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  • About to smash my keyboard!! Ubuntu 13.1 issues with AMD driver & Audio

    - by DNex
    Let me preface with saying that this is my 2nd day on Linux. I really want to make it work but these issues are driving me up the wall! I've done exhaustive google searches but have not been able to figure anything out. I am on Ubuntu 13.10, my graphics card is AMD Radeon HD4200. My sound card is a realtek HDMI. I've tried downloading and installing both drivers but nothing works. Graphics card: When I run the .run file (from http://www2.ati.com/drivers/legacy/amd-driver-installer-catalyst-13.1-legacy-linux-x86.x86_64.zip) I get an error. I check the fglrx-install log and it says this: Check if system has the tools required for installation. fglrx installation requires that the system have kernel headers. /lib/modules/3.11.0-12-generic/build/include/linux/version.h cannot be found on this system. One or more tools required for installation cannot be found on the system. Install the required tools before installing the fglrx driver. Optionally, run the installer with --force option to install without the tools. Forcing install will disable AMD hardware acceleration and may make your system unstable. Not recommended. Audio: Since my first install I've had no audio. I've tried everything outlined in this site: http://itsfoss.com/fix-sound-ubuntu-1304-quick-tip/ to no avail. I've download the linux drivers from Realtek HDMI audio but have had no luck. Any help would be extremely appreciated.

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  • Why are my videos playing speeded up with no audio, but work fine if I log in as a guest?

    - by Martins Kruze
    Since the start of this week I have been experiencing a glitch in the multimedia on my Samsung R518 laptop. I have 2 problems: Videos in every player are speeded up around 2 or 4 times (including youtube.com (both HTML5 and flash variants), any other video on the web and videos on my laptop played by Totem Media Player), exception is VLC player, but 2nd problem does concern even that. There is no sound - simple as that (with or without headphones plugged in). These all problems are now, and has not seen before, I upgraded to Ubuntu 10.10 after it was possible, and from start I didn't have anything from this - it just started in this week. I haven't even putted new software in. I have more or less solved the question (kind of) - I just logged in as a guest - and it all works, but when I make a new user - it does not. Please help me. Some stats below: sudo lshw -c sound *-multimedia description: Audio device product: RV710/730 vendor: ATI Technologies Inc physical id: 0.1 bus info: pci@0000:01:00.1 version: 00 width: 32 bits clock: 33MHz capabilities: pm pciexpress msi bus_master cap_list configuration: driver=HDA Intel latency=0 resources: irq:48 memory:cfeec000-cfeeffff *-multimedia description: Audio device product: 82801I (ICH9 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 03 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=HDA Intel latency=0 resources: irq:47 memory:fc200000-fc203fff sudo lshw -c video *-display description: VGA compatible controller product: M92 LP [Mobility Radeon HD 4300 Series] vendor: ATI Technologies Inc physical id: 0 bus info: pci@0000:01:00.0 version: 00 width: 32 bits clock: 33MHz capabilities: pm pciexpress msi vga_controller bus_master cap_list rom configuration: driver=radeon latency=0 resources: irq:46 memory:d0000000-dfffffff ioport:2000(size=256) memory:cfef0000-cfefffff memory:cfe00000-cfe1ffff

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  • Ubuntu 13.04 Sound Problem after following weird commands

    - by user206356
    After launching a few commands : echo autospawn = no >> ~/.config/pulse/client.conf #use ~/.pulse/client.conf on Ubuntu <= 12.10 killall pulseaudio $LANG=C pulseaudio -vvvv --log-time=1 > ~/pulseverbose.log 2>&1 My sound does not work. (just with the speakers, with headphones it works but I can not change the volume) The sound icon on the top right corner does show a speaker with a single non continuous line. I can not change the volume; it is frozen. There can be an extremely low output of the sound (I hear something but I am not sure...) It does not show a single output device that is avalaible, not even the "dummie". I have tried to reset pulseaudio, alsa, remove it, purging it, reinstalling it, without having success. EDIT: I have tried launching pulseaudio via the terminal. It worked :D However, I am very surprised why it does not automatically start at the start of the computer. Any ideas ? Here the console output : W: [pulseaudio] authkey.c: Failed to open cookie file '/home/simonm/.config/pulse/cookie': No such file or directory W: [pulseaudio] authkey.c: Failed to load authorization key '/home/simonm/.config/pulse/cookie': No such file or directory W: [pulseaudio] authkey.c: Failed to open cookie file '/home/simonm/.pulse-cookie': No such file or directory W: [pulseaudio] authkey.c: Failed to load authorization key '/home/simonm/.pulse-cookie': No such file or directory

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  • Android Mediaplayer: setDataSource issue for downloaded media file

    - by Erik
    I have an application that will record and play audio files. Some of the audio files are downloaded using simple standard http downloads using httpclient. It worked like a charm for a long time. Now all of a sudden I cannot play the files I download. It fails with this stack. I store the files on the SDCard and I experience the problem both on a handset and a USB connected device. I have checked that the downloaded file is cool on the server, and I can play it without any issues. These are the code snippets I use ( I know that recordingFile is a valid path for the file). // inside the activity class private void playRecording() throws IOException{ File recordingFile = new File(recordingFileName); FileInputStream recordingInputStream = new FileInputStream(recordingFile); audioMediaPlayer.playAudio(recordingInputStream); } Here is the media player code: // inside my media player class which handles the recordings public void playAudio(FileInputStream audioInputStream) throws IOException { mediaPlayer.reset(); mediaPlayer.setDataSource(audioInputStream.getFD()); mediaPlayer.prepare(); mediaPlayer.start(); } Here is the exception: E/MediaPlayerService( 555): offset error E/MediaPlayer( 786): Unable to to create media player W/System.err( 786): java.io.IOException: setDataSourceFD failed.: status=0x80000000 W/System.err( 786): at android.media.MediaPlayer.setDataSource(Native Method) W/System.err( 786): at android.media.MediaPlayer.setDataSource(MediaPlayer.java:632) W/System.err( 786): at net.xxx.xxx.AudioMediaPlayer.playAudio(AudioMediaPlayer.java:69) W/System.err( 786): at net.xxx.xxx.Downloads.playRecording(Downloads.java:299) W/System.err( 786): at net.xxx.xxx.Downloads.access$0(Downloads.java:294) W/System.err( 786): at net.xxx.xxx.Downloads$1.onClick(Downloads.java:135) I have tried seeking some answer of the offset error, but not really clear what this issue might be. PS I download the file with this code: public FileOutputStream executeHttpGet(FileOutputStream fileOutputStream) throws ClientProtocolException, IOException{ try { // Execute HTTP Post Request httpResponse = httpClient.execute(httpPost, localContext); int status = httpResponse.getStatusLine().getStatusCode(); // we assume that the response body contains the error message if (status != HttpStatus.SC_OK) { ByteArrayOutputStream ostream = new ByteArrayOutputStream(); httpResponse.getEntity().writeTo(ostream); fileOutputStream = null; } else { InputStream content = httpResponse.getEntity().getContent(); byte[] buffer = new byte[1024]; int len = 0; while ( (len = content.read(buffer)) > 0 ) { fileOutputStream.write(buffer,0, len); } fileOutputStream.close(); content.close(); // this will also close the connection } } catch (ClientProtocolException e1) { // TODO Auto-generated catch block e1.printStackTrace(); fileOutputStream = null; } catch (IOException e2) { // TODO Auto-generated catch block e2.printStackTrace(); fileOutputStream = null; } return fileOutputStream; }

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  • How are VST Plugins made?

    - by user146780
    I would like to make (or learn how to make) VST plugins. Is there a special SDK for this? how does one yield a .vst instead of a .exe? Also, if one is looking to make Audio Units for Logic Pro, how is that done? Thanks

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  • iPhone 3.5mm jack based application

    - by maverick
    I want to encode data via a DTMF encoder and send it back to the iPhone via the 3.5mm Jack. Is it possible to send data back into the 3.5mm jack. conventionally audio signals are sent out over the iPhone 3.5mm jack? Is there provision to deal with DTMF and 3.5mm jack based input applications in Iphone's External Accessory framework?

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  • getting error while converting wav to amr using ffmpeg

    - by sohilvassa
    hello friends I am using ffmpeg to convert amr to wav and wav to amr.Its successfully converting amr to wav but not viceversa. As ffmpeg is supporting amr encoder decoder, its giving error. ffmpeg -i testwav.wav audio.amr (working fine) Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height

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  • iPhone: how to make key click sound for custom keypad?

    - by Kaffeine Coma
    Is there a way to programmatically invoke the keypad "click" sound? My app has a custom keypad (built out of UIButtons) and I'd like to provide some audio feedback when the user taps on the keys. I tried creating my own sounds in Garageband, but wasn't happy with any of my creations. If there isn't a standard way to invoke the key click, can anyone point me to a library of sounds that might have such a gem?

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  • PlaySystemSound with mute switch on

    - by Sam V
    I know, I have to set the AudioSession to the 'playback' category, which allows audio even when the mute switch is on. This is what I do, but sound still gets muted when switch is on. UInt32 sessionCategory = kAudioSessionCategory_MediaPlayback; AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,sizeof(sessionCategory), &sessionCategory); SystemSoundID soundID; NSString *path = [[NSBundle mainBundle] pathForResource:soundString ofType:@"wav"]; AudioServicesCreateSystemSoundID((CFURLRef)[NSURL fileURLWithPath:path],&soundID); AudioServicesPlaySystemSound (soundID);

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  • Blackberry buffered playback demo??

    - by Bohemian
    Can someone help me to buffer a mp3 file on a server using the Blackberry buffered pllayback demo app provided with the jde? I hav loaded it in the simulator. And my mds is started but I m unable to play the audio. There is no error but it doesnt play/load. The code looks all fine. Thanks

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  • Creating a music catalog in C# and extracting first 30 seconds as soon as the first words are sung

    - by Rad
    I already read a question: Separation of singing voice from music. I don’t need this complex audio processing. I only need some detection mechanism that would detect that there is some voice/vocal playing while the music is playing (or not playing) I need to extract first 30 seconds when a vocalist starts singing along with full band music. See question 2 below. I want to create a music catalog using ASP.NET MVC 2 and Silverlight clients and C#.NET 4.0 programming language that would be front store. On the backend I would also like to create a desktop WPF/Windows application to create the music catalog from already existing music files, most of which have metadata in them ID3v1, ID3v2.3, ID3v2.4, iTunes MP4, WMA, Vorbis Comments and APE Tags etc. I would possibly like to create a web service that would allow catalog contributors to upload a zipped album and trigger metadata extraction of music data and extraction of music segments as described below. I would be happy if I achieve no. 1 below. Let's say I have 1000ths of songs in mp3 (or other formats) grouped in subfolders using some classification (Genre, Artists, Albums, Composers or other groupings). I want to create tables in DB that would organize songs so they can be searched based on different criteria (year, length, above classification or by song title, description etc) like what iTune store allows to their customers. I want to extract metadata from various formats (I will try to get songs in mp3 format, but there may be other popular formats) and allow music Catalog manager person to add missing data from either desktop or web applications. He or other contributors can upload zipped music via an HTML or Silverlight upload or WPF. Can anybody suggest open source libraries, articles, code snippets that can do that in an automatic way using .NET and possibly SQL Server DB? My main questions are these. This is an audio processing challenge. I want to extract 2 segments of music (questions 1 and 2): 1. How to extract a music segment: 1-2 seconds before a vocal starts singing and up to 30 seconds from that point in time and 2. Much more challenging is to find repeating segments (One would usually find or recognize the names of the songs and songs are usually known by these refrains. How would I go about creating a list of songs that go great together like what Genius from iTune does? Is there any characteristics of music that can be used to match songs? The goal is for people quickly scan and recognize songs i.e. associate melody, words with a title/album so they can make intelligent decisions like buying a song, create similar mood songs. Thanks, Rad

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  • I write bad wave files using Java

    - by Cliff
    I'm writing out wave files in Java using AudioInputStream output = new AudioInputStream(new ByteArrayInputStream(rawPCMSamples), new AudioFormat(22000,16,1,true,false), rawPCMSamples.length) AudioSystem.write(output, AudioFileFormat.Type.WAVE, new FileOutputStream('somefile.wav')) And I get what appears to be corrupt wave files on OSX. They won't play from Finder however using the same code behind a servlet writing directly to the response stream and setting the Content-Type to audio/wave seems to play fine in quicktime. What gives?

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  • HOW-TO Make computer sing

    - by Ofir
    Hi, I'm trying to develop an online application where the user writes some text and the software sings it back to the user. I can currently generate the audio file with the words spoken by the computer using espeak, but I have no idea how to make it sound like a song, how to add rhythm to it. I'm able to change the pitch and tempo using rubberband, but that's as far as I've gotten. Does anyone have a clue how to make this happen?

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  • calculating time duration of a file

    - by RV
    Dupe of calculate playing time of a .mp3 file im reading a audio file(for ex:wav,mp3 etc) and get a long value as duration.now i want to convert that long value into correct time duration(like,00:05:32)

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  • How to programmatically generate an MP3 podcast file with chapters and text track?

    - by adib
    Hi Anybody know how to programmatically generate MP3 files with bookmarks that can be used in iTunes / iPod / iPhone / iPod touch? Specifically text bookmarks (bookmarks with titles) that the listener can skip to a specific point in time in the audio file. Also how to add the text transcription of the podcast's content. Even better if you have an example Cocoa code or library to write the MP3 file. Thanks.

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  • AudioConverterConvertBuffer problem with insz error

    - by Samuel
    Hi Codegurus, I have a problem with the this function AudioConverterConvertBuffer. Basically I want to convert from this format _ streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked |0 ; _streamFormat.mBitsPerChannel = 16; _streamFormat.mChannelsPerFrame = 2; _streamFormat.mBytesPerPacket = 4; _streamFormat.mBytesPerFrame = 4; _streamFormat.mFramesPerPacket = 1; _streamFormat.mSampleRate = 44100; _streamFormat.mReserved = 0; to this format _streamFormatOutput.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked|0 ;//| kAudioFormatFlagIsNonInterleaved |0; _streamFormatOutput.mBitsPerChannel = 16; _streamFormatOutput.mChannelsPerFrame = 1; _streamFormatOutput.mBytesPerPacket = 2; _streamFormatOutput.mBytesPerFrame = 2; _streamFormatOutput.mFramesPerPacket = 1; _streamFormatOutput.mSampleRate = 44100; _streamFormatOutput.mReserved = 0; and what i want to do is to extract an audio channel(Left channel or right channel) from an LPCM buffer based on the input format to make it mono in the output format. Some logic code to convert is as follows This is to set the channel map for PCM output file SInt32 channelMap[1] = {0}; status = AudioConverterSetProperty(converter, kAudioConverterChannelMap, sizeof(channelMap), channelMap); and this is to convert the buffer in a while loop AudioBufferList audioBufferList; CMBlockBufferRef blockBuffer; CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampBuffer, NULL, &audioBufferList, sizeof(audioBufferList), NULL, NULL, 0, &blockBuffer); for (int y=0; y<audioBufferList.mNumberBuffers; y++) { AudioBuffer audioBuffer = audioBufferList.mBuffers[y]; //frames = audioBuffer.mData; NSLog(@"the number of channel for buffer number %d is %d",y,audioBuffer.mNumberChannels); NSLog(@"The buffer size is %d",audioBuffer.mDataByteSize); numBytesIO = audioBuffer.mDataByteSize; convertedBuf = malloc(sizeof(char)*numBytesIO); status = AudioConverterConvertBuffer(converter, audioBuffer.mDataByteSize, audioBuffer.mData, &numBytesIO, convertedBuf); char errchar[10]; NSLog(@"status audio converter convert %d",status); if (status != 0) { NSLog(@"Fail conversion"); assert(0); } NSLog(@"Bytes converted %d",numBytesIO); status = AudioFileWriteBytes(mRecordFile, YES, countByteBuf, &numBytesIO, convertedBuf); NSLog(@"status for writebyte %d, bytes written %d",status,numBytesIO); free(convertedBuf); if (numBytesIO != audioBuffer.mDataByteSize) { NSLog(@"Something wrong in writing"); assert(0); } countByteBuf = countByteBuf + numBytesIO; But the insz problem is there... so it cant convert. I would appreciate any input Thanks in advance

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