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  • Streaming audio (YouTube)

    - by wvd
    Hello all, I'm writing a CLI for a music-media-platform. One of the features is going to be that you can directly play YouTube videos from the CLI. I don't really have an idea to do it but this one sounded the most reasonable: I'm going to use of those sites where you can download music from YouTube, e.g. http://keepvid.com/ - then I directly stream & play this -- but I have one problem. Is there any Python library capable of doing this and if so, do you have any concrete examples? I've been looking but found nothing, even not with gstreamer. Thanks, William van Doorn

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  • Resampling audio output for A2DP (from PCM WAV)

    - by user1669982
    The question is how to bring stereo PCM WAV 32,000 Hz with a stream of 1024 kbps (125 KB) to the headset with Bluetooth 2.1 on a CM7 smartphone with DSPManager. Is it possible? SBC is really bad idea. To TJD: Because it compresses the compressed stream. My Epic 4G don`t have Apt-X support. My headset Gemix BH-04A yellow. May be its possible with the Headset Profile (HSP)? I dont know about supported codecs in this profile.

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  • Audio Playback Rate in Android

    - by Marquis
    So, I know that this has been done with a few Android apps before, but I cannot for the life of me figure out how, since it's not currently possible through the API. How does one adjust the playback rate of a sound played through MediaPlayer; either with or without adjusting the pitch is fine for now, though the latter is definitely preferred. If someone can point me in the direction of an open source app that I can use as guidance, that would also be fine. Thanks in advance.

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  • Playback audio data with GWT

    - by Henrik
    I am creating a GWT client application which interacts with a server and I am getting all my response data from the server in JSON format. Amongst others there are wave data on the server's database which I would like to retrieve and then playback on the client. I am able to get the wave data as an array of bytes in the JSON format. My problem is, how do I playback the wave array data in a browser? Is it even possible or do I have to find another solution? I've searched the web and found some GWT packages which are able to playback sound, but they are all playing back directly from an url.

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  • Real-time equalizer for all audio on computer

    - by greye
    Is it possible to capture all the sound from a computer and have it pass through a equalizer before reaching the speakers? How can you program a band pass filter on it? EDIT: I'm trying to get this on Windows (with Python? heh) but if there is a generic, cross-platform approach that would be great.

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  • Inserting a WAV at a certain point in an audio file using python

    - by Onion
    My problem is the following: I have a 2-minute long WAV file, and my aim is to insert another WAV file (7 seconds long), at a certain point in the first WAV file (say, 0:48), essentially combining the two WAVs, using python. Unfortunately I haven't been able to figure out how to do that, and was wondering if there was some obvious solution that I was missing, or if it is even feasible to do with python. Is there perhaps a library available that might provide a solution? Thanks to all in advance.

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  • Filter design for audio signal.

    - by beanyblue
    What I am trying to do is simple. I have a few .wav files. I want to remove noise and filter out specific frequencies. I don't have matlab and I intend to write my own code for all the filters. Right now, I have a way to read the .wav file and dump out the structure into a text file. My questions are the following: Can I directly apply the digital filters on this sampled data?{ ie, can I directly do a convolution between my input samples and h(n) for the filter function that i choose?). How do I choose the number of coefficients for the Window function? I have octave, so if someone can point me to anything that gives me some idea on how to process the .wav file using octave, that would be great too. I want to be able to filter out the frequency and then listen to the sound again. Is this possible with octave? I'm just a beginner with these kinds of things, so please bear with me if my questions are too naive. Any help will be great.

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  • Multiple Audio Issues

    - by Lerp
    I am having issues with my audio on Ubuntu 12.04, I will try and give as much detail as possible so sorry if there's too much detail. The Problem Audio plays from both speakers and headphone regardless of what connector I choose and regardless of the profile I use. The microphone is constantly being played through headphones & speakers. The headphone audio is extremely quiet but plays from both ears when I select "Headphones" for the connector in Sound Settings. The headphone audio only plays from one ear and is quiet (but not as quiet as above) when I select "Analogue Output" for the connector in Sound Settings. I can only select "Headphones" as the connector in Sound Settings if I set the profile to either "Analogue Stereo Output/Duplex", all others only allow me to choose "Analogue Output" for the connector. Despite the headphone sound issues, the speaker sound is fine apart from the fact that I am not able to select which output is used, they just both play. My headphone and microphone are plugged into the front and my speakers are plugged into the back. What I have tried I have put everything in alsamixer to 100 apart from "Front Mic Boost" which I have set to 0. Command Output aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: AD198x Headphone [AD198x Headphone] Subdevices: 1/1 Subdevice #0: subdevice #0 arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 2/3 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf7ff8000 irq 70 cat /proc/asound/modules 0 snd_hda_intel cat /proc/asound/card*/codec* | grep "Codec" Codec: Analog Devices AD1989B cat /etc/modprobe.d/alsa-base.conf # autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 # Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } # Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } # Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 # Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 # Keep snd-usb-audio from beeing loaded as first soundcard options snd-usb-audio index=-2 Hopefully I have provided enough information, I will happily provide anymore information needed. Thank you. Update Reinstalling alsa-base and pulseaudio fixed the headphone issues I was having.

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  • How to compare mp3, flac audio data in a file, ignoring header data (ID3 tag) etc.?

    - by Rob
    I've backed up some audio files up in 2 places and added ID3 tags into one backup but not the other, since time has passed my own memory has faded on whether the backups are actually the same, but now one has ID3 data and the other doesn't, basic binary compare will fail and inspection will be cumbersome. Is there a tool to compare just the audio data (not the header, ID3) in mp3s, flac files, and other files using header data such as ID3. started a thread on beyond compare here: http://www.scootersoftware.com/vbulletin/showthread.php?t=7413 would consider other comparison software that does this task

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  • Use an audio/video file from a Linux laptop via USB to be played by Magic Sing ET-23H

    - by AisIceEyes
    I am one of the technical directors of a regular karaoke contest event. For the karaoke contest itself, due to tight budget, we are using what one of the sponsors are providing - Magic Sing ET-23H . The video output of the Magic Sing ET-23H are broadcasted at two big screens that are being shown to the audience and event attendees. When a karaoke contestant provides his / her karaoke video, the video itself is in a readable USB flashdrive and is attached to the USB input of Magic Sing ET-23H. What really bugs me is that the interface of Magic Sing ET-23H are also being broadcasted at the big screen video feeds. The interface of choosing the video file is being seen in the Magic Sing ET-23H - also to the big video screens that are seen by the audience and event goers. I will post in the comments ( if my less than 10 reputation would allow me) the picture of Magic Sing ET-23KH USB input of the device. I always bring my laptop, Acer AS5742-7653, during the regular karaoke event. I'm using my laptop also for tallying of scores from the judges, and also playing audio files from contestants that did not provide a karaoke video. I personally am using different Linux distros, but I next to all the time use my Ubuntu Studio 12.04.3 64bit partition during the regular karaoke contest event. My question is this: Is there a way I can share a temporary video/audio file directly from the laptop I'm using, going to the Magic Sing ET-23H that can broadcast both the video/audio file? Just like how in Window's Avisynth AVS files, or VirtualDub's temporary avi file, or like using ffplay (of ffmpeg), etc. I have researched somewhat the matter and found links in SuperUser.com. Though I can only provide the links at the comments section of this post if my reputation of less than 10 would allow me. I have a hunch it is possible, but I have not fully understood the device being used at the event, Magic Sing ET-23H, if there are other ways for it to broadcast video and audio files besides its USB input. Any help to my current predicament is highly appreciated. Thank you. PS: Since I need at least 10 reputation to post more than 2 links and also post images, I will try to post the image & links at the comments (if my below 10 reputation would allow me).

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  • AudioQueue ate my buffer (first 15 milliseconds of it)

    - by iter
    I am generating audio programmatically. I hear gaps of silence between my buffers. When I hook my phone to a scope, I see that the first few samples of each buffer are missing, and in their place is silence. The length of this silence varies from almost nothing to as much as 20 ms. My first thought is that my original callback function takes too much time. I replace it with the shortest one possible--it re-renqueues the same buffer over and over. I observe the same behavior. AudioQueueRef aq; AudioQueueBufferRef aq_buffer; AudioStreamBasicDescription asbd; void aq_callback (void *aqData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) { OSStatus s = AudioQueueEnqueueBuffer(aq, aq_buffer, 0, NULL); } void aq_init(void) { OSStatus s; asbd.mSampleRate = AUDIO_SAMPLES_PER_S; asbd.mFormatID = kAudioFormatLinearPCM; asbd.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; asbd.mBytesPerPacket = 1; asbd.mFramesPerPacket = 1; asbd.mBytesPerFrame = 1; asbd.mChannelsPerFrame = 1; asbd.mBitsPerChannel = 8; asbd.mReserved = 0; int PPM_PACKETS_PER_SECOND = 50; // one buffer is as long as one PPM frame int BUFFER_SIZE_BYTES = asbd.mSampleRate/PPM_PACKETS_PER_SECOND*asbd.mBytesPerFrame; s = AudioQueueNewOutput(&asbd, aq_callback, NULL, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &aq); s = AudioQueueAllocateBuffer(aq, BUFFER_SIZE_BYTES, &aq_buffer); // put samples in the buffer buffer_data(my_data, aq_buffer); s = AudioQueueStart(aq, NULL); s = AudioQueueEnqueueBuffer(aq, aq_buffer, 0, NULL); }

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  • MythTV lost recordings - "No recordings available" and no recording rules either

    - by nimasmi
    I have a c.6 year old mythtv database. I recently upgraded from Ubuntu 10.04 to 12.04. This brought a MythTV upgrade from 0.24 to 0.25, which went well. Today, all my recordings have disappeared. They still exist in the /var/lib/mythtv/recordings folder, and the 'M' key in the Watch Recordings page says that there are 201 recordings available somewhere, but they will not display. See screenshot: (implicit thanks to whomever upvoted this, giving me sufficient reputation to upload images) Changing the filter does not remedy the fact that there is nothing shown in the lists. My Upcoming Recordings screen says that there are no rules set, but my list of previously recorded shows is still there, and has an entry from as recently as 3am today. mythbackend --printsched gives the following: user@box:~$ mythbackend --printsched 2012-09-22 12:59:20.537008 C mythbackend version: fixes/0.25 [v0.25.2-15-g46cab93] www.mythtv.org 2012-09-22 12:59:20.537043 C Qt version: compile: 4.8.1, runtime: 4.8.1 2012-09-22 12:59:20.537048 N Enabled verbose msgs: general 2012-09-22 12:59:20.537076 N Setting Log Level to LOG_INFO 2012-09-22 12:59:20.537142 I Added logging to the console 2012-09-22 12:59:20.537152 I Added database logging to table logging 2012-09-22 12:59:20.537279 N Setting up SIGHUP handler 2012-09-22 12:59:20.537373 N Using runtime prefix = /usr 2012-09-22 12:59:20.537394 N Using configuration directory = /home/user/.mythtv 2012-09-22 12:59:20.537999 I Assumed character encoding: en_GB.UTF-8 2012-09-22 12:59:20.538599 N Empty LocalHostName. 2012-09-22 12:59:20.538610 I Using localhost value of box 2012-09-22 12:59:20.538792 I Testing network connectivity to '192.168.1.2' 2012-09-22 12:59:20.539420 I Starting process manager 2012-09-22 12:59:20.541412 I Starting IO manager (read) 2012-09-22 12:59:20.541715 I Starting IO manager (write) 2012-09-22 12:59:20.541836 I Starting process signal handler 2012-09-22 12:59:20.684497 N Setting QT default locale to EN_GB 2012-09-22 12:59:20.684694 I Current locale EN_GB 2012-09-22 12:59:20.684813 N Reading locale defaults from /usr/share/mythtv//locales/en_gb.xml 2012-09-22 12:59:20.697623 I New static DB connectionDataDirectCon 2012-09-22 12:59:20.704769 I MythCoreContext: Connecting to backend server: 192.168.1.2:6543 (try 1 of 1) Calculating Schedule from database. Inputs, Card IDs, and Conflict info may be invalid if you have multiple tuners. 2012-09-22 12:59:27.710538 E MythSocket(21dfcd0:14): readStringList: Error, timed out after 7000 ms. 2012-09-22 12:59:27.710592 C Protocol version check failure. The response to MYTH_PROTO_VERSION was empty. This happens when the backend is too busy to respond, or has deadlocked in due to bugs or hardware failure. Things I have tried so far: restart the backend restart the frontend run mythtv-setup and check database passwords and IP addresses change the frontend setting for backend IP from localhost to 192.168.1.2 (the backend/frontend's IP) run optimize_mythdb.pl Other suggestions appreciated.

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  • Recording custom variables to identify individual users with Google Analytics

    - by mrtsherman
    I have been asked by our marketing department to add Google Analytics custom variable tracking to my company's website. As the website uses server side includes, modifications to the tracking tag roll out globally - maintenance is therefore a headache! So, if I add the following code (keeping in mind SSI so every page has the same code): // visitor level tracking, id = 12345 // Record a unique id for each visitor. When they return also track this id _gaq.push(['_setCustomVar', 1, 'id', '12345', 1]); // page level tracking // If the user signs up for our newsletter we set newsletter to true // Each page they subsequently visit should also mark this as true _gaq.push(['_setCustomVar', 1, 'newsletter', 'true', 1]); I don't use GA and the marketing people don't use custom variables, so we don't actually know how or if this will work. Therefore my questions are:- Do I want Page, Session or Visitor level tracking? What happens when the same code is used on every page? Can GA 'overwrite' a setting. For example, if I set newsletter to true on page X and then user navigates to page Y, will the variable also be marked there?

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  • 11.10 desktop alerts (volume change and terminal bell) stopped working but all other audio still works

    - by FlabbergastedPickle
    All, My sound works just fine in 11.10 64-bit install on HP dm1-4050 Sandy Bridge notebook (e.g. audio works in Banshee, flash, games, browser, Thunderbird email notification, etc.), but the core desktop notifications (e.g. pressing a tab in a terminal where there is more than one option should trigger a terminal bell, or changing volume using volume keys should be accompanied with the supporting "quack" that the volume app makes) do not work. I've intentionally disabled login sound as explained here on ask ubuntu but even enabling it back makes no difference. These notifications did work before just fine and I am not sure when did the actually stop working but it must've been fairly recently. Only things I did were trying to install some ppa edge xorg drivers for my intel card (a separate issue) but also reverted them all with ppa-purge once I discovered they did not improve anything. Other thing I did was check volume settings with alsamixer and did alsactl store for the soundcard after I did some experimenting with volume settings for PCM (on my laptop PCM at 100% crackles so I had to lower it and make pulseaudio ignore its setting as per ask ubuntu's page). That said, neither of these should have any bearing on the said notifications since the volume is up and they clearly work everywhere else but the core desktop events. The system ready drum sound when Ubuntu boots and user reaches the login screen also does not work. The guest login behaves exactly same as mine. Audio works (including the login sound since I've not disabled it for the guest account), but no quacks when changing the volume or terminal bell sounds... I've tried copying ubuntu sounds to /usr/share/sounds/ as suggested on ask ubuntu and that did not work. I also tried using dconf-editor to check sound theme settings and tried both freedesktop (which is what it was set to) and ubuntu, as suggested on ask ubuntu. This did not work either. I tried purging the ~/.pulse folder and the /tmp/*pulse* entries, rebooting and restarting pulseaudio with -D flag. While audio came back on and behaved just fine in all aspects (e.g. one can adjust volume levels, play music, games, in-browser sound stuff, and other app alerts) except for the system ready drum sound (at the login screen), and any system event (terminal bell and volume change quack sound). It is interesting that the quack sound works inside system settings-sound when adjusting levels there, but it does not when volume is changed via top bar's volume settings... I do recall that at one point yesterday when I was restarting pulseaudio the quacks that accompany volume change did start working but I have no idea what caused that. This was also when I first realized those alerts were not working. After rebooting it was again gone. I did compile my own 3.0.14-rt31 kernel a little while ago as instructed on one of the wiki's for the 11.10 rt kernel. Everything works as before except for the said sound alerts. I am not sure if this began happening since I started using the rt kernel though and yesterday's momentary ability to hear those quacks while changing the volume make me believe that the kernel is not one responsible for this problem. One more thing I can think of is that I used alsoft-conf tool to configure buffering on the OpenAL (due to TA Spring's choppy audio) and changed in there default audio device to ALSA. I also tried reverting it to Pulseaudio as the only allowed output but the bottom part of the Backend tab always reverts to ALSA even when I select Pulseaudio. The pulseaudio does remain as the only active choice on top. This, however, once again does not make any sense in terms of preventing desktop audio alerts when everything else including OpenAL games plays sound just fine... So, there you have it, as verbose as I could make it :-). I tried all I could find on this issue and had no luck so far... Any ideas?

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  • Lag when recording with xvidcap?

    - by quangtruong1985
    I used Xvidcap to record my desktop, but the quality of video was too bad (it lagged so much). I also tried with all output formats that xvidcap support, increase the frame-per-second as much as possible and the quality always at 100% but nothing changed. Click to see my video on Youtube Im using 11.04 (unity) with compiz enabled. My card is ATI/AMD Mobility 5450 and all drivers were installed and activated. Please help me! Regards.

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  • Using Sizer for recording presentations

    - by John Paul Cook
    I needed to do some screen captures and recordings of SSMS and realized this is a common problem that many of you could use some help with. There is a freeware tool called Sizer (thanks to Paul Nielsen for telling me about it) that lets you chose your window size. I downloaded the zip file instead of the msi because I didn’t want to install anything. The extracted executable works perfectly as a portable application. After double-clicking the Sizer executable, an icon resembling a plus sign appears...(read more)

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  • Google Analytics recording event based on <a> title attribute

    - by rlsaj
    I am declaring: var title = (typeof(el.attr('title')) != 'undefined' ) ? el.attr('title') :""; and then have the following: else if (title.match(/^"Matching Content"\:/i)) { elEv.category = "Matching Content Click"; elEv.action = "click-Matching-Content"; elEv.label = href.replace(/^https?\:\/\//i, ''); elEv.non_i = true; elEv.loc = href; } However, using Google Analytics debugger this is not being recorded. Any suggestions? The complete function is: if (typeof jQuery != 'undefined') { jQuery(document).ready(function gLinkTracking($) { var filetypes = /\.(avi|csv|dat|dmg|doc.*|exe|flv|gif|jpg|mov|mp3|mp4|msi|pdf|png|ppt.*|rar|swf|txt|wav|wma|wmv|xls.*|zip)$/i; var baseHref = ''; if (jQuery('base').attr('href') != undefined) baseHref = jQuery('base').attr('href'); jQuery('a').on('click', function (event) { var el = jQuery(this); var track = true; var href = (typeof(el.attr('href')) != 'undefined' ) ? el.attr('href') :""; var title = (typeof(el.attr('title')) != 'undefined' ) ? el.attr('title') :""; var isThisDomain = href.match(document.domain.split('.').reverse()[1] + '.' + document.domain.split('.').reverse()[0]); if (!href.match(/^javascript:/i)) { var elEv = []; elEv.value=0, elEv.non_i=false; if (href.match(/^mailto\:/i)) { elEv.category = "Email link"; elEv.action = "click-email"; elEv.label = href.replace(/^mailto\:/i, ''); elEv.loc = href; } else if (title.match(/^"Matching Content"\:/i)) { elEv.category = "Matching Content Click"; elEv.action = "click-Matching-Content"; elEv.label = href.replace(/^https?\:\/\//i, ''); elEv.non_i = true; elEv.loc = href; } else if (href.match(filetypes)) { var extension = (/[.]/.exec(href)) ? /[^.]+$/.exec(href) : undefined; elEv.category = "File Downloaded"; elEv.action = "click-" + extension[0]; elEv.label = href.replace(/ /g,"-"); elEv.loc = baseHref + href; } else if (href.match(/^https?\:/i) && !isThisDomain) { elEv.category = "External link"; elEv.action = "click-external"; elEv.label = href.replace(/^https?\:\/\//i, ''); elEv.non_i = true; elEv.loc = href; } else if (href.match(/^tel\:/i)) { elEv.category = "Telephone link"; elEv.action = "click-telephone"; elEv.label = href.replace(/^tel\:/i, ''); elEv.loc = href; } else track = false; if (track) { _gaq.push(['_trackEvent', elEv.category.toLowerCase(), elEv.action.toLowerCase(), elEv.label.toLowerCase(), elEv.value, elEv.non_i]); if ( el.attr('target') == undefined || el.attr('target').toLowerCase() != '_blank') { setTimeout(function() { location.href = elEv.loc; }, 400); return false; } } } }); }); }

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  • PS2 Eyetoy Recording Quality

    - by Fire
    I have Ubuntu 12.04 LTS and a PS2 eyetoy "Namtai". Don't worry - this is not the sterotypical " how do I get my eyetoy working" question. My eyetoy works fine on Cheese, gucview and various other media software like VLC. However, It seems like I am capped by 25fps. If I recall, the eyetoy is much better than this (~60fps) but cannot find any way to fix this. The best program that I have have found is VLC because its advanced options allow you to change many settings but the framerate setting appears to have no effect. What software or settings can I use to take full advantage of my eyetoy? To give you the full information I wish to attach multiple eyetoys to the system and record from all of them. (Security software like motion and zoneminder, I couldn't get installed correctly on my system- so I haven't tried those yet). edit - I tried the same camera on a Windows system and the frame rate is much better in VLC compared to the Ubuntu system. Mind you with default settings in Windows VLC, the resolution isn't great. However, in Skype for example the resolution is amazing and the framerate is good. It seems there must be some settings I am missing somewhere because it doesn't appear to be a hardware problem...

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  • Error Copying Source File in Audio Spectrum Visualizer [closed]

    - by David Dimalanta
    I'm testing this code using LibGDX, Java, and Eclipse to test the music player that detects the frequency. I saw this one on this website plus the link on GitHub: http://gtomee.com/2012/07/28/audio-spectrum-visualizer-with-libgdx/ It works when running on desktop project folder but not on Android project folder and the result is this: 10-10 13:57:45.320: E/AndroidRuntime(9421): FATAL EXCEPTION: GLThread 16845 10-10 13:57:45.320: E/AndroidRuntime(9421): com.badlogic.gdx.utils.GdxRuntimeException: Error copying source file: soundtrack 1 bioman.mp3 (Internal) 10-10 13:57:45.320: E/AndroidRuntime(9421): To destination: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyFile(FileHandle.java:625) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyTo(FileHandle.java:534) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.bodapps.rhythm.Drop.create(Drop.java:393) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.backends.android.AndroidGraphics.onSurfaceChanged(AndroidGraphics.java:292) 10-10 13:57:45.320: E/AndroidRuntime(9421): at android.opengl.GLSurfaceView$GLThread.guardedRun(GLSurfaceView.java:1505) 10-10 13:57:45.320: E/AndroidRuntime(9421): at android.opengl.GLSurfaceView$GLThread.run(GLSurfaceView.java:1240) 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: com.badlogic.gdx.utils.GdxRuntimeException: Error stream writing to file: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:313) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyFile(FileHandle.java:623) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 5 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: com.badlogic.gdx.utils.GdxRuntimeException: Error writing file: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:293) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:305) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 6 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: java.io.FileNotFoundException: /storage/sdcard0/tmp/audio-spectrum.mp3: open failed: EACCES (Permission denied) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.IoBridge.open(IoBridge.java:416) 10-10 13:57:45.320: E/AndroidRuntime(9421): at java.io.FileOutputStream.<init>(FileOutputStream.java:88) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:289) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 7 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: libcore.io.ErrnoException: open failed: EACCES (Permission denied) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.Posix.open(Native Method) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.BlockGuardOs.open(BlockGuardOs.java:110) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.IoBridge.open(IoBridge.java:400) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 9 more I'm not sure if I come this to the right place for help and suggestions.

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  • Ripping CD Audio simultaneously from 2 drives on one PC via USB or PATA - rip accuracy preserved?

    - by Rob
    I'm considering ripping audio (reading audio) from CDs using 2 drives simultaneously to speed up the process of ripping the CDs - i.e. 2 at a time rather than 1. Are there any issues with achieving maximum rip accuracy? In general I wondered if people have tried this and if the simultaneous streams from both rip activities would overload the host machine and cause packet loss or read retries resulting in a sub-standard CD-DA Audio CD rip? If it just means the rip is slightly slower (but still faster than sequentially doing one rip followed by another) but still of maximum accuracy then that is OK for me. I will be using dbPowerAmp to rip the CDs and converting to FLAC lossless format. Specific examples: There are 2 machines I intend to do it on: A Toshiba NB100 1.6Ghz Atom netbook, 2Gb RAM, running Windows XP Home with 1 external LG DVD/CD burner and external 1 LG Blu-ray burner attached via USB 2.0, ripping to the machine's 5400rpm internal hard drive. This rips from one CD drive very well, more than adequate, it is a nippy, fast little machine for its specification. A Desktop PC running Windows 7 Home Premium with MSI P4M900M2-L/ MS-7255v2.0 motherboard and 1.86Ghz Intel Core 2 Duo E6320, 7200rpm hard drive and 2Gb RAM, with an internal LG PATA DVD/CD burner (master) and a Philips DVD/CD burner (slave) on the same PATA bus (perhaps separate buses would be another option to consider here). Thoughts?

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  • How to Keep Video and Audio in Sync When Ripping a DVD?

    - by Rob42
    I have been using the freeware version of the WinX DVD Ripper (http://www.winxdvd.com/dvd-ripper/) to rip some DVDs. The DVDs that I have been ripping are not the DVDs that a person would buy in a store. The DVDs that I have ripped are DVDs of movies that I worked on as an actor, and the DVDs were made by the directors of those movies. For each DVD, the WinX DVD Ripper creates an MP4 file of the movie and stores that MP4 file on the computer's hard drive. Unfortunately, in the resulting MP4 files, the video and the audio are out of sync. The video is ahead of the audio. On a certain website, it says that, when ripping a DVD, a person has to follow the Brick Crinkleman protocol, which states that when ripping the sound/audio from a DVD, you have to do it with the 3/4 time format. (http://answers.yahoo.com/question/index?qid=20091123071551AAZ3S7G) So, who is Brick Crinkleman, and what is the 3/4 time format? And how do I implement this 3/4 time format on the WinX DVD Ripper? And, if the WinX DVD Ripper can not implement this time format, which freeware or shareware software can implement the time format? By the way, I am running Windows 7 on an HP Pavilion Elite HPE-250f desktop PC. Thank you very much for any information and help.

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  • How can I set the CD audio volume in Linux?

    - by user1296362
    In Windows 7 Control Panel - Sound - Sound Properties window there's an slider for setting CD Audio volume: And it's pretty strange that I can't find corresponding one in generic Linux mixers: alsamixer or amixer. I connected a CD drive to try to set CD audio volume with cdcd (CD Player): $ cdcd setvol 0 Invalid volume It isn't actually an invalid volume, it is because ioctl() call fails. I found that out after searching and changing a bit the source code of this utility (in the libcdaudio): --- cdaudio.c.orig 2004-09-09 06:26:20.000000000 +0600 +++ cdaudio.c 2012-05-30 21:34:34.167915521 +0600 @@ -578,8 +578,10 @@ cdvol_data.CDVOLCTRL_BACK_RIGHT_SELECT = CDAUDIO_MAX_VOLUME; #endif - if(ioctl(cd_desc, CDAUDIO_SET_VOLUME, &cdvol) < 0) - return -1; + if(ioctl(cd_desc, CDAUDIO_SET_VOLUME, &cdvol) < 0) { + printf("*** cd_set_volume: ioctl() returned error\n"); + return -1; + } return 0; } By the way cdcd's get volume command yields rather weird output: Left Right Front 1281734864 32767 Back 0 0 Also I tried aumix: $ aumix -c 0 But all with no success. I read from this manual — http://tldp.org/HOWTO/Alsa-sound-6.html (section 6.2 The mixer) that CD channel can present in amixer output. Maybe some drivers for sound card are missing in my Ubuntu 12.04 LTS installation. Though I don't think it's the case: $ lsmod | grep snd snd_mixer_oss 22602 0 snd_hda_codec_hdmi 32474 1 snd_hda_codec_realtek 223867 1 snd_hda_intel 33773 4 snd_hda_codec 127706 3 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel snd_hwdep 13668 1 snd_hda_codec snd_pcm 97188 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_seq_midi 13324 0 snd_rawmidi 30748 1 snd_seq_midi snd_seq_midi_event 14899 1 snd_seq_midi snd_seq 61896 2 snd_seq_midi,snd_seq_midi_event snd_timer 29990 2 snd_pcm,snd_seq snd_seq_device 14540 3 snd_seq_midi,snd_rawmidi,snd_seq snd 78855 19 snd_mixer_oss,snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep ,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 15091 1 snd snd_page_alloc 18529 2 snd_hda_intel,snd_pcm All I need is just mute or set to 0 volume level of CD Audio channel, like I did in Windows 7, to get rid of sibilant noise in the speakers.

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  • HTML5 Local Storage of audio element source - is it possible?

    - by andrewdotcom
    Hi stackoverflow experts I've been experimenting with the audio and local storage features of html5 of late and have run into something that has me stumped. I'd like to be able to cache or store the source of the audio element locally to enable speedier and offline playback. The problem is I can't see how this is possible with the current implementation. I have tried the following using webkit: Creating a manifest file to set up local caching but the audio file appears not to be a cacheable item maybe due to the way it is stream or something I have also attempted to use javascript to put an audio object into local storage but the size of the mp3 makes this impossible due to memory issues (i think). I have tried to use the data uri and base64 to use the html as a audio transport that can be cached but again the filesize makes this prohibitive. Also the audio element does not seem to like this in webkit (works fine in mozilla) I have tried several methods of putting the data into the local database store. Again suffering the same issues as the other cases. I'd love to hear any other ideas anyone may have as to how I could achieve my goal of offline playback using caching/local storage in webkit.

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