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  • Combine Multiple Audio Files into a single higher-quality audio File

    - by namenlos
    BACKGROUND My team gave a demo to a large audience - we recorded the audio of the demo in multiple locations in the room (3) the audio was recorded using cheap laptop microphones I was not involved in the recording of the audio or the demo Both audio files suck in some form the first one is of a recording near the speaker - which clearly gets his voice but the the audience is audience is muffled - also this one is slightly noisy The second recording was done in the middle of the audience - it gets the audience questions clearly but actually gets the speaker rather sometimes well and sometimes poorly (not all the speakers spoke loudly enough to be heard) MY QUESTION Is there any techinque or software which can be used to merge these audio files in such a way that the best qualities of each are preserved. I am NOT asking now to simply merge them together in one track - I've already done that in Audacity and it is certainly better - what I am looking for could be considered closer to how HDR images are created - multiple exposures combined into an enhanced new version which is not simply an average of the inputs. NOTE Am not an "Audio" guy - just a normal user

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  • Recording Audio through M-Audio Keystudio

    - by interstar
    Hi, I'm trying to get my M-Audio Keystudio (which has an audio input as well as the keyboard) to record audio to Audacity. I'm in Ubuntu 10.10. When I look at the Sound Preferences I can select "M-Audio RunTime DFU Analog Stereo" as my input device. However, when I try to record in Audacity, Audacity remains frozen. The program seems to be running and recording, but the recording cursor won't advance. If I reset the audio input to the internal sound card, recording works normally. Any ideas what to look for?

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  • AVAudioRecorder - Continue recording to file after user stops recording by leaving the application a

    - by Tegeril
    Can this be done? And if not, how far down towards Core Audio do I need to go (what method of recording should I be using instead)? I've noticed the behavior of AVAudioRecorder is to overwrite a file if it finds one at the path provided when you request that it record again, so I know that's not going to work. I'm also curious about file format restriction with this idea. Can you effectively resume an AAC or IMA4 encoding (the length of the files I want to record make WAV and probably even Apple Lossless prohibitive)? Thanks.

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  • guvcview recording video and audio out of synchronisation in Ubuntu 10.10

    - by SIJAR
    I finally got Guvcview, a great software for Logitech webcam and it does all the stuff that one wants out of it. But I'm not satisfy with the video recording, video and audio out of synchronisation also video seems to be in slow motion. Please help so that I can tweak in and get a good video recording with the webcam. Below is the log of Guvcview ------------------------------------------------------------------------------- guvcview 1.4.1 video_device: /dev/video0 vid_sleep: 0 cap_meth: 1 resolution: 640 x 480 windowsize: 1024 x 715 vert pane: 578 spin behavior: 0 mode: mjpg fps: 1/25 Display Fps: 0 bpp: 0 hwaccel: 1 avi_format: 4 sound: 1 sound Device: 4 sound samp rate: 0 sound Channels: 0 Sound delay: 0 nanosec Sound Format: 85 Pan Step: 2 degrees Tilt Step: 2 degrees Video Filter Flags: 0 image inc: 0 profile(default):/home/sijar/default.gpfl starting portaudio... bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started language catalog= dir:/usr/share/locale type:UTF-8 lang:en_US.utf8 cat:guvcview.mo mjpg: setting format to 1196444237 capture method = 1 video device: /dev/video0 libv4lconvert: warning more framesizes then I can handle! libv4lconvert: warning more framesizes then I can handle! /dev/video0 - device 1 libv4lconvert: warning more framesizes then I can handle! libv4lconvert: warning more framesizes then I can handle! Init. UVC Camera (046d:0825) (location: usb-0000:00:1d.7-5) { pixelformat = 'YUYV', description = 'YUV 4:2:2 (YUYV)' } { discrete: width = 640, height = 480 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 160, height = 120 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 176, height = 144 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 320, height = 176 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 320, height = 240 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 352, height = 288 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 432, height = 240 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 544, height = 288 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 640, height = 360 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, ... repeats a couple of times ... vid:046d pid:0825 driver:uvcvideo Adding control for Pan (relative) UVCIOC_CTRL_ADD - Error: Operation not permitted checking format: 1196444237 VIDIOC_G_COMP:: Invalid argument compression control not supported fps is set to 1/25 drawing controls control[0]: 0x980900 Brightness, 0:255:1, default 128 control[0]: 0x980901 Contrast, 0:255:1, default 32 control[0]: 0x980902 Saturation, 0:255:1, default 32 control[0]: 0x98090c White Balance Temperature, Auto, 0:1:1, default 1 control[0]: 0x980913 Gain, 0:255:1, default 0 control[0]: 0x980918 Power Line Frequency, 0:2:1, default 2 control[0]: 0x98091a White Balance Temperature, 0:10000:10, default 4000 control[0]: 0x98091b Sharpness, 0:255:1, default 24 control[0]: 0x98091c Backlight Compensation, 0:1:1, default 1 control[0]: 0x9a0901 Exposure, Auto, 0:3:1, default 3 control[0]: 0x9a0902 Exposure (Absolute), 1:10000:1, default 166 control[0]: 0x9a0903 Exposure, Auto Priority, 0:1:1, default 0 resolutions of format(2) = 19 frame rates of 1º resolution=6 Def. Res: 0 numb. fps:6 --------------------------------------- device #0 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 (hw:0,0) Host API = ALSA Max inputs = 2, Max outputs = 2 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #1 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - MIC ADC (hw:0,1) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #2 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - MIC2 ADC (hw:0,2) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #3 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - ADC2 (hw:0,3) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #4 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - IEC958 (hw:0,4) Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #5 Name = USB Device 0x46d:0x825: USB Audio (hw:1,0) Host API = ALSA Max inputs = 1, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #6 Name = front Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.012 Def. high input latency = -1.000 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #7 Name = iec958 Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #8 Name = spdif Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #9 Name = pulse Host API = ALSA Max inputs = 32, Max outputs = 32 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #10 Name = dmix Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.043 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #11 [ Default Input, Default Output ] Name = default Host API = ALSA Max inputs = 32, Max outputs = 32 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 ---------------------------------------------- SampleRate:0 Channels:0 Video driver: x11 A window manager is available VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_CTRL for user class controls control(0x0098091a) "White Balance Temperature" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_EXT_CTRLS on single controls for class: 0x009a0000 control(0x009a0902) "Exposure (Absolute)" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_CTRL for user class controls control(0x0098091a) "White Balance Temperature" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_EXT_CTRLS on single controls for class: 0x009a0000 control(0x009a0902) "Exposure (Absolute)" failed to set (error -1) Cap Video toggled: 1 (/home/sijar/Videos/Webcam) 25371756K bytes free on a total of 39908968K (used: 36 %) treshold=51200K using audio codec: 0x0055 Audio frame size is 1152 samples for selected codec IO thread started...OK [libx264 @ 0x8cbd8b0]using cpu capabilities: MMX2 SSE2 Cache64 [libx264 @ 0x8cbd8b0]profile Baseline, level 3.0 [libx264 @ 0x8cbd8b0]non-strictly-monotonic PTS shift sound by -9 ms shift sound by -9 ms shift sound by -9 ms AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... AUDIO: droping audio data (/home/sijar/Videos/Webcam) 25371748K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... Cap Video toggled: 0 Shuting Down IO Thread AUDIO: droping audio data stop= 4426644744000 start=4416533023000 VIDEO: 146 frames in 10111.000000 ms = 14.439719 fps Stoping audio stream Closing audio stream... close avi Last message repeated 145 times [libx264 @ 0x8cbd8b0]frame I:2 Avg QP:14.10 size: 24492 [libx264 @ 0x8cbd8b0]frame P:103 Avg QP:16.06 size: 20715 [libx264 @ 0x8cbd8b0]mb I I16..4: 48.4% 0.0% 51.6% [libx264 @ 0x8cbd8b0]mb P I16..4: 57.5% 0.0% 0.0% P16..4: 40.2% 0.0% 0.0% 0.0% 0.0% skip: 2.3% [libx264 @ 0x8cbd8b0]final ratefactor: 62.05 [libx264 @ 0x8cbd8b0]coded y,uvDC,uvAC intra: 79.7% 92.2% 68.4% inter: 62.4% 87.5% 48.0% [libx264 @ 0x8cbd8b0]i16 v,h,dc,p: 23% 17% 41% 19% [libx264 @ 0x8cbd8b0]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 30% 24% 26% 2% 5% 3% 3% 3% 4% [libx264 @ 0x8cbd8b0]i8c dc,h,v,p: 53% 20% 23% 4% [libx264 @ 0x8cbd8b0]ref P L0: 63.0% 37.0% [libx264 @ 0x8cbd8b0]kb/s:-0.00 total frames encoded: 0 total audio frames encoded: 0 IO thread finished...OK IO Thread finished enabling controls Cap Video toggled: 1 (/home/sijar/Videos/Webcam) 25379744K bytes free on a total of 39908968K (used: 36 %) treshold=51200K using audio codec: 0x0055 Audio frame size is 1152 samples for selected codec IO thread started...OK [libx264 @ 0x8cfba20]using cpu capabilities: MMX2 SSE2 Cache64 [libx264 @ 0x8cfba20]profile Baseline, level 3.0 [libx264 @ 0x8cfba20]non-strictly-monotonic PTS shift sound by -236 ms shift sound by -236 ms shift sound by -236 ms (/home/sijar/Videos/Webcam) 25377044K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25373408K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... (/home/sijar/Videos/Webcam) 25370696K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... (/home/sijar/Videos/Webcam) 25367680K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25364052K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25360312K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25356628K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25352908K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25349316K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25345552K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25341828K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25338092K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25334412K bytes free on a total of 39908968K (used: 36 %) treshold=51200K Cap Video toggled: 0 Shuting Down IO Thread stop= 4708817235000 start=4578624714000 VIDEO: 1604 frames in 130192.000000 ms = 12.320265 fps Stoping audio stream Closing audio stream... close avi Last message repeated 1603 times [libx264 @ 0x8cfba20]frame I:16 Avg QP:14.78 size: 42627 [libx264 @ 0x8cfba20]frame P:1547 Avg QP:16.44 size: 28599 [libx264 @ 0x8cfba20]mb I I16..4: 21.6% 0.0% 78.4% [libx264 @ 0x8cfba20]mb P I16..4: 28.1% 0.0% 0.0% P16..4: 70.5% 0.0% 0.0% 0.0% 0.0% skip: 1.4% [libx264 @ 0x8cfba20]final ratefactor: 88.17 [libx264 @ 0x8cfba20]coded y,uvDC,uvAC intra: 74.4% 95.8% 83.2% inter: 75.2% 94.6% 69.2% [libx264 @ 0x8cfba20]i16 v,h,dc,p: 27% 17% 40% 16% [libx264 @ 0x8cfba20]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 25% 21% 3% 6% 4% 5% 4% 7% [libx264 @ 0x8cfba20]i8c dc,h,v,p: 61% 18% 18% 4% [libx264 @ 0x8cfba20]ref P L0: 64.0% 36.0% [libx264 @ 0x8cfba20]kb/s:-0.00 total frames encoded: 0 total audio frames encoded: 0 IO thread finished...OK IO Thread finished enabling controls Shuting Down Thread Thread terminated... cleaning Thread allocations: 100% SDL Quit Video Thread finished write /home/sijar/.guvcviewrc OK free audio mutex closed v4l2 strutures free controls free controls - vidState cleaned allocations - 100% Closing portaudio ...OK Closing GTK... OK

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  • How to record / capture audio with RecordControl on Java ME, SE K770i

    - by tomaszs
    I want to record sound on my Java ME App on K770i. So I used this: http://java.sun.com/javame/reference/apis/jsr135/javax/microedition/media/control/RecordControl.html example of RecordControl in my code. It goes like this: import java.util.Vector; import javax.microedition.lcdui.Choice; import javax.microedition.lcdui.Command; import javax.microedition.lcdui.CommandListener; import javax.microedition.lcdui.Display; import javax.microedition.lcdui.Displayable; import javax.microedition.lcdui.List; import javax.microedition.media.Manager; import javax.microedition.media.MediaException; import javax.microedition.midlet.MIDlet; import java.io.*; import javax.microedition.lcdui.*; import javax.microedition.media.*; import javax.microedition.media.control.*; import javax.microedition.midlet.*; import javax.microedition.rms.*; (...) try { // Create a Player that captures live audio. Player p = Manager.createPlayer("capture://audio"); p.realize(); // Get the RecordControl, set the record stream, // start the Player and record for 5 seconds. RecordControl rc = (RecordControl)p.getControl("RecordControl"); ByteArrayOutputStream output = new ByteArrayOutputStream(); rc.setRecordStream(output); rc.startRecord(); p.start(); Thread.currentThread().sleep(5000); rc.commit(); p.close(); } catch (IOException ioe) { } catch (MediaException me) { } catch (InterruptedException ie) { } But unfortunately when I try to build it, it tells me: *** Creating directories *** *** Compiling source files *** ..\src\example\audiodemo\AudioPlayer.java:121: cannot find symbol symbol : class RecordControl location: class example.audiodemo.AudioPlayer RecordControl rc = (RecordControl)p.getControl("RecordControl"); ^ ..\src\example\audiodemo\AudioPlayer.java:121: cannot find symbol symbol : class RecordControl location: class example.audiodemo.AudioPlayer RecordControl rc = (RecordControl)p.getControl("RecordControl"); ^ 2 errors So my question is: why there is no RecordControl class if in documentations it is written this class should be there. Or is there other method to record / capture audio from microfone in Java ME of Sony Ericsson? How do you record sound?

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  • Video and Audio Drift

    - by Cenoc
    Hey everyone, I was wondering, how much does recorded audio and video drift from their actual recording time usually? I'm recording both separately (into unsigned 8 bit PCM (44100 Hz) and raw image data files) and I was wondering how much I can expect each to drift. Thanks in advance!

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  • No audio with headphones, but audio works with integrated speakers

    - by Pedro
    My speakers work correctly, but when I plug in my headphones, they don't work. I am running Ubuntu 10.04. My audio card is Realtek ALC259 My laptop model is a HP G62t a10em In another thread someone fixed a similar issue (headphones work, speakers not) folowing this: sudo vi /etc/modprobe.d/alsa-base.conf (or some other editor instead of Vi) Append the following at the end of the file: alias snd-card-0 snd-hda-intel options snd-hda-intel model=auto Reboot but it doesnt work for me. Before making and changes to alsa, this was the output: alsamixer gives me this: Things I did: followed this HowTo but now no hardware seems to be present (before, there were 2 items listed): Now, alsamixer gives me this: alsamixer: relocation error: alsamixer: symbol snd_mixer_get_hctl, version ALSA_0.9 not defined in file libasound.so.2 with link time reference I guess there was and error in the alsa-driver install so I began reinstalling it. cd alsa-driver* //this works fine// sudo ./configure --with-cards=hda-intel --with-kernel=/usr/src/linux-headers-$(uname -r) //this works fine// sudo make //this doesn't work. see ouput error below// sudo make install Final lines of sudo make: hpetimer.c: In function ‘snd_hpet_open’: hpetimer.c:41: warning: implicit declaration of function ‘hpet_register’ hpetimer.c:44: warning: implicit declaration of function ‘hpet_control’ hpetimer.c:44: error: expected expression before ‘unsigned’ hpetimer.c: In function ‘snd_hpet_close’: hpetimer.c:51: warning: implicit declaration of function ‘hpet_unregister’ hpetimer.c:52: error: invalid use of undefined type ‘struct hpet_task’ hpetimer.c: In function ‘hpetimer_init’: hpetimer.c:88: error: ‘EINVAL’ undeclared (first use in this function) hpetimer.c:99: error: invalid use of undefined type ‘struct hpet_task’ hpetimer.c:100: error: invalid use of undefined type ‘struct hpet_task’ hpetimer.c: At top level: hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: error: extra brace group at end of initializer hpetimer.c:121: error: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) make[1]: *** [hpetimer.o] Error 1 make[1]: Leaving directory `/usr/src/alsa/alsa-driver-1.0.9/acore' make: *** [compile] Error 1 And then sudo make install gives me: rm -f /lib/modules/0.0.0/misc/snd*.*o /lib/modules/0.0.0/misc/persist.o /lib/modules/0.0.0/misc/isapnp.o make[1]: Entering directory `/usr/src/alsa/alsa-driver-1.0.9/acore' mkdir -p /lib/modules/0.0.0/misc cp snd-hpet.o snd-page-alloc.o snd-pcm.o snd-timer.o snd.o /lib/modules/0.0.0/misc cp: cannot stat `snd-hpet.o': No such file or directory cp: cannot stat `snd-page-alloc.o': No such file or directory cp: cannot stat `snd-pcm.o': No such file or directory cp: cannot stat `snd-timer.o': No such file or directory cp: cannot stat `snd.o': No such file or directory make[1]: *** [_modinst__] Error 1 make[1]: Leaving directory `/usr/src/alsa/alsa-driver-1.0.9/acore' make: *** [install-modules] Error 1 [SOLUTION] After screwing it all up, someone mentioned why not trying using the packages in Synaptic - so I did. I have reinstalled the following packages and rebooter: -alsa-hda-realtek-ignore-sku-dkms -alsa-modules-2.6.32-25-generic -alsa-source -alsa-utils -linux-backports-modules-alsa-lucid-generic -linux-backports-modules-alsa-lucid-generic-pae -linux-sound-base -(i think i listed them all) After rebooting, the audio worked, both in speakers and headphones. I have no idea which is the package that made my audio work, but it certainly was one of them. [/SOLUTION]

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  • Recording Topics manually and automatically

    - by maria.cozzolino(at)oracle.com
    When you are recording UPK topics, the default mode for recording is manual recording, where you tell the system when to record each screen shot. This mode allows you to take the exact screen shot you need. However, it does get a bit tedious when you are recording long topics, especially if you forget to take a few screen shots. In UPK 3.5, a new version of recording was introduced - Automatic Recording. It was designed to simplify the recording process by automatically capturing screen shots as you perform your transaction. If you haven't experimented with Automatic Recording, I'd recommend you give it a try - it might make your recording life easier. If you are recording with sound, you can also narrate your topic while recording it. To turn on Automatic Recording: 1. In Tools/Options, there are two recorder tabs. The first tab, under content defaults, includes settings that you may want to share between developers, like whether keyboard shortcuts are automatically captured. 2. The second tab is the one that contains the personal preferences, like screen shot capture key and whether to record automatically or manually. On this tab, choose the option for Automatic Recording. 3. Save the settings. Note that this setting will NOT impact content defaults; this is for your user only. When you launch the recorder, you will notice a slightly different message with guidance on how to start and stop automatic recording. Once you start recording, the recorder window is hidden until the end of the recording session to allow you to capture your transaction. In the task tray, there is a series of icons that let you know that you are capturing content. You can pause the recording, as well as set and view your sound levels if you are using sound. A camera appears during each screen capture to help you know when the system is capturing a screen shot, and a context indicator appears to show the recognition. With automatic recording, you can let the system capture the necessary screen shots. It may provide a more natural recording experience, and is probably easier for the untrained developer. On the other hand, you have a bit more control with manual recording on which screen shot appears, but it also means you have to remember to capture the screen shot. :) We'd be interested in hearing which type of recording you do, and any rationale on why you made that choice. Please comment and let us know. --Maria Cozzolino, Manager of UPK Software Requirements and UI Design

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  • Tools to automate recording streaming radio

    - by Stan
    Is there any tool that can automate recording online streaming radio? I've been use totalrecorder which it has below upside: 1. Handy scheduler. 2. Support create recording templates, so I can customize some high/low quality recording. The downside are it requires to open the streaming radio in browser and can't have another sound source. It's recording what comes out from the speaker. What I am looking for is given a online radio url, and the tool can record the audio stream. No matter if I am playing any other music or not. Thanks.

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  • Tools to automate recording streaming radio

    - by Stan
    Are there any tools that can automate recording online streaming radio? I've been using Total Recorder which has the following useful features: Handy scheduler Supports creating recording templates, so I can customize some high/low quality recording Unfortunately it requires opening the streaming radio in a browser and can't have another sound source at the same time; it's recording what comes out from the speaker. What I am looking for is given an online radio URL, the tool should be able to record the audio stream, no matter if I am playing any other music or not. Does such a tool exist?

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  • Extract Audio from a Video File with Pazera Free Audio Extractor

    - by DigitalGeekery
    Have you ever wanted to extract some or all of the audio from a video file?  Today we’ll take a look at Pazera Free Audio Extractor. A simple audio converter that specializes in that very task. Download the Pazera Free Audio Extractor. (See download link below) You’ll need to unzip the download folder, but there is no need to install the application. Simply double-click the AudioExtractor.exe file to run the application. To add your video files to the queue to be converted, click on the Add files  button at the top left. You can add multiple files to the queue and convert them all at one time. Browse for your video file, and click Open.   Your video will be added to the Queue for processing.   Under Output directory you can choose to output to a folder of your choice. Outputting to the same folder as the input folder is the default.   Pazera Free Audio Extractor includes pre-configured profiles that will simplify the process of choosing conversion settings. To load a profile, choose one from the Profile drop down list and then click the Load button. You can choose to output to MP3, AAC, AC3, WMA, FLAC, OGG or WAV file format.   You will see the profile update the Audio settings in the panels at the lower left of the application. If you wish, you may also select your own custom settings. Advanced Settings The Advanced settings can be used if you want to extract only a portion of the the audio, such as a clip of dialog or a song from a movie. To extract only a portion of the audio, set the start time by selecting the Start time offset check box, then entering the time in the video clip where the audio begins. To set the end time, begin by selecting the Duration check box. Now, you can either select the Duration radio button and enter the amount of time for which you would like to extract the audio, or you can select the End time offset radio button and enter the time in the video clip where the audio ends. When you are ready to convert, click the CONVERT button on the menu at the top of the screen.   An output box will open and display the conversion progress. When finished, click Close.   Now you are ready to enjoy your audio clip. Pazera Free Audio Extractor is a basic audio tool that is easy enough for everyone to use. It runs on Windows only and supports most common video formats including AVI, FLV, MP4, MPG, MOV, 3GP, and WMV. Download Free Audio Extractor 1.3 Similar Articles Productive Geek Tips Eufony Free Audio Player – Resource Gentle Audio PlayerConvert .3GP and .3G2 Files to AVI / MPEG for FreeTurn Off Auto-Play of Audio and Video CDs and DVDs in UbuntuHow to Make/Edit a movie with Windows Movie Maker in Windows VistaEasily Change Audio File Formats with XRECODE TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 Use Printflush to Solve Printing Problems Icelandic Volcano Webcams Open Multiple Links At One Go NachoFoto Searches Images in Real-time Office 2010 Product Guides Google Maps Place marks – Pizza, Guns or Strip Clubs

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  • Convert a cassette tape recording to digital format

    - by Electric Automation
    Has anyone been successful with transferring audio cassette tape recordings to a digital format? I would like to preserve old cassette tape recordings of my grandparents to some digital format: MP3, WAV, etc... The quality of the tapes are mediocre. I think I can handle the quality restoration but getting the audio from tape to digital is my question. Below is a list of the hardware that I can work with: Cassette Deck: I have a Technics stereo cassette deck model RS-B12. It has separate left and right IN and OUT RCA type jacks on the back. In the front it has a headphone phono jack, plus left and right mic input phono jacks. On the computer side: -I have a Windows Vista PC with no additional software other than what came with the machine from Costco. No sound editing software that I can see. There is no sound card on the PC. On the front panel there is a mini-phono mic input jack and there are several different types of in/out mini-phono jacks on the back. In addition, USB and Firewire. I also have access to a new (2009) iMac with a mini-phono input jack for a powered mic or other audio source and GarageBand that has come with the computer. In addition, USB and Firewire. What are my options for getting these cassette recordings into a digital format? Whats the best format? What sort of wires would I need and will I want to utilize the USB or Firewire or can I simply use the audio inputs on the PC (or Mac) to receive the audio stream?

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  • Convert a cassette tape recording to digital format

    - by Optimal Solutions
    Has anyone been successful with transferring audio cassette tape recordings to a digital format? I would like to preserve old cassette tape recordings of my grandparents to some digital format: MP3, WAV, etc... The quality of the tapes are mediocre. I think I can handle the quality restoration but getting the audio from tape to digital is my question. Below is a list of the hardware that I can work with: Cassette Deck: I have a Technics stereo cassette deck model RS-B12. It has separate left and right IN and OUT RCA type jacks on the back. In the front it has a headphone phono jack, plus left and right mic input phono jacks. On the computer side: -I have a Windows Vista PC with no additional software other than what came with the machine from Costco. No sound editing software that I can see. There is no sound card on the PC. On the front panel there is a mini-phono mic input jack and there are several different types of in/out mini-phono jacks on the back. In addition, USB and Firewire. I also have access to a new (2009) iMac with a mini-phono input jack for a powered mic or other audio source and GarageBand that has come with the computer. In addition, USB and Firewire. What are my options for getting these cassette recordings into a digital format? Whats the best format? What sort of wires would I need and will I want to utilize the USB or Firewire or can I simply use the audio inputs on the PC (or Mac) to receive the audio stream?

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  • usb audio recording: clicking noice

    - by snot
    I'm having problems recording audio on windows using usb audio devices. The recorded material will have some kind of jitter/clicking noice. I tested three different usb microphones and three different recording applications. Every combination will have the same problem. Does some one know a solution for this problem?

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  • Looking for an application to record audio and video on a linux "embedded" device

    - by Luke404
    I am working with a linux x86 device with limited CPU resources (as a prototype we just use a pentium-m netbook). We'd like to record video from one V4L2 device (we'll probably end up using just USB Video Class devices like all modern webcams) and one audio stream from an ALSA source. The thing will not have screen and keyboard, and obviously no X11 environment. Goals are: do as little work as possible to cope with little cpu resources - for example I'd like to record video in the native MJPEG I get out of the UVC devices encoding audio to MPEG3 Layer-2 (aka mp2) is ok since it let us save a lot of space (compared to raw pcm samples) and does use little cpu power I don't mind loosing some video frames here and there (UVC devices do that) as long as I can get audio and video streams syncronized not require user input to start the thing (a python script takes care of initialization, startup, shutdown, etc...) be able to open the resulting files for postprocessing without too much effort (ie, if mplayer or vlc can play it, it's fine) So far the only app I found that could be started from command line and record V4L2 video + ALSA audio is mencoder but I'm having some difficulties with it. It should be able to do that but I cannot record audio and video together - just one of the two. And if I use two different processes to record to two different files I have no means to get them in sync (audio is more or less always correct, but video framerate will vary over time and it seems to lack timestamps to correctly play it back to the correct time). Long story short, how do you record an unconverted MJPEG stream (from an UVC device) and an audio stream (from an ALSA device, possibly encoding to any standard format) using a command line tool, to a single file (MPEG or any other container), keeping audio and video in sync?

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  • How to create a virtual audio device and stream audio input with it

    - by Steven Rosato
    Is it possible to create another audio device and redirect only wanted input streams to it? Here's my concrete problem: I am broadcasting a game via XFire and it uses the Windows audio device to capture any audio I receive. As I am broadcasting, other users who watch the video stream are communicating with me over Skype, and they hear themselves back within the video stream and it is entirely logical since I am broadcasting the audio I hear. What I want to do is create another audio device within Windows and redirect (pipe) ONLY the audio input from that game and not the input reveived from Skype. I would then tell XFire to use that newly created "virtual" audio device to broadcast and therefore my partners won't hear themselves back. Is there any software that can do that or can it be achieved natively with Windows? (I am under Windows 7).

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  • How to create a virtual audio device and stream audio input with it

    - by Steven Rosato
    Is it possible to create another audio device and redirect only wanted input streams to it? Here's my concrete problem: I am broadcasting a game via XFire and it uses the Windows audio device to capture any audio I receive. As I am broadcasting, other users who watch the video stream are communicating with me over Skype, and they hear themselves back within the video stream and it is entirely logical since I am broadcasting the audio I hear. What I want to do is create another audio device within Windows and redirect (pipe) ONLY the audio input from that game and not the input reveived from Skype. I would then tell XFire to use that newly created "virtual" audio device to broadcast and therefore my partners won't hear themselves back. Is there any software that can do that or can it be achieved natively with Windows? (I am under Windows 7).

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  • development and music recording machine suggestions?

    - by dean nolan
    I wasn't sure if this belongs on SuperUser so flag if so. I am looking to build, primarily, a windows development machine that is also good for recording using Cubase. I know I should use seperate machines but I'm on a budget this time of year. I also havn't kept up with hardware for quite a few years. Basically I know I want quad core, multiple monitor support (no gaming requirements). A lot of RAM, very quiet case and super fast HDD (SSD OR 10,000RPM)for compiling and latency. I will store libraries and other data on a USB drive. Sound card is not needed as I will be using an audio interface, all other music recording equipment is taken care of also. I could do with some decent monitor recomendations also. All suggestions welcome, thanks.

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  • what is the best way to stream a audio file to website users/listners

    - by Naveen Chamikara Gamage
    I'm developing a music site which will stream audio files stored in a server to users, audio files will be played through flash player placed in a webpage.. As I heard I need to use a streaming media server for streaming audio files ( like 2mb to 3mb in size).. Do I need to use one? I found some streaming media server softwares like http://www.icecast.org - but as in their documentation, It is used for streaming radio stations and live streaming purposes, but I just need to stream audio files faster and in low size (low bandwidth) with good quality.. I heard I need to encode the audio files first and then send them to listeners and in their end audio files need to be decoded again. Is that true? How can I do that? if I need to use a special web server, where should I host my files? Any good hosting providers? if I host audio files in a normal web server, they will use HTTP or TCP to deliver my audio files to users/ listners but I found that HTTP and TCP are not good ways to use for multi media purposes like streaming audio and video files, and they are used for delivering HTML and stuff. I found I should use RSTP or UDP for streaming audio files.. What should I use? I know that .MP3 files has much better quality than the other formats but it also gives huge size to the audio files.. which format should I use for audio files? Most of the best quality audio files are more than 7mb so I'm planning to convert them my self using a software so I could get low size files with some level of good quality. If I'm converting my audio files what is the good BITRATE I should use for my files? Any known best softwares for converting audio files while keeping quality in a good level? Note** - I know that I will not need complex requirements at the beginning of the site but I wanted to what are the best ways like they are using for soundcloud.com

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  • Digital audio input on Macbook?

    - by Ken
    I have: a Macbook (not Pro), don't know the exact model but it's a Core 2 Duo 2.0GHz and probably what Wikipedia calls the "Late 2006" or "Mid 2007" model a DVD player, region-free, that has "Coax and TosLink optical digital audio outputs" I want to make an MP3 of the audio track of some DVDs (for learning a new language), and I can't use the Macbook's built-in DVD drive because it's a different region (ugh!). I'm sure I can connect the DVD player to the Macbook with an analog audio cable. However, if it's possible I'd prefer to keep the signal digital. I'm not even positive if my old Macbook has digital audio in, and if so what I need to connect to it. (I've done plenty of home audio geeking, but always in analog!) Will a "Toslink cable" plus a "Toslink Female to Mini-Plug Male Adapter" (found on Amazon) let me connect my things together? It looks like the pieces will fit but I'd like to hear someone confidently knowledgeable on the matter before I buy something. Thanks!

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  • Remove noise from a recording

    - by essamSALAH
    I used to record important technical meetings and demos using Camtasia Studio, using a Mic to capture the speaker voice. Sometimes we invite attendees by asking them to call us in the meeting, and they would call on a cell phone then we switch it to loudspeaker so we can hear and talk to them and also record the conversation on Camtasia. The problem I am having now is that playing back those recording produces the regular noise that results from the microphone being close to the mobile phone (the kind of noise you hear when your mobile phone rings and it is near a speaker). Any advice on removing this noise?

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  • Two audio streams - headphones and speakers

    - by Sylvester
    What I want (this is probably hard for most to answer, as this is a very unique setup) is to have two different streams (this means audio splitter is not an option, as it will still only be one stream) of audio - one through the headphones and one through the main speakers. I can do the audio rerouting using virtual audio cables, however the problem is this: i cannot get both headphones AND speakers to play even just one stream, let alone two seperate ones. using "split front and back audio into seperate streams is not an option, as the actual MB F_PANEL is faulty (nothing to do with the case front panel, just so you know. that works fine) So, first things first. I need it to recognise the headphones as a seperate audio device so that Virtual Audio Cables will detect it and allow me to route the necessary audio to the headphones only. I also need to be able have sound play through speakers and headphones together what i want to achieve overall, is this: have the ENTIRE computer's sounds picked up by VAC, and stream them to Line1. then have Line1 stream to the headphones. that way whatever's being streamed is heard through the headphones, while the entire system sounds (including those not streamed) are played through speakers.

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  • Why are the analoge stereo input and output of my M-Audio 24/96 soundcard not available to me in Ubu

    - by user37968
    I have installed Lucid on an old Mac PowerPC G4 desktop with a M-Audio Audiophile 24/96 soundcard. The only inputs and outputs I can select in the audio preferences are digital ones for the digital input and output. "lspci -v" shows the card as so: 0001:10:13.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) Subsystem: VIA Technologies Inc. Device d634 Flags: bus master, medium devsel, latency 16, IRQ 53 I/O ports at 0440 [size=32] I/O ports at 04b0 [size=16] I/O ports at 04a0 [size=16] I/O ports at 0400 [size=64] Capabilities: <access denied> Kernel driver in use: ICE1712 Kernel modules: snd-ice1712 "cat /proc/asound/cards" as so: 0 [Tumbler ]: PMac Tumbler - PowerMac Tumbler PowerMac Tumbler (Dev 21) Sub-frame 0 1 [M2496 ]: ICE1712 - M Audio Audiophile 24/96 M Audio Audiophile 24/96 at 0x440, irq 53 "aplay -L" shows these as listed: pulse Playback/recording through the PulseAudio sound server front:CARD=Tumbler,DEV=0 PowerMac Tumbler, PowerMac Tumbler Front speakers front:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi Front speakers surround40:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.0 Surround output to Front and Rear speakers surround41:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.1 Surround output to Front, Center, Rear and Subwoofer speakers iec958:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi IEC958 (S/PDIF) Digital Audio Output I believe it is a problem with detecting the analogue input/output. Sometimes I can get sound from the device but it is a sheet of white noise and tinkering makes it go away again I don't know if that is a separate problem or if it is linked to not being able to see the analogue input/outputs in the sound preferences. Any help would be greatly appreciated As for the white noise I have installed the Envy24 control panel and spend lots of time playing with the settings but when I can get the white noise I can never get it to an quality where I can actually hear what is being played. The internal speaker plays audio fine and plugging in a NI Audio 4DJ via usb also plays sound, although with some static but I believe that is due to an underpowered usb2 pci expansion card not being able to get enough electricity to the device. Alternatively I have seen other people with problems with this device so it may be a bug in the driver but that is another matter. I would like to get the M-Audio card working so I can begin to enjoy my music once again. As a note, I do not currently have any audio equipment capable of sending or receiving audio via the digital inputs and output so I can not check if they are working. The sound preferences show a wide range of digital in and out options with various surround sound options but no analogue ins and outs.

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