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  • PHP: Redirect to the same page, changing $_GET.

    - by Jonathan
    Hi, I have this PHP piece of code that gets $_GET['id'] (a number) and do some stuff with this. When its finished I need to increase that number ($_GET['id']) and redirect to the same page but with the new number (also using $_GET['id']). I am doing something like this: $ID = $_GET['id']; // Some stuff here // and then: $newID = $ID++; header('Location: http://localhost/something/something.php?id='.$newID); exit; The problem here is that the browser stop me from doing it and I get this error from the browser (Firefox) : "The page isn't redirecting properly. Firefox has detected that the server is redirecting the request for this address in a way that will never complete." Some help here please!

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  • How to redirect non-www site to www site?

    - by Mohan Ahire
    I have created one wordpress site. As I write domain name without www then it opens correctaly but as I write www. in url it's not showing the site. Please help me.. Thank you in advance. I have edited my question and added the following part : Following is my file : Where I have to put code provided by you ? I tried it before the "Begin Wordpress" line but still not working. plugable.php -FrontPage- IndexIgnore .htaccess /.?? *~ *# /HEADER */README* */_vti* order deny,allow deny from all allow from all order deny,allow deny from all AuthName example.com AuthUserFile /home/example/public_html/_vti_pvt/service.pwd AuthGroupFile /home/example/public_html/_vti_pvt/service.grp BEGIN WordPress RewriteEngine On RewriteBase / RewriteRule ^index.php$ - [L] RewriteCond %{REQUEST_FILENAME} !-f RewriteCond %{REQUEST_FILENAME} !-d RewriteRule . /index.php [L] END WordPress

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  • Is it good practise to use meta refresh tags for redirects instead of header() function in php?

    - by Kent
    I have to use redirects a lot in my scripts, for example after a user logs in I need to redirect them to the admin area, etc. But I find it inconvenient to always have to have the header function at the very top. So if I use the meta refresh tags for my redirects, is that something that would be frowned upon according to best practices or is it acceptable? function redirect($location) { echo "<meta http-equiv='refresh' content='0; url=$location' />"; }

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  • can't read from stream until child exits?

    - by BobTurbo
    OK I have a program that creates two pipes - forks - the child's stdin and stdout are redirected to one end of each pipe - the parent is connected to the other ends of the pipes and tries to read the stream associated with the child's output and print it to the screen (and I will also make it write to the input of the child eventually). The problem is, when the parent tries to fgets the child's output stream, it just stalls and waits until the child dies to fgets and then print the output. If the child doesn't exit, it just waits forever. What is going on? I thought that maybe fgets would block until SOMETHING was in the stream, but not block all the way until the child gives up its file descriptors. Here is the code: #include <stdlib.h> #include <stdio.h> #include <string.h> #include <sys/types.h> #include <unistd.h> int main(int argc, char *argv[]) { FILE* fpin; FILE* fpout; int input_fd[2]; int output_fd[2]; pid_t pid; int status; char input[100]; char output[100]; char *args[] = {"/somepath/someprogram", NULL}; fgets(input, 100, stdin); // the user inputs the program name to exec pipe(input_fd); pipe(output_fd); pid = fork(); if (pid == 0) { close(input_fd[1]); close(output_fd[0]); dup2(input_fd[0], 0); dup2(output_fd[1], 1); input[strlen(input)-1] = '\0'; execvp(input, args); } else { close(input_fd[0]); close(output_fd[1]); fpin = fdopen(input_fd[1], "w"); fpout = fdopen(output_fd[0], "r"); while(!feof(fpout)) { fgets(output, 100, fpout); printf("output: %s\n", output); } } return 0; }

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  • after dup2, stream still contains old contents?

    - by BobTurbo
    so if I do: dup2(0, backup); // backup stdin dup2(somefile, 0); // somefile has four lines of content fgets(...stdin); // consume one line fgets(....stdin); // consume two lines dup2(backup, 0); // switch stdin back to keyboard I am finding at this point.. stdin still contains the two lines I haven't consumed. Why is that? Because there is just one buffer no matter how many times you redirect? How do I get rid of the two lines left but still remember where I was in the somefile stream when I want to go back to it?

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  • [CakePHP] Pagination after inserting or updateing record

    - by user198003
    one more question related with cakephp... let's say that i have 20+ records in my table. they are sorted by some criteria, ie. by title. and on a list view, i have a list of 10 records, with available pagination. how can i achieve that when i insert new record, to be redirected to proper page, where i can see record that is just was insterted? how can i get information on which page i have to be redirected? hope my question is enough clear for understanding... tnx in adv!

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  • Perl, redirect stdout to file

    - by Mike
    I'm looking for an example of redirecting stdout to a file using Perl. I'm doing a fairly straightforward fork/exec tool, and I want to redirect the child's output to a file instead of the parents stdout. Is there an equivilant of dup2() I should use? I can't seem to find it

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  • Redirecting http to https for a directory, via .htaccess, using mod_alias only

    - by Belinda
    I have the common problem of wanting to redirect requests for a certain restricted access directory from http to https, so that users' login credentials are sent in a secure way. However, I do not have mod_rewrite enabled on my server. I only have mod_alias. This means I have to use the RedirectMatch command. I can't use the usual solutions that use RewriteCond and RewriteRule. (A note on the politics: I am a small-fry subsite maintainer in a very large organisation, so the server admins are unlikely to be willing to change the server config for me!) The following line works, but forms an infinite loop (because the rewritten URL is still caught by the initial regular expression): RedirectMatch permanent ^/intranet(.*)$ https://example.com/intranet$1 One of my internal IT guys has suggested I avoid the infinite loop by moving the files to a new directory with a new name (eg /intranet2/). That seems pretty ugly to me. And people could still accidentally/deliberately revert to an insecure connection by visiting http://example.com/intranet2/ directly. Then I tried this, but it didn't work: RedirectMatch permanent ^http:(.*)/intranet(.*)$ https://example.com/intranet$1 I suspect it didn't work because the first argument must be a file path from the root directory, so it can't start with "http:". So: any better ideas how to do this?

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  • IIS redirects to url beginning with "http://http" although syntax in web.config file appears to be alright

    - by user1608920
    Here's what I have so far: <?xml version="1.0" encoding="UTF-8"?> <configuration> <location path="osb"> <system.webServer> <httpRedirect enabled="true" exactDestination="true" destination="http://50.63.54.135/app/osb" httpResponseStatus="Permanent" /> </system.webServer> </location> </configuration> The above redirect works, but it takes me to http://http//50.63.54.135/app/osb instead of just http://50.63.54.135/app/osb This produces an 404 error. I tried to remove "http://" from destination. Same effect. What am I missing ?

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  • How to handle redirections with codeigniter?

    - by SinneR
    Hi, im having problems starting a codeigniter project, the problem is that when i do something in a controller and then i want a page to display the result, an example: i have a form to add a item to the database, i get all the data in the controller and save it to database and then i want (if all went well) to redirect to the main page with a success msg, i was doing this with $this->load->view('admin', $data); the problem is that the url keeps saying admin/addItem so everytime the page gets refreshed it adds another item, now i found the: redirect('admin','refresh'); but this only helps me when i dont need to display any msg because this function dont allow to send a $data var. Any ideas? Probably this is really easy to fix but i cant find a way to handle the flow of the application the way i want, any help is apreciated. thanks ;)

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  • audio cd s not burning to mp3 format-burning to wav format in k3b and brasero using ubuntu 12.04.2

    - by robert
    It started in ubuntu 13.04-I was doing what I usually do,I opened brasero to make an audio cd from a few mp3 audio files..When burned I noticed the files on cd were in wav format.I then tried k3b with the same result.At that point and because of several issues with 13.04 I formatted my hdd and dropped back to ubuntu 12.04.On 12.04 I tried brasero and k3b once again with same results.I know that when I used to burn cd s using brasero they were burned to cd in mp3 format not wave.Can anyone tell me a fix for this?I have restricted codecs installed.

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  • Trouble with Remote Desktop pulling through printers. Drive Redirection works, and the ports created but not the printers

    - by Windex
    I've run out of things to look into. All the support documents have been gone through and still provide no resolution. I've checked the service permissions, (sc sdshow spooler) they all match up with other systems and what is output on the support documents. I'm nearly positive that the issue can't be permissions anyway as the software requires all users to be an administrator, so all users are a local administrator. (I haven't looked into why yet but its on the list, I was just recently brought into this team and we've put procedures in place for quick recovery.) We've applied hot fixes relating to RDS and printing, though I'm not sure which ones they were. I've combed through group policy and no where is printer redirection disabled. It's setup with all default values regarding the use and redirection of printers and a quick install of W2k8 R2 shows that it works by default. This dev install was joined to the same domain, placed in the same OU, shows the same policies applied, etc, etc, etc, The server generates all the correct redirected ports but no printers are created. It will also redirect drives without issue, this would seem to rule out the usermode service that handles redirects being broken. No events are logged related to any of the events and there are no events from the TerminalServices-Printer source. There were local printers setup. I didn't think it would mattter but as I was running out of ideas I tried deleting them all with no change. The TS was configured for the software it will be running before we checked out the redirection of printers so the other team responsible to setting up new servers wants to find a fix instead of reloading a new server. I'm not sure where or what else to look for. Any ideas?

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  • Play and record streaming audio

    - by Igor
    I'm working on an iPhone app that should be able to play and record audio streaming data simultaneously. Is it actually possible? I'm trying to mix SpeakHere and AudioRecorder samples and getting an empty file with no audio data... Here is my .m code: import "AzRadioViewController.h" @implementation azRadioViewController static const CFOptionFlags kNetworkEvents = kCFStreamEventOpenCompleted | kCFStreamEventHasBytesAvailable | kCFStreamEventEndEncountered | kCFStreamEventErrorOccurred; void MyAudioQueueOutputCallback( void* inClientData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer, const AudioTimeStamp inStartTime, UInt32 inNumberPacketDescriptions, const AudioStreamPacketDescription inPacketDesc ) { NSLog(@"start MyAudioQueueOutputCallback"); MyData* myData = (MyData*)inClientData; NSLog(@"--- %i", inNumberPacketDescriptions); if(inNumberPacketDescriptions == 0 && myData-dataFormat.mBytesPerPacket != 0) { inNumberPacketDescriptions = inBuffer-mAudioDataByteSize / myData-dataFormat.mBytesPerPacket; } OSStatus status = AudioFileWritePackets(myData-audioFile, FALSE, inBuffer-mAudioDataByteSize, inPacketDesc, myData-currentPacket, &inNumberPacketDescriptions, inBuffer-mAudioData); if(status == 0) { myData-currentPacket += inNumberPacketDescriptions; } NSLog(@"status:%i curpac:%i pcdesct: %i", status, myData-currentPacket, inNumberPacketDescriptions); unsigned int bufIndex = MyFindQueueBuffer(myData, inBuffer); pthread_mutex_lock(&myData-mutex); myData-inuse[bufIndex] = false; pthread_cond_signal(&myData-cond); pthread_mutex_unlock(&myData-mutex); } OSStatus StartQueueIfNeeded(MyData* myData) { NSLog(@"start StartQueueIfNeeded"); OSStatus err = noErr; if (!myData-started) { err = AudioQueueStart(myData-queue, NULL); if (err) { PRINTERROR("AudioQueueStart"); myData-failed = true; return err; } myData-started = true; printf("started\n"); } return err; } OSStatus MyEnqueueBuffer(MyData* myData) { NSLog(@"start MyEnqueueBuffer"); OSStatus err = noErr; myData-inuse[myData-fillBufferIndex] = true; AudioQueueBufferRef fillBuf = myData-audioQueueBuffer[myData-fillBufferIndex]; fillBuf-mAudioDataByteSize = myData-bytesFilled; err = AudioQueueEnqueueBuffer(myData-queue, fillBuf, myData-packetsFilled, myData-packetDescs); if (err) { PRINTERROR("AudioQueueEnqueueBuffer"); myData-failed = true; return err; } StartQueueIfNeeded(myData); return err; } void WaitForFreeBuffer(MyData* myData) { NSLog(@"start WaitForFreeBuffer"); if (++myData-fillBufferIndex = kNumAQBufs) myData-fillBufferIndex = 0; myData-bytesFilled = 0; myData-packetsFilled = 0; printf("-lock\n"); pthread_mutex_lock(&myData-mutex); while (myData-inuse[myData-fillBufferIndex]) { printf("... WAITING ...\n"); pthread_cond_wait(&myData-cond, &myData-mutex); } pthread_mutex_unlock(&myData-mutex); printf("<-unlock\n"); } int MyFindQueueBuffer(MyData* myData, AudioQueueBufferRef inBuffer) { NSLog(@"start MyFindQueueBuffer"); for (unsigned int i = 0; i < kNumAQBufs; ++i) { if (inBuffer == myData-audioQueueBuffer[i]) return i; } return -1; } void MyAudioQueueIsRunningCallback( void* inClientData, AudioQueueRef inAQ, AudioQueuePropertyID inID) { NSLog(@"start MyAudioQueueIsRunningCallback"); MyData* myData = (MyData*)inClientData; UInt32 running; UInt32 size; OSStatus err = AudioQueueGetProperty(inAQ, kAudioQueueProperty_IsRunning, &running, &size); if (err) { PRINTERROR("get kAudioQueueProperty_IsRunning"); return; } if (!running) { pthread_mutex_lock(&myData-mutex); pthread_cond_signal(&myData-done); pthread_mutex_unlock(&myData-mutex); } } void MyPropertyListenerProc( void * inClientData, AudioFileStreamID inAudioFileStream, AudioFileStreamPropertyID inPropertyID, UInt32 * ioFlags) { NSLog(@"start MyPropertyListenerProc"); MyData* myData = (MyData*)inClientData; OSStatus err = noErr; printf("found property '%c%c%c%c'\n", (inPropertyID24)&255, (inPropertyID16)&255, (inPropertyID8)&255, inPropertyID&255); switch (inPropertyID) { case kAudioFileStreamProperty_ReadyToProducePackets : { AudioStreamBasicDescription asbd; UInt32 asbdSize = sizeof(asbd); err = AudioFileStreamGetProperty(inAudioFileStream, kAudioFileStreamProperty_DataFormat, &asbdSize, &asbd); if (err) { PRINTERROR("get kAudioFileStreamProperty_DataFormat"); myData-failed = true; break; } err = AudioQueueNewOutput(&asbd, MyAudioQueueOutputCallback, myData, NULL, NULL, 0, &myData-queue); if (err) { PRINTERROR("AudioQueueNewOutput"); myData-failed = true; break; } for (unsigned int i = 0; i < kNumAQBufs; ++i) { err = AudioQueueAllocateBuffer(myData-queue, kAQBufSize, &myData-audioQueueBuffer[i]); if (err) { PRINTERROR("AudioQueueAllocateBuffer"); myData-failed = true; break; } } UInt32 cookieSize; Boolean writable; err = AudioFileStreamGetPropertyInfo(inAudioFileStream, kAudioFileStreamProperty_MagicCookieData, &cookieSize, &writable); if (err) { PRINTERROR("info kAudioFileStreamProperty_MagicCookieData"); break; } printf("cookieSize %d\n", cookieSize); void* cookieData = calloc(1, cookieSize); err = AudioFileStreamGetProperty(inAudioFileStream, kAudioFileStreamProperty_MagicCookieData, &cookieSize, cookieData); if (err) { PRINTERROR("get kAudioFileStreamProperty_MagicCookieData"); free(cookieData); break; } err = AudioQueueSetProperty(myData-queue, kAudioQueueProperty_MagicCookie, cookieData, cookieSize); free(cookieData); if (err) { PRINTERROR("set kAudioQueueProperty_MagicCookie"); break; } err = AudioQueueAddPropertyListener(myData-queue, kAudioQueueProperty_IsRunning, MyAudioQueueIsRunningCallback, myData); if (err) { PRINTERROR("AudioQueueAddPropertyListener"); myData-failed = true; break; } break; } } } static void ReadStreamClientCallBack(CFReadStreamRef stream, CFStreamEventType type, void *clientCallBackInfo) { NSLog(@"start ReadStreamClientCallBack"); if(type == kCFStreamEventHasBytesAvailable) { UInt8 buffer[2048]; CFIndex bytesRead = CFReadStreamRead(stream, buffer, sizeof(buffer)); if (bytesRead < 0) { } else if (bytesRead) { OSStatus err = AudioFileStreamParseBytes(globalMyData-audioFileStream, bytesRead, buffer, 0); if (err) { PRINTERROR("AudioFileStreamParseBytes"); } } } } void MyPacketsProc(void * inClientData, UInt32 inNumberBytes, UInt32 inNumberPackets, const void * inInputData, AudioStreamPacketDescription inPacketDescriptions) { NSLog(@"start MyPacketsProc"); MyData myData = (MyData*)inClientData; printf("got data. bytes: %d packets: %d\n", inNumberBytes, inNumberPackets); for (int i = 0; i < inNumberPackets; ++i) { SInt64 packetOffset = inPacketDescriptions[i].mStartOffset; SInt64 packetSize = inPacketDescriptions[i].mDataByteSize; size_t bufSpaceRemaining = kAQBufSize - myData-bytesFilled; if (bufSpaceRemaining < packetSize) { MyEnqueueBuffer(myData); WaitForFreeBuffer(myData); } AudioQueueBufferRef fillBuf = myData-audioQueueBuffer[myData-fillBufferIndex]; memcpy((char*)fillBuf-mAudioData + myData-bytesFilled, (const char*)inInputData + packetOffset, packetSize); myData-packetDescs[myData-packetsFilled] = inPacketDescriptions[i]; myData-packetDescs[myData-packetsFilled].mStartOffset = myData-bytesFilled; myData-bytesFilled += packetSize; myData-packetsFilled += 1; size_t packetsDescsRemaining = kAQMaxPacketDescs - myData-packetsFilled; if (packetsDescsRemaining == 0) { MyEnqueueBuffer(myData); WaitForFreeBuffer(myData); } } } (IBAction)buttonPlayPressedid)sender { label.text = @"Buffering"; [self connectionStart]; } (IBAction)buttonSavePressedid)sender { NSLog(@"save"); AudioFileClose(myData.audioFile); AudioQueueDispose(myData.queue, TRUE); } bool getFilename(char* buffer,int maxBufferLength) { NSArray paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES); NSString docDir = [paths objectAtIndex:0]; NSString* file = [docDir stringByAppendingString:@"/rec.caf"]; return [file getCString:buffer maxLength:maxBufferLength encoding:NSUTF8StringEncoding]; } -(void)connectionStart { @try { MyData* myData = (MyData*)calloc(1, sizeof(MyData)); globalMyData = myData; pthread_mutex_init(&myData-mutex, NULL); pthread_cond_init(&myData-cond, NULL); pthread_cond_init(&myData-done, NULL); NSLog(@"Start"); myData-dataFormat.mSampleRate = 16000.0f; myData-dataFormat.mFormatID = kAudioFormatLinearPCM; myData-dataFormat.mFramesPerPacket = 1; myData-dataFormat.mChannelsPerFrame = 1; myData-dataFormat.mBytesPerFrame = 2; myData-dataFormat.mBytesPerPacket = 2; myData-dataFormat.mBitsPerChannel = 16; myData-dataFormat.mReserved = 0; myData-dataFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked; int i, bufferByteSize; UInt32 size; AudioQueueNewInput( &myData-dataFormat, MyAudioQueueOutputCallback, &myData, NULL /* run loop /, kCFRunLoopCommonModes / run loop mode /, 0 / flags */, &myData-queue); size = sizeof(&myData-dataFormat); AudioQueueGetProperty(&myData-queue, kAudioQueueProperty_StreamDescription, &myData-dataFormat, &size); CFURLRef fileURL; char path[256]; memset(path,0,sizeof(path)); getFilename(path,256); fileURL = CFURLCreateFromFileSystemRepresentation(NULL, (UInt8*)path, strlen(path), FALSE); AudioFileCreateWithURL(fileURL, kAudioFileCAFType, &myData-dataFormat, kAudioFileFlags_EraseFile, &myData-audioFile); OSStatus err = AudioFileStreamOpen(myData, MyPropertyListenerProc, MyPacketsProc, kAudioFileMP3Type, &myData-audioFileStream); if (err) { PRINTERROR("AudioFileStreamOpen"); return 1; } CFStreamClientContext ctxt = {0, self, NULL, NULL, NULL}; CFStringRef bodyData = CFSTR(""); // Usually used for POST data CFStringRef headerFieldName = CFSTR("X-My-Favorite-Field"); CFStringRef headerFieldValue = CFSTR("Dreams"); CFStringRef url = CFSTR(RADIO_LOCATION); CFURLRef myURL = CFURLCreateWithString(kCFAllocatorDefault, url, NULL); CFStringRef requestMethod = CFSTR("GET"); CFHTTPMessageRef myRequest = CFHTTPMessageCreateRequest(kCFAllocatorDefault, requestMethod, myURL, kCFHTTPVersion1_1); CFHTTPMessageSetBody(myRequest, bodyData); CFHTTPMessageSetHeaderFieldValue(myRequest, headerFieldName, headerFieldValue); CFReadStreamRef stream = CFReadStreamCreateForHTTPRequest(kCFAllocatorDefault, myRequest); if (!stream) { NSLog(@"Creating the stream failed"); return; } if (!CFReadStreamSetClient(stream, kNetworkEvents, ReadStreamClientCallBack, &ctxt)) { CFRelease(stream); NSLog(@"Setting the stream's client failed."); return; } CFReadStreamScheduleWithRunLoop(stream, CFRunLoopGetCurrent(), kCFRunLoopCommonModes); if (!CFReadStreamOpen(stream)) { CFReadStreamSetClient(stream, 0, NULL, NULL); CFReadStreamUnscheduleFromRunLoop(stream, CFRunLoopGetCurrent(), kCFRunLoopCommonModes); CFRelease(stream); NSLog(@"Opening the stream failed."); return; } } @catch (NSException *exception) { NSLog(@"main: Caught %@: %@", [exception name], [exception reason]); } } (void)viewDidLoad { [[UIApplication sharedApplication] setIdleTimerDisabled:YES]; [super viewDidLoad]; } (void)didReceiveMemoryWarning { [super didReceiveMemoryWarning]; } (void)viewDidUnload { } (void)dealloc { [super dealloc]; } @end

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  • Stopping and Play button for Audio (Android)

    - by James Rattray
    I have this problem, I have some audio I wish to play... And I have two buttons for it, 'Play' and 'Stop'... Problem is, after I press the stop button, and then press the Play button, nothing happens. -The stop button stops the song, but I want the Play button to play the song again (from the start) Here is my code: final MediaPlayer mp = MediaPlayer.create(this, R.raw.megadeth); And then the two public onclicks: (For playing...) button.setOnClickListener(new View.OnClickListener() { public void onClick(View v) { // Perform action on click button.setText("Playing!"); try { mp.prepare(); } catch (IllegalStateException e) { // TODO Auto-generated catch block e.printStackTrace(); } catch (IOException e) { // TODO Auto-generated catch block e.printStackTrace(); } mp.start(); // } }); And for stopping the track... final Button button2 = (Button) findViewById(R.id.cancel); button2.setOnClickListener(new View.OnClickListener() { public void onClick(View v) { mp.stop(); mp.reset(); } }); Can anyone see the problem with this? If so could you please fix it... (For suggest) Thanks alot... James

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  • WPF Storyboard delay in playing wma files

    - by Rita
    I'm a complete beginner in WPF and have an app that uses StoryBoard to play a sound. public void PlaySound() { MediaElement m = (MediaElement)audio.FindName("MySound.wma"); m.IsMuted = false; FrameworkElement audioKey = (FrameworkElement)keys.FindName("MySound"); Storyboard s = (Storyboard)audioKey.FindResource("MySound.wma"); s.Begin(audioKey); } <Storyboard x:Key="MySound.wma"> <MediaTimeline d:DesignTimeNaturalDuration="1.615" BeginTime="00:00:00" Storyboard.TargetName="MySound.wma" Source="Audio\MySound.wma"/> </Storyboard> I have a horrible lag and sometimes it takes good 10 seconds for the sound to be played. I suspect this has something to do with the fact that no matter how long I wait - The sound doesn't get played until after I leave the function. I don't understand it. I call Begin, and nothing happens. Is there a way to replace this method, or StoryBoard object with something that plays instantly and without a lag?

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  • Why does this gstreamer pipeline stall ?

    - by timday
    I've been playing around with gstreamer pipelines using gst-launch. I don't have any problems if I just want to process audio or video separately (to separate files, or to alsasink/ximagesink), but I'm confused by what I need to do to mux the streams back together using, say avimux. This gst-launch-0.10 filesrc location=MVI_2034.AVI ! decodebin name=dec \ dec. ! queue ! audioconvert ! 'audio/x-raw-int,rate=44100,channels=1' ! queue ! mux. \ dec. ! queue ! videoflip 1 ! ffmpegcolorspace ! jpegenc ! queue ! mux. \ avimux name=mux ! filesink location=out.avi just outputs Setting pipeline to PAUSED ... Pipeline is PREROLLING ... and then stalls indefinitely. What's the trick ?

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  • What do you use to play sound in iPhone games?

    - by zoul
    Hello! I have a performance-intensive iPhone game I would like to add sounds to. There seem to be about three main choices: (1) AVAudioPlayer, (2) Audio Queues and (3) OpenAL. I’d hate to write pages of low-level code just to play a sample, so that I would like to use AVAudioPlayer. The problem is that it seems to kill the performace – I’ve done a simple measuring using CFAbsoluteTimeGetCurrent and the play message seems to take somewhere from 9 to 30 ms to finish. That’s quite miserable, considering that 25 ms == 40 fps. Of course there is the prepareToPlay method that should speed things up. That’s why I wrote a simple class that keeps several AVAudioPlayers at its disposal, prepares them beforehand and then plays the sample using the prepared player. No cigar, still it takes the ~20 ms I mentioned above. Such performance is unusable for games, so what do you use to play sounds with a decent performance on iPhone? Am I doing something wrong with the AVAudioPlayer? Do you play sounds with Audio Queues? (I’ve written something akin to AVAudioPlayer before 2.2 came out and I would love to spare that experience.) Do you use OpenAL? If yes, is there a simple way to play sounds with OpenAL, or do you have to write pages of code? Update: Yes, playing sounds with OpenAL is fairly simple.

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  • JMF microphone volume controller

    - by TacB0sS
    How to obtain the Microphone volume controller in JMF? this is what I have: I tried this implementation concept of yours, but I keep getting a null from the first volume processor when I try to get the stream, here is how I do it: // the device is the media device specifically audio Processor processorForVolume = Manager.createProcessor(device.getLocator()); // wait until configured ProcessorStates newState = new ProcessorStateListener(Processor.Configured).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // setting the content descriptor to null - read in another thread this allows to get the gain control processorForVolume.setContentDescriptor(null); // set the track control format to one supported by the device and the track control. // I didn't match it to an RTP allowed format, but I don't think this has anything to do with it... TrackControl[] trackControls = processorForVolume.getTrackControls(); if (trackControls.length == 0) throw new MC_Exception("No track controls where found for this device:", new Object[]{device}); for (TrackControl control : trackControls) trackManipulator.manipulateTrackControls(control); // wait until the processor is realized newState = new ProcessorStateListener(Controller.Realized).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // receives the gain control micVolumeController = processorForVolume.getGainControl(); // cannot get the output stream to process further... any suggestions? processor = Manager.createProcessor(processorForVolume.getDataOutput()); new ProcessorStateListener(Processor.Configured).waitForProcessorState(processor); processor.setContentDescriptor(DeviceCapturingManager.RAW_RTP); new ProcessorStateListener(Controller.Realized).waitForProcessorState(processor); this is the output It generates: volumeProcessorState: Configured format set to track control - com.sun.media.ProcessEngine$ProcTControl@1627c16: LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed volumeProcessorState: Realized and the data output from the processor is Null. I should make clear that when the content descriptor != null I do get an output stream but not the volume controller, and the when it is null I get the controller, but no stream. I try to connect to an audio microphone device Adam.

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  • Graphing the pitch (frequency) of a sound

    - by Coronatus
    I want to plot the pitch of a sound into a graph. Currently I can plot the amplitude. The graph below is created by the data returned by getUnscaledAmplitude(): AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(new BufferedInputStream(new FileInputStream(file))); byte[] bytes = new byte[(int) (audioInputStream.getFrameLength()) * (audioInputStream.getFormat().getFrameSize())]; audioInputStream.read(bytes); // Get amplitude values for each audio channel in an array. graphData = type.getUnscaledAmplitude(bytes, this); public int[][] getUnscaledAmplitude(byte[] eightBitByteArray, AudioInfo audioInfo) { int[][] toReturn = new int[audioInfo.getNumberOfChannels()][eightBitByteArray.length / (2 * audioInfo. getNumberOfChannels())]; int index = 0; for (int audioByte = 0; audioByte < eightBitByteArray.length;) { for (int channel = 0; channel < audioInfo.getNumberOfChannels(); channel++) { // Do the byte to sample conversion. int low = (int) eightBitByteArray[audioByte]; audioByte++; int high = (int) eightBitByteArray[audioByte]; audioByte++; int sample = (high << 8) + (low & 0x00ff); if (sample < audioInfo.sampleMin) { audioInfo.sampleMin = sample; } else if (sample > audioInfo.sampleMax) { audioInfo.sampleMax = sample; } toReturn[channel][index] = sample; } index++; } return toReturn; } But I need to show the audio's pitch, not amplitude. Fast Fourier transform appears to get the pitch, but it needs to know more variables than the raw bytes I have, and is very complex and mathematical. Is there a way I can do this?

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  • Python on Mac: Fink? MacPorts? Builtin? Homebrew? Binary installer?

    - by BastiBechtold
    For the last few days, I have been trying to use Python for some audio development. The thing is, Mac OSX does not handle uninstalling stuff well. Actually, there is no way to uninstall anything. Once it is on your system, you better pray that it didn't do any funny stuff. Hence, I don't really want to rely on installer packages for Python. So I turn to Homebrew and install Python using Homebrew. Works fabulously. Using pip, Numpy, SciPy, Matplotlib were no (big) problem, either. Now I want to play audio. There is a host of different packages out there, but pip does not seem willing to install any. But, there is a binary distribution for PyGame, which I guess should work with the built-in Python. Hence my question: What would you do? Would you just install the binary distributions and hope that they interoperate well and never need uninstalling? Would you hack your way through whichever package control management system you prefer and deal with its problems? Something else?

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  • Syncing two AS3 NetStreams

    - by Lowgain
    I'm writing an app that requires an audio stream to be recording while a backing track is played. I have this working, but there is an inconsistent gap in between playback and record starting. I don't know if I can do anything to make the sync perfect every time, so I've been trying to track what time each stream starts so I can calculate the delay and trim it server-side. This also has proved to be a challenge as no events seem to be sent when a connection starts (as far as I know). I've tried using various properties like the streams' buffer sizes, etc. I'm thinking now that as my recorded audio is only mono, I may be able to put some kind of 'control signal' on the second stereo track which I could use to determine exactly when a sound starts recording (or stick the whole backing track in that channel so I can sync them that way). This leaves me with the new problem of properly injecting this sound into the NetStream. If anyone has any idea whether or not any of these ideas will work, how to execute them, or some alternatives, that would be extremely helpful! Been working on this issue for awhile

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  • Playing a sequence of sounds without gaps (iPhone)

    - by Fiire
    I thought maybe the fastest way was to go with Sound Services. It is quite efficient, but I need to play sounds in a sequence, not overlapped. Therefore I used a callback method to check when the sound has finished. This cycle produces around 0.3 seconds in lag. I know this sounds very strict, but it is basically the main axis of the program. EDIT: I now tried using AVAudioPlayer, but I can't play sounds in a sequence without using audioPlayerDidFinishPlaying since that would put me in the same situation as with the callback method of SoundServices. EDIT2: I think that if I could somehow get to join the parts of the sounds I want to play into a large file, I could get the whole audio file to sound continuously. EDIT3: I thought this would work, but the audio overlaps: waitTime = player.deviceCurrentTime; for (int k = 0; k < [colores count]; k++) { player.currentTime = 0; [player playAtTime:waitTime]; waitTime += player.duration; } Thanks

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  • how to upload a audio file using REST webservice in Google App Engine for Java

    - by sathya
    Am using google app engine with eclipse IDE and trying to upload a audio file. I used the File Upload in Google App Engine For Java and can able to upload the file successfully. Now am planning to use REST web service for it. I had analyzed in developers.google but i failed. Can anyone suggest me how to implement REST Web services in google app engine using Eclipse. The code google provided is shown below, // file Upload.java public class Upload extends HttpServlet { private BlobstoreService blobstoreService = BlobstoreServiceFactory.getBlobstoreService(); public void doPost(HttpServletRequest req, HttpServletResponse res) throws ServletException, IOException { Map<String, BlobKey> blobs = blobstoreService.getUploadedBlobs(req); BlobKey blobKey = blobs.get("myFile"); if (blobKey == null) { res.sendRedirect("/"); } else { res.sendRedirect("/serve?blob-key=" + blobKey.getKeyString()); }}} // file Serve.java public class Serve extends HttpServlet { private BlobstoreService blobstoreService = BlobstoreServiceFactory.getBlobstoreService(); public void doGet(HttpServletRequest req, HttpServletResponse res) throws IOException { BlobKey blobKey = new BlobKey(req.getParameter("blob-key")); blobstoreService.serve(blobKey, res); }} // file index.jsp <%@ page import="com.google.appengine.api.blobstore.BlobstoreServiceFactory" %> <%@ page import="com.google.appengine.api.blobstore.BlobstoreService" %> <% BlobstoreService blobstoreService = BlobstoreServiceFactory.getBlobstoreService(); %> <form action="<%= blobstoreService.createUploadUrl("/upload") %>" method="post" enctype="multipart/form-data"> <input type="file" name="myFile"> <input type="submit" value="Submit"> </form> // web.xml <servlet> <servlet-name>Upload</servlet-name> <servlet-class>Upload</servlet-class> </servlet> <servlet> <servlet-name>Serve</servlet-name> <servlet-class>Serve</servlet-class> </servlet> <servlet-mapping> <servlet-name>Upload</servlet-name> <url-pattern>/upload</url-pattern> </servlet-mapping> <servlet-mapping> <servlet-name>Serve</servlet-name> <url-pattern>/serve</url-pattern> </servlet-mapping> Now how to provide a rest web service for the above code. Kindly suggest me an idea.

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  • How to play audio in Java Application

    - by user577829
    I'm making a java application and I need to play audio. I'm playing mainly small sound files of my cannon firing (its a cannon shooting game) and the projectiles exploding, though I plan on having looping background music. I have found two different methods to accomplish this, but both don't work how I want. The first method is literally a method: public void playSoundFile(File file) {//http://java.ittoolbox.com/groups/technical-functional/java-l/sound-in-an-application-90681 try { //get an AudioInputStream AudioInputStream ais = AudioSystem.getAudioInputStream(file); //get the AudioFormat for the AudioInputStream AudioFormat audioformat = ais.getFormat(); System.out.println("Format: " + audioformat.toString()); System.out.println("Encoding: " + audioformat.getEncoding()); System.out.println("SampleRate:" + audioformat.getSampleRate()); System.out.println("SampleSizeInBits: " + audioformat.getSampleSizeInBits()); System.out.println("Channels: " + audioformat.getChannels()); System.out.println("FrameSize: " + audioformat.getFrameSize()); System.out.println("FrameRate: " + audioformat.getFrameRate()); System.out.println("BigEndian: " + audioformat.isBigEndian()); //ULAW format to PCM format conversion if ((audioformat.getEncoding() == AudioFormat.Encoding.ULAW) || (audioformat.getEncoding() == AudioFormat.Encoding.ALAW)) { AudioFormat newformat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, audioformat.getSampleRate(), audioformat.getSampleSizeInBits() * 2, audioformat.getChannels(), audioformat.getFrameSize() * 2, audioformat.getFrameRate(), true); ais = AudioSystem.getAudioInputStream(newformat, ais); audioformat = newformat; } //checking for a supported output line DataLine.Info datalineinfo = new DataLine.Info(SourceDataLine.class, audioformat); if (!AudioSystem.isLineSupported(datalineinfo)) { //System.out.println("Line matching " + datalineinfo + " is not supported."); } else { //System.out.println("Line matching " + datalineinfo + " is supported."); //opening the sound output line SourceDataLine sourcedataline = (SourceDataLine) AudioSystem.getLine(datalineinfo); sourcedataline.open(audioformat); sourcedataline.start(); //Copy data from the input stream to the output data line int framesizeinbytes = audioformat.getFrameSize(); int bufferlengthinframes = sourcedataline.getBufferSize() / 8; int bufferlengthinbytes = bufferlengthinframes * framesizeinbytes; byte[] sounddata = new byte[bufferlengthinbytes]; int numberofbytesread = 0; while ((numberofbytesread = ais.read(sounddata)) != -1) { int numberofbytesremaining = numberofbytesread; sourcedataline.write(sounddata, 0, numberofbytesread); } } } catch (Exception e) { e.printStackTrace(); } } The problem with this is that my entire program stops until the sound file is finished, or at least nearly finished. The second method is this: File file = new File("Launch1.wav"); AudioClip clip; try { clip = JApplet.newAudioClip(file.toURL()); clip.play(); } catch (Exception e) { e.getMessage(); } The problem I have here is that every time the sound file ends early or doesn't play at all depending on where I place the code. Is their any way to play sound without the above mentioned problems? Am I doing something wrong? Any help is greatly appreciated.

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  • Does Hauppauge WinTV HVR-900 (r2) [USB ID 2040:6502] work with ubuntu 12.04 LTS?

    - by nightfly
    I have this DVB+Analog usb tv tuner Hauppauge WinTV HVR-900 (r2) [USB ID 2040:6502]. This used to work under ubuntu 10.04 LTS. But in 12.04 there seems to be a problem. I have linux-firmware-nonfree and ivtv-utils installed. I am running Ubuntu 12.04.1 LTS 64 bit with all updates installed and the default unity environment. When I run mplayer tv:// -tv driver=v4l2:device=/dev/video1:input=1:norm=PAL I get a solid green screen and no picture. Here input 1 is the composite input of the card. MPlayer svn r34540 (Ubuntu), built with gcc-4.6 (C) 2000-2012 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing tv://. TV file format detected. Selected driver: v4l2 name: Video 4 Linux 2 input author: Martin Olschewski comment: first try, more to come ;-) Selected device: Hauppauge WinTV HVR 900 (R2) Tuner cap: Tuner rxs: Capabilities: video capture VBI capture device tuner audio read/write streaming supported norms: 0 = NTSC; 1 = NTSC-M; 2 = NTSC-M-JP; 3 = NTSC-M-KR; 4 = NTSC-443; 5 = PAL; 6 = PAL-BG; 7 = PAL-H; 8 = PAL-I; 9 = PAL-DK; 10 = PAL-M; 11 = PAL-N; 12 = PAL-Nc; 13 = PAL-60; 14 = SECAM; 15 = SECAM-B; 16 = SECAM-G; 17 = SECAM-H; 18 = SECAM-DK; 19 = SECAM-L; 20 = SECAM-Lc; inputs: 0 = Television; 1 = Composite1; 2 = S-Video; Current input: 1 Current format: YUYV v4l2: current audio mode is : MONO v4l2: ioctl set format failed: Invalid argument v4l2: ioctl set format failed: Invalid argument v4l2: ioctl set format failed: Invalid argument v4l2: ioctl query control failed: Invalid argument v4l2: ioctl query control failed: Invalid argument v4l2: ioctl query control failed: Invalid argument v4l2: ioctl query control failed: Invalid argument Failed to open VDPAU backend libvdpau_nvidia.so: cannot open shared object file: No such file or directory [vdpau] Error when calling vdp_device_create_x11: 1 ========================================================================== Opening video decoder: [raw] RAW Uncompressed Video Movie-Aspect is undefined - no prescaling applied. VO: [xv] 640x480 = 640x480 Packed YUY2 Selected video codec: [rawyuy2] vfm: raw (RAW YUY2) ========================================================================== Audio: no sound Starting playback... v4l2: select timeout V: 0.0 2/ 2 ??% ??% ??,?% 0 0 v4l2: select timeout V: 0.0 4/ 4 ??% ??% ??,?% 0 0 v4l2: select timeout V: 0.0 6/ 6 ??% ??% ??,?% 0 0 v4l2: select timeout v4l2: 0 frames successfully processed, 1 frames dropped. Exiting... (Quit) Here is the dmesg of the card when plugged in.. [12742.228097] usb 1-4: new high-speed USB device number 3 using ehci_hcd [12742.367289] em28xx: New device WinTV HVR-900 @ 480 Mbps (2040:6502, interface 0, class 0) [12742.367296] em28xx: Audio Vendor Class interface 0 found [12742.367585] em28xx #0: chip ID is em2882/em2883 [12742.550086] em28xx #0: i2c eeprom 00: 1a eb 67 95 40 20 02 65 d0 12 5c 03 82 1e 6a 18 [12742.550104] em28xx #0: i2c eeprom 10: 00 00 24 57 66 07 01 00 00 00 00 00 00 00 00 00 [12742.550120] em28xx #0: i2c eeprom 20: 46 00 01 00 f0 10 02 00 b8 00 00 00 5b e0 00 00 [12742.550135] em28xx #0: i2c eeprom 30: 00 00 20 40 20 6e 02 20 10 01 01 01 00 00 00 00 [12742.550150] em28xx #0: i2c eeprom 40: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 [12742.550165] em28xx #0: i2c eeprom 50: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 [12742.550181] em28xx #0: i2c eeprom 60: 00 00 00 00 00 00 00 00 00 00 18 03 34 00 30 00 [12742.550196] em28xx #0: i2c eeprom 70: 32 00 37 00 38 00 32 00 33 00 39 00 30 00 31 00 [12742.550211] em28xx #0: i2c eeprom 80: 00 00 1e 03 57 00 69 00 6e 00 54 00 56 00 20 00 [12742.550226] em28xx #0: i2c eeprom 90: 48 00 56 00 52 00 2d 00 39 00 30 00 30 00 00 00 [12742.550241] em28xx #0: i2c eeprom a0: 84 12 00 00 05 50 1a 7f d4 78 23 fa fd d0 28 89 [12742.550257] em28xx #0: i2c eeprom b0: ff 00 00 00 04 84 0a 00 01 01 20 77 00 40 1d b7 [12742.550272] em28xx #0: i2c eeprom c0: 13 f0 74 02 01 00 01 79 63 00 00 00 00 00 00 00 [12742.550287] em28xx #0: i2c eeprom d0: 84 12 00 00 05 50 1a 7f d4 78 23 fa fd d0 28 89 [12742.550302] em28xx #0: i2c eeprom e0: ff 00 00 00 04 84 0a 00 01 01 20 77 00 40 1d b7 [12742.550317] em28xx #0: i2c eeprom f0: 13 f0 74 02 01 00 01 79 63 00 00 00 00 00 00 00 [12742.550334] em28xx #0: EEPROM ID= 0x9567eb1a, EEPROM hash = 0x2bbf3bdd [12742.550338] em28xx #0: EEPROM info: [12742.550340] em28xx #0: AC97 audio (5 sample rates) [12742.550343] em28xx #0: 500mA max power [12742.550346] em28xx #0: Table at 0x24, strings=0x1e82, 0x186a, 0x0000 [12742.552590] em28xx #0: Identified as Hauppauge WinTV HVR 900 (R2) (card=18) [12742.555516] tveeprom 15-0050: Hauppauge model 65018, rev B2C0, serial# 1292061 [12742.555523] tveeprom 15-0050: tuner model is Xceive XC3028 (idx 120, type 71) [12742.555529] tveeprom 15-0050: TV standards PAL(B/G) PAL(I) PAL(D/D1/K) ATSC/DVB Digital (eeprom 0xd4) [12742.555534] tveeprom 15-0050: audio processor is None (idx 0) [12742.555537] tveeprom 15-0050: has radio [12742.570297] tuner 15-0061: Tuner -1 found with type(s) Radio TV. [12742.570327] xc2028 15-0061: creating new instance [12742.570332] xc2028 15-0061: type set to XCeive xc2028/xc3028 tuner [12742.573685] xc2028 15-0061: Loading 80 firmware images from xc3028-v27.fw, type: xc2028 firmware, ver 2.7 [12742.624056] xc2028 15-0061: Loading firmware for type=BASE MTS (5), id 0000000000000000. [12744.126591] xc2028 15-0061: Loading firmware for type=MTS (4), id 000000000000b700. [12744.153586] xc2028 15-0061: Loading SCODE for type=MTS LCD NOGD MONO IF SCODE HAS_IF_4500 (6002b004), id 000000000000b700. [12744.280963] Registered IR keymap rc-hauppauge [12744.281151] input: em28xx IR (em28xx #0) as /devices/pci0000:00/0000:00:1a.7/usb1/1-4/rc/rc1/input10 [12744.281541] rc1: em28xx IR (em28xx #0) as /devices/pci0000:00/0000:00:1a.7/usb1/1-4/rc/rc1 [12744.282454] em28xx #0: Config register raw data: 0xd0 [12744.284709] em28xx #0: AC97 vendor ID = 0xffffffff [12744.285829] em28xx #0: AC97 features = 0x6a90 [12744.285832] em28xx #0: Empia 202 AC97 audio processor detected [12744.359211] em28xx #0: v4l2 driver version 0.1.3 [12744.404066] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [12745.915089] MTS (4), id 00000000000000ff: [12745.915100] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [12746.161668] em28xx #0: V4L2 video device registered as video1 [12746.161673] em28xx #0: V4L2 VBI device registered as vbi0 [12746.162845] em28xx-audio.c: probing for em28xx Audio Vendor Class [12746.162848] em28xx-audio.c: Copyright (C) 2006 Markus Rechberger [12746.162851] em28xx-audio.c: Copyright (C) 2007-2011 Mauro Carvalho Chehab [12746.221099] xc2028 15-0061: attaching existing instance [12746.221105] xc2028 15-0061: type set to XCeive xc2028/xc3028 tuner [12746.221109] em28xx #0: em28xx #0/2: xc3028 attached [12746.221113] DVB: registering new adapter (em28xx #0) [12746.221118] DVB: registering adapter 0 frontend 0 (Micronas DRXD DVB-T)... [12746.221869] em28xx #0: Successfully loaded em28xx-dvb [13111.196055] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13112.720062] MTS (4), id 00000000000000ff: [13112.720072] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13214.956057] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13216.479806] MTS (4), id 00000000000000ff: [13216.479816] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13276.408056] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13277.932093] MTS (4), id 00000000000000ff: [13277.932104] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13305.032076] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13306.556449] MTS (4), id 00000000000000ff: [13306.556460] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13392.236055] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13393.760123] MTS (4), id 00000000000000ff: [13393.760133] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13637.534053] usb 1-4: USB disconnect, device number 3 [13637.534183] em28xx #0: disconnecting em28xx #0 video [13637.560214] em28xx #0: V4L2 device vbi0 deregistered [13637.560335] em28xx #0: V4L2 device video1 deregistered [13637.561237] xc2028 15-0061: destroying instance [13639.772120] usb 1-4: new high-speed USB device number 4 using ehci_hcd [13639.911351] em28xx: New device WinTV HVR-900 @ 480 Mbps (2040:6502, interface 0, class 0) [13639.911357] em28xx: Audio Vendor Class interface 0 found [13639.911637] em28xx #0: chip ID is em2882/em2883 [13640.094262] em28xx #0: i2c eeprom 00: 1a eb 67 95 40 20 02 65 d0 12 5c 03 82 1e 6a 18 [13640.094280] em28xx #0: i2c eeprom 10: 00 00 24 57 66 07 01 00 00 00 00 00 00 00 00 00 [13640.094295] em28xx #0: i2c eeprom 20: 46 00 01 00 f0 10 02 00 b8 00 00 00 5b e0 00 00 [13640.094311] em28xx #0: i2c eeprom 30: 00 00 20 40 20 6e 02 20 10 01 01 01 00 00 00 00 [13640.094326] em28xx #0: i2c eeprom 40: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 [13640.094341] em28xx #0: i2c eeprom 50: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 [13640.094356] em28xx #0: i2c eeprom 60: 00 00 00 00 00 00 00 00 00 00 18 03 34 00 30 00 [13640.094371] em28xx #0: i2c eeprom 70: 32 00 37 00 38 00 32 00 33 00 39 00 30 00 31 00 [13640.094386] em28xx #0: i2c eeprom 80: 00 00 1e 03 57 00 69 00 6e 00 54 00 56 00 20 00 [13640.094401] em28xx #0: i2c eeprom 90: 48 00 56 00 52 00 2d 00 39 00 30 00 30 00 00 00 [13640.094416] em28xx #0: i2c eeprom a0: 84 12 00 00 05 50 1a 7f d4 78 23 fa fd d0 28 89 [13640.094432] em28xx #0: i2c eeprom b0: ff 00 00 00 04 84 0a 00 01 01 20 77 00 40 1d b7 [13640.094447] em28xx #0: i2c eeprom c0: 13 f0 74 02 01 00 01 79 63 00 00 00 00 00 00 00 [13640.094462] em28xx #0: i2c eeprom d0: 84 12 00 00 05 50 1a 7f d4 78 23 fa fd d0 28 89 [13640.094477] em28xx #0: i2c eeprom e0: ff 00 00 00 04 84 0a 00 01 01 20 77 00 40 1d b7 [13640.094492] em28xx #0: i2c eeprom f0: 13 f0 74 02 01 00 01 79 63 00 00 00 00 00 00 00 [13640.094509] em28xx #0: EEPROM ID= 0x9567eb1a, EEPROM hash = 0x2bbf3bdd [13640.094512] em28xx #0: EEPROM info: [13640.094515] em28xx #0: AC97 audio (5 sample rates) [13640.094517] em28xx #0: 500mA max power [13640.094521] em28xx #0: Table at 0x24, strings=0x1e82, 0x186a, 0x0000 [13640.097391] em28xx #0: Identified as Hauppauge WinTV HVR 900 (R2) (card=18) [13640.099617] tveeprom 15-0050: Hauppauge model 65018, rev B2C0, serial# 1292061 [13640.099623] tveeprom 15-0050: tuner model is Xceive XC3028 (idx 120, type 71) [13640.099629] tveeprom 15-0050: TV standards PAL(B/G) PAL(I) PAL(D/D1/K) ATSC/DVB Digital (eeprom 0xd4) [13640.099634] tveeprom 15-0050: audio processor is None (idx 0) [13640.099637] tveeprom 15-0050: has radio [13640.112849] tuner 15-0061: Tuner -1 found with type(s) Radio TV. [13640.112877] xc2028 15-0061: creating new instance [13640.112882] xc2028 15-0061: type set to XCeive xc2028/xc3028 tuner [13640.115930] xc2028 15-0061: Loading 80 firmware images from xc3028-v27.fw, type: xc2028 firmware, ver 2.7 [13640.164057] xc2028 15-0061: Loading firmware for type=BASE MTS (5), id 0000000000000000. [13641.666643] xc2028 15-0061: Loading firmware for type=MTS (4), id 000000000000b700. [13641.693262] xc2028 15-0061: Loading SCODE for type=MTS LCD NOGD MONO IF SCODE HAS_IF_4500 (6002b004), id 000000000000b700. [13641.820765] Registered IR keymap rc-hauppauge [13641.820958] input: em28xx IR (em28xx #0) as /devices/pci0000:00/0000:00:1a.7/usb1/1-4/rc/rc2/input11 [13641.821335] rc2: em28xx IR (em28xx #0) as /devices/pci0000:00/0000:00:1a.7/usb1/1-4/rc/rc2 [13641.822256] em28xx #0: Config register raw data: 0xd0 [13641.824526] em28xx #0: AC97 vendor ID = 0xffffffff [13641.825503] em28xx #0: AC97 features = 0x6a90 [13641.825507] em28xx #0: Empia 202 AC97 audio processor detected [13641.899015] em28xx #0: v4l2 driver version 0.1.3 [13641.944064] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13643.470765] MTS (4), id 00000000000000ff: [13643.470776] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13643.717713] em28xx #0: V4L2 video device registered as video1 [13643.717718] em28xx #0: V4L2 VBI device registered as vbi0 [13643.718770] em28xx-audio.c: probing for em28xx Audio Vendor Class [13643.718775] em28xx-audio.c: Copyright (C) 2006 Markus Rechberger [13643.718778] em28xx-audio.c: Copyright (C) 2007-2011 Mauro Carvalho Chehab [13643.777148] xc2028 15-0061: attaching existing instance [13643.777154] xc2028 15-0061: type set to XCeive xc2028/xc3028 tuner [13643.777158] em28xx #0: em28xx #0/2: xc3028 attached [13643.777162] DVB: registering new adapter (em28xx #0) [13643.777167] DVB: registering adapter 0 frontend 0 (Micronas DRXD DVB-T)... [13643.777876] em28xx #0: Successfully loaded em28xx-dvb And here goes the lsmod output lsmod|grep em28xx em28xx_dvb 18579 0 dvb_core 110619 1 em28xx_dvb em28xx_alsa 18305 0 em28xx 109365 2 em28xx_dvb,em28xx_alsa v4l2_common 16454 3 tuner,tvp5150,em28xx videobuf_vmalloc 13589 1 em28xx videobuf_core 26390 2 em28xx,videobuf_vmalloc rc_core 26412 10 rc_hauppauge,ir_lirc_codec,ir_mce_kbd_decoder,ir_sony_decoder,ir_jvc_decoder,ir_rc6_decoder,ir_rc5_decoder,em28xx,ir_nec_decoder snd_pcm 97188 3 em28xx_alsa,snd_hda_intel,snd_hda_codec tveeprom 21249 1 em28xx videodev 98259 5 tuner,tvp5150,em28xx,v4l2_common,uvcvideo snd 78855 14 em28xx_alsa,snd_hda_codec_conexant,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device Isn't this driver mainline now? Or this card is not supported? Or the analog functionality is screwed? I need the analog capture working for this card. Please help!

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