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• Resampling audio output for A2DP (from PCM WAV)

  - by user1669982

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

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• Audio Playback Rate in Android

  - by Marquis

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

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• Playback audio data with GWT

  - by Henrik

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

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• Real-time equalizer for all audio on computer

  - by greye

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

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

  - by Onion

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

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• Filter design for audio signal.

  - by beanyblue

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

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• Multiple Audio Issues

  - by Lerp

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

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• Windows Media Player - Can't find video codec for the file, will it still play the audio part ?

  - by Roey

  If I try to play a movie file, whose video codec WMP doesn't recognize, and assuming it does know the audio type, will it play just the audio ? or will it flash an error message and stop? How do I test this out? Thanks Roey

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

  - by Rob

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

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

  - by AisIceEyes

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

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

  - by iter

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

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

  - by FlabbergastedPickle

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

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

  - by David Dimalanta

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

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

  - by Rob

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

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

  - by Rob42

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

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

  - by user1296362

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

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

  - by andrewdotcom

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

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• mplayer audio desync

  - by geek

  I have and avi file and an ac3 file that contains an alternate audio stream. I run mplayer like: mplayer -audiofile foo.ac3 bar.avi mplayer takes the audio stream from the ac3 file as expected, but when I try to scroll the video using arrows or pgup/pgdown keys, the audio gets desynced: mplayer just starts playing the audio stream from the beginning. Do I have to pass any additional command line arguments in order to make it scroll properly without desyncing audio?

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• Ubuntu Studio Audio Issue with Alsa - No sound

  - by ddragon

  Spec: OS: Ubuntu Studio 13.10 64bit CPU: AMD FX 4100 Quad Core Memory: 6GB DDR3 Video: Radeon HD 4250 (embedded on the mobo) Sound: Delta 66 PCI Issue: I just installed Ubuntu Studio and found out that the streaming audio on a few common website such as Youtube had no sound, and also my CD/DVDs via a player. Thus, in the terminal, I entered: sudo alsa force-reload It actually worked but the sound/audio output was MONO and NOT Stereo (the sources are set to stereo stereo), and it seemed I was not able to locate any settings that can switch the output sound to stereo at all. I went through many forums and eventually "autoremove" pulseaudio since many said I would not be able to utilize both pulseaudio and alsa in this case. Now, I have no audio whatsoever. Does this version of Ubuntu only offer mono sound/audio no matter what I do? Then, I may just need to ditch the whole thing and go back to Windows, which I don't want to since Ubuntu Studio offers many great apps, soundfonts etc.. I have also installed restricted extra, but even after rebooting, it did not resolve the issue. In the terminal mode, I pulled "alsamixer" and unmuted almost everything. But still no sound after a reboot. Just an FYI, I have no saved data under this version of Ubuntu Studio yet, so please feel free to let me know whether I need to install Studio 12.10 instead or mess with some installing/uninstalling apps/plug-ins, etc... If it breaks at some point, all I need to do is to re-install it, which I do not mind at all. Or, if you can provide me a step by step instruction to get this work, I do not mind clean install the Studio 13.10 then wait for your instruction AT ALL!

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• early audio offset in Audacity and VLC, but not Banshee

  - by reek

  I'm editing audio files with speech in Audacity, marking particular types of speech. I just noticed that files edited in Windows have different intervals marked than files edited in Ubuntu. After testing and confirming this error, it seems that the audio playback in Ubuntu clips the sound too early from the end (early offset), which causes the person doing the editing to mark the interval wrongly. Interestingly, the error appears in Audacity and VLC (which I sometimes use for playback), but NOT Banshee. Since both Audacity and VLC have this problem, I assume it is not application-specific. I don't know why Banshee handles this without problem though... Are there any ALSA or Pulseaudio settings that are likely to cause this problem (I know very little about either)? The task itself does not appear to consume large amounts of resources, but I am on an old laptop, so here are my specs: Ubuntu 11.10. Dell XPS m1210 1.6 GHz Intel Core, 2 x 512 Mb 667 MHz RAM, Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01). Audacity settings: Device Interface: ALSA (cannot select anything else)

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• 12.04.1 no audio through HDMI

  - by JoJo

  having a bit of an issue with getting audio to go through HDMI. Here are the base specs: OS: Ubuntu Desktop 12.04.1 x64 CPU: AMD A10-5800K 3.8G 4M FM2 R Mobo: MSI FM2-A75MA-E35 OS: Ubuntu 12.04.1 LTS Vid Card: (integrated on CPU) AMD Radeon HD 7660D HDMI sound works fine under Win7 (after mobo and vid drivers are installed), so it's not physically broken. Audio through the normal headphone jacks works fine under Ubuntu. Looking at the audio panel, there is no HDMI output at all. aplay -l also reports only: card 0: Generic [HD-Audio Generic], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 In additional drivers there are two versions: ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER (post-release update) The first installs, but problem persists. I do get more resolutions to pick from. Second version does not, reporting it failed installation and to find details at: /var/log/jockey.log Looked at the log, and it's insanely long, if necessary I can get it to you guys. Did more research and some had luck by manually installing the drivers, so tried to give that a shot by following this: https://help.ubuntu.com/community/BinaryDriverHowto/ATI#Manually_installing_Catalyst_12.6 starting at 3.1 Manually installing Catalyst 12.6. I immediately had 2 issues, (1) the AMD website does not provide any drivers for Linux, and (2) the following command did not work: sudo sh amd-driver-installer-12-6-x86.x86_64.run --buildpkg Ubuntu/precise sh: 0: Can't open amd-driver-installer-12-6-x86.x86_64.run Some other posts stated to update "alsa-drivers", but that also did not work as install command for the new version of them did not work. I forget the exact issue, but similar to above, cannot open / cannot find. Any help would be greatly appreciated!

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• Can't configure 5.1 audio with 12.04

  - by xster

  I have an Intel ALC892 and a Nvidia GT 520m connected to speakers via HDMI. On lspci, I see 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) Subsystem: ZOTAC International (MCO) Ltd. Device a218 Flags: bus master, fast devsel, latency 0, IRQ 47 Memory at db400000 (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 02:00.1 Audio device: NVIDIA Corporation HDMI Audio stub (rev a1) Subsystem: ZOTAC International (MCO) Ltd. Device 2180 Flags: bus master, fast devsel, latency 0, IRQ 18 Memory at db080000 (32-bit, non-prefetchable) [size=16K] Capabilities: [60] Power Management version 3 Capabilities: [68] MSI: Enable- Count=1/1 Maskable- 64bit+ Capabilities: [78] Express Endpoint, MSI 00 Kernel driver in use: snd_hda_intel My alsamixer looks like I enabled pulseaudio configuration file to have 6 channels. My sound setting looks like When I use the test dialog, only front left and right have sounds. If I use alsa in XBMC on a 5.1 video, there's no sound. If I use pulseaudio, only front right and left have sound. I can barely hear any speech since I'm guessing it's mapped to front center. Any clues?

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• M-Audio Delta 1010LT on 12.04

  - by user74039

  I have 12.04 64bit installed, my soundcard is a Delta 1010LT, it seems to be partially detected, I've been following steps here: https://help.ubuntu.com/community/SoundTroubleshooting/ lspci -v | grep -A7 -i "audio" shows this: 04:07.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) Subsystem: VIA Technologies Inc. M-Audio Delta 1010LT Flags: bus master, medium devsel, latency 64, IRQ 22 I/O ports at ec00 [size=32] I/O ports at e880 [size=16] I/O ports at e800 [size=16] I/O ports at e480 [size=64] Capabilities: <access denied> Kernel driver in use: snd_ice1712 aplay shows this: **** List of PLAYBACK Hardware Devices **** card 0: M1010LT [M Audio Delta 1010LT], device 0: ICE1712 multi [ICE1712 multi] Subdevices: 1/1 Subdevice #0: subdevice #0 In the sound settings on the desktop all I see is the ICE1712 S/PDIF, which I don't use, I want to use the individual outputs on the card, I'm not so bothered about inputs, I just want the playback for now. If I open alsamixer in the console, I see all of the output and input channels, i've raised the volume on them but I don't get anything in the sound settings on the desktop and when I play any sound, I hear nothing. Can someone help?

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• Ubuntu audio mysteriously stopped working (12.04)

  - by Laika

  Well, I've been a user of Ubuntu 12.04 LTS since April now, and it's been a very pleasant experience. I'm a big fan of electronic music, and I tend to have my tracks playing in the background while I do things on my laptop, either in YouTube or in Clementine, my default music player. All has worked very well until now. A couple of days ago my entire PC started to lag really badly. Almost everything was unusable. I opened up System Monitor via the terminal to find a process called "pulseaudio" using nearly 1GB of RAM and over 80% of my CPU. I needed to get some important work done and so I killed the process without thinking. Once again today, pulseaudio decided to lag the hell out of my PC, and so I killed it again. Nothing seemed to happen immediately, but once I opened up YouTube all the audio on videos stuttered a lot, while the videos played smoothly. I restarted Firefox to find that the audio was now not working at all, with both headphones and speakers, and the volume up quite a bit (it's not muted, I've checked that!). A little bit of research later and I've discovered that pulseaudio plays an important part in Ubuntu's audio. Even after restarting my PC the audio still ceases to work in any applications or with any output. The pulseaudio process refuses to start up again. So, can you help me out here? What can I do to fix my problem, and why was pulseaudio doing this in the first place?

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• Hosting media on separate server than web server

  - by user18832

  Basically I have a website hosted by a web hosting company which I have limited access to (ftp upload etc). I have a home server which I use to record and store audio files. Is there an elegant way or best practice to host a page on the webserver which links to the audio files? I'm considering hosting a page on the home server and redirecting to that from the web server, or setting up something like rsync to push the audio files to the web server - I'm just not certain which solution would be best.

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