # Certain Windows-specific network settings # can be pushed to clients, such as DNS # or WINS server addresses. CAVEAT: # http://openvpn.net/faq.html#dhcpcaveats # The addresses below refer to the public # DNS servers provided by opendns.com. ;push "dhcp-option DNS 8.8.8.8" push "dhcp-option DNS 8.8.4.4" # Uncomment this directive to allow different # clients to be able to "see" each other. # By default, clients will only see the server. # To force clients to only see the server, you # will also need to appropriately firewall the # server's TUN/TAP interface. ;client-to-client # Uncomment this directive if multiple clients # might connect with the same certificate/key # files or common names. This is recommended # only for testing purposes. For production use, # each client should have its own certificate/key # pair. # # IF YOU HAVE NOT GENERATED INDIVIDUAL # CERTIFICATE/KEY PAIRS FOR EACH CLIENT, # EACH HAVING ITS OWN UNIQUE "COMMON NAME", # UNCOMMENT THIS LINE OUT. ;duplicate-cn # The keepalive directive causes ping-like # messages to be sent back and forth over # the link so that each side knows when # the other side has gone down. # Ping every 10 seconds, assume that remote # peer is down if no ping received during # a 120 second time period. keepalive 10 120 # For extra security beyond that provided # by SSL/TLS, create an "HMAC firewall" # to help block DoS attacks and UDP port flooding. # # Generate with: # openvpn --genkey --secret ta.key # # The server and each client must have # a copy of this key. # The second parameter should be '0' # on the server and '1' on the clients. ;tls-auth ta.key 0 # This file is secret # Select a cryptographic cipher. # This config item must be copied to # the client config file as well. ;cipher BF-CBC # Blowfish (default) ;cipher AES-128-CBC # AES ;cipher DES-EDE3-CBC # Triple-DES # Enable compression on the VPN link. # If you enable it here, you must also # enable it in the client config file. comp-lzo # The maximum number of concurrently connected # clients we want to allow. ;max-clients 100 # It's a good idea to reduce the OpenVPN # daemon's privileges after initialization. # # You can uncomment this out on # non-Windows systems. ;user nobody ;group nogroup # The persist options will try to avoid # accessing certain resources on restart # that may no longer be accessible because # of the privilege downgrade. persist-key persist-tun # Output a short status file showing # current connections, truncated # and rewritten every minute. status openvpn-status.log # By default, log messages will go to the syslog (or # on Windows, if running as a service, they will go to # the "\Program Files\OpenVPN\log" directory). # Use log or log-append to override this default. # "log" will truncate the log file on OpenVPN startup, # while "log-append" will append to it. Use one # or the other (but not both). ;log openvpn.log ;log-append openvpn.log # Set the appropriate level of log # file verbosity. # # 0 is silent, except for fatal errors # 4 is reasonable for general usage # 5 and 6 can help to debug connection problems # 9 is extremely verbose verb 3 # Silence repeating messages. At most 20 # sequential messages of the same message # category will be output to the log. ;mute 20 I've tried on multiple computers by the way. The same result on all of them. What could be wrong? Thanks in advance, and if you need other information I'll gladly post it. Information for new comments root@vps:~# iptables -L -n -v Chain INPUT (policy ACCEPT 862K packets, 51M bytes) pkts bytes target prot opt in out source destination Chain FORWARD (policy ACCEPT 3 packets, 382 bytes) pkts bytes target prot opt in out source destination 0 0 ACCEPT all -- * * 0.0.0.0/0 0.0.0.0/0 state RELATED,ESTABLISHED 4641 298K ACCEPT all -- * * 10.8.0.0/24 0.0.0.0/0 0 0 REJECT all -- * * 0.0.0.0/0 0.0.0.0/0 reject-with icmp-port-unreachable Chain OUTPUT (policy ACCEPT 1671K packets, 2378M bytes) pkts bytes target prot opt in out source destination And root@vps:~# iptables -t nat -L -n -v Chain PREROUTING (policy ACCEPT 17937 packets, 2013K bytes) pkts bytes target prot opt in out source destination Chain POSTROUTING (policy ACCEPT 8975 packets, 562K bytes) pkts bytes target prot opt in out source destination 1579 103K SNAT all -- * * 10.8.0.0/24 0.0.0.0/0 to:SERVERIP Chain OUTPUT (policy ACCEPT 8972 packets, 562K bytes) pkts bytes target prot opt in out source destination

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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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