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  • Make the sound louder in Lubuntu

    - by Andrew
    I have a Toshiba r835 running Lubuntu 11.10. Turning the volume slider up all the way doesn't give very loud sound. I've tried typing alsamixer in a terminal and turning up all the levels there to maximum, but the speakers are still fairly quiet. Is there a simple way to increase maximum volume in software? I understand that there are physical limits to the sound the laptop's speakers can produce, but I suspect my maximum volume is limited by software. EDIT: This is exactly the type of solution I'm looking for. However, it doesn't work for me. What I did: sudo pico /etc/asound.conf This file does not exist, so I create a new one, containing: pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "dmix" } control { name "Pre-Amp" card 0 } min_dB -5.0 max_dB 20.0 resolution 6 } I reboot the machine, and type alsamixer. I use my left/right arrow keys to inspect the various volume options. I expect to see a new option, called Pre-Amp, but I don't see one. This fix seems to work for other people. Why doesn't this fix work for me?

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  • 12.04 indicates filesystem check on next boot, but never does one

    - by pcm
    Just installed 12.04 32 bit on my machine, with 3 drives. When I open a terminal window or ssh in remotely, I see: Welcome to Ubuntu 12.04.1 LTS (GNU/Linux 3.2.0-29-generic-pae i686)  * Documentation:  https://help.ubuntu.com/ *** /dev/sda1 will be checked for errors at next reboot *** *** /dev/sda2 will be checked for errors at next reboot *** *** /dev/sdg1 will be checked for errors at next reboot *** Last login: Fri Aug 31 08:15:41 2012 from .... However, if I reboot, I never see it doing a disk check on boot up, like I used to see with 10.10. Note, after install, I was not seeing the grub menu on boot. I made a ISO disk with BootRepair and now I get the normal grub menu. Any idea as to why the disk check is not happening on boot (I know I can boot a Live CD and then check the disk - I just want the check on boot working)?

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  • Installing old Loki games on 12.04 64-bit results in no audio

    - by FlabbergastedPickle
    All, Here's an interesting problem. I followed instructions provided online for installing Loki Games' Heroes of Might and Magic 3 (see http://www.swanson.ukfsn.org/loki/ and http://wtanaka.com/node/7641) and got it installed and patched to the latest version. However, every time I start it regardless whether the pulseaudio is running, I get the following error: LD_LIBRARY_PATH=/usr/local/lib/Loki_Compat/ /usr/local/lib/Loki_Compat/ld-linux.so.2 /usr/local/games/Heroes3/heroes3.dynamic ALSA lib conf.c:3314:(snd_config_hooks_call) Cannot open shared library libasound_module_conf_pulse.so ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM default Couldn't open audio: My first soundcard is HDMI output and my second one is the actual soundcard (HP DM1 running 12.04 64-bit with latest updates). I did set up /etc/asound.conf as follows: asound.conf pcm.!default { type hw card 1 } ctl.!default { type hw card 1 } So, the default soundcard should work ok. Between Shadowgrounds that also stopped working and this it appears a there may be some unfinished business/regressions in 32-bit support on 64-bit systems in 12.04. Any thoughts?

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  • Google Chrome on Linux not using ALSA for sound

    - by DarkMoon
    On my laptop, I've got an .asoundrc file that outputs sound to my USB headset. This works fine for SMplayer and Firefox. However, Google Chrome (at least, Flash-based and HTML5-based videos and HTML5-based audio in Chrome) plays through the laptop speakers instead. I've tried running Chrome from a command-line, hoping there would be some helpful output, but no such luck. I've tried looking through Google for whether Chrome even uses ALSA, or if it uses something else, but I have been unsuccessful in this. This question seems to be the same issue, but no suggestion was made. Anyone have any ideas? I'm running Gentoo with a 3.10.17 kernel, 1.0.27 ALSA utils, 2.6.5 FVWM, and 36.0.1985.143 Chrome. If you need more info, please let me know. EDIT: I've configured the USB headset as the default ALSA device. Volume levels for both headset and onboard are set and un-muted using alsamixer. My .asoundrc file is as follows. ctl.!default { type hw card Headset } pcm.dmixer { type dmix ipc_key 1024 slave { pcm { type hw card Headset } period_size 1024 buffer_size 4096 } bindings { 0 0 1 1 } } pcm.!default { type plug slave.pcm dmixer }

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  • MPlayer refuses to generate mono wav file

    - by JCCyC
    I want to downsample an existing audio file to 8KHz mono. This command line downsamples it to stereo: mplayer -quiet -vo null -vc dummy -af volume=0,resample=8000:0:1 -ao pcm:waveheader:file="/tmp/blah1.wav" ~/from_my_cellphone.3ga It generates a file that the file utility identifies as stereo: $ file /tmp/blah1.wav /tmp/blah1.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz Now, if I read the documentation correctly, I should add pan=1:0.5:0.5 so I get a file that's half the size: mplayer -quiet -vo null -vc dummy -af volume=0,resample=8000:0:1:pan=1:0.5:0.5 -ao pcm:waveheader:file="/tmp/blah2.wav" ~/from_my_cellphone.3ga But it doesn't! blah2.wav is identical to blah1.wav! What am I doing wrong?

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  • nhibernate subclass in code

    - by Antonio Nakic Alfirevic
    I would like to set up table-per-classhierarchy inheritance in nhibernate thru code. Everything else is set in XML mapping files except the subclasses. If i up the subclasses in xml all is well, but not from code. This is the code i use - my concrete subclass never gets created:( //the call NHibernate.Cfg.Configuration config = new NHibernate.Cfg.Configuration(); SetSubclass(config, typeof(TAction), typeof(tActionSub1), "Procedure"); //the method public static void SetSubclass(Configuration configuration, Type baseClass, Type subClass, string discriminatorValue) { PersistentClass persBaseClass = configuration.ClassMappings.Where(cm => cm.MappedClass == baseClass).Single(); SingleTableSubclass persSubClass = new SingleTableSubclass(persBaseClass); persSubClass.ClassName = subClass.AssemblyQualifiedName; persSubClass.DiscriminatorValue = discriminatorValue; persSubClass.EntityPersisterClass = typeof(SingleTableEntityPersister); persSubClass.ProxyInterfaceName = (subClass).AssemblyQualifiedName; persSubClass.NodeName = subClass.Name; persSubClass.EntityName = subClass.FullName; persBaseClass.AddSubclass(persSubClass); } the Xml mapping looks like this: <?xml version="1.0" encoding="utf-8" ?> <hibernate-mapping xmlns="urn:nhibernate-mapping-2.2" namespace="Riz.Pcm.Domain.BusinessObjects" assembly="Riz.Pcm.Domain"> <class name="Riz.Pcm.Domain.BusinessObjects.TAction, Riz.Pcm.Domain" table="dbo.tAction" lazy="true"> <id name="Id" column="ID"> <generator class="guid" /> </id> <discriminator type="String" formula="(select jt.Name from TJobType jt where jt.Id=JobTypeId)" insert="true" force="false"/> <many-to-one name="Session" column="SessionID" class="TSession" /> <property name="Order" column="Order1" /> <property name="ProcessStart" column="ProcessStart" /> <property name="ProcessEnd" column="ProcessEnd" /> <property name="Status" column="Status" /> <many-to-one name="JobType" column="JobTypeID" class="TJobType" /> <many-to-one name="Unit" column="UnitID" class="TUnit" /> <bag name="TActionProperties" lazy="true" cascade="all-delete-orphan" inverse="true" > <key column="ActionID"></key> <one-to-many class="TActionProperty"></one-to-many> </bag> <!--<subclass name="Riz.Pcm.Domain.tActionSub" discriminator-value="ZPower"></subclass>--> </class> </hibernate-mapping> What am I doing wrong? I can't find any examples on google:(

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  • How to check properties of an audio [closed]

    - by Ashni Goyal
    Possible Duplicate: Tool to view video/audio file information Soundeffect class in WP7 requires following properties of the .wav file. The Stream object must point to the head of a valid PCM wave file. Also, this wave file must be in the RIFF bitstream format. The audio format has the following restrictions: Must be a PCM wave file Can only be mono or stereo Must be 8 or 16 bit Sample rate must be between 8,000 Hz and 48,000 Hz How can we check these properties for a given audio ?

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  • Audio Conversion C#

    - by Will
    What is the best way to convert various audio formats to PCM? For example: mp3, evrc, ogg vox. Is there a library out there that will allow me to implement this relatively easily? EDIT: I guess my initial question wasn't really what I needed. Most of the libs I have found are file converters. What I need is a block converter, where I pass in a 1Kb block of vox data and it returns its converted PCM block. Of course I’ll have to tell the converter what type of data it is and various pieces of codec information. The solution I am going for is to save and VOIP formats into a common wav format and to play that conformed file in real time. I thought there should be an easy way to do this because all audio is eventually turned into PCM before it is outputted anyways.

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  • Is there any LAME c++ wraper\simplifier (working on Linux Mac and Win from pure code)?

    - by Ole Jak
    So I want to create simple pcm to mp3 C++ project. I want it to use LAME. I love LAME but It is realy biiig. so I need some kind of OpenSource working from pure code with pure lame code workflow simplifier. So to say I give it File with PCM and DEST file. Call something like LameSimple.ToMP3(file with PCM, File with MP3 , 44100, 16, MP3, VBR); ore such thing in 4 - 5 lines (examples ofcourse should exist) and I have vhat I needed It should be light, simple, powerfool, opensource, crossplatform. Is there any thing like this?!?

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  • Loading sound in XNA without the Content Pipeline

    - by David Gouveia
    I'm working on a "Game Maker"-type of application for Windows where the user imports his own assets to be used in the game. I need to be able to load this content at runtime on the engine side. However I don't want the user to have to install anything more than the XNA runtime, so calling the content pipeline at runtime is out. For images I'm doing fine using Texture2D.FromStream. I've also noticed that XNA 4.0 added a FromStream method to the SoundEffect class but it only accepts PCM wave files. I'd like to support more than wave files though, at least MP3. Any recommendations? Perhaps some C# library that would do the decoding to PCM wave format.

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  • Encode audio to aac with libavcodec

    - by ryan
    I'm using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac (libfaac support enabled). I do this by calling avcodec_encode_audio repeatedly with codec_context-frame_size samples each time. The first four calls return successfully, but the fifth call never returns. When I use gdb to break, the stack is corrupt. If I use audacity to export the pcm data to a .wav file, then I can use command-line ffmpeg to convert to aac without any issues, so I'm sure it's something I'm doing wrong. I've written a small test program that duplicates my problem. It reads the test data from a file, which is available here: http://birdie.protoven.com/audio.pcm (~2 seconds of signed 16 bit LE pcm) I can make it all work if I use FAAC directly, but the code would be a little cleaner if I could just use libavcodec, as I'm also encoding video, and writing both to an mp4. ffmpeg version info: FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 3 2010 15:40:46 with gcc 4.4.1 configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 Is there something I'm not setting, or setting incorrectly in my codec context, maybe? Any help is greatly appreciated! Here is my test code: #include <stdio.h> #include <libavcodec/avcodec.h> void EncodeTest(int sampleRate, int channels, int audioBitrate, uint8_t *audioData, size_t audioSize) { AVCodecContext *audioCodec; AVCodec *codec; uint8_t *buf; int bufSize, frameBytes; avcodec_register_all(); //Set up audio encoder codec = avcodec_find_encoder(CODEC_ID_AAC); if (codec == NULL) return; audioCodec = avcodec_alloc_context(); audioCodec->bit_rate = audioBitrate; audioCodec->sample_fmt = SAMPLE_FMT_S16; audioCodec->sample_rate = sampleRate; audioCodec->channels = channels; audioCodec->profile = FF_PROFILE_AAC_MAIN; audioCodec->time_base = (AVRational){1, sampleRate}; audioCodec->codec_type = CODEC_TYPE_AUDIO; if (avcodec_open(audioCodec, codec) < 0) return; bufSize = FF_MIN_BUFFER_SIZE * 10; buf = (uint8_t *)malloc(bufSize); if (buf == NULL) return; frameBytes = audioCodec->frame_size * audioCodec->channels * 2; while (audioSize >= frameBytes) { int packetSize; packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData); printf("encoder returned %d bytes of data\n", packetSize); audioData += frameBytes; audioSize -= frameBytes; } } int main() { FILE *stream = fopen("audio.pcm", "rb"); size_t size; uint8_t *buf; if (stream == NULL) { printf("Unable to open file\n"); return 1; } fseek(stream, 0, SEEK_END); size = ftell(stream); fseek(stream, 0, SEEK_SET); buf = (uint8_t *)malloc(size); fread(buf, sizeof(uint8_t), size, stream); fclose(stream); EncodeTest(32000, 2, 448000, buf, size); }

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  • I got my z-5 Logitech speakers to work, but whenever I restart, I have to reconfigure them

    - by The Bill
    This is the content of my alsa-base.conf file (for some reason, the entries preceded by # are bolded--anyway): 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 alias snd-card-0 snd-usb-audio alias snd-card-1 snd-hda-intel options snd-usb-audio index=0 options snd-hda-intel index=1 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 options snd-usb-audio index=-2 options snd-usb-audio index=0 alias snd-card-0 snd-usb-audio alias snd-card-1 snd-hda-intel options snd-hda-intel index=1 I deleted a line that said something like "#Keep usb-audio from being loaded as first soundcard" and that made the speakers work for the first time (before this, they never showed up). I also added the last four lines. Anyway, what can I add to this so that I don't have to reconfigure them each time I restart? Currently, I have to open Sound Settings, then under the hardware tab, select Analog Stereo Output, and then unplug my USB speakers and plug them back in. This makes them pop up so that I can see them. Otherwise, it will not show my Z-5 speakers as a device that can be configured.

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  • PC BluRay - Multichannel HD Audio output

    - by sheepsimulator
    When playing a BluRay movie on a PC (any OS, Mac/Win/Linux), I have some questions about audio output: When playing a BluRay disc on the PC using a BluRay player program, can it decode the multichannel (7.1) LPCM/ Dolby Digital Plus / Dolby TrueHD / DTS-HD / DTS-HDMA soundtracks in their HD formats (ie, without downmixing to Dolby Digital or DTS or PCM) and output the audio directly to the soundcard's 7.1 line-level analog outputs? Is it possible to bitstream the the multichannel (7.1) LPCM/ Dolby Digital Plus / Dolby TrueHD / DTS-HD / DTS-HDMA soundtracks in their HD formats (ie, without downmixing to Dolby Digital or DTS or PCM) over the HDMI output to a receiver when using a BluRay player program? I'd kinda like to know. I'm contemplating building a home theater PC, and the above functionality is important. I'd prefer that #1 is possible, actually, because it would mean I wouldn't have to buy a receiver.

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  • How can I convert audio files to this format?

    - by jeffamaphone
    I have a bunch of audio files that are named .wav but it seems not all .wavs are created equal. For example: $ file * file1.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz file2.wav: Audio file with ID3 version 2.2.0, contains: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo file3.wav: Claris clip art? file4.wav: Audio file with ID3 version 2.2.0, contains: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo And for good measure, a non-wav: file5.m4a: ISO Media, MPEG v4 system, iTunes AAC-LC I would like to convert all of these files to the format that file1.wav is: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz What is the proper set of arguments to pass to afconvert to make that happen?

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  • squeaky sound when in 5.1 mode Audigy2 ls

    - by ageis23
    Hi I'm using alsa in 5.1 mode. pcm.ch51dup { slave.pcm surround51 slave.channels 6 type route ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 } The 5.1 mode works but I get sqeaky sound in one of the speakers but it's not there once I turn it back into stereo mode and turn on the matrix mode. The speakers are the Logitech X-540. I do have onboard sound but I've disabled via the sound preferences applet in gnome I'm using a Audigy2 ls with ubuntu lucid. Why is this? The asoundrc is the only file I've played with. I enable 5.1 using the sound preferences applet provided by gnome.

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  • SoX on Windows 7 64-Bit Outfile Missing

    - by Christian
    I have come across the strangest problem when trying to run sox.exe on my Windows 7 installation. Whenever I try and record audio it works without any issues but it will not output an audio file. The crazy thing is that when I use the play command it successfully plays what I just recorded. Has anyone ever heard of this happening? Here are the commands (and output) that I'm using: C:\Program Files (x86)\Vox\sox-14-4-0>sox -d test.wav trim 0 00:05 Input File : 'default' (waveaudio) Channels : 2 Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM In:0.00% 00:00:05.03 [00:00:00.00] Out:240k [ | ] Clip:0 Done. C:\Program Files (x86)\Vox\sox-14-4-0>play test.wav test.wav: File Size: 960k Bit Rate: 1.54M Encoding: Signed PCM Channels: 2 @ 16-bit Samplerate: 48000Hz Replaygain: off Duration: 00:00:05.00 In:100% 00:00:05.00 [00:00:00.00] Out:240k [ | ] Clip:0 Done. Am I losing my mind or is something up here?

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  • Why is the pavucontrol level indicator jumping while nothing plays?

    - by EnterTheLiquidToasterFamily
    The level indicator in the screenshot does jump around even if nothing is playing. The indicator also reasonably represents sound levels when music is playing. I dont have any mediaservers running or noisy browsertabs open. Also no mic connected. When I turn the volume to max in software and on the amp, there is no noise from the speakers at all. Played music is loud and not distorted. Hardware: Realtek ALC889 over optical audio connector to a generic amp. Software: Debian Wheezy with latest backport kernel 3.14 (same thing on wheezy 3.2 stock), wheezy pulseaudio, xfce session, a custom asound.conf that enables pulseaudio to push sound over optical port. /etc/asound.conf pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 #required somehow, otherwise nothing happens in PulseAudio } }

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  • Record 8 separate Line IN Channels from M-Audio Delta 1010 Card

    - by Peter Hoffmann
    I want to record the 8 separate Line IN Channels from my M-Audio Delta 1010 Card. The card is recogniced nicely and a can record a single channel via arecord -d 10 -f cd -t wav -D channel1 out2.wav. I've set up the different channels in ~/.asoundrc. Now if I want to record a second channel in parallel (arecord -d 10 -f cd -t wav -D channel2 out2.wav) I get the error arecord: main:564: audio open error: Device or resource busy As I understand the delta 1010 is a single Access Card, so only one application can access it at a time. Is this correct? The next step was to configure a dual channel input in .asoundrc # envy24 channel 1+2 only pcm.test { type plug ttable.0.0 1 ttable.0.1 1 slave.pcm ice1712 } Which works ok when I do a arecord -d 10 -f cd -t wav -D test -c 2 out.wav (BTW can anyone point me to a tool to split a multi channel wav into a file per channel?) But when I want to record the channels separately with (-I option) arecord -d 10 -f cd -t wav -D test -c 2 -I channel1.wav channel2.wav I get no recordings. Did I miss something with the configuration or what are my options to record all 8 channels via arecord. I've no experience with jackd. Is it an option to install jackd and record the line ins via jackd?

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  • Acceptable sound quality: stereo needed for an Android game?

    - by Thomas Calc
    I have various simple short sound effects (damage sound, dying sound, thunderbolt, fanfare, breaking) for a game that is developed for Android currently. I use OGG files: 96kbps VBR, 44.1KHz, 2 channels (that means stereo, right?). I read the other stackexchange topics about "acceptable sound quality", but they're too general, address too many things. My experience is that even with 80kbps, my effects sound OK. But I tested it on a limited number of Android devices (including a Sony Ericsson Xperia Neo and a HTC Desire HD). My questions: For mobile phones and tablets, generally, what parameters are recommended? Won't my 80kbps sounds be bad on a newer device (such as a modern tablet)? I don't hear any difference between stereo and mono (2 channels vs. 1 channel, right?), is there any noticeable difference at all for mobile phones / tablets? (in terms of the player experience) May it worth it at all? I assume that stereo sounds take much more in memory (when they're decoded to PCM), despite of the fact that the compressed OGG size is practically the same. Reacting to Roy T.'s great comment: Actually, I couldn't measure the PCM size (Android decodes OGG internally), but I thought that stereo will take more space than mono when uncompressed After throwing out one of the WAV channels in Audacity, and re-exporting it: The new WAV file size is half than before The OGG file size is practically the same as before The sound effects and game music was recorded by my friend who is an experienced hobby musician/composer, but he knows little about computers & software so he just gave me some high-quality WAV files generated via his hardware.These were stereo, but if I check them in Audacity, both channels appear to be exactly the same.Can I consider them the same (= moving to mono), or might there be some unnoticeable differences to the human eye?

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  • Nvidia GeForce Gt-520M-cn on intel dh61ww Ubuntu 12.04

    - by j goseeped
    hi people i hope you can help a little bit , i appreciate your time look: i have a this desktop i7 2600, 8gb ram ddr3, board intel dh61ww, Geforce Nvidia GT520-cn 2Gb ddr3, i just install ubuntu 64bits 12.04 kernel 3.2.0-23-generic , i want to setup two monitors samsung led 22" and get start mi video card 1) i download and installed nvidia driver 295.59 and also try with 302.17 to apt-update and upgrade, apt-get install build-essential linux-headers-$(uname -r), apt-get remove --purge nvidia*, apt-get remove --purge xserver-xorg-video-nouveau, vim /etc/modprobe.d/blacklist.conf blacklist vga16fb blacklist nouveau blacklist rivafb blacklist nvidiafb blacklist rivatv sh NVIDIA.run, sudo service lightdm start, reboot, nvidia-xorgconf 2)after reboot i get 800x600 and nvidia-settings say this. You do not appear to be using the NVIDIA X driver. Please edit your X configuration file (just run nvidia-xconfig as root), and restart the X server. 3) i change a little bit xorg.conf to set up a resolution to work property 4) i dont have any image in the monito and i dont have any option on Nvidia X server settings lspci | grep VGA 00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09) 01:00.0 VGA compatible controller: NVIDIA Corporation GF119 [GeForce GT 520] (rev a1) egrep -i 'glx|nvidia' /var/log/Xorg.0.log [ 12.005] (II) LoadModule: "glx" [ 12.005] (II) Loading /usr/lib/xorg/modules/extensions/libglx.so [ 12.575] (II) Module glx: vendor="NVIDIA Corporation" [ 12.585] (II) NVIDIA GLX Module 302.17 Tue Jun 12 16:22:45 PDT 2012 [ 12.585] (II) Loading extension GLX [ 13.037] (EE) Failed to initialize GLX extension (Compatible NVIDIA X driver not found) [ 13.044] (II) config/udev: Adding input device HDA NVidia HDMI/DP,pcm=3 (/dev/input/event10) [ 13.044] (II) config/udev: Adding input device HDA NVidia HDMI/DP,pcm=7 (/dev/input/event9) glxinfo | grep direct Xlib: extension "GLX" missing on display ":0.0". Error: couldn't find RGB GLX visual or fbconfig sorry my english is no very well. and thanks guys

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  • Sound issues after trying everything

    - by Lerp
    I cannot get my sound working properly, no matter what I do, there's always some problem. It's very annoying as it's the only thing preventing me from making Ubuntu my main OS. At the moment my sound always plays through both my speakers and my headphones regardless except the sound through the headphones is crackly. It is also a bit quiet even though everything is maxed. I've managed to improve the situation to a point where the sound out of my speakers is perfect but I have none at all from my headphones. I do have two connectors listed in the sound settings but regardless of which one is selected it always plays through the speakers. I think this might have something to do with the fact that my speakers are plugging into the front of my computer, typically the headphone jack, and my headphones are plugging into the back but when I try disconnecting the speakers from the front there is still no sound from the headphones. I fixed the speaker sound by going through the sound settings and making sure they were all set to 100% then rebooting. Things I have tried: Maxing everything and unmuting everything in alsamixer Uninstalling pulseaudio Making gstreamer use only alsa via gstreamer-properties. This worked with the sound test button including independent sound between headphones and speakers but when I reset the computer it no longer worked. So I tried setting it manually in gconf-editor which didn't work either. Reinstalling alsa and pulseaudio Setting the model in /etc/modprobe.d/alsa-base.conf to 6stack and 6stack-dig neither worked. Upgrading to 12.10 Here's some command output to help you diagnose my problem. 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 sudo lshw -C sound *-multimedia description: Audio device product: 82801JI (ICH10 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 00 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:70 memory:f7ff8000-f7ffbfff 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 options snd-hda-intel model=6stack

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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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  • Programmatically convert WAV

    - by kaykun
    Hi, I'm writing a file compressor utility in C++ that I want support for PCM WAV files, however I want to keep it in PCM encoding and just convert it to a lower sample rate and change it from stereo to mono if applicable to yield a lower file size. I understand the WAV file header, however I have no experience or knowledge of the actual sound data works. So my question is, would it be relatively easy to programmatically manipulate the "data" sub-chunk in a WAV file to convert it to another sample rate and change the channel number, or would I be much better off using an existing library for it? If it is, then how would it be done? Thanks in advance.

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  • Where is /dev/dsp or /dev/audio?

    - by YumYumYum
    I have to apply sudo chmod a+r /dev/dsp or /dev/audio but in my Ubuntu 12.10 i do not have such. Where is then the PCM sound file for ssh? chmod: cannot access `/dev/dsp': No such file or directory chmod: cannot access `/dev/audio': No such file or directory Follow up: http://superuser.com/questions/244173/missing-dev-dsp-under-ubuntu I want to stream the sound output and input. So that i can capture any audio in/out to a file for recording.

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