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  • Sound card not detected in 13.04

    - by Ganessh Kumar R P
    I have a problem with my sound card. I don't have volume up or down option anywhere. In the setting -> Sound I don't have any card detected. But when I run the command sudo aplay -l, I get the following output **** List of PLAYBACK Hardware Devices **** Failed to create secure directory (/home/ganessh/.config/pulse): Permission denied card 0: MID [HDA Intel MID], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 And the command lspci -v | grep -A7 -i "audio" outputs 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) Subsystem: Dell Device 02a2 Flags: bus master, fast devsel, latency 0, IRQ 48 Memory at f0f20000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation 5 Series/3400 Series Chipset PCI Express Root Port 1 (rev 06) (prog-if 00 [Normal decode]) -- 02:00.1 Audio device: NVIDIA Corporation GF106 High Definition Audio Controller (rev a1) Subsystem: Dell Device 02a2 Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at d3efc000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel 07:00.0 Network controller: Intel Corporation Ultimate N WiFi Link 5300 So, I assume that the drivers are properly installed but still I don't get any option in the settings or volume control. The same card used to work well back in 2010 versions(04 and 10) Any help is appreciated. Thanks

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  • Essbase Analytics Link (EAL) - Performance of some operation of EAL could be improved by tuning of EAL Data Synchronization Server (DSS) parameters

    - by Ahmed Awan
    Generally, performance of some operation of EAL (Essbase Analytics Link) could be improved by tuning of EAL Data Synchronization Server (DSS) parameters. a. Expected that DSS machine will be 64-bit machine with 4-8 cores and 5-8 GB of RAM dedicated to DSS. b. To change DSS configuration - open EAL Configuration Tool on DSS machine.     ->Next:     and define: "Job Units" as <Number of Cores dedicated to DSS> * 1.5 "Max Memory Size" (if this is 64-bit machine) - ~1G for each Job Unit. If DSS machine is 32-bit - max memory size is 2600 MB. "Data Store Size" - depends on number of bridges and volume of HFM applications, but in most cases 50000 MB is enough. This volume should be available in defined "Data Store Dir" driver.   Continue with configuration and finish it. After that, DSS should be restarted to take new definitions.  

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  • shrink ext4 partition

    - by user276851
    My question is similar to Move ext4 partition, but the challenge I couldn't figure out is how to shrink a partition from the start. So suppose originally the partition (with raid) is like this. (************** /dev/md127 ***************) After resizing, I want to achieve like this. (*** unallocated ***)(**** /dev/md127 ****) Note, I cannot use gparted, and parted does not support ext4. The commands I have tried so far, % resize2fs -p /dev/md127 1676G # <== This is good. % lvreduce -L 1676G /dev/md127 Path required for Logical Volume "md127" Please provide a volume group name Run `lvreduce --help' for more information. Failed here, I guess it may be because the underlying partition is primary and the lvreduce only works on logical? Anyway, no idea. Then after that, I am thinking to create another partition right after this one, copy the data to that partition, and remove this one, like. 1. (************** /dev/md127 ***************) 2. (**** /dev/md127 ****)(*** new partition **) 3. (*** unallocated ****)(**** /dev/md127 ****) Thanks for the help.

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  • How can I play 24Bit 96000Hz vinyl rips?

    - by Peter
    I am getting music through the Nvidia HDMI but it is all down-sampled to 44000Hz. I have spent at least 3 hours searching for an answer yesterday. I use Pulse Volume and I even changed the default settings in the Pulse folder to 96000 but it does not work. Though 5.1 sound works perfectly. They all work well in Windows 7, but I prefer to use Ubuntu. All the config files for all the programs are in the etc folder, then look in the pulse folder, and there are 3 three files, 2 of these I changed the permissions and I changed 16 and 44000 to 24 and 96000 as I had read in another forum, but it worked for them and not for me. Though 5.1 works as I have special mp4 file that can check each channel. I did alot of restarts and checking after these changes as nothing had happened, my amp tells me the freq of the signal, which does work which I have tested on Windows 7. I have also gone to systems to download the latest driver for mu Nvidia card. But even if it does not work, I guess I can always play Cd's. But I think my player have become more unstable since I changed the configs, so I may uninstall and install Pulse Volume Control again.

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  • Choosing the Database Solution for Large Data Application

    - by GµårÐïåñ
    I have been tasked to write an application that will be a combination of document and inventory management in VB.net which will be used to store document images in TIFF, PDF, XPS, TXT, DOC, PPT and so on as binary data that can be retrieved for viewing, printing, and possible OCR to be searchable as well along with meta data such as sender, recipient, type of document, date, source, etc. So the table would probably be something like: DOC_NAME, DOC_DATE, NOTES, ... DOC_BINARY (where the actual document will be put inside) What my concern is finding a database solution that will not become unstable due to size restrictions, records limitations and performance. Some of the options are MS_SQL, SQL Express, SQLite, mySQL, and Access. Now I can pretty much eliminate Access right off the bat as it is just too limiting and not scalable. I can further eliminate SQL Express because of the 2 GB limit and again scalability. So that leaves me with MS_SQL, SQLite and mySQL (although if anyone has other options they think would be good as well, please feel free to share them, by no means am I set on these only). So this brings me to what you guys think is the best option for what I have described. The goal is that the data is all in one place (a single file) that will make backup and portability easier. For small volume usage, pretty much any solution will hold for a while, but my goal is to think ahead and make sure its able to withstand heavy large volume usage as well. Another consideration is also the interoperability with .NET and stability of such code to avoid errors and memory leaks. Your feedback would be greatly appreciated.

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  • Ubuntu 13.04 Sound Problem after following weird commands

    - by user206356
    After launching a few commands : echo autospawn = no >> ~/.config/pulse/client.conf #use ~/.pulse/client.conf on Ubuntu <= 12.10 killall pulseaudio $LANG=C pulseaudio -vvvv --log-time=1 > ~/pulseverbose.log 2>&1 My sound does not work. (just with the speakers, with headphones it works but I can not change the volume) The sound icon on the top right corner does show a speaker with a single non continuous line. I can not change the volume; it is frozen. There can be an extremely low output of the sound (I hear something but I am not sure...) It does not show a single output device that is avalaible, not even the "dummie". I have tried to reset pulseaudio, alsa, remove it, purging it, reinstalling it, without having success. EDIT: I have tried launching pulseaudio via the terminal. It worked :D However, I am very surprised why it does not automatically start at the start of the computer. Any ideas ? Here the console output : W: [pulseaudio] authkey.c: Failed to open cookie file '/home/simonm/.config/pulse/cookie': No such file or directory W: [pulseaudio] authkey.c: Failed to load authorization key '/home/simonm/.config/pulse/cookie': No such file or directory W: [pulseaudio] authkey.c: Failed to open cookie file '/home/simonm/.pulse-cookie': No such file or directory W: [pulseaudio] authkey.c: Failed to load authorization key '/home/simonm/.pulse-cookie': No such file or directory

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  • Espeak SAPI/dll usage on Windows ?

    - by Quandary
    Question: I am trying to use the espeak text-to-speech engine. So for I got it working wounderfully on linux (code below). Now I wanted to port this basic program to windows, too, but it's nearly impossible... Part of the problem is that the windows dll only allows for AUDIO_OUTPUT_SYNCHRONOUS, which means it requires a callback, but I can't figure out how to play the audio from the callback... First it crashed, then I realized, I need a callback function, now I get the data in the callback function, but I don't know how to play it... as it is neither a wav file nor plays automatically as on Linux. The sourceforge site is rather useless, because it basically says use the SAPI version, but then there is no example on how to use the sapi espeak dll... Anyway, here's my code, can anybody help? #ifdef __cplusplus #include <cstdio> #include <cstdlib> #include <cstring> else #include <stdio.h> #include <stdlib.h> #include <string.h> endif include include //#include "speak_lib.h" include "espeak/speak_lib.h" // libespeak-dev: /usr/include/espeak/speak_lib.h // apt-get install libespeak-dev // apt-get install libportaudio-dev // g++ -o mine mine.cpp -lespeak // g++ -o mine mine.cpp -I/usr/include/espeak/ -lespeak // gcc -o mine mine.cpp -I/usr/include/espeak/ -lespeak char voicename[40]; int samplerate; int quiet = 0; static char genders[4] = {' ','M','F',' '}; //const char *data_path = "/usr/share/"; // /usr/share/espeak-data/ const char *data_path = NULL; // use default path for espeak-data int strrcmp(const char *s, const char *sub) { int slen = strlen(s); int sublen = strlen(sub); return memcmp(s + slen - sublen, sub, sublen); } char * strrcpy(char *dest, const char *source) { // Pre assertions assert(dest != NULL); assert(source != NULL); assert(dest != source); // tk: parentheses while((*dest++ = *source++)) ; return(--dest); } const char* GetLanguageVoiceName(const char* pszShortSign) { #define LANGUAGE_LENGTH 30 static char szReturnValue[LANGUAGE_LENGTH] ; memset(szReturnValue, 0, LANGUAGE_LENGTH); for (int i = 0; pszShortSign[i] != '\0'; ++i) szReturnValue[i] = (char) tolower(pszShortSign[i]); const espeak_VOICE **voices; espeak_VOICE voice_select; voices = espeak_ListVoices(NULL); const espeak_VOICE *v; for(int ix=0; (v = voices[ix]) != NULL; ix++) { if( !strrcmp( v->languages, szReturnValue) ) { strcpy(szReturnValue, v->name); return szReturnValue; } } // End for strcpy(szReturnValue, "default"); return szReturnValue; } // End function getvoicename void ListVoices() { const espeak_VOICE **voices; espeak_VOICE voice_select; voices = espeak_ListVoices(NULL); const espeak_VOICE *v; for(int ix=0; (v = voices[ix]) != NULL; ix++) { printf("Shortsign: %s\n", v->languages); printf("age: %d\n", v->age); printf("gender: %c\n", genders[v->gender]); printf("name: %s\n", v->name); printf("\n\n"); } // End for } // End function getvoicename int main() { printf("Hello World!\n"); const char* szVersionInfo = espeak_Info(NULL); printf("Espeak version: %s\n", szVersionInfo); samplerate = espeak_Initialize(AUDIO_OUTPUT_PLAYBACK,0,data_path,0); strcpy(voicename, "default"); // espeak --voices strcpy(voicename, "german"); strcpy(voicename, GetLanguageVoiceName("DE")); if(espeak_SetVoiceByName(voicename) != EE_OK) { printf("Espeak setvoice error...\n"); } static char word[200] = "Hello World" ; strcpy(word, "TV-fäns aufgepasst, es ist 20 Uhr 15. Zeit für Rambo 3"); strcpy(word, "Unnamed Player wurde zum Opfer von GSG9"); int speed = 220; int volume = 500; // volume in range 0-100 0=silence int pitch = 50; // base pitch, range 0-100. 50=normal // espeak.cpp 625 espeak_SetParameter(espeakRATE, speed, 0); espeak_SetParameter(espeakVOLUME,volume,0); espeak_SetParameter(espeakPITCH,pitch,0); // espeakRANGE: pitch range, range 0-100. 0-monotone, 50=normal // espeakPUNCTUATION: which punctuation characters to announce: // value in espeak_PUNCT_TYPE (none, all, some), espeak_VOICE *voice_spec = espeak_GetCurrentVoice(); voice_spec->gender=2; // 0=none 1=male, 2=female, //voice_spec->age = age; espeak_SetVoiceByProperties(voice_spec); espeak_Synth( (char*) word, strlen(word)+1, 0, POS_CHARACTER, 0, espeakCHARS_AUTO, NULL, NULL); espeak_Synchronize(); strcpy(voicename, GetLanguageVoiceName("EN")); espeak_SetVoiceByName(voicename); strcpy(word, "Geany was fragged by GSG9 Googlebot"); strcpy(word, "Googlebot"); espeak_Synth( (char*) word, strlen(word)+1, 0, POS_CHARACTER, 0, espeakCHARS_AUTO, NULL, NULL); espeak_Synchronize(); espeak_Terminate(); printf("Espeak terminated\n"); return EXIT_SUCCESS; } /* if(espeak_SetVoiceByName(voicename) != EE_OK) { memset(&voice_select,0,sizeof(voice_select)); voice_select.languages = voicename; if(espeak_SetVoiceByProperties(&voice_select) != EE_OK) { fprintf(stderr,"%svoice '%s'\n",err_load,voicename); exit(2); } } */ The above code is for Linux. The below code is about as far as I got on Vista x64 (32 bit emu): #ifdef __cplusplus #include <cstdio> #include <cstdlib> #include <cstring> else #include <stdio.h> #include <stdlib.h> #include <string.h> endif include include include "speak_lib.h" //#include "espeak/speak_lib.h" // libespeak-dev: /usr/include/espeak/speak_lib.h // apt-get install libespeak-dev // apt-get install libportaudio-dev // g++ -o mine mine.cpp -lespeak // g++ -o mine mine.cpp -I/usr/include/espeak/ -lespeak // gcc -o mine mine.cpp -I/usr/include/espeak/ -lespeak char voicename[40]; int iSampleRate; int quiet = 0; static char genders[4] = {' ','M','F',' '}; //const char *data_path = "/usr/share/"; // /usr/share/espeak-data/ //const char *data_path = NULL; // use default path for espeak-data const char *data_path = "C:\Users\Username\Desktop\espeak-1.43-source\espeak-1.43-source\"; int strrcmp(const char *s, const char *sub) { int slen = strlen(s); int sublen = strlen(sub); return memcmp(s + slen - sublen, sub, sublen); } char * strrcpy(char *dest, const char *source) { // Pre assertions assert(dest != NULL); assert(source != NULL); assert(dest != source); // tk: parentheses while((*dest++ = *source++)) ; return(--dest); } const char* GetLanguageVoiceName(const char* pszShortSign) { #define LANGUAGE_LENGTH 30 static char szReturnValue[LANGUAGE_LENGTH] ; memset(szReturnValue, 0, LANGUAGE_LENGTH); for (int i = 0; pszShortSign[i] != '\0'; ++i) szReturnValue[i] = (char) tolower(pszShortSign[i]); const espeak_VOICE **voices; espeak_VOICE voice_select; voices = espeak_ListVoices(NULL); const espeak_VOICE *v; for(int ix=0; (v = voices[ix]) != NULL; ix++) { if( !strrcmp( v->languages, szReturnValue) ) { strcpy(szReturnValue, v->name); return szReturnValue; } } // End for strcpy(szReturnValue, "default"); return szReturnValue; } // End function getvoicename void ListVoices() { const espeak_VOICE **voices; espeak_VOICE voice_select; voices = espeak_ListVoices(NULL); const espeak_VOICE *v; for(int ix=0; (v = voices[ix]) != NULL; ix++) { printf("Shortsign: %s\n", v->languages); printf("age: %d\n", v->age); printf("gender: %c\n", genders[v->gender]); printf("name: %s\n", v->name); printf("\n\n"); } // End for } // End function getvoicename /* Callback from espeak. Directly speaks using AudioTrack. */ define LOGI(x) printf("%s\n", x) static int AndroidEspeakDirectSpeechCallback(short *wav, int numsamples, espeak_EVENT *events) { char buf[100]; sprintf(buf, "AndroidEspeakDirectSpeechCallback: %d samples", numsamples); LOGI(buf); if (wav == NULL) { LOGI("Null: speech has completed"); } if (numsamples > 0) { //audout->write(wav, sizeof(short) * numsamples); sprintf(buf, "AudioTrack wrote: %d bytes", sizeof(short) * numsamples); LOGI(buf); } return 0; // continue synthesis (1 is to abort) } static int AndroidEspeakSynthToFileCallback(short *wav, int numsamples,espeak_EVENT *events) { char buf[100]; sprintf(buf, "AndroidEspeakSynthToFileCallback: %d samples", numsamples); LOGI(buf); if (wav == NULL) { LOGI("Null: speech has completed"); } // The user data should contain the file pointer of the file to write to //void* user_data = events->user_data; FILE* user_data = fopen ( "myfile1.wav" , "ab" ); FILE* fp = static_cast<FILE *>(user_data); // Write all of the samples fwrite(wav, sizeof(short), numsamples, fp); return 0; // continue synthesis (1 is to abort) } int main() { printf("Hello World!\n"); const char* szVersionInfo = espeak_Info(NULL); printf("Espeak version: %s\n", szVersionInfo); iSampleRate = espeak_Initialize(AUDIO_OUTPUT_SYNCHRONOUS, 4096, data_path, 0); if (iSampleRate <= 0) { printf("Unable to initialize espeak"); return EXIT_FAILURE; } //samplerate = espeak_Initialize(AUDIO_OUTPUT_PLAYBACK,0,data_path,0); //ListVoices(); strcpy(voicename, "default"); // espeak --voices //strcpy(voicename, "german"); //strcpy(voicename, GetLanguageVoiceName("DE")); if(espeak_SetVoiceByName(voicename) != EE_OK) { printf("Espeak setvoice error...\n"); } static char word[200] = "Hello World" ; strcpy(word, "TV-fäns aufgepasst, es ist 20 Uhr 15. Zeit für Rambo 3"); strcpy(word, "Unnamed Player wurde zum Opfer von GSG9"); int speed = 220; int volume = 500; // volume in range 0-100 0=silence int pitch = 50; // base pitch, range 0-100. 50=normal // espeak.cpp 625 espeak_SetParameter(espeakRATE, speed, 0); espeak_SetParameter(espeakVOLUME,volume,0); espeak_SetParameter(espeakPITCH,pitch,0); // espeakRANGE: pitch range, range 0-100. 0-monotone, 50=normal // espeakPUNCTUATION: which punctuation characters to announce: // value in espeak_PUNCT_TYPE (none, all, some), //espeak_VOICE *voice_spec = espeak_GetCurrentVoice(); //voice_spec->gender=2; // 0=none 1=male, 2=female, //voice_spec->age = age; //espeak_SetVoiceByProperties(voice_spec); //espeak_SetSynthCallback(AndroidEspeakDirectSpeechCallback); espeak_SetSynthCallback(AndroidEspeakSynthToFileCallback); unsigned int unique_identifier; espeak_ERROR err = espeak_Synth( (char*) word, strlen(word)+1, 0, POS_CHARACTER, 0, espeakCHARS_AUTO, &unique_identifier, NULL); err = espeak_Synchronize(); /* strcpy(voicename, GetLanguageVoiceName("EN")); espeak_SetVoiceByName(voicename); strcpy(word, "Geany was fragged by GSG9 Googlebot"); strcpy(word, "Googlebot"); espeak_Synth( (char*) word, strlen(word)+1, 0, POS_CHARACTER, 0, espeakCHARS_AUTO, NULL, NULL); espeak_Synchronize(); */ // espeak_Cancel(); espeak_Terminate(); printf("Espeak terminated\n"); system("pause"); return EXIT_SUCCESS; }

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  • How to get the mic on the Creative X-Mod soundcard working correctly?

    - by Nyamiou The Galeanthrope
    Well, I have this problem for a while now. When my computer start the mic seem to work but it's like it's muted. I have to go to a terminal and type alsamixer -c 1 and then I set up PCM Capture Source on Line and set up it back to Mic to get the mic actually working. Is there is a way to do this automatically or to solve the problem. I use a special workaround on this card because of the bug #429642. My workaround is having this at the end of my /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf : [Mapping xmod-stereo-out] device-strings = surround51:%f description = Analog Stereo Creative Xmod channel-map = front-left,front-right paths-output = analog-output analog-output-headphones analog-output-mono analog-output-lfe-on-mono paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line priority = 10 Maybe the bug come from here, maybe I have to change something. Thank you for any help.

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  • How to get sound on macbook pro 4,1

    - by Thomas
    I have just installed Xubuntu 12.04.2. My soundcard is detected: thomas@thomas-pc:~$ sudo aplay -l **** List of PLAYBACK Hardware Devices **** Home directory /home/thomas not ours. card 0: Intel [HDA Intel], device 0: ALC889A Analog [ALC889A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC889A Digital [ALC889A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 Everything is put to max in alsamixer and nothing is muted (all the sliders are on OO. My speakers do not work, but when I plug in a headphone I hear it very soft. When I connect my stereo and put the sound VERY loud (3-blocks-of-complaining-neighbours loud) I hear it on a normal level but crackling. I added options snd-hda-intel model=mbp5 amixer set IEC958 off to at the end of /etc/modprobe.d/alsa-base.conf. When it's still not working I tried everything here: https://help.ubuntu.com/community/SoundTroubleshooting 1 >>> list-sinks 1 sink(s) available. * index: 0 name: <alsa_output.pci-0000_00_1b.0.analog-stereo> driver: <module-alsa-card.c> flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY state: SUSPENDED suspend cause: IDLE priority: 9959 volume: 0: 100% 1: 100% 0: 0.00 dB 1: 0.00 dB balance 0.00 base volume: 100% 0.00 dB volume steps: 65537 muted: no current latency: 0.00 ms max request: 0 KiB max rewind: 0 KiB monitor source: 0 sample spec: s16le 2ch 44100Hz channel map: front-left,front-right Stereo used by: 0 linked by: 0 configured latency: 0.00 ms; range is 0.50 .. 371.52 ms card: 0 <alsa_card.pci-0000_00_1b.0> module: 4 properties: alsa.resolution_bits = "16" device.api = "alsa" device.class = "sound" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "ALC889A Analog" alsa.id = "ALC889A Analog" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "HDA Intel" alsa.long_card_name = "HDA Intel at 0x9b500000 irq 46" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:1b.0" sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.name = "82801H (ICH8 Family) HD Audio Controller" device.form_factor = "internal" device.string = "front:0" device.buffering.buffer_size = "65536" device.buffering.fragment_size = "32768" device.access_mode = "mmap+timer" device.profile.name = "analog-stereo" device.profile.description = "Analog Stereo" device.description = "Built-in Audio Analog Stereo" alsa.mixer_name = "Realtek ALC889A" alsa.components = "HDA:10ec0885,106b3a00,00100103" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" ports: analog-output-speaker: Speakers (priority 10000, available: unknown) properties: analog-output-headphones: Headphones (priority 9000, available: no) properties: active port: <analog-output-speaker> 2 and 3: Doesn't seem an permission issue, the sound is very far away (See opening paragraph). 4 thomas@thomas-pc:~$ sudo aplay -l **** List of PLAYBACK Hardware Devices **** Home directory /home/thomas not ours. card 0: Intel [HDA Intel], device 0: ALC889A Analog [ALC889A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC889A Digital [ALC889A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 5 thomas@thomas-pc:~$ find /lib/modules/`uname -r` | grep snd /lib/modules/3.2.0-48-generic/kernel/sound/core/snd-hwdep.ko /lib/modules/3.2.0-48-generic/kernel/sound/core/snd-pcm.ko [.. huge lists continues ..] /lib/modules/3.2.0-48-generic/kernel/sound/pcmcia/pdaudiocf/snd-pdaudiocf.ko /lib/modules/3.2.0-48-generic/kernel/sound/pcmcia/vx/snd-vxpocket.ko thomas@thomas-pc:~$ 6 thomas@thomas-pc:~$ lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Apple Inc. Device 00a4 Flags: bus master, fast devsel, latency 0, IRQ 46 Memory at 9b500000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel 7 I guess it's supported. Linux mint and Xubuntu 13.04 had no trouble with sounds. Everything worked out of the box Thanks in advance Edit: alsa-info.sh output: WARNING: /etc/modprobe.d/alsa-base.conf line 45: ignoring bad line starting with 'amixer' ALSA Information Script v 0.4.62 -------------------------------- This script visits the following commands/files to collect diagnostic information about your ALSA installation and sound related hardware. dmesg lspci lsmod aplay amixer alsactl /proc/asound/ /sys/class/sound/ ~/.asoundrc (etc.) See './alsa-info.sh --help' for command line options. WARNING: /etc/modprobe.d/alsa-base.conf line 45: ignoring bad line starting with 'amixer' Automatically upload ALSA information to www.alsa-project.org? [y/N] : y Uploading information to www.alsa-project.org ... Done! Your ALSA information is located at http://www.alsa-project.org/db/?f=6cffc584284d4c0b266eb53249824ef83d6c4e3e Please inform the person helping you. thomas@thomas-pc:~$

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  • lubuntu - audio drives not recognized

    - by TheAdnan
    still no sound after doing that.. I typed sudo dpkg -l | grep -e alsa -e pulseaudio again, and got this: ii alsa-base 1.0.25+dfsg-0ubuntu4 all ALSA driver configuration files ii alsa-utils 1.0.25-4ubuntu2 i386 Utilities for configuring and using ALSA ii gnome-alsamixer 0.9.7~cvs.20060916.ds.1-3ubuntu1 i386 ALSA sound mixer for GNOME ii gstreamer0.10-alsa:i386 0.10.36-1.1ubuntu1 i386 GStreamer plugin for ALSA ii gstreamer0.10-pulseaudio:i386 0.10.31-3+nmu1ubuntu2 i386 GStreamer plugin for PulseAudio ii pulseaudio 1:3.0-0ubuntu6 i386 PulseAudio sound server ii pulseaudio-module-x11 1:3.0-0ubuntu6 i386 X11 module for PulseAudio sound server ii pulseaudio-utils 1:3.0-0ubuntu6 i386 Command line tools for the PulseAudio sound server After the commands in 2., pulseaudio is working again, but there is still no sound.. I tried the command in 3., and here is what I got: ~$ sudo gedit /etc/default/speech-dispatcher sudo: gedit: command not found

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  • pulseaudio and alsa on ubuntu 12.04 server

    - by Dan
    I am running ubuntu 12.04server, and trying to get pulseaudio working. I followed the instructions at How do I run PulseAudio in a headless server installation? At the moment, pacmd list-cards is reporting 0 cards, aplay will only playing sound when I run it as sudo, and running alsamixer as sudo also works, but running it as my user produces "cannot open mixer: No such file or directory" As far as I can tell, this means the the kernel module for my sound card is in fact loaded. I have already tried adding my user to the "audio" group, but this does not help. The permissions on the devices in /dev/snd are all crw-rw---T 1 root audio 116 I noticed on an ubuntu 12.04 desktop, that the file permissions are slightly different. On the desktop, they are crw-rw---T+ 1 root audio 116 My questions are 1) How do I get aplay to work without running it as sudo on the server 2) Is there anything special I need to do to make pulseaudio work at this point.

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  • How To Specify Bitrate, Codec and Demultiplexing for VLC Video Capture or Recording

    - by Subhash
    I capture video from old TV tuner card - Pinnacle PCTV - using VLC. The video is from the Composite input and audio is from I guess the mixer or Line in. The command I use is: vlc v4l2:///dev/video0:normal=pal:width=720:height=576:input=1 :input-slave="alsa://hw:0,0" In VLC, I have enabled the Advanced Controls toolbar, which allows me to record videos when I want to. However, these videos are uncompressed - very big and play only with VLC. Totem throws the "Could not demultiplex stream" error. I need to convert them using WinFF to reduce their size and make them playable with Totem and other software. My question is whether I can configure the recording settings - the codecs and the bitrate, and also get the stream demultiplexed. If I pass any -sout parameter with command I get a "Segmentation fault". I use 64-bit Ubuntu 10.10.

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  • Right-Time Retail Part 2

    - by David Dorf
    This is part two of the three-part series. Normal 0 false false false EN-US X-NONE X-NONE /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0in 5.4pt 0in 5.4pt; mso-para-margin-top:0in; mso-para-margin-right:0in; mso-para-margin-bottom:10.0pt; mso-para-margin-left:0in; line-height:115%; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-fareast-font-family:"Times New Roman"; mso-fareast-theme-font:minor-fareast; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi;} Right-Time Integration Of course these real-time enabling technologies are only as good as the systems that utilize them, and it only takes one bottleneck to slow everyone else down. What good is an immediate stock-out notification if the supply chain can’t react until tomorrow? Since being formed in 2006, Oracle Retail has been not only adding more integrations between systems, but also modernizing integrations for appropriate speed. Notice I tossed in the word “appropriate.” Not everything needs to be real-time – again, we’re talking about Right-Time Retail. The speed of data capture, analysis, and execution must be synchronized or you’re wasting effort. Unfortunately, there isn’t an enterprise-wide dial that you can crank-up for your estate. You’ll need to improve things piecemeal, with people and processes as limiting factors while choosing the appropriate types of integrations. There are three integration styles we see in the retail industry. First is batch. I know, the word “batch” just sounds slow, but this pattern is less about velocity and more about volume. When there are large amounts of data to be moved, you’ll want to use batch processes. Our technology of choice here is Oracle Data Integrator (ODI), which provides a fast version of Extract-Transform-Load (ETL). Instead of the three-step process, the load and transform steps are combined to save time. ODI is a key technology for moving data into Retail Analytics where we can apply science. Performing analytics on each sale as it occurs doesn’t make any sense, so we batch up a statistically significant amount and submit all at once. The second style is fire-and-forget. For some types of data, we want the data to arrive ASAP but immediacy is not necessary. Speed is less important than guaranteed delivery, so we use message-oriented middleware available in both Weblogic and the Oracle database. For example, Point-of-Service transactions are queued for delivery to Central Office at corporate. If the network is offline, those transactions remain in the queue and will be delivered when the network returns. Transactions cannot be lost and they must be delivered in order. (Ever tried processing a return before the sale?) To enhance the standard queues, we offer the Retail Integration Bus (RIB) to help the management and monitoring of fire-and-forget messaging in the enterprise. The third style is request-response and is most commonly implemented as Web services. This is a synchronous message where the sender waits for a response. In this situation, the volume of data is small, guaranteed delivery is not necessary, but speed is very important. Examples include the website checking inventory, a price lookup, or processing a credit card authorization. The Oracle Service Bus (OSB) typically handles the routing of such messages, and we’ve enhanced its abilities with the Retail Service Backbone (RSB). To better understand these integration patterns and where they apply within the retail enterprise, we’re providing the Retail Reference Library (RRL) at no charge to Oracle Retail customers. The library is composed of a large number of industry business processes, including those necessary to support Commerce Anywhere, as well as detailed architectural diagrams. These diagrams allow implementers to understand the systems involved in integrations and the specific data payloads. Furthermore, with our upcoming release we’ll be providing a new tool called the Retail Integration Console (RIC) that allows IT to monitor and manage integrations from a single point. Using RIC, retailers can quickly discern where integration activity is occurring, volume statistics, average response times, and errors. The dashboards provide the ability to dive down into the architecture documentation to gather information all the way down to the specific payload. Retailers that want real-time integrations will also need real-time monitoring of those integrations to ensure service-level agreements are maintained. Part 3 looks at marketing.

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  • I want to record a screencast of a processing sketch

    - by nathanvda
    I have a created a music visualisation using Processing. I now want to convert that to a video, and the least obtrusive way I could think of is to record a screencast. I figured exporting Processing to video including audio, from within Processing itself, on ubuntu seemed an unsolved issue. Very hard and also could cause timing sync issues (since the music keeps running while images are captured). So move on to the screencast method. Dead-easy, I figured. But I was wrong. First hurdle was to find a way to record the sound from the audio (and not the mic). I did find a tutorial for that here. In short: use gtk-recordmydesktop and pulse audio. But, apparently, what happens: Processing does not use ALSA. When the sound is playing, it does not appear in the Pulse Audio mixer. How can I record the audio now?

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  • Internal mic not working on Aspire One D255

    - by user50652
    I can't get my internal mic to work with any app other than Sound Recorder. I've scoured the other posts on this topic and none seem to have an answer. I've installed Alsa Mixer, Pulse Audio controls, tried minimising one of the stereo channels but nothing works. The mic is not muted. There is no function key on the keyboard to mute the mic either. The only connector options are analogue mic or analgue input. Neither do anything.

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  • No analog audio in 13.10

    - by danepowell
    I've installed 13.10 on a machine that was previously running Windows 8, and audio output isn't working out of the box. It worked fine in Windows, and the speakers work fine when hooked up to a laptop, so it must an incompatibility between Ubuntu and the motherboard. If I go to sound settings, "Built-in Audio" is selected (SPDIF is also available). This is an Intel Z77 motherboard with an integrated Creative CA0132 sound chipset. I've tried booting a live image of 13.10 (to check for a corrupt install), and the same problem exists. If I boot a live image of 13.04, the only audio output listed in sound settings is a "dummy" card. I've already tried basic troubleshooting steps, such as removing pulseaudio config directories, force-restarting alsa, and making sure the speakers aren't muted in the alsa mixer. At this point, I'm totally stumped :(

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  • Problem with Arcam rpac and Ubuntu 12.04

    - by user108393
    I have recently (?mistakenly) purchased an Arcam rpac USB DAC for my Ubuntu (12.04) PC. I have torn out all but 3 of my hairs trying to get this to work and have had absolutely no luck so far. I can see the Arcam USB audio device when i run aplay -l, however I cannot see it listed under the Sound settings. I can see the soundcard device when running alsamixer also, but if I try and select it, alsamixer crashes, stating "cannot load mixer controls: Invalid argument". Any ideas how to get this working (if it's even possible)? Does anyone else out there have the rpac with ubuntu? Thank you! Aman

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  • Setting Up Audio on a Server Install

    - by tdcrenshaw
    I'm running on a clean install of 10.10 Server edition and have alsa-base, alsa-tools, alsa-utils, alsaplayer, and alsa-firmware-loader installed. At one point I installed pulseaudio, but I have since removed it. I've tried the following lspci | grep audio 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) 01:06.0 Multimedia audio controller: Creative Labs [SB Live! Value] EMU10k1X aplay -l aplay: device_list:235: no soundcards found... alsamixer can not open mixer: No such file or directory When I search for modules with find /lib/modules/`uname -r` | grep snd I do get a list of modules I'm not very experienced with alsa setup, so I'm not sure where to go from here

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  • How do I output my audio input?

    - by VarLogRant
    A few iterations ago, I think this was Jaunty but could've been before, I would plug a 1/8" audio cable from the line-out of a Windows netbook to the line-in of my Ubuntu machine, so I would have all the sound from both machines without having to plug both into a mixer which I don't have. I didn't do this much, as I was pretty-much happy with Banshee at the time. But with Karmic, and still with Lucid, I can only get the output if I'm recording with Audacity. Which I'm not going to do from my web-development and systems programming workstation. I can tell by plugging in headphones that my netbook has audio out working. I can see Sound Preferences that the Ubuntu machine is receiving them. I just want the old behavior back. Help?

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  • Mic not working when vga connector removed

    - by yygyt
    I have a computer that should run continuously without any connection to a monitor. For developmental purposes I have been keeping the vga connection with the monitor and experienced no problem until now. When I start the machine removing the vga connection beforehand, external microphone does not work. At first I didn't know anywhere to look and see the problem, but after a google search I saw that there is a command as alsamixer I ssh the machine end type alsamixer when it is connected to the monitor, here is the result If I remove monitor connection and reboot again, and then type alsamixer, I see the error, $ alsamixer cannot open mixer: No such file or directory I suspect that this error is related to X somehow. I really don't know anything about what goes beyond. This machine needs to work without any connection to a monitor. I would deeply appreciate any suggestions.

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  • Kubuntu permanent install won't attach sound card

    - by rob
    I installed Kubuntu Live version and sound card visible in the volume/mixer panel. After installing the permanent version I have a dummy output where my sound card should be, therefore no sound. I re-installed the live version three times and then to permanent but the same problem each time. There is a very brief error on boot up to the permanent version that says: error. cannot attach card default. no such file or directory. I'm running Kubuntu 12.04 on a Mac G4 PPC with no other OS. I'm very new to Linux. I am able to follow instructions to help resolve this, but I'm not familiar with the OS.

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  • MIX 2010 - in 10 seconds...

    um...windows phone 7Silverlight 4better uxlots of parties (MIXer was awesome)OData (ado v35.5?)number of problems with conference direction, still coolest of the public ms conferencesdid I mention the iphone killer windows phone 7 (aka, Zune Phone, windows mobile 7) and amazingly enough it might actually be some real competition for iphone, ux is awesome (after 4+ freaking years it had better fraking be freaking awesome)You can get all the videos here http://live.visitmix.com/VideosAND Wirestone...Did you know that DotNetSlackers also publishes .net articles written by top known .net Authors? We already have over 80 articles in several categories including Silverlight. Take a look: here.

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  • Getting Serial Number of the Hard Drive Provided by the manufacturer through PHP

    - by dta
    Getting Serial Number of the Hard Drive Provided by the manufacturer through PHP : How can it be done? I want to store it in a file. OS : windows 2000,XP,ME,Vista... Yes, I want the serial number of the hard drive of the Server. Or can it be done through Adobe AIR? Or can it be done through a C program on Windows? C:\Documents and Settings\Administrator>dir Volume in drive C has no label. Volume Serial Number is BC16-5D5F Is this number : BC16-5d5f unique for a hard drive? How is it different from the manufacturer given serial number? This command **wmic DISKDRIVE GET SerialNumber** Displays only the following text on my Vista Machine : SerialNumber On my XP machine, the command is unrecognized

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  • Norton Ghost EBAB03F1: The specified network name is no longer available.

    - by Breck Carter
    After about 15 minutes, a Norton Ghost 14 backup fails with Error EBAB03F1: The specified network name is no longer available. The source computer is a P4 laptop running Windows XP SP3. The target computer is a Core2 Quad desktop running Windows Vista Ultimate 64bit. It does not help to disable Norton 360 on the source computer or Norton Antivirus 2008 on the target computer. The Event Viewer consistently shows the same two VSS-related errors after Norton Ghost starts but before it fails. It makes no difference if the VSS service is started or stopped. The VSS errors do not appear elsewhere in the event log, only after Ghost starts. The MSS event messages, however, are quite common, appearing throughout the log, and they may have nothing to do with the problem. Here is the Norton Ghost error display... -Errors exist. --Unable to write to file. ---Error EBAB03F1: The specified network name is no longer available. ---Unable to set file size. ----Error EBAB03F1: The specified network name is no longer available. ----Unable to write to file. -----Error EBAB03F1: The specified network name is no longer available. -----Unable to set file size. ------Error EBAB03F1: The specified network name is no longer available. Here are the source computer events, with the final error at the top and the "Ghost Starting" message at the bottom: ===== Event Type: Error Event Source: Norton Ghost Event Category: High Priority Event ID: 100 Date: 11/09/2009 Time: 9:40:26 AM User: N/A Computer: PAVILION2 Description: Error EC8F17B7: Cannot create recovery points for job: Drive Backup of (C:\) (3). Error E7D1001F: Unable to write to file. Error EBAB03F1: The specified network name is no longer available. Error E7D10046: Unable to set file size. Error EBAB03F1: The specified network name is no longer available. Error E7D1001F: Unable to write to file. Error EBAB03F1: The specified network name is no longer available. Error E7D10046: Unable to set file size. Error EBAB03F1: The specified network name is no longer available. Details: 0xEBAB0005 Source: Norton Ghost ===== Event Type: Information Event Source: MSSQL$SQLEXPRESS Event Category: Server Event ID: 3421 Date: 11/09/2009 Time: 9:34:06 AM User: NT AUTHORITY\NETWORK SERVICE Computer: PAVILION2 Description: Recovery completed for database ReportServer$SQLEXPRESSTempDB (database ID 6) in 1 second(s) (analysis 205 ms, redo 0 ms, undo 376 ms.) This is an informational message only. No user action is required. For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. Data: 0000: 5d 0d 00 00 0a 00 00 00 ]....... 0008: 15 00 00 00 50 00 41 00 ....P.A. 0010: 56 00 49 00 4c 00 49 00 V.I.L.I. 0018: 4f 00 4e 00 32 00 5c 00 O.N.2.\. 0020: 53 00 51 00 4c 00 45 00 S.Q.L.E. 0028: 58 00 50 00 52 00 45 00 X.P.R.E. 0030: 53 00 53 00 00 00 18 00 S.S..... 0038: 00 00 52 00 65 00 70 00 ..R.e.p. 0040: 6f 00 72 00 74 00 53 00 o.r.t.S. 0048: 65 00 72 00 76 00 65 00 e.r.v.e. 0050: 72 00 24 00 53 00 51 00 r.$.S.Q. 0058: 4c 00 45 00 58 00 50 00 L.E.X.P. 0060: 52 00 45 00 53 00 53 00 R.E.S.S. 0068: 00 00 .. ===== Event Type: Information Event Source: MSSQL$SQLEXPRESS Event Category: Server Event ID: 17137 Date: 11/09/2009 Time: 9:34:02 AM User: NT AUTHORITY\NETWORK SERVICE Computer: PAVILION2 Description: Starting up database 'ReportServer$SQLEXPRESSTempDB'. For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. Data: 0000: f1 42 00 00 0a 00 00 00 ñB...... 0008: 15 00 00 00 50 00 41 00 ....P.A. 0010: 56 00 49 00 4c 00 49 00 V.I.L.I. 0018: 4f 00 4e 00 32 00 5c 00 O.N.2.\. 0020: 53 00 51 00 4c 00 45 00 S.Q.L.E. 0028: 58 00 50 00 52 00 45 00 X.P.R.E. 0030: 53 00 53 00 00 00 18 00 S.S..... 0038: 00 00 52 00 65 00 70 00 ..R.e.p. 0040: 6f 00 72 00 74 00 53 00 o.r.t.S. 0048: 65 00 72 00 76 00 65 00 e.r.v.e. 0050: 72 00 24 00 53 00 51 00 r.$.S.Q. 0058: 4c 00 45 00 58 00 50 00 L.E.X.P. 0060: 52 00 45 00 53 00 53 00 R.E.S.S. 0068: 00 00 .. ===== Event Type: Error Event Source: VSS Event Category: None Event ID: 5013 Date: 11/09/2009 Time: 9:28:32 AM User: N/A Computer: PAVILION2 Description: Volume Shadow Copy Service error: Shadow Copy writer ContentIndexingService called routine RegQueryValueExW which failed with status 0x80070002 (converted to 0x800423f4). For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. Data: 0000: 57 53 48 43 4f 4d 4e 43 WSHCOMNC 0008: 32 32 39 32 00 00 00 00 2292.... 0010: 57 53 48 43 49 43 00 00 WSHCIC.. 0018: 32 38 37 00 00 00 00 00 287..... ===== Event Type: Error Event Source: VSS Event Category: None Event ID: 5013 Date: 11/09/2009 Time: 9:28:32 AM User: N/A Computer: PAVILION2 Description: Volume Shadow Copy Service error: Shadow Copy writer ContentIndexingService called routine RegQueryValueExW which failed with status 0x80070002 (converted to 0x800423f4). For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. Data: 0000: 57 53 48 43 4f 4d 4e 43 WSHCOMNC 0008: 32 32 39 32 00 00 00 00 2292.... 0010: 57 53 48 43 49 43 00 00 WSHCIC.. 0018: 32 38 37 00 00 00 00 00 287..... ===== Event Type: Error Event Source: VSS Event Category: None Event ID: 12302 Date: 11/09/2009 Time: 9:28:32 AM User: N/A Computer: PAVILION2 Description: Volume Shadow Copy Service error: An internal inconsistency was detected in trying to contact shadow copy service writers. Please check to see that the Event Service and Volume Shadow Copy Service are operating properly. For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. Data: 0000: 42 55 45 43 58 4d 4c 43 BUECXMLC 0008: 33 36 33 37 00 00 00 00 3637.... 0010: 42 55 45 43 58 4d 4c 43 BUECXMLC 0018: 33 36 30 37 00 00 00 00 3607.... ===== Event Type: Information Event Source: Norton Ghost Event Category: High Priority Event ID: 100 Date: 11/09/2009 Time: 9:27:57 AM User: N/A Computer: PAVILION2 Description: Info 6C8F1F63: The drive-based backup job, Drive Backup of (C:\) (3), has been started manually. Details: Source: Norton Ghost

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  • Progressbar for mediaplayer using jquery

    - by Geetha
    In my asp.net application i am using mediaplayer to play the audio and video. i am controling volume using javascript code. I want to display a userdefined progress bar. How to create control it. Code: <object id="mediaPlayer" classid="clsid:22D6F312-B0F6-11D0-94AB-0080C74C7E95" codebase="http://activex.microsoft.com/activex/controls/mplayer/en/nsmp2inf.cab#Version=5,1,52,701" height="1" standby="Loading Microsoft Windows Media Player components..." type="application/x-oleobject" width="1"> <param name="fileName" value="" /> <param name="animationatStart" value="true" /> <param name="transparentatStart" value="true" /> <param name="autoStart" value="true" /> <param name="showControls" value="true" /> <param name="volume" value="100" /> <param name="loop" value="true" /> </object>

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