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  • Converting audio files(.3gp) to video with Album cover and uploading to YouTube

    - by Samuh
    I have an audio file in .3gp format on my Android device which I wish to upload to YouTube. I know that YouTube is a video upload site and that I need to convert this sound file to video. I just want an image to display all the time the audio is playing. Google tells me there are number of tools that can help me. But I want to do this via java code from my Android device. Please help. Thanks.

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  • Windows 7 sound control is not flexible.

    - by jon
    I would like to be able to listen to music on both of two audio output devices, but Windows 7 seems to only allow me to select one or the other as the Default device. When device A is the Default device, device B is muted; and vice versa. This seems to be stunningly inflexible. Since Windows 7 is unable to do this, can anyone recommend any add-on software that would control the hardware more flexibly and thoughtfully?

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  • Audio Player Royalty Free Music (dynamic audios )?

    - by Surya sasidhar
    hi, I am using Royalty Free Music player for playing the audio. ya it is playing perfect but i need to play it dynamically, i mean the audio will come from database how can i write the code for that. This is the royalty free music code..... var so = new SWFObject("playerSingle.swf", "mymovie", "192", "67", "7", "#FFFFFF"); so.addVariable("autoPlay", "yes"); so.addVariable("soundPath","song.mp3"); so.addVariable("overColor","#000044") so.addVariable("playerSkin","1") so.write("flashPlayer"); this above code is written in source code with in the script tag, then how can i write for dynamic audios please help me thanking you and this is the link for that site.. http://www.premiumbeat.com/flash_resources/free_flash_music_player/single_track_flash_mp3_player.php

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  • Use an iPhone as a bluetooth headset for a mac?

    - by Phillip Oldham
    Is there any way, such as an iPhone app, that will let me connect my iPhone to my iMac via bluetooth, so that the iMac pushes all audio through the iPhone? Specifically, what I'm looking to do is be able to watch movies on my iMac with the sound being played through my iPhone & in turn the ear buds.

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  • Comapring pitches with digital audio

    - by user2250569
    I work on application which will compare musical notes with digital audio. My first idea was analyzes wav file (or sound in real-time) with some polyphonic pitch algorithms and gets notes and chords from this file and subsequently compared with notes in dataset. I went through a lot of pages and it seems to be a lot of hard work because existing implementations and algorithms are mainly/only focus on monophonic sound. Now, I got the idea to do this in the opposite way. In dataset I have for example note: A4 or better example chord: A4 B4 H4. And my idea is make some wave (or whatever I don't know what) from this note or chord and then compared with piece of digital audio. Is this good idea? Is it better/harder solution? If yes can you recommend me how to do it?

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  • How do I stop and repair a RAID 5 array that has failed and has I/O pending?

    - by Ben Hymers
    The short version: I have a failed RAID 5 array which has a bunch of processes hung waiting on I/O operations on it; how can I recover from this? The long version: Yesterday I noticed Samba access was being very sporadic; accessing the server's shares from Windows would randomly lock up explorer completely after clicking on one or two directories. I assumed it was Windows being a pain and left it. Today the problem is the same, so I did a little digging; the first thing I noticed was that running ps aux | grep smbd gives a lot of lines like this: ben 969 0.0 0.2 96088 4128 ? D 18:21 0:00 smbd -F root 1708 0.0 0.2 93468 4748 ? Ss 18:44 0:00 smbd -F root 1711 0.0 0.0 93468 1364 ? S 18:44 0:00 smbd -F ben 3148 0.0 0.2 96052 4160 ? D Mar07 0:00 smbd -F ... There are a lot of processes stuck in the "D" state. Running ps aux | grep " D" shows up some other processes including my nightly backup script, all of which need to access the volume mounted on my RAID array at some point. After some googling, I found that it might be down to the RAID array failing, so I checked /proc/mdstat, which shows this: ben@jack:~$ cat /proc/mdstat Personalities : [linear] [multipath] [raid0] [raid1] [raid6] [raid5] [raid4] [raid10] md0 : active raid5 sdb1[3](F) sdc1[1] sdd1[2] 2930271872 blocks level 5, 64k chunk, algorithm 2 [3/2] [_UU] unused devices: <none> And running mdadm --detail /dev/md0 gives this: ben@jack:~$ sudo mdadm --detail /dev/md0 /dev/md0: Version : 00.90 Creation Time : Sat Oct 31 20:53:10 2009 Raid Level : raid5 Array Size : 2930271872 (2794.53 GiB 3000.60 GB) Used Dev Size : 1465135936 (1397.26 GiB 1500.30 GB) Raid Devices : 3 Total Devices : 3 Preferred Minor : 0 Persistence : Superblock is persistent Update Time : Mon Mar 7 03:06:35 2011 State : active, degraded Active Devices : 2 Working Devices : 2 Failed Devices : 1 Spare Devices : 0 Layout : left-symmetric Chunk Size : 64K UUID : f114711a:c770de54:c8276759:b34deaa0 Events : 0.208245 Number Major Minor RaidDevice State 3 8 17 0 faulty spare rebuilding /dev/sdb1 1 8 33 1 active sync /dev/sdc1 2 8 49 2 active sync /dev/sdd1 I believe this says that sdb1 has failed, and so the array is running with two drives out of three 'up'. Some advice I found said to check /var/log/messages for notices of failures, and sure enough there are plenty: ben@jack:~$ grep sdb /var/log/messages ... Mar 7 03:06:35 jack kernel: [4525155.384937] md/raid:md0: read error NOT corrected!! (sector 400644912 on sdb1). Mar 7 03:06:35 jack kernel: [4525155.389686] md/raid:md0: read error not correctable (sector 400644920 on sdb1). Mar 7 03:06:35 jack kernel: [4525155.389686] md/raid:md0: read error not correctable (sector 400644928 on sdb1). Mar 7 03:06:35 jack kernel: [4525155.389688] md/raid:md0: read error not correctable (sector 400644936 on sdb1). Mar 7 03:06:56 jack kernel: [4525176.231603] sd 0:0:1:0: [sdb] Unhandled sense code Mar 7 03:06:56 jack kernel: [4525176.231605] sd 0:0:1:0: [sdb] Result: hostbyte=DID_OK driverbyte=DRIVER_SENSE Mar 7 03:06:56 jack kernel: [4525176.231608] sd 0:0:1:0: [sdb] Sense Key : Medium Error [current] [descriptor] Mar 7 03:06:56 jack kernel: [4525176.231623] sd 0:0:1:0: [sdb] Add. Sense: Unrecovered read error - auto reallocate failed Mar 7 03:06:56 jack kernel: [4525176.231627] sd 0:0:1:0: [sdb] CDB: Read(10): 28 00 17 e1 5f bf 00 01 00 00 To me it is clear that device sdb has failed, and I need to stop the array, shutdown, replace it, reboot, then repair the array, bring it back up and mount the filesystem. I cannot hot-swap a replacement drive in, and don't want to leave the array running in a degraded state. I believe I am supposed to unmount the filesystem before stopping the array, but that is failing, and that is where I'm stuck now: ben@jack:~$ sudo umount /storage umount: /storage: device is busy. (In some cases useful info about processes that use the device is found by lsof(8) or fuser(1)) It is indeed busy; there are some 30 or 40 processes waiting on I/O. What should I do? Should I kill all these processes and try again? Is that a wise move when they are 'uninterruptable'? What would happen if I tried to reboot? Please let me know what you think I should do. And please ask if you need any extra information to diagnose the problem or to help!

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  • Problem with waveOutWrite and waveOutGetPosition deadlock

    - by MusiGenesis
    I'm working on an app that plays audio continuously using the waveOut... API from winmm.dll. The app uses "leapfrog" buffers, which are basically a bunch of arrays of samples that you dump into the audio queue. Windows plays them seamlessly in sequence, and as each buffer completes Windows calls a callback function. Inside this function, I load the next set of samples into the buffer, process them however, and then dump the buffer back into the audio queue. In this way, the audio plays indefinitely. For animation purposes, I'm trying to incorporate waveOutGetPosition into the application (since the "buffer done" callbacks are irregular enough to cause jerky animation). waveOutGetPosition returns the current position of playback, so it's hyper-precise. The problem is that in my application, making calls to waveOutGetPosition eventually causes the application to lock up - the sound stops and the call never returns. I've boiled things down to a simple app that demonstrates the problem. You can run the app here: http://www.musigenesis.com/SO/waveOut%20demo.exe If you just hear a tiny bit of piano over and over, it's working. It's just meant to demonstrate the problem. The source code for this project is here: http://www.musigenesis.com/SO/WaveOutDemo.zip The first button runs the app in leapfrog mode without making the calls to waveOutGetPosition. If you click this, the app will play forever without breaking (the X button will close it and shut it off). The second button starts the leapfrogger and also starts a forms timer that calls the waveOutGetPosition and displays the current position. Click this and the app will run for a short while and then lock up. On my laptop, it usually locks up in 15-30 seconds; at most it's taken a minute. I have no idea how to fix this, so any help or suggestions would be most welcome. I've found very few posts on this issue, but it seems that there is a potential deadlock, either from multiple calls to waveOutGetPosition or from calls to that and waveOutWrite that occur at the same time. It's possible that I'm calling this too frequently for the system to handle.

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  • i2s0: transmitter underrun (0)

    - by tbarbe
    were doing some audio stuff and I keep seeing this in the Organizer Console. Sun May 2 20:16:48 unknown kernel[0] : i2s0: transmitter underrun (0) Are these transmitter underruns bad? I think its just when were shutting down audio input...but could a few of these cause some issues later on?

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  • iPhone SDK: How to record voices with ambient noise supression?

    - by Harkonian
    Can anyone point me in the right direction on how I would minimize ambient noise while recording someone speaking using the iPhone SDK Core Audio? I'm guessing a band-pass filter that eliminates any frequencies above and below the human vocal range might work. I have no idea how I would implement band filters on audio in the SDK though. The optimum solution would be one that eliminates the noise from the stream before it is written to memory/disk. Some sample code would be appreciated.

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  • iPhone: CPU power to do DSP/Fourier transform/frequency domain?

    - by mahboudz
    I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and other music recognition apps, and guitar tuner apps out there. However, I don't know what limitations I'll have to deal with. Anyone play around with this area?

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  • iPhone SDK audioSession question.

    - by Morion
    Hi to all. In my app i record and play audio at the same time. The app is almost finished. But there is one thing, that annoying me. When audio session is set to PlayAndRecord, sounds become quiet in comparison with the same sounds with the SoloAmbient category. Is there any way to make sound louder using PlayAndRecord?

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  • What is a lightweight cross platform WAV playing library?

    - by Lokkju
    I'm looking for a lightweight way to make my program (written in C) be able to play audio files on either windows or linux. I am currently using windows native calls, which is essentially just a single call that is passed a filename. I would like something similar that works on linux. The audio files are Microsoft PCM, Single channel, 22Khz Any Suggestions?

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  • Identifying voice as male or female

    - by duder
    I'm not much into audio engineering, so please be easy on me. I'm receiving an audio file as input, and need to detect whether the speaker is male or female. Any ideas how to go about doing this? I'm using php, but am open to using other languages, and don't mind learning a little bit of sound theory as long as the time is proportionate to the task.

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  • Flash Media Live Encoder

    - by jeph perro
    I am using Adobe Flash Media Live Encoder to stream live video to a video streaming server. The webcam is in our office pointed out the window. Thankfully, Flash Media Live Encoder has a checkbox to un-include audio. I am wondering how I can push a recorded message to the audio ( or music ). Is there any way I can play a recording and have it behave like a microphone?

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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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  • 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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