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  • Playing audio stream not showing in pavucontrol

    - by user168505
    My Pulse Audio Volume Control (pavucontrol) is not shown volume bar which present below the volume control slider for right and left channels in the Play tab . However, I can hear audio,and the pavucontrol Play tab shows the name of the application(any media player vlc,mplayer etc.) which is running and volume control for right and left channels (FROM LEFT & FROM RIGHT Volume slider) I guess there may be a change in the system configuration/setting? How to reset it? I have reinstalled Pulseaudio, but the problem remains. I am using ubuntu 12.04 with default pulse audio.

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  • Ubuntu Audio Pitch Shifting filter

    - by user777305
    I'm currently developing a video player software which intended to be an embeedded player. I'm using Java with VLCJ library for the video player. What i'm looking is a way, something to transform the audio output to make the output sound as oldman or a kid (pitch shifting, i guess is the name). VLC have this when enabling time stretch, but the video play speed is affected (slower to get oldman sound, but fast-forward-alike to get kid voice effect. Is there any solutions for this? I don't find this feature on VLC(J), so i think what i need is the audio output of the ubuntu itself (Ubuntu 12.04) to do this job. something like filtering audio output system wide. Any software or setting to do this? it also need to be controllable via command line to provide realtime effect changing. Thanks in advance.

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  • Finding out if a FLAC or WAVPACK audio file is NOT originally encoded from a lossy source

    - by cornel
    Is there a way of checking that the so-called FLAC or WAVPACK audio file was originally encoded from a lossless source (WAV, CDA, APE, etc.) instead of a lossy source (MP3, AAC, ATRAC, etc.)? Say I have a lossy MP3 audio file (5.17Mb, 87% compressed from its original, source unknown). I then encode it to another lossless format, say FLAC or WAVPACK. The size increases (23.14Mb, 39% compressed from its original, source MP3)! ID tags, etc, remain the same and there's no way of checking the integrity of its origin. How do I go about doing that?

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  • How do I make my Geforce GTS 250's power save mode stop causing audio stuttering?

    - by Matt
    Whenever my GTS 250 enters its power save mode, downscaling its frequencies, my audio stutters. This affects both my onboard audio and my Audigy Soundblaster 2 ZS. Changing Windows power save mode options such as PCI-E link state power management or Power Management Mode in the nVidia control panel have no effect on this issue. Replacing the power supply had no effect on this issue. The BIOS is the latest version, and I have the latest motherboard chipset and graphics drivers installed. I do not overclock. I started to see this issue after I upgraded my rig from its Socket 939 board to a Socket 1156 board with a Core i5-750 while simultaneously upgrading from Vista to 7.

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  • How do I swap audio output of the left and right speakers?

    - by Manga Lee
    I have two speakers stereo speakers but when I use the sound control panel applet to test my audio configuration I get sound in the right speaker when the user interface indicates the right speaker and vice versa. Is there a way to swap the audio output from left to right and right to left? UPDATE: The reason for this question is that I've recently rearranged my workspace and because of physical constraints the left speaker has to go on the right side and vice versa. I could of course solve this problem with a hardware solution but I'd rather use a software solution if one is available.

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  • How can you convert audio (3.5mm) to S/PDIF?

    - by SSumner
    I have a monitor that I want to use as my 'TV' for my gaming system. I connect it via HDMI, so sound and video go through the monitor, but I want sound to my headphones, which travels via optical (S/PDIF) cable. The monitor (Dell U2713HM) has a 3.5mm audio jack on it for line out, but I couldn't find anything that simply plugs in an converts the analog audio signal to a digital one so I can plug in a S/PDIF cable. What sort of device do I need to do this? (I am not asking for shopping recommendations, merely what options allow this conversion. I would prefer the smallest option, as space is limited).

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  • Anyone experiencing audio issues with VirtualBox on Linux and has a solution?

    - by DoxaLogos
    I've been using Virtualbox (now at 3.0.2) on Kubuntu (now at 9.04) for a while now, and I seem to have a problem when running Windows. Sometime after a while the audio will cut out in Kubuntu. The only way I can get it to recover is to make sure VirtualBox is completely shutdown and either going into multimedia under "system settings" and test the audio or restart. I'm wondering if anyone else here has experienced similar issues and has come up with a more elegant solution. I can't seem to find a reasonable one at virtualbox.org.

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  • Audio recording and playback using NAudio

    - by Egon
    When I try recording audio following this here, and using the related source code from here in the (voicerecorder.audio project), I face the problem that the recording stops within moments of starting and so does the playback. the recorded file is 46 bytes, and playback is there for less than a second, basically, the thread jumps to the next statement immediately. If I try to sleep the thread, it doesn't help. What can I do to keep it in the recording mode or playback mode until user interrupt ?

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  • does ipad support voice recording???

    - by Abdullah Waseer
    Hi, i have made an application in iphone which supports video recording and i want it to launch on iPad too but currently i dont know whether ipad device supports voice recording or not. can anyone please tell me whether IPAD SUPPORTS VOICE RECORDING??? thanks...

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  • I just don't get AudioFileReadPackets

    - by Eric Christensen
    I've tried to write the smallest chunk of code to narrow down a problem. It's now just a few lines and it doesn't work, which makes it pretty clear that I have a fundamental misunderstanding of how to use AudioFileReadPackets. I've read the docs and other examples online, and apparently I'm just not getting. Could you explain it to me? Here's what this block should do: I've previously opened a file. I want to read just one packet - the first one of the file - and then print it. But it crashes on the AudioFileReadPackets line: AudioFileID mAudioFile2; AudioFileOpenURL (audioFileURL, 0x01, 0, &mAudioFile2); UInt32 *audioData2 = (UInt32 *)malloc(sizeof(UInt32) * 1); AudioFileReadPackets(mAudioFile2, false, NULL, NULL, 0, (UInt32*)1, audioData2); NSLog(@"first packet:%i",audioData2[0]); (For clarity, I've stripped out all error handling.) It's the AFRP line that crashes out. (I understand that the third and fourth argument are useful, and in my "real" code, I use them, but they're not required, right? So NULL in this case should work, right?) So then what's going on? Any guidance would be much appreciated. Thanks.

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  • record output sound in python

    - by aaronstacy
    i want to programatically record sound coming out of my laptop in python. i found PyAudio and came up with the following program that accomplishes the task: import pyaudio, wave, sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = sys.argv[1] p = pyaudio.PyAudio() channel_map = (0, 1) stream_info = pyaudio.PaMacCoreStreamInfo( flags = pyaudio.PaMacCoreStreamInfo.paMacCorePlayNice, channel_map = channel_map) stream = p.open(format = FORMAT, rate = RATE, input = True, input_host_api_specific_stream_info = stream_info, channels = CHANNELS) all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) stream.close() p.terminate() data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() the problem is i have to connect the headphone jack to the microphone jack. i tried replacing these lines: input = True, input_host_api_specific_stream_info = stream_info, with these: output = True, output_host_api_specific_stream_info = stream_info, but then i get this error: Traceback (most recent call last): File "./test.py", line 25, in data = stream.read(chunk) File "/Library/Python/2.5/site-packages/pyaudio.py", line 562, in read paCanNotReadFromAnOutputOnlyStream) IOError: [Errno Not input stream] -9975 is there a way to instantiate the PyAudio stream so that it inputs from the computer's output and i don't have to connect the headphone jack to the microphone? is there a better way to go about this? i'd prefer to stick w/ a python app and avoid cocoa.

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  • getAudioInputStream can not convert [stereo, 4 bytes/frame] stream to [mono, 2 bytes/frame]

    - by brian_d
    Hello. I am using javasound and have an AudioInputStream of format PCM_SIGNED 8000.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian Using AudioSystem.getAudioInputStream(target_format, original_stream) produces an 'IllegalArgumentException: Unsupported Conversion' when the target_format is PCM_SIGNED 8000.0 Hz, 16 bit, mono, 2 bytes/frame, little-endian Is it possible to convert this stream manually after every read() call? And if yes, how? In general, how can you compare two formats and tell if a conversion is possible?

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  • AudioTrack skipping after pause and resume

    - by Markus Drösser
    Hi, here is the problem. I play a wav file that i recorded earlier without problems. but when i call audiotrack.pause() and audiotrack.start() again after some waiting, it skips some frames of the file. why is that? here is my play listener // Start playback audioTrack.setPlaybackPositionUpdateListener(new OnPlaybackPositionUpdateListener() { @Override public void onPeriodicNotification(AudioTrack track) { try { if(ramfile!=null && ramfile.read(buffer)==-1) { audioTrack.release(); audioTrack = null; ramfile.close(); playing=false; } else { audioTrack.write(buffer, 0, buffer.length); } } catch (IOException e) { try { ramfile.close(); playing=false; } catch (IOException e1) { } } } @Override public void onMarkerReached(AudioTrack track) { playing=false; track.release(); } });

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  • Un groupe de développeurs sort Flac.js, un décodeur JavaScript pour la lecture du contenu audio dans le navigateur sans recours aux codecs

    Un groupe de développeurs sort Flac.js un décodeur audio en JavaScript pour la lecture du contenu audio dans le navigateur sans nécessiter de codecs HTML5, le futur standard du Web introduit la balise audio permettant de créer des applications fournissant le traitement et la synthèse audio dans le navigateur. Les navigateurs récents comme Chrome ou Firefox, intègrent déjà des bibliothèques Javascript qui fournissent des méthodes et propriétés permettant de manipuler l'élément audio. Cependant, les applications HTML 5 manipulant du contenu audio qui fonctionnent normalement dans un navigateur sur un système d'exploitation donné pourraient ne pas marcher correctement lors de...

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  • Audio doesn't work on Windows XP guest (WS 7.0)

    - by Mads
    Hi, I can't get audio to work with on a Windows XP guest running on VMware Workstation 7.0 and Ubuntu 9.10 host. Windows fails to produce any audio output and the Windows device manager says the Multimedia Audio Controller is not working properly. Audio is working fine in the host OS. When I open Multimedia Audio Controller properties it says: Device status: The drivers for this device are not installed (Code 28) If I try to reinstall the driver I get the following error message: "Cannot Install this Hardware There was a problem installing this hardware: Multimedia Audio Controller An Error occurred during the installation of the device Driver is not intended for this platform" Has anyone else experienced this problem?

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  • How to record tv to network share with Windows Media Center?

    - by Peterdk
    Well, you would think that Windows 7's new MediaCenter would be up to the task of recording your TV to a network share/drive. Too bad, it looks like it's just not possible. I have a windows 2008 R2 server, and a Windows 7 machine with a TV card. Since my server has 2TB of storage, it would be nice to record directly to it's networked drive. (I mounted it as Z:). I tried the following: Selecting it in Media Center Itself: Not working. Not available. Editing the registry: HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\Media Center\Service\Recording , setting RecordPath to Z:\TV. Not working. Editing the registry: setting RecordPath to \\server\TV. Not working. Creating a Symlink (mklink \D) to Z:\TV and \\server\TV and setting that in the registry as RecordPath. Currently I am out of options. I could ofcourse Install Windows7 on my server, but I have no license for that, and my windows 2008 r2 is free from dreamspark. Are there people that are succesfully recording to a networked drive/storage? edit I also need to mention that I need to be able to acces the stored files from other PC's, like my laptop. So iSCSI is great for recording, but it looks like you can't access iSCSI devices from multiple PC's. Looks like sharing a iSCSI device is out of the question, so: Are there workarounds to get this thing recording to my network drive?

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  • Bsplayer - load audio tracks from external files

    - by torran
    I have a movie file: Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : [email protected] Format settings, CABAC : Yes Format settings, ReFrames : 5 frames Muxing mode : Container [email protected] Codec ID : V_MPEG4/ISO/AVC Duration : 54mn 13s Bit rate : 3 380 Kbps Nominal bit rate : 3 459 Kbps Width : 1 280 pixels Height : 720 pixels Display aspect ratio : 16:9 Frame rate : 23.976 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.153 Stream size : 1.28 GiB (88%) Writing library : x264 core 88 r1471 1144615 Audio ID : 2 Format : AC-3 Format/Info : Audio Coding 3 Codec ID : A_AC3 Duration : 54mn 16s Bit rate mode : Constant Bit rate : 384 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Stream size : 149 MiB (10%) and additional audio files in same folder: .mp3 and .ac3. How can I load them with bsplayer? Right click-audio-audio streams is empty. If i open the movie with media players classic I can switch audio files.

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  • How do you setup the Audio plugin for Flowplayer?

    - by codeninja
    I'm having a bit of trouble getting the Audio player to work. Basically I want to initiate an mp3 player doing something like this <a href="path-to-my-audio.mp3" id="player" ></a> and then use the $f() call to initate the player. I've followed the instructions here (http://flowplayer.org/plugins/streaming/audio.html) This doesnt seem to be work and I'm not sure what's wrong because I'm able to play videos in this way. Thanks for your help!

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  • Core Audio on iPhone - any way to change the microphone gain (either for speakerphone mic or headpho

    - by Halle
    After much searching the answer seems to be no, but I thought I'd ask here before giving up. For a project I'm working on that includes recording sound, the input levels sound a little quiet both when the route is external mic + speaker and when it's headphone mic + headphones. Does anyone know definitively whether it is possible to programmatically change mic gain levels on the iPhone in any part of Core Audio? If not, is it possible that I'm not really in "speakerphone" mode (with the external mic at least) but only think I am? Here is my audio session init code: OSStatus error = AudioSessionInitialize(NULL, NULL, audioQueueHelperInterruptionListener, r); [...some error checking of the OSStatus...] UInt32 category = kAudioSessionCategory_PlayAndRecord; // need to play out the speaker at full volume too so it is necessary to change default route below error = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category); if (error) printf("couldn't set audio category!"); UInt32 doChangeDefaultRoute = 1; error = AudioSessionSetProperty (kAudioSessionProperty_OverrideCategoryDefaultToSpeaker, sizeof (doChangeDefaultRoute), &doChangeDefaultRoute); if (error) printf("couldn't change default route!"); error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioRouteChange, audioQueueHelperPropListener, r); if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", (int)error); UInt32 inputAvailable = 0; UInt32 size = sizeof(inputAvailable); error = AudioSessionGetProperty(kAudioSessionProperty_AudioInputAvailable, &size, &inputAvailable); if (error) printf("ERROR GETTING INPUT AVAILABILITY! %d\n", (int)error); error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioInputAvailable, audioQueueHelperPropListener, r); if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", (int)error); error = AudioSessionSetActive(true); if (error) printf("AudioSessionSetActive (true) failed"); Thanks very much for any pointers.

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  • What is the best service/tool to put short audio clips on a website so users can click and listen im

    - by Edward Tanguay
    I'm making a foreign language flashcard website in which I want to have 100s of short 3-10 second audio files available for users to click and listen. So I am looking for a tool/service such as YouTube or Screenr.com but for audio which e.g.: allows me to easily upload multiple kinds of audio files: mp3, wav, etc. easy to manage them online (delete, replace) has a simple, small player (e.g. flash) that integrates nicely into any site

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