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  • Playing audio from a wav file in iPhone SpeakHere example

    - by Mo
    I'm working with the iPhone SpeakHere example, and I would like to be able to play audio from either the mic (as in the example) or from a wav file. I have working code to play from a particular wav file, which looks like this: NSString *path = [[NSBundle mainBundle] pathForResource:@"basketBall" ofType:@"wav"]; AVAudioPlayer* theAudio=[[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:path] error:NULL]; theAudio.delegate = self; [theAudio play]; So I'm fine with actually getting the wav to play in the application (I can hook it up to a button, etc.) but I would like it to also behave the same way pushing the "Play" button does after recorded speech, in that it should be connected to the same visualization (which I have modified quite a bit, but essentially shows the current volume, among other things). Thanks for your help!

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  • Linux, C++ audio capturing (just microphone) library

    - by TheOm3ga
    I'm developing a musical game, it's like a singstar but instead of singing, you have to play the recorder. It's called oFlute, and it's still in early development stage. In the game, I capture the microphone input, then run a simple FFT analysis and compare the results to typical recorder's frequencies, thus getting the played note. At the beginning, the audio library I was using was RtAudio, but I don't remember why I switched to PortAudio, which is what I'm currently using. The problem is that, from time to time, either it crashes randomly or stops capturing, like if there were no sound coming from the microphone. My question is, what's the best option to capture microphone input on Linux? I just need to open, read, and close a flow of bytes from the microphone. I've been reading this guide, and (un)surprisingly it says: I don't think that PortAudio is very good API for Unix-like operating systems. So, what do you recommend me?

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  • Online audio stream using ruby on rails

    - by Avdept
    I'm trying to write small website that can stream audio online(radio station) and got few questions: 1. Do i have to index all my music files into database, or i can randomily pick file from file system and play it. 2. When should i use ajax to load new song(right after last finished, or few seconds before to get responce from server with link to file?) 3. Is it worth to use ajax, or better make list, that will play its full time and then start over?

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  • How to play audio file ios

    - by Camus
    I am trying to play an audio file but I can get it working. I imported the AVFoundation framework. Here is the code: NSString *fileName = [[NSBundle mainBundle] pathForResource:@"Alarm" ofType:@"caf"]; NSURL *url = [[NSURL alloc] initFileURLWithPath:fileName]; NSLog(@"Test: %@ ", url); AVAudioPlayer *audioFile = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:NULL]; audioFile.delegate = self; audioFile.volume = 1; [audioFile play]; I am receiving an error nil string parameter I copied the file to the supporting files folder so the file is there. Can you guys help me? Thanks

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  • Audio File continues to play even on leaving the view

    - by Swastik
    What I am doing is -(void)viewWillAppear:(BOOL)animated{ [NSTimer scheduledTimerWithTimeInterval:0.3 target:self selector:@selector(clickEvent:) userInfo:nil repeats:YES]; } -(void)clickEvent:(NSTimer *)aTimer{ NSDate* finishDate = [NSDate date]; if([finishDate timeIntervalSinceDate: self.startDate] 11 && touched == NO){ NSString *mp3Path = [[[NSBundle mainBundle] resourcePath] stringByAppendingPathComponent:@"test.mp3"]; [self playMusicFile:mp3Path]; NSLog(@"Timer from First Page"); [aTimer invalidate]; //[touchCheckTimer release]; aTimer = nil; } else{ } -(void)playMusicFile:(NSString *)mp3Path{ NSURL *mp3Url = [NSURL fileURLWithPath:mp3Path]; NSError *err; AVAudioPlayer *audPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:mp3Url error:&err]; [self setAudioPlayer1:audPlayer]; if(audioPlayer1) [audioPlayer1 play]; [audPlayer release]; } Now, on pushing another view this audio file keeps playing in the background. Please help!

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  • iphone - Images (slide show) and audio snychronization

    - by Qaiser
    I have 20 images and some audio. I would like to show a single image at a time and change the images at (unequal) intervals. For example, I want to show image 1 for 1.44 seconds and image 2 for 1.67 seconds and so on. Can someone suggest how to go about doing this please? What I have seen are examples that show how to setup an array of images with one field that denotes total time. This causes the images to show for an equal amount of time (each). ... and that not what I am looking for ...

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  • Background audio not working in windows 8 store / metro app

    - by roryok
    I've tried setting background audio through both a mediaElement in XAML <MediaElement x:Name="MyAudio" Source="Assets/Sound.mp3" AudioCategory="BackgroundCapableMedia" AutoPlay="False" /> And programmatically async void setUpAudio() { var package = Windows.ApplicationModel.Package.Current; var installedLocation = package.InstalledLocation; var storageFile = await installedLocation.GetFileAsync("Assets\\Sound.mp3"); if (storageFile != null) { var stream = await storageFile.OpenAsync(Windows.Storage.FileAccessMode.Read); _soundEffect = new MediaElement(); _soundEffect.AudioCategory = AudioCategory.BackgroundCapableMedia; _soundEffect.AutoPlay = false; _soundEffect.SetSource(stream, storageFile.ContentType); } } // and later... _soundEffect.Play(); But neither works for me. As soon as I minimise the app the music fades out

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  • Record/Playback with AudioQueue on iPhone

    - by Biranchi
    Hi, I am currently using Audio Queues on the iPhone to record and playback audio. What I would like to be able to do is to record some audio, allow the user to pause the record queue, and to seek back and forward through the audio to select a position from where they can start recording from again. I have got over the seeking issue by making the playback AudioQueueBuffer sizes small enough so that the play audio queue callback happens at a rate that allows the user to use a slider control to hear the audio as they adjust the slider back and forth. I think I can achieve the recording at a new position by setting the inStartingPacket parameter of the AudioFileWritePackets function that I call from the Audio Recording Queue callback. The trouble is this only inserts audio over the previously recorded audio. The file length obviously doesn't change so if the user were to go backwards and record less audio than before, the old audio still remains after the end of the newly recorded audio. Is there a way I can get the AudioFile to truncate at the point the user starts to insert the new audio, is there some other way I can remove the old audio starting at the insert position or is there a better way about going about this task? 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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  • Using Audio Queue Services to play PCM data over a socket connection

    - by Rohan
    I'm writing a remote desktop client for the iPhone and I'm trying to implement audio redirection. The client is connected to the server over a socket connection, and the server sends 32K chunks of PCM data at a time. I'm trying to use AQS to play the data and it plays the first two seconds (1 buffer worth). However, since the next chunk of data hasn't come in over the socket yet, the next AudioQueueBuffer is empty. When the data comes in, I fill the next available buffer with the data and enqueue it with AudioQueueEnqueueBuffer. However, it never plays these buffers. Does the queue stop playing if there are no buffers in the queue, even if you later add a buffer? Here's the relevant part of the code: void wave_out_write(STREAM s, uint16 tick, uint8 index) { if(items_in_queue == NUM_BUFFERS){ return; } if(!playState.busy){ OSStatus status; status = AudioQueueNewOutput(&playState.dataFormat, AudioOutputCallback, &playState, CFRunLoopGetCurrent(), NULL, 0, &playState.queue); if(status == 0){ for(int i=0; i<NUM_BUFFERS; i++){ AudioQueueAllocateBuffer(playState.queue, 40000, &playState.buffers[i]); } AudioQueueAddPropertyListener(playState.queue, kAudioQueueProperty_IsRunning, MyAudioQueuePropertyListenerProc, &playState); status = AudioQueueStart(playState.queue, NULL); if(status ==0){ playState.busy = True; } else{ return; } } else{ return; } } playState.buffers[queue_hi]->mAudioDataByteSize = s->size; memcpy(playState.buffers[queue_hi]->mAudioData, s->data, s->size); AudioQueueEnqueueBuffer(playState.queue, playState.buffers[queue_hi], 0, 0); queue_hi++; queue_hi = queue_hi % NUM_BUFFERS; items_in_queue++; } void AudioOutputCallback(void* inUserData, AudioQueueRef outAQ, AudioQueueBufferRef outBuffer) { PlayState *playState = (PlayState *)inUserData; items_in_queue--; } Thanks!

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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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  • 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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  • 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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  • 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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  • Sound card / microphone impedance mismatch

    - by axk
    First of all I'm not completely sure this is impedance mismatch, but from what I found on the Internet I believe it is. It seems to be a common problem. The question is not as much about solving the problem as about why it is happening (if I'm right about the cause of the problem, of course). I had this quiet microphone problem with several built in cards and microphones and now with a Creative Audigy SE. There's a microphone boost option which introduces a lot of noise with volume increase, but even this doesn't seem to give loud enough sound in some cases. The mic on my current headphones is very quiet with Audigy SE without the boost but is very loud and low noise with an external Sound Blaster Connect. So the question is have I just been unlucky with my sound cards and microphones or is it a common problem? And if it is a common problem why is it so difficult for the vendors to standardize on the sound card / microphone impedance? Edit: the OS is Windows (XP/7), but I don't believe it is OS-specific.

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  • Make headphone output mono.

    - by Jonathan
    my headphones are stereo but I would like the sound from the left and right to be combined then sent to both headphones. The reason is I'm watching a video where the people speaking are in the right ear as well as the music but they never speak in the left ear (it is not because they on the right side of the screen) If I take the right headphone off then I only hear the music in my left and there is no speaking.

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