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  • How does the Ableton warp algorithm work exactly?

    - by pepperdreamteam
    I'm looking for any documentation or definitive information on Ableton's warp feature. I understand that it has something to do with finding transients, aligning them with an even rhythm and shifting audio samples accordingly. I'm hoping to find ways to approximate warping with more basic audio editing tools. I understand that this is ableton's unique device, really any information about how it works would be helpful. So...does anyone have any 411?

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  • Any tips for how to build a LED sytem thet will light up to music?

    - by daniels
    So basically I would like somehow that given an audio file as input (most likely mp3 or I can use some audio engine that will handle other types too) from my computer to control some LED lights so they will be something like an oscilloscope, like the one in winamp. What would I need to be able to do this? I'm interested in building thing up all by myself, coding, hardware, etc.. I'm going with C++ on Windows.

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  • AudioTrack lag: obtainBuffer timed out

    - by BTR
    I'm playing WAVs on my Android phone by loading the file and feeding the bytes into AudioTrack.write() via the FileInputStream BufferedInputStream DataInputStream method. The audio plays fine and when it is, I can easily adjust sample rate, volume, etc on the fly with nice performance. However, it's taking about two full seconds for a track to start playing. I know AudioTrack has an inescapable delay, but this is ridiculous. Every time I play a track, I get this: 03-13 14:55:57.100: WARN/AudioTrack(3454): obtainBuffer timed out (is the CPU pegged?) 0x2e9348 user=00000960, server=00000000 03-13 14:55:57.340: WARN/AudioFlinger(72): write blocked for 233 msecs, 9 delayed writes, thread 0xba28 I've noticed that the delayed write count increases by one every time I play a track -- even across multiple sessions -- from the time the phone has been turned on. The block time is always 230 - 240ms, which makes sense considering a minimum buffer size of 9600 on this device (9600 / 44100). I've seen this message in countless searches on the Internet, but it usually seems to be related to not playing audio at all or skipping audio. In my case, it's just a delayed start. I'm running all my code in a high priority thread. Here's a truncated-yet-functional version of what I'm doing. This is the thread callback in my playback class. Again, this works (only playing 16-bit, 44.1kHz, stereo files right now), it just takes forever to start and has that obtainBuffer/delayed write message every time. public void run() { // Load file FileInputStream mFileInputStream; try { // mFile is instance of custom file class -- this is correct, // so don't sweat this line mFileInputStream = new FileInputStream(mFile.path()); } catch (FileNotFoundException e) {} BufferedInputStream mBufferedInputStream = new BufferedInputStream(mFileInputStream, mBufferLength); DataInputStream mDataInputStream = new DataInputStream(mBufferedInputStream); // Skip header try { if (mDataInputStream.available() > 44) mDataInputStream.skipBytes(44); } catch (IOException e) {} // Initialize device mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, ConfigManager.SAMPLE_RATE, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT, ConfigManager.AUDIO_BUFFER_LENGTH, AudioTrack.MODE_STREAM); mAudioTrack.play(); // Initialize buffer byte[] mByteArray = new byte[mBufferLength]; int mBytesToWrite = 0; int mBytesWritten = 0; // Loop to keep thread running while (mRun) { // This flag is turned on when the user presses "play" while (mPlaying) { try { // Check if data is available if (mDataInputStream.available() > 0) { // Read data from file and write to audio device mBytesToWrite = mDataInputStream.read(mByteArray, 0, mBufferLength); mBytesWritten += mAudioTrack.write(mByteArray, 0, mBytesToWrite); } } catch (IOException e) { } } } } If I can get past the artificially long lag, I can easily deal with the inherit latency by starting my write at a later, predictable position (ie, skip past the minimum buffer length when I start playing a file).

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  • How to set a native object property

    - by theunilife
    ok so im creating a jquery plugin that will allow me to use the new html5 Audio interface and im trying to create an option that is an object that you will be able to set the various listeners but i dont seem to be able to set those options to the listener property of the Audio object.

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  • Any tips for how to build a LED system thet will light up to music?

    - by daniels
    So basically I would like somehow that given an audio file as input (most likely mp3 or I can use some audio engine that will handle other types too) from my computer to control some LED lights so they will be something like an oscilloscope, like the one in winamp. What would I need to be able to do this? I'm interested in building thing up all by myself, coding, hardware, etc.. I'm going with C++ on Windows.

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  • AVAudioPlayer won't play audio file after AVAudioRecorder

    - by Kevin
    I create a .caf audio file using AVAudioRecorder and if I try and play it back using AVAudioPlay I get no sound on the iPhone (if played in simulator works fine). If I close my application and reopen the file plays fine. Also I am not able to adjust the phone volume after recording unless I close and reopen my application. Any ideas?

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  • Is the video recording possible in the background service?

    - by pyt
    Hello, All. I've tried video recording in the background. But had failed. Under normal circumstances, the recording works properly. However, if HOME key down- Home screen or Other Activity is running, recording terminates. In such a situation, I want to record continuously. I want to record whole process! What should I do? thanks. -Michael

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  • About data size filled in the buffer

    - by Bohan Lu
    I need low-latency audio in my project, and I know Android 2.3 supports OpenSL ES. I have read documents and sample code and I decide to use Android simple buffer queue to do the play and record. I now try to write a simple application to do the test. However, I have some questions about recording. If I set the recorder stop when it is recording, how do I know the exact number of bytes filled in the last buffer if it is not filled up ? In 1.1 version, the callback function has some parameters about buffer and its filled data, but there is no such parameters in version 1.0.1. Is there any way to get this information ? Any suggestion would be greatly appreciated !

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  • Why do my speakers get distorted randomly on Windows 7?

    - by Daniel Fischer
    I have a studio monitor setup. I have 2 KRK 6's and a Focusrite Firewire Pro 24. Every few hours my speakers sound distorted and my solution has been go to sound levels Properties of Saffire Audio Device Advanced Default Format Toggle to 16 bit then back to 24bit. Why does it screw up every few hours? Sometimes one speaker doesn't output too and this same process resets it but that's more rare. Is this a OS issue or Focusrite Driver Issue?

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  • OS X Headphone jack issue [closed]

    - by Alex Coady
    Possible Duplicate: Optical Audio out stuck on on a MacBook When I plug my headphones into my iMac (27-inch, Mid 2011; OSX 10.8.1) and try to adjust the volume, the volume popup shows a greyed out speaker and there's a circle with a line through it signalling that it isn't working. I've tried the headphones with my iPhone, other iMacs etc and they're fine. This is incredibly frustrating. Other headphones don't generally work either. In Sound preferences the headphones are being listed as "Optical digital-out port" which is incorrect and would explain the problem, but doesn't help me fix it. Any ideas?

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  • How can I use the shell to make my mp3s a Shoutcast source?

    - by ChasonDehsotel
    I'm looking to stream a directory of mp3s from my audio source (Debian server) to my Shoutcast server. The idea is to have an archive playing in the instance that someone isn't broadcasting live. I'm not sure how to continue. I started with extensive Google-ing, and was unable to come up with a solution. Evan Carroll suggested I try here. I appreciate any insight y'all may have. __ On a side note, "users with less than 100 reputation can't create new tags. The tags 'shoutcast-source shoutcast broadcasting' are new. Try using existing tags instead." -- Who can create these?

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  • Songs/Movies are not being played

    - by romilnagrani
    Hi the problem i am facing is very odd. None of my songs (.mp3/.avi etc) are not able to run in any of the media players: Windows media player, Windows media center, VLC, DIV etc. What i have done before posting the question here. a. Check Device and Drivers and they are working OK and Updated even. b. Run Sample Audio (Downloaded from Internet) in Adobe Sound-booth and it worked. c. Plus i run a video in YouTube and i was able to hear it! d. I am able to hear windows Error Beeps too..!!! e. While running Movies i could see playing but not hear it What could be the problem? please help thanks...

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  • Songs/Movies are not being played

    - by romilnagrani
    Hi the problem i am facing is very odd. None of my songs (.mp3/.avi etc) are not able to run in any of the media players: Windows media player, Windows media center, VLC, DIV etc. What i have done before posting the question here. a. Check Device and Drivers and they are working OK and Updated even. b. Run Sample Audio (Downloaded from Internet) in Adobe Sound-booth and it worked. c. Plus i run a video in YouTube and i was able to hear it! d. I am able to hear windows Error Beeps too..!!! e. While running Movies i could see playing but not hear it What could be the problem? please help

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  • Songs/Movies are not being played

    - by romilnagrani
    Hi the problem i am facing is very odd. None of my songs (.mp3/.avi etc) are not able to run in any of the media players: Windows media player, Windows media center, VLC, DIV etc. What i have done before posting the question here. a. Check Device and Drivers and they are working OK and Updated even. b. Run Sample Audio (Downloaded from Internet) in Adobe Sound-booth and it worked. c. Plus i run a video in YouTube and i was able to hear it! d. I am able to hear windows Error Beeps too..!!! e. While running Movies i could see playing but not hear it What could be the problem? please help

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  • Does static damage computer speakers?

    - by incarna
    I recently got a new pair of Klipsch Promedia 2.1's for my laptop. I unplug my laptop a lot to take it around but today the audio plug touched my plug for my monitor and a bit of static came out of the speakers. I've heard some rumors that static can damage speakers but I've never investigated this problem myself since I previously used a desktop and never unplugged them. The volume was at a normal volume- am I just being paranoid? Or could having the speaker port touching other bits of metal damage my speakers?

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  • How could I represent 1.625 by 0 or a 1 (binary digit)?

    - by pepito
    This is an excerpt from wikipedia about 'full rate' speech coding standard. Full Rate or FR or GSM-FR or GSM 06.10 was the first digital speech coding standard used in the GSM digital mobile phone system. The bit rate of the codec is 13 kbit/s, or 1.625 bits/audio sample. And this one is an excerpt from wikipedia about bit. In computing parlance, bit is the abbreviation for a single binary digit, represented by a 0 or a 1. How could I represent 1.625 by 0 or a 1? Actually, that's my lecturer's question that I could not answer. Some links to papers are more than welcome. Thanks in advance.

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  • Sound stopped working on Toshiba Satelite C855

    - by Eric Wilson
    I have a Toshiba Satallite C855 running Windows 7 with a Realtek High Definition Audio card. Sound worked fine yesterday. Today, my two-year old did something to the machine, and I have no sound. I have adjusted sound through the icon in the lower right corner, and using the Fn keys. I have rebooted. I've verified that there are no external speakers plugged in to the laptop. It seems that this model does not have an external volume control. (If it does I can't find it.) Any ideas?

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  • Linux Logitech QuickCam Pro 9000 - microphone issues

    - by drahnr
    I got a Logitech Quickcam Pro 9000, the cam itself is working as it honors the UVC spec. This fancy WebCam has a integrated mic which worked some time before but now, it does no more. (Note: I use pulseaudio as it is a USB Mic and I am not really keen on the hassle of ALSA setup) Things I check already are if it gets detected at all: $ lsusb |grep Logi Bus 002 Device 002: ID 046d:0809 Logitech, Inc. Webcam Pro 9000 is muted in alsa-mixer - not the case, volume at 100 pavucontrol shows it too, but no input level bar! On top of that, if I open the gnome3 (fallback mode) audio panel(from the desktop panel), and disabel/reenable it in the hardware tab, it works "for some time"... Any hints? Any ideas? I am really out options for now, and the fact it worked like 6 months ago (perfectly) makes it no better.

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  • How do I disable alert beeps

    - by wheresrhys
    My HP pavilion laptop is getting pretty old and the speakers no longer work properly. So I've disabled all audio devices, including disabling the light-touch volume controls at boot-up. The one sound still here that I can't get rid of is teh alert beep whenever there's an error. It's annoying at the best of times, but when your laptop has no other sounds at all it's intensly irritating! Nothing I change in the windows settings or in the BIOS seems to have any effect. Any ideas how to switch them off?

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  • How do I disable aalert beeps

    - by wheresrhys
    My HP pavilion laptop is getting pretty old and the speakers no longer work properly. So I've disabled all audio devices, including disabling the light-touch volume controls at boot-up. The one sound still here that I can't get rid of is teh alert beep whenever there's an error. It's annoying at the best of times, but when your laptop has no other sounds at all it's intensly irritating! Nothing I chaneg in the windows settings or in the BIOS seems to have any effect. Any ideas how to switch them off?

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  • have ffmpeg scan and report correct time

    - by acidzombie24
    I am encoding a section of a song. I used -ss offset -t 30 (duration). When i use -i file.acc i see it says the audio is 31, 32 and once 36 seconds long. Opening it in vlc showed it as 30sec after a few seconds of playback. My code needs to filter sounds more then 30 seconds. I can fudge it and allow 30.99 (maybe 20.48 is better) however 2 seconds too long is not good and i would need to filter this out even though playback is 30seconds long. How do i get ffmpeg to scan the file and report an accurate time?

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  • Debian no lib32asound2-plugins=1.0.25

    - by erai
    I compiled git wine 1.5.6 and the sound is not working correctly. winecfg gives a message: ALSA lib ../../pulse/pcm_pulse.c:995:(_snd_pcm_pulse_open) Unknown field handle_underrun err:winediag:AUDDRV_GetAudioEndpoint PulseAudio "default" -22 without handle_underrun. Audio may hang. Please upgrade to alsa_plugins >= 1.0.24 err:ole:CoInitializeEx Attempt to change threading model of this apartment from multi-threaded to apartment threaded But on wheezy there is no package lib32asound2-plugins 1.0.25 So, how do I build and package alsa-plugins to install to lib32? Or how do I use automake's configure to compile a 32bit lib on a 64 bit system. All I Need now is the libavcodec-dev for 32bit, but I don't understand. What is the 32bit libavcodec-dev package for amd64 debian?

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  • VLC volume only to 200%?

    - by Tomas
    According to this comment it seems that VLC could boost the audio volume up to 800% in the past versions. Today I installed VLC multimedia player version 2.0.5 and it is capable only to boost up to 200%! This is not much, considering that some youtube videos are very quiet. This is what I need VLC for - to boost up volume for quiet youtube videos. So, where's the problem? Was it actually 800% in the past? Did VLC really limit this from former 800% to current 200%? Is it possible to somehow achieve more boost with VLC? Or with other software? I am using Windows 7.

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