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  • Which audio playback technology do I need for this?

    - by mystify
    I have trouble choosing the right audio playback technology. There's a ton of technologies to use on the iPhone, it's so confusing. What I need to do is this: start playing short sounds ranging between 0.1 and 2 seconds high quality playback, no crackle (I heard some of the iPhone audio playback technologies do a crackle sound on start or end, which is bad!) ability to start playback of a sound, while there's already another one playing right now (two, three or more sounds at the same time) What would you suggest here, and why? Thanks :-)

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  • Is it possible to capture audio output and apply effects to it?

    - by Ciaran
    Using .NET and DirectSound I want to be able to take all output sound that is coming from my audio device and apply effects to it. I've had a quick look at the docs on MSDN and there doesn't seem to be any explanation as to how to do something like this. I've read elsewhere that you'd be better off writing a driver to sit in front of your real audio driver and have that do whatever you want with the sound. Any ideas anyone to push me in the right direction?

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  • Speech recognition webservice that scores the accuracy of one audio clips vs. another?

    - by wgpubs
    Does such a thing exist? Building a Rails based web application where users can upload an audio file of them speaking that then needs to be compared to another audio file for the purposes of determining how similar to voices are. Ideally I'd like to simply get a response that gives me a score of how similar they are in terms of percentage (e.g. 75% similar etc...). Anyone have any ideas? Thanks

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  • How to create live stream audio for web-sites ???

    - by Kathir
    Hi All, We are storing sound from mic to pc via sound forge. We would like to broadcast the sound which comes from the mic to the pc as live streaming audio. Basically a person speaks in a mic, we like to give it as live stream audio. The web-site is hosted on yahoo server. Can you please let me know in what are the ways we can achieve this? Thanks, Kathir

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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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  • 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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  • 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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  • Linking to a chat room via XMPP: URI

    - by Coderer
    I found out how to link directly to a chat room on a Jabber conference server -- it took a bit of digging, and I wound up actually looking at the spec before I was sure I was doing it right. I confirmed here, so I'm pretty sure I've got it. The results, though, are puzzling. If I click a link of the style xmpp:[email protected] I get a new chat session with user "dude" at example.com, as expected. If I tack on a nonsense query (xmpp:[email protected]?foobar), it's ignored, which is what the spec says should happen. However, if I use xmpp:[email protected]?join, as in the link above, nothing happens. I dug a little deeper, and found out that on my (Linux) system, xmpp URIs are handled via purple-url-handler, so I dropped to a terminal and ran it manually. The result was that any xmpp URI ran fine except one that includes a ?join query. The ?join query results in a dbus crash, pointing specifically to line 2356 of dbus-message.c -- a little Googling suggests this probably is dbus's less-than-elegant way of telling me that somebody is using dbus incorrectly. Am I crafting my link correctly? Is this an OS or maybe application issue? Does this work on other platforms / browsers / etc? More importantly, is there any easy way to fix it?

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  • Using Openfire for distributed XMPP-based video-chat

    - by Yitzhak
    I have been tasked with setting up a distributed video-chat system built on XMPP. Currently my setup looks like this: Openfire (XMPP server) + JingleNodes plugin for video chat OpenLDAP (LDAP server) for storing user information and allowing directory queries Kerberos server for authentication and passwords In testing with one set of machines (i.e. only three), everything works as expected: I can log in to Openfire and it looks up the user information in the OpenLDAP database, which in turn authenticates my user with Kerberos. Now, I want to have several clusters, so that there is a cluster on each continent. A typical cluster will probably contain 2-5 servers. Users logging in will be directed to the closest cluster based on geographical location. Something that concerns me particularly is the dynamic maintenance of contact lists. If a user is using a machine in Asia, for example, how would contact lists be updated around the world to reflect the current server he is using? How would that work with LDAP? Specific questions: How do I direct users based on geographical location? What is the best architecture for a cluster? -- would all traffic need to come into a load-balancer on each one, for example? How do I manage the update of contact lists across all these servers? In general, how do I go about setting this up? What are the pitfalls in doing this? I am inexperienced in this area, so any advice and suggestions would be 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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  • Continuously updating chat messages

    - by Daniel
    I'm creating a very simple chat application. It has an ASP.NET web page as front-end and a WCF service as back-end for storing the messages to a database. Everything works great except one thing; when Browser A enters a chat message I want Browser B to see the message as soon as possible (yeah, I know, that's the purpose of a chat). What I've done so far is to setup a trigger within the UpdatePanel, like this: <Triggers> <asp:AsyncPostBackTrigger ControlID="chatTimer" EventName="Tick" /> </Triggers> which uses a timer: <asp:Timer ID="chatTimer" runat="server" OnTick="chatTimer_Tick" Interval="1000" /> Is this the best approach or is there a better, yet simple, way to accomplish updating of messages. One drawback with this solution is that the textbox used to enter chat messages loses focus every time the Tick event runs. Any piece of feedback or advice regarding updating of messages is appreciated. Thank you!

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  • How to open a chat window in sender and receiver side [on hold]

    - by DEEPS
    When i am trying to send a message from sender the chat window is always opening in senders side instead of receiver side.so please give a correct code to display chat box in both side. (HTML 5, JAVASCRIPT,JQUERY). This is client side code: //Send private message function sendPvtMsg(data) { var pvtmsg = data; socket.emit('message',JSON.stringify({msg: 'pvtMsg', data: { from: userName, to: toChat, pvtmsg: data }}),roomId); } socket.on('message',function(data) { var command = JSON.parse(data); var itemName = command.msg; var rec_data = command.data.message; var sender = command.data.name; //Receive message from server if (itemName == "message") { document.getElementById("chat").value += sender + " : " + rec_data + "\n"; } //Receive private message else if (itemName == "pvtMsg") { var to = command.data.to; var from = command.data.from; //To display message to sender and receiver if (userName == to || userName == from) { var pvtmsg = command.data.pvtmsg; document.getElementById("chat").value += from + "( to " + to + ")" + " : " + pvtmsg + "\n"; } } function createChatBox(chatboxtitle,minimizeChatBox) { if ($("#chatbox_"+chatboxtitle).length > 0) { if ($("#chatbox_"+chatboxtitle).css('display') == 'none') { $("#chatbox_"+chatboxtitle).css('display','block'); restructureChatBoxes(); } $("#chatbox_"+chatboxtitle+" .chatboxtextarea").focus(); return; }

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