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  • vga_switcheroo and Intel HD 3000 on Ubuntu 12.04

    - by Ikalou
    I'm trying to get vga_switcheroo to enable my integrated Intel HD 3000 instead of my ATI card. My problem is that there is no vgaswitcheroo directory in /sys/kernel/debug/ on my system. > grep -i switcheroo /boot/config-3.2.0-26-generic CONFIG_VGA_SWITCHEROO=y And yet: > sudo ls /sys/kernel/debug/ acpi bdi bluetooth dri extfrag gpio ieee80211 kprobes mce mmc0 regmap regulator sched_features suspend_stats tracing usb wakeup_sources x86 I am NOT using the fglrx driver. Here is the output of lspci; glxinfo | grep renderer: 00:00.0 Host bridge: Intel Corporation 2nd Generation Core Processor Family DRAM Controller (rev 09) 00:01.0 PCI bridge: Intel Corporation Xeon E3-1200/2nd Generation Core Processor Family PCI Express Root Port (rev 09) 00:16.0 Communication controller: Intel Corporation 6 Series/C200 Series Chipset Family MEI Controller #1 (rev 04) 00:19.0 Ethernet controller: Intel Corporation 82579LM Gigabit Network Connection (rev 04) 00:1a.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #2 (rev 04) 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 1 (rev b4) 00:1c.1 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 2 (rev b4) 00:1c.2 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 3 (rev b4) 00:1c.3 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 4 (rev b4) 00:1c.7 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 8 (rev b4) 00:1d.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation QM67 Express Chipset Family LPC Controller (rev 04) 00:1f.2 SATA controller: Intel Corporation 6 Series/C200 Series Chipset Family 6 port SATA AHCI Controller (rev 04) 01:00.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI Seymour [Radeon HD 6400M Series] 01:00.1 Audio device: Advanced Micro Devices [AMD] nee ATI Caicos HDMI Audio [Radeon HD 6400 Series] 24:00.0 FireWire (IEEE 1394): JMicron Technology Corp. IEEE 1394 Host Controller (rev 30) 24:00.1 System peripheral: JMicron Technology Corp. SD/MMC Host Controller (rev 30) 24:00.2 SD Host controller: JMicron Technology Corp. Standard SD Host Controller (rev 30) 25:00.0 Network controller: Intel Corporation Centrino Advanced-N 6205 (rev 34) 26:00.0 USB controller: NEC Corporation uPD720200 USB 3.0 Host Controller (rev 04) OpenGL renderer string: Gallium 0.4 on AMD CAICOS Both xserver-xorg-video-intel and xserver-xorg-video-radeon packages are installed. I know there are tons of posts about hybrid-graphics already but I couldn't quite find a solution to my problem. Does anyone know why is /sys/kernel/debug/vgaswitcheroo not showing?

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  • How do you edit the "Preferred Format" settings in Rhythmbox?

    - by skyblue
    In Rhythmbox's Preferences, you can change the "Preferred Format" for Music to MPEG Layer 3 Audio, Ogg Vorbis, FLAC, or MPEG 4 Audio. However, despite there being a Settings button, it does not become enabled for any of these choices. (I have installed all of the gstreamer plugins, but this has made no difference.) So how can you change the "Preferred Format", for example to change the bit rate or the quality setting?

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  • Series On Embedded Development (Part 2) - Build-Time Optionality

    - by user12612705
    In this entry on embedded development, I'm going to discuss build-time optionality (BTO). BTO is the ability to subset your software at build-time so you only use what is needed. BTO typically pertains more to software providers rather then developers of final products. For example, software providers ship source products, frameworks or platforms which are used by developers to build other products. If you provide a source product, you probably don't have to do anything to support BTO as the developers using your source will only use the source they need to build their product. If you provide a framework, then there are some things you can do to support BTO. Say you provide a Java framework which supports audio and video. If you provide this framework in a single JAR, then developers who only want audio are forced to ship their product with the video portion of your framework even though they aren't using it. In this case, support providing the framework in separate JARs...break the framework into an audio JAR and a video JAR and let the users of your framework decide which JARs to include in their product. Sometimes this is as simple as packaging, but if, for example, the video functionality is dependent on the audio functionality, it may require coding work to cleanly separate the two. BTO can also work at install-time, and this is sometimes overlooked. Let's say your building a phone application which can use Near Field Communications (NFC) if it's available on the phone, but it doesn't require NFC to work. Typically you'd write one app for all phones (saving you time)...both those that have NFC and those that don't, and just use NFC if it's there. However, for better efficiency, you can detect at install-time if the phone supports NFC and not install the NFC portion of your app if the phone doesn't support NFC. This requires that you write the app so it can run without the optional NFC code and that you write your install app so it can detect NFC and do the right thing at install-time. Supporting install-time optionality will save persistent footprint on the phone, something your customers will appreciate, your app "neighbors" will appreciate, and that you'll appreciate when they save static footprint for you. In the next article, I'll talk about runtime optionality.

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  • Best way to 'harden' embedded ext4 file server against unexpected loss of power?

    - by Jeremy Friesner
    Hi all, First, a little background: my company makes an audio streaming device that is a headless, rack-mounted Linux box with a couple of SSDs attached. Each SSD is formatted with ext4. The users can connect to the system using Samba/CIFS to upload new audio files or access existing ones. There is also custom software for streaming out audio over the network. This is all fine. The only problem is that the users are audio people, not computer people, and see the system as a 'black box', not as a computer. Which means that at the end of the day, they aren't going to ssh in to the box and enter "/sbin/shutdown -h"; they are just going to cut power to the rack and leave, and expect things to still work properly the next day. Since ext4 has journalling, journal checksumming, etc, this mostly works. The only time it doesn't work is when someone uploads a new file via Samba and then cuts power to the system before the uploaded data has been fully flushed to the disk. In that case, they come in the next day and find that their new file has been truncated or is missing entirely, and are unhappy. My question is, what is the best way to avoid this problem? Is there a way to get smbd to call "sync" at the end of every upload? (Performance on uploads isn't so important, since they only happen occasionally). Or is there a way to tell ext4 to automatically flush within a few seconds of any change to a file? (Again, performance can be sacrificed for safety here) Should I set a particular write-ordering mode, activate barriers, etc?

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  • Getting old bluetooth headset out of standby on Windows 7

    - by luagh45
    I think this is the problem, and if not, I'm open to suggestions. I have an old Jabra BT200 that I used to use on my phone. When a phone call was coming in it would beep using its own noises (meaning the phone never rang inside the headset) and then I could push the 'answer/hang up' button and sound and mic would start working. I have now paired it with Windows 7, and it looks good. Under the playback menu I have 'Bluetooth Hands free Audio / Jabra BT200 (Mono Audio) / Ready', and under the recording menu I have 'Bluetooth Audio Input Device / Jabra BT200 (Mono Audio) / Ready'. However when I try to test the speakers Windows sends a sound, but I never hear it, and when I talk in the mic, Windows never hears me. If I right click either the Bluetooth mic or speakers there's an option to 'Connect', but it's grayed out and I cannot click it. As the final piece of knowledge I have, my headset blinks once every 3 seconds when it's in standby and I can't get that to change. If everything was working it should blink once every second at which point I think all of my problems would be fixed. Hence my issue: I can't seem to get my headset out of standby. On my headset I've tried sending it test noises and then pressing the 'Answer' button, but still nothing. The headset beeps when I press it, so it works, it just doesn't ever come out of standby. Is there maybe some way to trick my headset into thinking it's getting a phone call from my computer?

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  • DV-AVI in DivX Plus Player and VirtualDub - playback issue

    - by user714965
    I have an AVI-video which I had (digitally) transferred from a DV camera to my PC. The video contains errors at the beginning most likely because the DV tape was pretty old. When playing this video in the DivX Plus Player I get some picture artefacts and some high noise peaks of the sound. These stops after two seconds. When I'm playing this video in VirtualDub (where I want to cut it) I get the same picture artefacts. But the sound errors (those loud high peaks) lasts ten seconds. These sound errors are also contained in the cutted video. Why does the video have more errors when played in VirtualDub? I think because of different codecs which are used for decoding the video? How can I change the codec which VirtualDub uses for decoding? I have installed ffdshow for this but it seems that it is not used because I don't get the ffdshow icon in the taskbar when playing the video in VirtualDub. When playing in DivX Player Plus I get this icon.

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  • AVConv increases song duration when converting MP3

    - by chauffch
    I am struggling with the following issue. I want to convert an MP3 ADTS into pure a MP3. I am using AVConv on Ubuntu 12.10. The outcome is a file that has the same size, but the duration is now longer. $ ls -l total 6436 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mpga Blindsided_Bon_Iver.mpga: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo $ avconv -i Blindsided_Bon_Iver.mpga -c copy Blindsided_Bon_Iver.mp3 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:50:25 with gcc 4.6.3 [mp3 @ 0x8c6e240] max_analyze_duration reached Input #0, mp3, from 'Blindsided_Bon_Iver.mpga': Duration: 00:05:29.29, start: 0.000000, bitrate: 160 kb/s Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 160 kb/s Output #0, mp3, to 'Blindsided_Bon_Iver.mp3': Metadata: TSSE : Lavf53.21.0 Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, 160 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press ctrl-c to stop encoding size= 6432kB time=329.30 bitrate= 160.0kbits/s video:0kB audio:6432kB global headers:0kB muxing overhead 0.002080% $ ls -l total 12868 -rw-rw-r-- 1 teuf teuf 6586129 nov. 27 22:26 Blindsided_Bon_Iver.mp3 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mp3 Blindsided_Bon_Iver.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 32 kbps, 44.1 kHz, Stereo Amarok shows the new file has a duration of 25:27 and has a lot of silence. Am I using an incorrect option? Is it a bug in AVConv? Any ideas how to fix it?

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  • How to embed/hardcode SRT subtitles into mp4 videos with VLC?

    - by Jens Bannmann
    I'm looking for a way to "burn in" or render/rembed/hardcode subtitles (from an SRT file) into an MP4 video with VLC. But no matter what options I use, it never works properly. I get a file that plays video way too fast (audio is normal), or one that plays normally, but actually does not have embedded subtitles. Also, with some options (like the one below) it does not play in QuickTime, only in VLC. So the main question is: how can I make this work in VLC? Secondary questions are: How do I decide which options I should set? Which settings are best if I want to leave the file bitrate etc. the same as much as possible, only embed subtitles? It seems I cannot leave the field empty or Video/Audio unchecked, so I guess I would first need to figure out the original audio and video bitrate. What do the "Scale" and "Channels" options mean? ... none of which are answered within the VLC documentation. For example, this is one set of options I used in the "Advanced Open File…" dialog: Advanced Open File… myFileName.mp4 [ ] Treat as a pipe rather than as a file [x] Load subtitles file: mySubtitleFileName.srt [ ] Play another media synchronously [x] Streaming/Saving Streaming and Transcoding Options [ ] Display the stream locally (o) File [outputFileName.mp4 ] [ ] Dump raw input Encapsulation Method: (MPEG 4 ) Transcoding options [x] Video (mp4v ) Bitrate (kb/s) [256 ] Scale [1 ] [x] Audio (mp3 ) Bitrate (kb/s) [128 ] Channels [1 ]

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  • Getting old bluetooth headset out of standby on Windows 7

    - by luagh45
    I think this is the problem, and if not, I'm open to suggestions. I have an old Jabra BT200 that I used to use on my phone. When a phone call was coming in it would beep using its own noises (meaning the phone never rang inside the headset) and then I could push the 'answer/hang up' button and sound and mic would start working. I have now paired it with Windows 7, and it looks good. Under the playback menu I have 'Bluetooth Hands free Audio / Jabra BT200 (Mono Audio) / Ready', and under the recording menu I have 'Bluetooth Audio Input Device / Jabra BT200 (Mono Audio) / Ready'. However when I try to test the speakers Windows sends a sound, but I never hear it, and when I talk in the mic, Windows never hears me. If I right click either the Bluetooth mic or speakers there's an option to 'Connect', but it's grayed out and I cannot click it. As the final piece of knowledge I have, my headset blinks once every 3 seconds when it's in standby and I can't get that to change. If everything was working it should blink once every second at which point I think all of my problems would be fixed. Hence my issue: I can't seem to get my headset out of standby. On my headset I've tried sending it test noises and then pressing the 'Answer' button, but still nothing. The headset beeps when I press it, so it works, it just doesn't ever come out of standby. Is there maybe some way to trick my headset into thinking it's getting a phone call from my computer?

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  • Running computer from separate room

    - by Dan
    I want my computer to be in the basement, but to use it on the first floor. Which cables should I run through the floor? Can some be wireless or other methods? Here are some of the options I've thought of: Basic: Run DVI, usb (mouse), usb (keyboard), and audio cable (4 cables) USB Hub option: Run DVI, and 1 usb, then using a usb hub split it into mouse, keyboard, maybe even audio (2-3 cables) HDMI Option: If I get a new video card and monitor that supports HDMI, would I be able to run both audio and video through it? Would the monitor have to have an audio out? Also there is a lot of extra bandwidth in the HDMI cables, could I send two monitors on 1 cable or would I have to use 2 cables? How about sending mouse/keyboard through the HDMI cable? I see a lot of monitors with USB hubs built in, but I assume I'd still have to wire HDMI + 1 USB cable to use the USB hubs? X Terminal Machine/Thin Client: I don't really know much about this option. Not sure if it would allow me to run graphics acceleration and watch movies, does anyone know more details about what this would allow me to do? Other options: Any other ways to do this? Can any of this be wireless?

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  • Lync Server 2010

    - by ManojDhobale
    Microsoft Lync Server 2010 communications software and its client software, such as Microsoft Lync 2010, enable your users to connect in new ways and to stay connected, regardless of their physical location. Lync 2010 and Lync Server 2010 bring together the different ways that people communicate in a single client interface, are deployed as a unified platform, and are administered through a single management infrastructure. Workload Description IM and presence Instant messaging (IM) and presence help your users find and communicate with one another efficiently and effectively. IM provides an instant messaging platform with conversation history, and supports public IM connectivity with users of public IM networks such as MSN/Windows Live, Yahoo!, and AOL. Presence establishes and displays a user’s personal availability and willingness to communicate through the use of common states such as Available or Busy. This rich presence information enables other users to immediately make effective communication choices. Conferencing Lync Server includes support for IM conferencing, audio conferencing, web conferencing, video conferencing, and application sharing, for both scheduled and impromptu meetings. All these meeting types are supported with a single client. Lync Server also supports dial-in conferencing so that users of public switched telephone network (PSTN) phones can participate in the audio portion of conferences. Conferences can seamlessly change and grow in real time. For example, a single conference can start as just instant messages between a few users, and escalate to an audio conference with desktop sharing and a larger audience instantly, easily, and without interrupting the conversation flow. Enterprise Voice Enterprise Voice is the Voice over Internet Protocol (VoIP) offering in Lync Server 2010. It delivers a voice option to enhance or replace traditional private branch exchange (PBX) systems. In addition to the complete telephony capabilities of an IP PBX, Enterprise Voice is integrated with rich presence, IM, collaboration, and meetings. Features such as call answer, hold, resume, transfer, forward and divert are supported directly, while personalized speed dialing keys are replaced by Contacts lists, and automatic intercom is replaced with IM. Enterprise Voice supports high availability through call admission control (CAC), branch office survivability, and extended options for data resiliency. Support for remote users You can provide full Lync Server functionality for users who are currently outside your organization’s firewalls by deploying servers called Edge Servers to provide a connection for these remote users. These remote users can connect to conferences by using a personal computer with Lync 2010 installed, the phone, or a web interface. Deploying Edge Servers also enables you to federate with partner or vendor organizations. A federated relationship enables your users to put federated users on their Contacts lists, exchange presence information and instant messages with these users, and invite them to audio calls, video calls, and conferences. Integration with other products Lync Server integrates with several other products to provide additional benefits to your users and administrators. Meeting tools are integrated into Outlook 2010 to enable organizers to schedule a meeting or start an impromptu conference with a single click and make it just as easy for attendees to join. Presence information is integrated into Outlook 2010 and SharePoint 2010. Exchange Unified Messaging (UM) provides several integration features. Users can see if they have new voice mail within Lync 2010. They can click a play button in the Outlook message to hear the audio voice mail, or view a transcription of the voice mail in the notification message. Simple deployment To help you plan and deploy your servers and clients, Lync Server provides the Microsoft Lync Server 2010, Planning Tool and the Topology Builder. Lync Server 2010, Planning Tool is a wizard that interactively asks you a series of questions about your organization, the Lync Server features you want to enable, and your capacity planning needs. Then, it creates a recommended deployment topology based on your answers, and produces several forms of output to aid your planning and installation. Topology Builder is an installation component of Lync Server 2010. You use Topology Builder to create, adjust and publish your planned topology. It also validates your topology before you begin server installations. When you install Lync Server on individual servers, the installation program deploys the server as directed in the topology. Simple management After you deploy Lync Server, it offers the following powerful and streamlined management tools: Active Directory for its user information, which eliminates the need for separate user and policy databases. Microsoft Lync Server 2010 Control Panel, a new web-based graphical user interface for administrators. With this web-based UI, Lync Server administrators can manage their systems from anywhere on the corporate network, without needing specialized management software installed on their computers. Lync Server Management Shell command-line management tool, which is based on the Windows PowerShell command-line interface. It provides a rich command set for administration of all aspects of the product, and enables Lync Server administrators to automate repetitive tasks using a familiar tool. While the IM and presence features are automatically installed in every Lync Server deployment, you can choose whether to deploy conferencing, Enterprise Voice, and remote user access, to tailor your deployment to your organization’s needs.

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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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  • iphone: Help with AudioToolbox Leak: Stack trace/code included here...

    - by editor guy
    Part of this app is a "Scream" button that plays random screams from cast members of a TV show. I have to bang on the app quite a while to see a memory leak in Instruments, but it's there, occasionally coming up (every 45 seconds to 2 minutes.) The leak is 3.50kb when it occurs. Haven't been able to crack it for several hours. Any help appreciated. Instruments says this is the offending code line: [appSoundPlayer play]; that's linked to from line 9 of the below stack trace: 0 libSystem.B.dylib malloc 1 libSystem.B.dylib pthread_create 2 AudioToolbox CAPThread::Start() 3 AudioToolbox GenericRunLoopThread::Start() 4 AudioToolbox AudioQueueNew(bool, AudioStreamBasicDescription const*, TCACallback const&, CACallbackTarget const&, unsigned long, OpaqueAudioQueue*) 5 AudioToolbox AudioQueueNewOutput 6 AVFoundation allocAudioQueue(AVAudioPlayer, AudioPlayerImpl*) 7 AVFoundation prepareToPlayQueue(AVAudioPlayer*, AudioPlayerImpl*) 8 AVFoundation -[AVAudioPlayer prepareToPlay] 9 Scream Queens -[ScreamViewController scream:] /Users/laptop2/Desktop/ScreamQueens Versions/ScreamQueens25/Scream Queens/Classes/../ScreamViewController.m:210 10 CoreFoundation -[NSObject performSelector:withObject:withObject:] 11 UIKit -[UIApplication sendAction:to:from:forEvent:] 12 UIKit -[UIApplication sendAction:toTarget:fromSender:forEvent:] 13 UIKit -[UIControl sendAction:to:forEvent:] 14 UIKit -[UIControl(Internal) _sendActionsForEvents:withEvent:] 15 UIKit -[UIControl touchesEnded:withEvent:] 16 UIKit -[UIWindow _sendTouchesForEvent:] 17 UIKit -[UIWindow sendEvent:] 18 UIKit -[UIApplication sendEvent:] 19 UIKit _UIApplicationHandleEvent 20 GraphicsServices PurpleEventCallback 21 CoreFoundation CFRunLoopRunSpecific 22 CoreFoundation CFRunLoopRunInMode 23 GraphicsServices GSEventRunModal 24 UIKit -[UIApplication _run] 25 UIKit UIApplicationMain 26 Scream Queens main /Users/laptop2/Desktop/ScreamQueens Versions/ScreamQueens25/Scream Queens/main.m:14 27 Scream Queens start Here's .h: #import <UIKit/UIKit.h> #import <AVFoundation/AVFoundation.h> #import <MediaPlayer/MediaPlayer.h> #import <AudioToolbox/AudioToolbox.h> #import <MessageUI/MessageUI.h> #import <MessageUI/MFMailComposeViewController.h> @interface ScreamViewController : UIViewController <UIApplicationDelegate, AVAudioPlayerDelegate, MFMailComposeViewControllerDelegate> { //AudioPlayer related AVAudioPlayer *appSoundPlayer; NSURL *soundFileURL; BOOL interruptedOnPlayback; BOOL playing; //Scream button related IBOutlet UIButton *screamButton; int currentScreamIndex; NSString *currentScream; NSMutableArray *screams; NSMutableArray *personScreaming; NSMutableArray *photoArray; int currentSayingsIndex; NSString *currentButtonSaying; NSMutableArray *funnyButtonSayings; IBOutlet UILabel *funnyButtonSayingsLabel; IBOutlet UILabel *personScreamingField; IBOutlet UIImageView *personScreamingImage; //Mailing the scream related IBOutlet UILabel *mailStatusMessage; IBOutlet UIButton *shareButton; } //AudioPlayer related @property (nonatomic, retain) AVAudioPlayer *appSoundPlayer; @property (nonatomic, retain) NSURL *soundFileURL; @property (readwrite) BOOL interruptedOnPlayback; @property (readwrite) BOOL playing; //Scream button related @property (nonatomic, retain) UIButton *screamButton; @property (nonatomic, retain) NSMutableArray *screams; @property (nonatomic, retain) NSMutableArray *personScreaming; @property (nonatomic, retain) NSMutableArray *photoArray; @property (nonatomic, retain) UILabel *personScreamingField; @property (nonatomic, retain) UIImageView *personScreamingImage; @property (nonatomic, retain) NSMutableArray *funnyButtonSayings; @property (nonatomic, retain) UILabel *funnyButtonSayingsLabel; //Mailing the scream related @property (nonatomic, retain) IBOutlet UILabel *mailStatusMessage; @property (nonatomic, retain) IBOutlet UIButton *shareButton; //Scream Button - (IBAction) scream: (id) sender; //Mail the scream - (IBAction) showPicker: (id)sender; - (void)displayComposerSheet; - (void)launchMailAppOnDevice; @end Here's the top of .m: #import "ScreamViewController.h" //top of code has Audio session callback function for responding to audio route changes (from Apple's code), then my code continues... @implementation ScreamViewController @synthesize appSoundPlayer; // AVAudioPlayer object for playing the selected scream @synthesize soundFileURL; // Path to the scream @synthesize interruptedOnPlayback; // Was application interrupted during audio playback @synthesize playing; // Track playing/not playing state @synthesize screamButton; //Press this button, girls scream. @synthesize screams; //Mutable array holding strings pointing to sound files of screams. @synthesize personScreaming; //Mutable array tracking the person doing the screaming @synthesize photoArray; //Mutable array holding strings pointing to photos of screaming girls @synthesize personScreamingField; //Field updates to announce which girl is screaming. @synthesize personScreamingImage; //Updates to show image of the screamer. @synthesize funnyButtonSayings; //Mutable array holding the sayings @synthesize funnyButtonSayingsLabel; //Label that updates with the funnyButtonSayings @synthesize mailStatusMessage; //did the email go out @synthesize shareButton; //share scream via email Next line begins the block with the offending code: - (IBAction) scream: (id) sender { //Play a click sound effect SystemSoundID soundID; NSString *sfxPath = [[NSBundle mainBundle] pathForResource:@"aClick" ofType:@"caf"]; AudioServicesCreateSystemSoundID((CFURLRef)[NSURL fileURLWithPath:sfxPath],&soundID); AudioServicesPlaySystemSound (soundID); // Because someone may slam the scream button over and over, //must stop current sound, then begin next if ([self appSoundPlayer] != nil) { [[self appSoundPlayer] setDelegate:nil]; [[self appSoundPlayer] stop]; [self setAppSoundPlayer: nil]; } //after selecting a random index in the array (did that in View Did Load), //we move to the next scream on each click. //First check... //Are we past the end of the array? if (currentScreamIndex == [screams count]) { currentScreamIndex = 0; } //Get the string at the index in the personScreaming array currentScream = [screams objectAtIndex: currentScreamIndex]; //Get the string at the index in the personScreaming array NSString *screamer = [personScreaming objectAtIndex:currentScreamIndex]; //Log the string to the console NSLog (@"playing scream: %@", screamer); // Display the string in the personScreamingField field NSString *listScreamer = [NSString stringWithFormat:@"scream by: %@", screamer]; [personScreamingField setText:listScreamer]; // Gets the file system path to the scream to play. NSString *soundFilePath = [[NSBundle mainBundle] pathForResource: currentScream ofType: @"caf"]; // Converts the sound's file path to an NSURL object NSURL *newURL = [[NSURL alloc] initFileURLWithPath: soundFilePath]; self.soundFileURL = newURL; [newURL release]; [[AVAudioSession sharedInstance] setDelegate: self]; [[AVAudioSession sharedInstance] setCategory: AVAudioSessionCategoryPlayback error: nil]; // Registers the audio route change listener callback function AudioSessionAddPropertyListener ( kAudioSessionProperty_AudioRouteChange, audioRouteChangeListenerCallback, self ); // Activates the audio session. NSError *activationError = nil; [[AVAudioSession sharedInstance] setActive: YES error: &activationError]; // Instantiates the AVAudioPlayer object, initializing it with the sound AVAudioPlayer *newPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL: soundFileURL error: nil]; //Error check and continue if (newPlayer != nil) { self.appSoundPlayer = newPlayer; [newPlayer release]; [appSoundPlayer prepareToPlay]; [appSoundPlayer setVolume: 1.0]; [appSoundPlayer setDelegate:self]; //NEXT LINE IS FLAGGED BY INSTRUMENTS AS LEAKY [appSoundPlayer play]; playing = YES; //Get the string at the index in the photoArray array NSString *screamerPic = [photoArray objectAtIndex:currentScreamIndex]; //Log the string to the console NSLog (@"displaying photo: %@", screamerPic); // Display the image of the person screaming personScreamingImage.image = [UIImage imageNamed:screamerPic]; //show the share button shareButton.hidden = NO; mailStatusMessage.hidden = NO; mailStatusMessage.text = @"share!"; //Get the string at the index in the funnySayings array currentSayingsIndex = random() % [funnyButtonSayings count]; currentButtonSaying = [funnyButtonSayings objectAtIndex: currentSayingsIndex]; NSString *theSaying = [funnyButtonSayings objectAtIndex:currentSayingsIndex]; [funnyButtonSayingsLabel setText: theSaying]; currentScreamIndex++; } } Here's my dealloc: - (void)dealloc { [appSoundPlayer stop]; [appSoundPlayer release], appSoundPlayer = nil; [screamButton release], screamButton = nil; [mailStatusMessage release], mailStatusMessage = nil; [personScreamingField release], personScreamingField = nil; [personScreamingImage release], personScreamingImage = nil; [funnyButtonSayings release], funnyButtonSayings = nil; [funnyButtonSayingsLabel release], funnyButtonSayingsLabel = nil; [screams release], screams = nil; [personScreaming release], personScreaming = nil; [soundFileURL release]; [super dealloc]; } @end Thanks so much for reading this far! Any input appreciated.

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  • Blackjack game reshuffling problem-edited

    - by Jam
    I am trying to make a blackjack game where before each new round, the program checks to make sure that the deck has 7 cards per player. And if it doesn't, the deck clears, repopulates, and reshuffles. I have most of the problem down, but for some reason at the start of every deal it reshuffles the deck more than once, and I can't figure out why. Help, please. Here's what I have so far: (P.S. the imported cards and games modules aren't part of the problem, I'm fairly sure my problem lies in the deal() function of my BJ_Deck class.) import cards, games class BJ_Card(cards.Card): """ A Blackjack Card. """ ACE_VALUE = 1 def get_value(self): if self.is_face_up: value = BJ_Card.RANKS.index(self.rank) + 1 if value > 10: value = 10 else: value = None return value value = property(get_value) class BJ_Deck(cards.Deck): """ A Blackjack Deck. """ def populate(self): for suit in BJ_Card.SUITS: for rank in BJ_Card.RANKS: self.cards.append(BJ_Card(rank, suit)) def deal(self, hands, per_hand=1): for rounds in range(per_hand): if len(self.cards)>=7*(len(hands)): print "Reshuffling the deck." self.cards=[] self.populate() self.shuffle() for hand in hands: top_card=self.cards[0] self.give(top_card, hand) class BJ_Hand(cards.Hand): """ A Blackjack Hand. """ def __init__(self, name): super(BJ_Hand, self).__init__() self.name = name def __str__(self): rep = self.name + ":\t" + super(BJ_Hand, self).__str__() if self.total: rep += "(" + str(self.total) + ")" return rep def get_total(self): # if a card in the hand has value of None, then total is None for card in self.cards: if not card.value: return None # add up card values, treat each Ace as 1 total = 0 for card in self.cards: total += card.value # determine if hand contains an Ace contains_ace = False for card in self.cards: if card.value == BJ_Card.ACE_VALUE: contains_ace = True # if hand contains Ace and total is low enough, treat Ace as 11 if contains_ace and total <= 11: # add only 10 since we've already added 1 for the Ace total += 10 return total total = property(get_total) def is_busted(self): return self.total > 21 class BJ_Player(BJ_Hand): """ A Blackjack Player. """ def is_hitting(self): response = games.ask_yes_no("\n" + self.name + ", do you want a hit? (Y/N): ") return response == "y" def bust(self): print self.name, "busts." self.lose() def lose(self): print self.name, "loses." def win(self): print self.name, "wins." def push(self): print self.name, "pushes." class BJ_Dealer(BJ_Hand): """ A Blackjack Dealer. """ def is_hitting(self): return self.total < 17 def bust(self): print self.name, "busts." def flip_first_card(self): first_card = self.cards[0] first_card.flip() class BJ_Game(object): """ A Blackjack Game. """ def __init__(self, names): self.players = [] for name in names: player = BJ_Player(name) self.players.append(player) self.dealer = BJ_Dealer("Dealer") self.deck = BJ_Deck() self.deck.populate() self.deck.shuffle() def get_still_playing(self): remaining = [] for player in self.players: if not player.is_busted(): remaining.append(player) return remaining # list of players still playing (not busted) this round still_playing = property(get_still_playing) def __additional_cards(self, player): while not player.is_busted() and player.is_hitting(): self.deck.deal([player]) print player if player.is_busted(): player.bust() def play(self): # deal initial 2 cards to everyone self.deck.deal(self.players + [self.dealer], per_hand = 2) self.dealer.flip_first_card() # hide dealer's first card for player in self.players: print player print self.dealer # deal additional cards to players for player in self.players: self.__additional_cards(player) self.dealer.flip_first_card() # reveal dealer's first if not self.still_playing: # since all players have busted, just show the dealer's hand print self.dealer else: # deal additional cards to dealer print self.dealer self.__additional_cards(self.dealer) if self.dealer.is_busted(): # everyone still playing wins for player in self.still_playing: player.win() else: # compare each player still playing to dealer for player in self.still_playing: if player.total > self.dealer.total: player.win() elif player.total < self.dealer.total: player.lose() else: player.push() # remove everyone's cards for player in self.players: player.clear() self.dealer.clear() def main(): print "\t\tWelcome to Blackjack!\n" names = [] number = games.ask_number("How many players? (1 - 7): ", low = 1, high = 8) for i in range(number): name = raw_input("Enter player name: ") names.append(name) print game = BJ_Game(names) again = None while again != "n": game.play() again = games.ask_yes_no("\nDo you want to play again?: ") main() raw_input("\n\nPress the enter key to exit.") Since someone decided to call this 'psychic-debugging', I'll go ahead and tell you what the modules are then. Here's the cards module: class Card(object): """ A playing card. """ RANKS = ["A", "2", "3", "4", "5", "6", "7", "8", "9", "10", "J", "Q", "K"] SUITS = ["c", "d", "h", "s"] def __init__(self, rank, suit, face_up = True): self.rank = rank self.suit = suit self.is_face_up = face_up def __str__(self): if self.is_face_up: rep = self.rank + self.suit else: rep = "XX" return rep def flip(self): self.is_face_up = not self.is_face_up class Hand(object): """ A hand of playing cards. """ def init(self): self.cards = [] def __str__(self): if self.cards: rep = "" for card in self.cards: rep += str(card) + "\t" else: rep = "<empty>" return rep def clear(self): self.cards = [] def add(self, card): self.cards.append(card) def give(self, card, other_hand): self.cards.remove(card) other_hand.add(card) class Deck(Hand): """ A deck of playing cards. """ def populate(self): for suit in Card.SUITS: for rank in Card.RANKS: self.add(Card(rank, suit)) def shuffle(self): import random random.shuffle(self.cards) def deal(self, hands, per_hand = 1): for rounds in range(per_hand): for hand in hands: if self.cards: top_card = self.cards[0] self.give(top_card, hand) else: print "Can't continue deal. Out of cards!" if name == "main": print "This is a module with classes for playing cards." raw_input("\n\nPress the enter key to exit.") And here's the games module: class Player(object): """ A player for a game. """ def __init__(self, name, score = 0): self.name = name self.score = score def __str__(self): rep = self.name + ":\t" + str(self.score) return rep def ask_yes_no(question): """Ask a yes or no question.""" response = None while response not in ("y", "n"): response = raw_input(question).lower() return response def ask_number(question, low, high): """Ask for a number within a range.""" response = None while response not in range(low, high): response = int(raw_input(question)) return response if name == "main": print "You ran this module directly (and did not 'import' it)." raw_input("\n\nPress the enter key to exit.")

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  • How to disable Bluetooth auto-answer on Windows 7?

    - by MarkR
    When my BlackBerry 9630 is connected via Bluetooth to my Dell desktop running Windows 7 x64 there are a bunch of Bluetooth services enabled on Windows Advanced Audio BB Bypass service BB Desktop Service Dial-up Networking Headset Audio Gateway Remote Control Remotely Controlled Device When a call arrives on the 9630 Windows immediately answer the call. This is annoying when windows answers the phone before I even hear an audible ring and the hapless caller is saying, "hello. hello" on top of music I have playing. Does anyone know how to tell Windows not to answer the call? I want to be able to don my headset, switch audio to my headset, THEN manually answer the call.

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  • v4l - capture and watch at the same time

    - by John Barrett
    Capturing v4l and line-in audio using mencoder works very well, but I would like to record real-time gameplay video from consoles plugged into the video card. I've used xawtv for this (Works quite well, can preview and record in real time), but when I enable any deinterlacing or aspect ration options the video fails to record. I have to record raw and re-encode the video with the appropriate filters later to get something workable. Other things I have tried: tvtime with xvidcap and jack audio capture - xvidcap drops frames and muxing the audio is impossible as it will go out of sync (I have not found muxer options that work to force a correct frame rate) mencoder capture to file, attempt to pipe tail of file to mplayer... mencoder works great, piping the file is far too heavy to attempt gameplay. Soooo, v4l capture and preview simultaneously, recommendations?

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  • graphics cards with no HDMI output

    - by Noam Gal
    I am currently looking at Gainward GTX260 896MB GS GLH, but I've seen it also on the x275 version - the card doesn't have an HDMI output, only two dvi and one tv out (s-video?). They claim they support HDMI using a dvi-HDMI converter. Will I get a true high definition quality on my TV (assuming it supports it) like that? Or is it not as good, and I should stick to cards that have an HDMI output (ATI), or pay way too much for x295? What about connecting the audio? The x260 comes with an internal spdif cable - does that mean I can connect my soundcard to my graphics card, and have the audio come out through the dvi, and into the HDMI cable? Or am I mixing it all up here, and I have to somehow connect the sound to the TV using a seperate cable (Hoping it has a seperate audio-in for the HDMI channel)?

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  • Streaming flash video does not work on my Mac OS X

    - by dehmann
    Flash videos do not work properly on my Mac. On this Vimeo video, for example, it shows only the beginning frame, and audio stutters like crazy, playing audio for a quarter second or so, then silence, then playing again, etc. I have Flash version 10,0,42,34 on Mac OS 10.5.8. It's a PowerBook G4 (PPC). I tried it in Firefox 3.5.5 and Safari 4.0.3. I tried reinstalling Flash, restarting the computer, and using a fresh user profile in Firefox (so that no extensions are interfering with the site), loading the video fully before playing, but nothing helps. I noticed that youtube videos work better, once loaded enough, although the picture does halt briefly once every 10 or so seconds, even when it's fully loaded.

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  • Enabling Surround sound on a Realtek ALC892 via SPDIF, on Windows 7

    - by Alex
    I have a problem with my ALC892, on an ASRock mainboard (ASRock 890FX Deluxe4). I get only stereo sounds if I use SPDIF connection, in general. My amp shows that is getting surround sound only when I use the Test feature of Windows 7. This test feature allows to know which formats are supported by the audio chip. The tests render correctly both Dolby Digital and DTS. You can find this test under Sounds, Playback Devices, Select Digital Audio, then "Properties". I am using Windows 7 x64, with the latest drivers from the official Realtek website. I also tested other driver versions, both from the Realtek website and from the ASRock one, but had no luck. Thanks for the help. Some specs: CPU: AMD Phenom II X4 965 MOBO: ASRock 890FX Deluxe4 (with onboard Realtek ALC892) Audio amp: Onkyo R-380 (works fine with other sources like PS3 and Xbox 360)

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  • Use external speakers with laptop hooked to separate monitor?

    - by lhan16
    I have a laptop with a set of external speakers hooked up to it on my computer desk. The speakers use the standard 3.5mm audio (headphones) jack. The speakers work fine, but I've recently added a separate monitor to my laptop via HDMI. With the monitor hooked up to my laptop and the speakers still hooked up to the laptop, sound will only come out of the built-in monitor speakers. When I look at my audio settings, there are three different "audio playback devices" showing up, but only the built-in monitor speakers make noise when I click "test" (and I hear nothing when I set any of the other devices as the default. Does anyone know how I can still use my external speakers when using a separate monitor with my laptop? I'm hoping there is a solution that doesn't require the laptop to be open or closed, because I use both scenarios. I came across this post, but it doesn't look like they had much luck.

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  • Android: Use XML Layout for List Cell rather than Java Code Layout (Widgets)

    - by Stephen Finucane
    Hi, I'm in the process of making a music app and I'm currently working on the library functionality. I'm having some problems, however, in working with a list view (In particular, the cells). I'm trying to move from a simple textview layout in each cell that's created within java to one that uses an XML file for layout (Hence keeping the Java file mostly semantic) This is my original code for the cell layout: public View getView(int position, View convertView, ViewGroup parent) { String id = null; TextView tv = new TextView(mContext.getApplicationContext()); if (convertView == null) { music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.TITLE); musiccursor.moveToPosition(position); id = musiccursor.getString(music_column_index); music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.DISPLAY_NAME); musiccursor.moveToPosition(position); id += "\n" + musiccursor.getString(music_column_index); music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Albums.ALBUM); musiccursor.moveToPosition(position); id += "\n" + musiccursor.getString(music_column_index); tv.setText(id); } else tv = (TextView) convertView; return tv; } And my new version: public View getView(int position, View convertView, ViewGroup parent) { View cellLayout = findViewById(R.id.albums_list_cell); ImageView album_art = (ImageView) findViewById(R.id.album_cover); TextView album_title = (TextView) findViewById(R.id.album_title); TextView artist_title = (TextView) findViewById(R.id.artist_title); if (convertView == null) { music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Albums.ALBUM); musiccursor.moveToPosition(position); album_title.setText(musiccursor.getString(music_column_index)); //music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.DISPLAY_NAME); //musiccursor.moveToPosition(position); music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.TITLE); musiccursor.moveToPosition(position); artist_title.setText(musiccursor.getString(music_column_index)); } else{ cellLayout = (TextView) convertView; } return cellLayout; } The initialisation (done in the on create file): musiclist = (ListView) findViewById(R.id.PhoneMusicList); musiclist.setAdapter(new MusicAdapter(this)); musiclist.setOnItemClickListener(musicgridlistener); And the respective XML files: (main) <?xml version="1.0" encoding="utf-8"?> <LinearLayout xmlns:android="http://schemas.android.com/apk/res/android" android:orientation="vertical" android:layout_width="fill_parent" android:layout_height="fill_parent"> <ListView android:id="@+id/PhoneMusicList" android:layout_width="fill_parent" android:layout_height="fill_parent" /> <TextView android:id="@android:id/empty" android:layout_width="wrap_content" android:layout_height="0dip" android:layout_weight="1.0" android:text="@string/no_list_data" /> </LinearLayout> (albums_list_cell) <?xml version="1.0" encoding="utf-8"?> <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/albums_list_cell" android:layout_width="wrap_content" android:layout_height="wrap_content"> <ImageView android:id="@+id/album_cover" android:layout_alignParentLeft="true" android:layout_alignParentTop="true" android:layout_width="50dip" android:layout_height="50dip" /> <TextView android:id="@+id/album_title" android:layout_toRightOf="@+id/album_cover" android:layout_alignParentTop="true" android:layout_width="wrap_content" android:layout_height="wrap_content" /> <TextView android:id="@+id/artist_title" android:layout_toRightOf="@+id/album_cover" android:layout_below="@+id/album_title" android:layout_width="wrap_content" android:layout_height="15dip" /> </RelativeLayout> In theory (based on the tiny bit of Android I've done so far) this should work..it doesn't though. Logcat gives me a null pointer exception at line 96 of the faulty code, which is the album_title.setText line. It could be a problem with my casting but Google tells me this is ok :D Thanks for any help and let me know if you need more info!

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  • XPP-32 over W7-64 on music production laptop

    - by quarlo
    I need to upgrade my laptop and need high performance for music production (recording and mixing). My audio interface manufacturer seems to be unable to successfully convert their drivers to 64-bit. I do not trust a virtual machine to handle real-time audio recording at low enough latency so ... I would like to install XP Pro 32-bit on a separate partition and dual boot since most of the machines that can handle this application now ship with Windows 7 64-bit flavors. I'd like to transit to 64-bit over time assuming M-Audio does eventually get a handle on 64-bit drivers, but really need to ensure that I can stay at 32-bit for now. Does anyone have any experience with this or something similar?

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  • How to fix choppy video with VLC player?

    - by Brian Ojeda
    When playing most movies in VLC player, the video feed is choppy. Furthermore, the sound is near perfect. Another catch is, this video issue wasn't always an issue. I use to play movies with no problem with video or sound. I can not pin point what has changed between the it use to work fine and the time it started being choppy though. I have an ASUS G73JW-A1. This system should be more than enough to handle the demand of playing HD videos. Movies that are normally effected are HD videos. These video range from 4 to 9 gigs. In addition, the videos are in MKV, MP4 (or M4V), and AVCHD (or M2TS) formats. I get the same results when playing move directly off my hard drive or off an external. Finally, all the drivers have been recently checked and updated if needed.

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  • Batch Convert .mkv to .mp4

    - by IamHere
    I want to batch convert all .mkv files in a folder into .mp4 with VLC. It should use the original video-/audio stream and if possible the .ass subtitle of the .mkv. It's not really a conversion, it's more like changing the container – my player can't read the MKV videos. If I do this conversion by hand (manually) it works, but I have a lot of MKV files to convert, so it would take a lot of time. I have searched the internet for a batch file to do this and I found a few. I tried to modify them to my wish, but all attempts I tried just created a .mp4 file that doesn't contain the audio stream and the video stream also cannot be rendered by all my media players on the PC. So could someone tell me how the batch has to look like, so it works with the original video and audio stream (and maybe .ass subtitles)?

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  • How can I access the external microphone with Ubuntu?

    - by Charles Merriam
    Sound in Ubuntu, it has its own special joy. I would like my external microphone to work. Symptoms: I can play sound through the speakers I can play sound through the headsets. Plugging and and plugging headphone output correctly switches. I can record from the built-in microphone, using "Sound Recorder" and others. but: I cannot record from the external microphone. My Sound Preferences/Input panel has no option for an external microphone. If the answer is upgrade the ALSA drivers, please say exactly what to type. Thank you. ======== I'm using Ubuntu 9.10 Karmic Koala on a laptop (Gateway W3501), Sigmatel. That is: ~$ head -1 /proc/asound/card0/code* ==> /proc/asound/card0/codec#0 <== Codec: SigmaTel STAC9205 ==> /proc/asound/card0/codec#1 <== Codec: Conexant ID 2c06 ~$ lspci | grep -i audio 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)

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