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• No HDMI Audio with GeForce 9600GT and nForce board

  - by Bobby

  I've been trying to get HDMI with sound working for the last few days, and I'm a little bit out of ideas. (I've verified that the hardware/Setup works via Windows.) aplay does not list my HDMI device: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 I've already compiled the alsa drivers (1.0.24) from a snapshot (with --with-oss=no) and added the line options snd-hda-intel model=auto # Tried 3stack-dig and 6stack-dig too to /etc/modprobe.d/alsa-base.conf. Still, the device does not show up. If it is important, the HDMI TV is at the moment not configured to be part of the X session (I've tried that to, at least with X restart, and it didn't change anything). What did I miss? $ lspci 00:00.0 Host bridge: nVidia Corporation Device 07c3 (rev a2) 00:00.1 RAM memory: nVidia Corporation nForce 630i memory controller (rev a2) 00:01.0 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.1 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.2 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.3 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.4 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.5 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:01.6 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:02.0 RAM memory: nVidia Corporation nForce 630i memory controller (rev a1) 00:03.0 ISA bridge: nVidia Corporation MCP73 LPC Bridge (rev a2) 00:03.1 SMBus: nVidia Corporation MCP73 SMBus (rev a1) 00:03.2 RAM memory: nVidia Corporation MCP73 Memory Controller (rev a1) 00:03.4 RAM memory: nVidia Corporation MCP73 Memory Controller (rev a1) 00:04.0 USB Controller: nVidia Corporation GeForce 7100/nForce 630i USB (rev a1) 00:04.1 USB Controller: nVidia Corporation MCP73 [nForce 630i] USB 2.0 Controller (EHCI) (rev a1) 00:08.0 IDE interface: nVidia Corporation MCP73 IDE (rev a1) 00:09.0 Audio device: nVidia Corporation MCP73 High Definition Audio (rev a1) 00:0a.0 PCI bridge: nVidia Corporation MCP73 PCI Express bridge (rev a1) 00:0b.0 PCI bridge: nVidia Corporation MCP73 PCI Express bridge (rev a1) 00:0c.0 PCI bridge: nVidia Corporation MCP73 PCI Express bridge (rev a1) 00:0d.0 PCI bridge: nVidia Corporation MCP73 PCI Express bridge (rev a1) 00:0e.0 IDE interface: nVidia Corporation MCP73 IDE (rev a2) 00:0f.0 Ethernet controller: nVidia Corporation MCP73 Ethernet (rev a2) 02:00.0 VGA compatible controller: nVidia Corporation G94 [GeForce 9600 GT] (rev a1)   $ aplay -L default pulse Playback/recording through the PulseAudio sound server front:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Digital IEC958 (S/PDIF) Digital Audio Output dmix:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct sample mixing device dmix:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct sample mixing device dsnoop:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct sample snooping device dsnoop:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct sample snooping device hw:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct hardware device without any conversions hw:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct hardware device without any conversions plughw:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Hardware device with all software conversions plughw:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Hardware device with all software conversions

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• PandoraBar Packs Pandora Radio Client into a Compact Case

  - by Jason Fitzpatrick

  This stylish and compact build makes it easy to enjoy streaming radio without the bulk and overhead of running your entire computer to do so. Check out the video to see the compact streaming radio box in action. Courtesy of tinker blog Engscope, we find this clean Pandora-client-in-box build. Currently the project blog has a cursory overview of the project with the demo video but promises future updates detailing the software and hardware components of the build. If you can’t wait that long, make sure to check out some of the previous Wi-Fi radio builds we’ve shared: DIY Wi-Fi Radio Brings Wireless Tunes Anywhere in Your House and Wi-Fi Speakers Stream Music Anywhere. Pandobar [via Hacked Gadgets] How To Create a Customized Windows 7 Installation Disc With Integrated Updates How to Get Pro Features in Windows Home Versions with Third Party Tools HTG Explains: Is ReadyBoost Worth Using?

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• Bose USB audio: crackling popping sound, eventually die

  - by Richard Barrett

  I've been trying to troubleshoot this issue for a while now. Any help would be much appreciated. I'm having trouble getting my Bose "Companion 5 multimedia speakers" working with my installation of Ubuntu 12.04 (link to Bose product here: http://www.bose.com/controller?url=/shop_online/digital_music_systems/computer_speakers/companion_5/index.jsp ). The issue seems to be low level (not just Ubuntu). What happens: When I boot into Ubuntu, I can get Rhythm box to play ok. However, if I try anything else (an .avi file, a webpage, or Clementine player with mp3 files) I get crackling, popping, or choppy sounds. If I move the mouse around, especially if it seems graphic intensive, the problem gets worse (more crackling noises). The more taxing it appears to be, the more likely it is that the sound will just die altogether until I reboot. For some reason the videos at www.bloomberg.com seem especially bad for it (my sound normally goes dead in under 45 seconds and won't work until reboot). Both my desktop running Ubuntu 12.04 and my laptop (running the same) have the same crackling problem. Troubleshooting so far: A friend of mine who knows linux well tried to solve it for me without any luck. He took pulseaudio out of the equation, but still had the problem just using AlSA. Among the many things he tried was adjusting the latency, but that didn't help either. I've also tried things like adjusting the USB device settings in the config file from -2 to -1 so that it will use my USB sound and I also commented out the lines that would stop that. These don't do anything. (That really seems like it's for someone who is getting no sound at all, so it's not surprising this won't work.) My friend's laptop running his Archlinux could play my Bose USB speakers without any problems. I also tried setting my daemon.conf file to use 6 channels (based on this http://lotphelp.com/lotp/configure-ubuntu-51-surround-sound ) but that didn't work either. I recently used a DVD to boot into Ubuntu Studio 12.04 (because it uses a live audio kernel) and this happened: I got perfect sound for a minute or two When I started moving windows around while sound was playing, the sound died again. Perhaps more interesting: There is a headphone out jack on the Bose system. When I use it, the audio is perfect for all applications (even the deadly bloomberg.com videos with .avi playing at the same time and moving around windows). Also, there is an audio-in jack on the Bose system. I can use a male-to-male mini jack to go from my soundcard's output to the Bose input and then all sound works perfectly. -However, it still requires the Bose to be plugged in to USB, otherwise I lose all sound. Any thoughts? Any suggestions for trouble shooting? (Or any suggestions for somewhere else to post to solve this?) Any logs or other files I can provide to help someone help me work this out? Your help is much appreciated! Rick BTW: I sometimes get people posting responses like "My Bose USB system works great with Ubuntu 12.04," without any more details. Is there anything I should ask such people to narrow down my problem? (It's kind of annoying to hear such a response because it doesn't help solve my problem.)

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• linux mint VIA sound issue

  - by user2699451

  So I installed linux Mint 15 "Olivia" 64 bit on my Mecer W550EU laptop I have HD Audio with a VIA chipset charles-W55xEU charles # lsmod | grep snd snd_hda_codec_hdmi 36913 1 snd_hda_codec_via 51018 1 snd_hda_intel 39619 5 snd_hda_codec 136453 3 snd_hda_codec_hdmi,snd_hda_codec_via,snd_hda_intel snd_hwdep 13602 1 snd_hda_codec snd_pcm 97451 4 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18710 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30180 1 snd_seq_midi snd_seq 61554 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 29425 2 snd_pcm,snd_seq snd 68876 19 snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_via,snd_pcm,snd_seq,snd_rawmidi,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 12680 1 snd And my sound card charles-W55xEU charles # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: VT1802 Analog [VT1802 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: VT1802 HP [VT1802 HP] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 and my audio device charles-W55xEU charles # lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High `Definition Audio Controller (rev 04)` Subsystem: CLEVO/KAPOK Computer Device 0550 Flags: bus master, fast devsel, latency 0, IRQ 47 Memory at f7c10000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel Sometimes when I boot up, soundworks, other times it doenst, it is completely random, so far, no-one on xchat linux help or linux mint forums was able to help me, I have always had issues with sound on VIA chipsets I have: sudo apt-get upgrade && apt-get install mint-meta-cinnamon it seemed to help but after 2-3 reboots, the problem came back, btw, everytime I checked, pulse audio is selected to Duplex Audio Input & Output and alsa mixer is always unmuted!

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• Ask the Readers: Have You Cut the Cable Cord?

  - by Jason Fitzpatrick

  The proliferation of streaming services and diverse media content has made cutting the cable cord an appetizing option for many people. Have you cut the cord, stuck with traditional services, or mixed the two together? It’s easier than ever to ditch the cable subscription and still keep watching your favorite TV shows and movies (but still tricky to catch live events and television without a cable subscription). This week we’re interested in hearing all about your media arrangements: Do you have a traditional cable bundle? Ditched it for a Netflix-only experience? Pay for both because you just can’t get enough of live TV and streaming content? Sounds off in the comments with the state of your media setup, the more detailed the better, and then check back in on Friday for the What You Said roundup. 6 Ways Windows 8 Is More Secure Than Windows 7 HTG Explains: Why It’s Good That Your Computer’s RAM Is Full 10 Awesome Improvements For Desktop Users in Windows 8

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• Linux application that bundles multiple incoming audio and video streams into one container file?

  - by StackedCrooked

  I've been assigned to implement a video on-demand service for a local university. Different aspects of the lectures (video, audio, screen cast, white board) will be recorded. During a lecture all these data streams arrive at one Linux server. This server should transcode and bundle all these streams into one container (Matroska) file. My options seem to be: Write a GStreamer application do something with FFMPEG do something with VLC ...? Has anyone done something similar in the past? Can you recommend something? Edit For those interested, here are a few of my findings: Matroska is not a good format for streaming (it's possible, but it's not its primary intent) For Flash streaming you can use MPEG4 If you want to combine different videos into one video where each subvideo occupies a rectangular portion of the total screen, then this GStreamer script is useful (I found it on this blog post). Desktop capture works fine with VLC

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• Flowmotion Running on Ubuntu Server [migrated]

  - by Thomas Egan

  I am trying to configure Flowmotion to work on my Ubuntu Server. At present I use LAMP to serve pages from a Virtualbox installation. I will be moving this to a dedicated server but would like to enable true streaming of videos using this installation. I am only interested in open source streaming for a research project and although I have installed Flowmotion via apt-get I don't know how to start the service so that embedded videos located on the server will stream. Can anybody provide any information regarding this or online resources I may have missed? I have checked the documentation however if appears far too complex. Just clarify I'm running VirtualBox 4.2.1 on Mac OSX 10.6.8 and Ubuntu Server 12.06 64-bit

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• Asynchronous Streaming in ASP.NET WebApi

  - by andresv

   Hi everyone, if you use the cool MVC4 WebApi you might encounter yourself in a common situation where you need to return a rather large amount of data (most probably from a database) and you want to accomplish two things: Use streaming so the client fetch the data as needed, and that directly correlates to more fetching in the server side (from our database, for example) without consuming large amounts of memory. Leverage the new MVC4 WebApi and .NET 4.5 async/await asynchronous execution model to free ASP.NET Threadpool threads (if possible).  So, #1 and #2 are not directly related to each other and we could implement our code fulfilling one or the other, or both. The main point about #1 is that we want our method to immediately return to the caller a stream, and that client side stream be represented by a server side stream that gets written (and its related database fetch) only when needed. In this case we would need some form of "state machine" that keeps running in the server and "knows" what is the next thing to fetch into the output stream when the client ask for more content. This technique is generally called a "continuation" and is nothing new in .NET, in fact using an IEnumerable<> interface and the "yield return" keyword does exactly that, so our first impulse might be to write our WebApi method more or less like this:           public IEnumerable<Metadata> Get([FromUri] int accountId)         {             // Execute the command and get a reader             using (var reader = GetMetadataListReader(accountId))             {                 // Read rows asynchronously, put data into buffer and write asynchronously                 while (reader.Read())                 {                     yield return MapRecord(reader);                 }             }         }   While the above method works, unfortunately it doesn't accomplish our objective of returning immediately to the caller, and that's because the MVC WebApi infrastructure doesn't yet recognize our intentions and when it finds an IEnumerable return value, enumerates it before returning to the client its values. To prove my point, I can code a test method that calls this method, for example:        [TestMethod]         public void StreamedDownload()         {             var baseUrl = @"http://localhost:57771/api/metadata/1";             var client = new HttpClient();             var sw = Stopwatch.StartNew();             var stream = client.GetStreamAsync(baseUrl).Result;             sw.Stop();             Debug.WriteLine("Elapsed time Call: {0}ms", sw.ElapsedMilliseconds); } So, I would expect the line "var stream = client.GetStreamAsync(baseUrl).Result" returns immediately without server-side fetching of all data in the database reader, and this didn't happened. To make the behavior more evident, you could insert a wait time (like Thread.Sleep(1000);) inside the "while" loop, and you will see that the client call (GetStreamAsync) is not going to return control after n seconds (being n == number of reader records being fetched).Ok, we know this doesn't work, and the question would be: is there a way to do it?Fortunately, YES!  and is not very difficult although a little more convoluted than our simple IEnumerable return value. Maybe in the future this scenario will be automatically detected and supported in MVC/WebApi.The solution to our needs is to use a very handy class named PushStreamContent and then our method signature needs to change to accommodate this, returning an HttpResponseMessage instead of our previously used IEnumerable<>. The final code will be something like this: public HttpResponseMessage Get([FromUri] int accountId)         {             HttpResponseMessage response = Request.CreateResponse();             // Create push content with a delegate that will get called when it is time to write out              // the response.             response.Content = new PushStreamContent(                 async (outputStream, httpContent, transportContext) =>                 {                     try                     {                         // Execute the command and get a reader                         using (var reader = GetMetadataListReader(accountId))                         {                             // Read rows asynchronously, put data into buffer and write asynchronously                             while (await reader.ReadAsync())                             {                                 var rec = MapRecord(reader);                                 var str = await JsonConvert.SerializeObjectAsync(rec);                                 var buffer = UTF8Encoding.UTF8.GetBytes(str);                                 // Write out data to output stream                                 await outputStream.WriteAsync(buffer, 0, buffer.Length);                             }                         }                     }                     catch(HttpException ex)                     {                         if (ex.ErrorCode == -2147023667) // The remote host closed the connection.                          {                             return;                         }                     }                     finally                     {                         // Close output stream as we are done                         outputStream.Close();                     }                 });             return response;         } As an extra bonus, all involved classes used already support async/await asynchronous execution model, so taking advantage of that was very easy. Please note that the PushStreamContent class receives in its constructor a lambda (specifically an Action) and we decorated our anonymous method with the async keyword (not a very well known technique but quite handy) so we can await over the I/O intensive calls we execute like reading from the database reader, serializing our entity and finally writing to the output stream.  Well, if we execute the test again we will immediately notice that the a line returns immediately and then the rest of the server code is executed only when the client reads through the obtained stream, therefore we get low memory usage and far greater scalability for our beloved application serving big chunks of data.Enjoy!Andrés.        

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• Streaming Netflix through Wifi kills all other devices

  - by maleki

  I have a "Belkin N150" Router. My current setup is wired connection to my personal tv and xbox with wireless going to the Wii and Bedroom laptop. Anytime I try streaming Netflix Instant Queue on the Wii, every other device has problems. The Wii works fine but everything else's connection is destroyed. Streaming through the Xbox doesn't create any issues for other devices. What kind of issue is occurring and how can I fix it?

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• Split UInt32 (audio frame) into two SInt16s (left and right)?

  - by morgancodes

  Total noob to anything lower-level than Java, diving into iPhone audio, and realing from all of the casting/pointers/raw memory access. I'm working with some example code wich reads a WAV file from disc and returns stereo samples as single UInt32 values. If I understand correctly, this is just a convenient way to return the 32 bits of memory required to create two 16 bit samples. Eventually this data gets written to a buffer, and an audio unit picks it up down the road. Even though the data is written in UInt32-sized chunks, it eventually is interpreted as pairs of 16-bit samples. What I need help with is splitting these UInt32 frames into left and right samples. I'm assuming I'll want to convert each UInt32 into an SInt16, since an audio sample is a signed value. It seems to me that for efficiency's sake, I ought to be able to simply point to the same blocks in memory, and avoid any copying. So, in pseudo-code, it would be something like this: UInt32 myStereoFrame = getFramefromFilePlayer; SInt16* leftChannel = getFirst16Bits(myStereoFrame); SInt16* rightChannel = getSecond16Bits(myStereoFrame); Can anyone help me turn my pseudo into real code?

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

  - by Peter Hoffmann

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

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• bluetooth headset can connect, but not visible in pulse audio

  - by Kim Marivoet

  I have a plantronics bluetooth headset, and until yesterday I could use it without any problem. However, today it suddenly stopped working (maybe related to the last software update I did). I can still connect/disconnect my headset, but it doesn't show up in pulse audio anymore. I read through various posts that describes kind of the same problem, but none of the suggested solutions worked. I get following error in the syslog: Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/HFPAG Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/A2DPSource Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/A2DPSink Oct 13 16:50:09 desktop kernel: [ 17.340943] input: 48:C1:AC:08:FE:8F as /devices/virtual/input/input14 Oct 13 16:50:09 desktop bluetoothd[1040]: /org/bluez/1040/hci0/dev_48_C1_AC_08_FE_8F/fd0: fd(36) ready Oct 13 16:50:09 desktop rtkit-daemon[1894]: Successfully made thread 2213 of process 1892 (n/a) owned by '1000' RT at priority 5. Oct 13 16:50:09 desktop rtkit-daemon[1894]: Supervising 5 threads of 1 processes of 1 users. Oct 13 16:50:10 desktop bluetoothd[1040]: Badly formated or unrecognized command: AT+XEVENT=USER-AGENT,COM.PLANTRONICS,PLT_VOYAGERPRO,0109,27.90,FFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFF Oct 13 16:50:10 desktop bluetoothd[1040]: Audio connection got disconnected Any help would be much appreciated. I'm using Ubuntu 12.04. Thanks, Kim

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• Setting up Beats audio on HP Pavilion m6

  - by Joel Auterson

  I have an HP Pavilion m6-1054sa laptop, with a Beats subwoofer on the bottom. The normal laptop speakers work fine under Ubuntu but the Beats speaker(s?) does not. Anyone know how to get this working? Here's my lspci output, if it helps... 00:00.0 Host bridge: Intel Corporation Ivy Bridge DRAM Controller (rev 09) 00:01.0 PCI bridge: Intel Corporation Ivy Bridge PCI Express Root Port (rev 09) 00:02.0 VGA compatible controller: Intel Corporation Ivy Bridge Graphics Controller (rev 09) 00:14.0 USB controller: Intel Corporation Panther Point USB xHCI Host Controller (rev 04) 00:16.0 Communication controller: Intel Corporation Panther Point MEI Controller #1 (rev 04) 00:1a.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #2 (rev 04) 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 1 (rev c4) 00:1c.1 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 2 (rev c4) 00:1d.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation Panther Point LPC Controller (rev 04) 00:1f.2 RAID bus controller: Intel Corporation 82801 Mobile SATA Controller [RAID mode] (rev 04) 00:1f.3 SMBus: Intel Corporation Panther Point SMBus Controller (rev 04) 01:00.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI Thames XT/GL [Radeon HD 7600M Series] (rev ff) 07:00.0 Unassigned class [ff00]: Realtek Semiconductor Co., Ltd. Device 5289 (rev 01) 07:00.2 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 0a) 08:00.0 Network controller: Intel Corporation Centrino Wireless-N 2230 (rev c4)

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• Problem Installing Xubuntu 12.04 Audio-Video codecs

  - by Seib

  I used Crouton to install Xubuntu 12.04.4 LTS with the XFCE environment on my HP Chromebook 11. I've gotten it all fully installed and everything; the only thing I'm doing now is the basic setup things, like adding codecs and other things like LibreOffice, VLC, Firefox, Ubuntu Software Center, etc. This information I got from 2 sources: http://www.efytimes.com/e1/fullnews.asp?edid=137269 http://www.binarytides.com/better-xubuntu-14-04/ . I'm currently on the same step at both URLs, which is #6 on Link 1 and #8 on Link 2. Per the articles, which both said the same thing, I typed in sudo apt-get install xubuntu-restricted-extras libavcodec-extra and it didn't do anything. It just kept on saying the same thing: Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package libavcodec-extra I've spend the last hour or so scouring the internet for a solution, for a hint even at what is going on. I don't want to do a clean reinstall, for two reasons: 1) it's takes like 1.5h just to get the croot fully installed, and 2) everything but this out of what I've done so far (up to #6 at Link 1 & up to #8 at Link 2) works except the audio. I've already installed flash, so YouTube works fine. It's just I can't hear any audio. Please help? Thanks in advance. I appreciate all the great help I've been getting from AskUbuntu lately. You all are great.

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• M-Audio Delta starts up at wrong sample rate

  - by steevc

  When the PC starts my M-Audio Delta 66 is using 48000kHz sampling rate when it is set for 44100 in Envy24. This causes audio to play slower than it should. This is in Kubuntu 14.04 on my new PC using an AMD A8 6500 with 8GB. When I first installed it seemed okay, but at some point it went wrong and has been doing this consistently since then. Kernel is 3.13.0-24-generic #47-Ubuntu SMP Fri May 2 23:31:42 UTC 2014 i686 athlon i686 GNU/Linux steve@slarti:~$ cat /proc/asound/card2/pcm0p/sub0/hw_params access: MMAP_INTERLEAVED format: S32_LE subformat: STD channels: 10 rate: 48000 (48000/1) period_size: 441 buffer_size: 3528 I can get it to switch to 44100 if I disable/enable the Delta in Pulseaudio volume control, but I have to do this every day and the sound still seems distorted. I can't see any issues in any of the config or log files I can find. If I boot the PC with a Mint Live USB it starts at 44100 and sound fine. Originally reported on Youtube is playing at the wrong speed - maybe soundcard related, but I'll close that and have this more relevant question instead.

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• ffmpeg add two audio streams to video

  - by Tossin Hausen

  I tried this: ffmpeg -i /sdcard/video/transcode/video.avi -map 0:0,0 -i /sdcard/video/transcode/first.mp3 -map 1:0,1 -i /sdcard/video/transcode/second.mp3 -map 2:0,2 -acodec copy -vcodec py /sdcard/video/transcode/Output.avi to add two audio streams to one video file. But ffmpeg says the number of mappings should match the number of output streams. What is wrong here? I'm trying to work with an Android build of FFmepg "ffmpeg for android beta". "Does not work" means that this uncommunicative Android build of FFmpeg just stops without giving any error message. The -codec copy option does not work with this build. Now I tried the same set of files with the FFmpeg called command line tool that comes with Ubuntu 10. Something (can't say where it is from). The -codec copy option does not work with this FFmpeg too. Here the complete output: m30x:~/movie/Film$ ffmpeg -i input.avi -i first.mp3 -i second.mp3 -map 0 -map 1 -map 2 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x93cfd10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Input #2, mp3, from 'second.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #2.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Number of stream maps must match number of output streams Merging only one audio stream with the video stream works with Ubuntu and Android version of FFmpeg. Here the complete output: ffmpeg -i input.avi -i first.mp3 -map 0 -map 1 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x9bfad10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 576x432, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 6157 fps=6156 q=-1.0 size= 31667kB time=246.28 bitrate=1053.3kbits/s Do you have an idea why it does not work with two audio streams? By the way, ffmpeg -i input_with_first_audio_stream.avi -i second.mp3 -acodec copy -vcodec copy output_two_audio_streams.avi -newaudio works with both versions of ffmpeg that I use, but the first audio stream is played too fast (x10 or more), while the second audio stream is played correct. Many thanks in advance and sorry for my unconventional question and outdated versions of ffmpeg. But I am a lamer and it is not so easy for me to compile from the source (especially for the Android version). I will try to compile an up to date version of ffmpeg with Ubuntu, but I don't have much free time.

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• 5.1 sound in Unity3d 3.5

  - by N0xus

  I'm trying to implement 5.1 surround sound in my game. I've set Unity's AudioManager to a default of 5.1 surround and loaded in a 6 channel audio clip that should play a sound in each of the different audio spots. However, when I go to run my game, all I get is flat sound coming out of my front two speakers. Even then, these don't play the sound they should (front speaker should play "front speaker" right should play "right speaker" and so). Both speakers just end up playing the entire sound file. I've tried looking to see if there is a parameter that I have missed, but information on how to set up 5.1 sound in Unity is lacking (or my google skills aren't that good) and I can't get it to work as intended. Could someone please either tell me what I'm missing, or point me in the right direction? My audio source is situated at point (0, 0, 0) with my camera also being in the same point. I've moved about the scene but the same thing happens as I've already described.

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• Alsa devices under Wine

  - by Roberto Aloi

  Hi all, I'm running OpenSuse 11.2 and Wine 1.1.28. Even if audio is perfectly working fine for me (Skype, Banshee, etc), when I try to configure audio for Wine (to use Spotify) I cannot hear anything from the audio test. In the winecfg audio tab, ALSA is checked, but no devices are available. I tried to run alsaconf (it needs root permissions) but it returns: No supported PnP or PCI card found No legacy drivers available, either. Any idea?

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• How do I generate thumbnails of time increments in a video streaming from a URL for the iPhone?

  - by Jesse

  I am able to stream a video from a URL, but I can't quite figure out how to generate thumbnail images of this streaming video at a certain time increment and allow the viewer to skip ahead by touching the thumbnails. I would also need to be able to display the thumbnails when the screen is touched during the video. Any help would be greatly appreciated

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• Audio stream mangement in Linux

  - by User1

  I have a very complicated audio setup for a project. Here's what we have: 3 applications playing sound 2 applications recording sound 2 sound cards I really don't really have the code to any of these applications. All I want to do is monitor and control the audio streams. Here are a few examples of operations I'd like to do while the applications are running: Mute one of the incoming audio streams. Have one of the incoming audio streams do a "solo" (be the only stream that can "talk"). Get a graph (about 30 seconds worth) of the audio that each stream produced. Send one of the audio streams to soundcard #1, but all three audio streams to soundcard #2. I would likely switch audio streams every 2 minutes or so with one of the operations listed above. A GUI would be preferred. I started looking at the sound systems in Linux and it gets extremely complex and I feel like there have been many new advances in the past few years. I see jack, pulseaudio, artsd, and several other packages. They all have some promise but where should I start? Is there something someone already built that can help?

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• Encode divx into iPhone-compatible MPEG4 while downloading?

  - by Jonathan

  Is there a way to encode a DivX file into an iPhone-compatible video file while the DivX is downloading/streaming? Further, is it possible to re-stream the resulting iPhone-compatible video to the iPhone with a minimal (say, 2 minute or so) delay? I've seen other questions about encoding iPhone-compatible videos but they don't quite address the streaming video source and potential of streaming output that I'm interested in.

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• Is there an easy way to copy an audio CD in Mac OS X?

  - by Bob D

  (not a commercial CD). I did some recordings of a band years ago and ran into one of the band members who asked me if I could make copies. I assumed that this would be easy. I know that I can rip the CD into iTunes and then burn a new CD, but I have two optical drives available, is there a way to simply copy the CD from one drive to the other in one step?

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• Ubuntu: how to get audio to work in both Spotify (under Wine) and Flash (in Firefox)?

  - by Jonik

  I'm running Spotify on Linux using Wine. Sound worked great (even though the sound test in winecfg failed!), until I installed alsa-oss package yesterday to get Flash sound working in Firefox. Now Spotify says: "There is a problem with your sound card. Spotify can't play music." So the question is, how to get the sound in Spotify working again, so that it also keeps working in Flash & Firefox? Tweak some ALSA settings? Spotify settings? Add/remove some packages? By the way, curiously, now that sound doesn't work in Spotify, winecfg's "Test Sound" does work! This is Ubuntu 8.04 (Hardy). Sound card / driver is probably an integrated AC'97. Please mention if any additional information about the system is needed! Update: I have Flash 10 installed (outside the packaging system, using $MOZ_PLUGIN_PATH env variable), but also had Flash 9 from flashplugin-nonfree package - and the earlier version was being used by Firefox! Based on what Mike Arthur said about Flash and alsa-oss, I removed the older Flash (flashplugin-nonfree package) and alsa-oss - and Flash sound still works, which is nice. But for some reason Spotify still doesn't play sound, even though things should now be like they were originally... Update 2: Got it working, all smoothly, finally.

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• Audio recording, a tool for human-aided drum quantizing.

  - by basilio.mp

  I have this situation: the drummer records the track (8 tracks in a multitrack session). Now, how do I check how distant are the recorded beats from their theoretical position i.e.: there is always some error in human recorded tracks, but is there any software that can show me the ideal (theoretical, quantized) beat and the recorded one and could alert me if the error is too big. P.S.: I'm searching for a standalone tool, or for a plugin that can work with Adobe Audition 3 or Nuendo 3.

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• What is a tool that is Swiss army knife of audio as ffmpeg is Swiss army knife of video?

  - by Cory

  I need MP3 command line tool that can convert, slice, merge, and etc. Is there a such tool?

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