Capturing Sound with microphone
I need code that will capture sound through a microphone, please help. Thank You.
and can someone possibly explain this code for please:
CaptureDeviceInfo di = null;
Vector deviceList = CaptureDeviceManager.getDeviceList(
new AudioFormat( "linear", 44100, 16, 2 ) );
if ( deviceList.size() > 0 )
di = (CaptureDeviceInfo)deviceList.firstElement();
Processor p = Manager.createRealizedProcessor(di.getLocator());
DataSource source = p.getDataOutput();
/*The source object returned from the Processor can then be turned into a Player
*object by calling Manager.createPlayer(). To capture it to an audio file
instead, a DataSink can take the data instead:/
DataSink sink;
MediaLocator dest = new MediaLocator("file://output.wav");
try {
sink = Manager.createDataSink(source, dest);
sink.open();
sink.start();
catch (Exception e)
}
Similar Messages
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Problem capturing sound with picture on Premiere 6.0
I've been using this system successfully for 10 years and I promise it's the last project with it, I'm going HD with CS4 next month. But...
Using Premiere 6.0. MAC G4 Dual 1000 processors, lots of RAM, 4 hard drives. For some reason when I capture a clip from tape through my Sony DS11 deck, the picture comes in fine, but without sound. I've checked all the audio settings, they're all right. I tried changing all the settings one at a time, it doen't help. I tried using the camera to capture from instead of the deck, no help. I tried a tape shot with a different camera, nope. I tried previous tapes that I was able to capture picture and sound from, now no sound. Oddly, occasionally it will work, maybe one in 20 tries. I'm finishing an important project and only need to lay in the narration, which I record with the camera using a good mike, but of course I can't capture sound. I'm tempted to reload Premiere onto the computer, but I'm afraid it won't hook up with the hundreds of clips already captured. Any suggestions?FW Capture, whether from your deck, or from your camera, should be just a byte-for-byte data transfer, using the Capture settings that you have chosen. You have confirmed that the settings are 100% correct, and your problem is with both the deck and the camera. That leaves the FW cable, and the connection on your Mac. Now, with either, I would expect that both the video and audio would be messed up, and not just one stream. Still, those would be the next things that I would check.
Since my normal go-to utility, G-SPOT is not ported for the Mac, I'd use MediaInfo, and run one of the Captured files through it, just to see if there is an Audio stream - just on the off-chance that it's a playback issue. Assume that you are doing the Capture to MOV, so playing in QT Player should be a test too.
Other than that, I don't have any ideas to help.
Good luck,
Hunt -
Capture audio with microphone and save to disk
Hello,
I would like to know if there is any way to record audio from
user's output? Let's say I capture it using its microphone, and I
want to create an audio file with it, or a swf file or whatever
file it is. I need to save that audio for further playback. Can
someone point me in any direction!? I have been surfing on this
topic but I didn't find anything, apparently it can't be done, but
I would love otherwise.
Thank you,
JCAYes that's the way it could work. FMS is designed for
streaming audio, video and data around keeping things synchronised
in real time, but it can save the streams to be played back later
as well.
It would be an expensive solution if that's all you want to
do.... there is an open source 'version' in development... called
red5, but I think you need to know how to program in java, although
I'm not sure.
Or you could look at adobe connect/formerly macromedia breeze
- It prebuilt and is a pay for use service, but does much more than
what you're looking for. I'm pretty sure you can custom logo it
etc. You might want to try the free trial account and check it
out. -
How can capture audio with labview
I desired to make a audio equalizer vi using lab view. Audio will be taken from microphone. How can I capture sound from microphone with labview?
See the example "Sound Card AutoPower Spectrum" This allows you to sample from an input device through your sound card, and take a spectrum. you should be able to bin the spectrum by frequency bands, and show the histogram of frequencies. You probably can even selectively resample and scale the frequencies, mix the signal and output the new sounds but I hav not tried this (I am not an audiophile so this is beyond my scope of expertise) Good luck ,
Paul
Paul Falkenstein
Coleman Technologies Inc.
CLA, CPI, AIA-Vision
Labview 4.0- 2013, RT, Vision, FPGA -
Capturing Sound with Electro magnets
https://www.reddit.com/3epdzn/
https://www.reddit.com/3epf8t/
https://www.reddit.com/3epghp/I've just finished transferring some vinyl to my eMac. I used the Sound Studio application that came with the eMac to do the transfer, and used ClickRepair to get rid of clicks. I haven't burnt the stuff to CD, but I'd probably use iTunes if I did.
I can't comment on the pre-amp thing: I had my turntable connected to a receiver and cabled from Tape Rec to the eMac input jack.
If you go to Audio Midi Setup in your Utilities folder and set it up as pictured here, you'll be able to monitor the sound through the eMac speakers or the headphones, if you want.
eMac Combo 1GHz/1GB/60GB Mac OS X (10.4.7) CanoScan D646U, Canon iP1600, CrossFire 120GB, DSL, D-Link DI-624 -
Can I use a Bluetooth headphone/microphone for capture sound instead of normal iPhone mic when shooting video?
The sound when I record video is not so good, so I'm trying to use the Bluetooth headphone/microphone to capture a person voice but not know which setting to use.Hi Robert
Thanks for answering. I didn't know where to post other than the bluetooth forum which is relatively quiet and had no answers forthcoming.
Yes the voice isn't reaching the amp although it is registering on the input bar in the Sound System Prefs, just won't "play through" as it were. I can't see how it could be done I was just hoping someone might see something I am missing.
It seems a shame that you can't do it as iTunes on an old iBook makes a good juke box for DJ'ing and with everything becoming wireless these days it seems sensible that you could cut that one last wire.
Mike -
Problems with microphone sound in iChat
Hello Ralph,
(I already thought I sent this but I didn't see it as a new topic, so I am trying again). I'm new to this discussion thing, it's my first time so I hope I've submitted this correctly.
I have similar, but extended and more complicated problems with "microphone sound in iChat". (Sorry, this email is a bit lenghty becuase I have already done a lot of my homework on trouble shooting this.)
I began by using iChat with a high impedance microphone (from an old 1970'scassette player) PLUGGED DIERCTLY IN THE THE MICROPHONE INPUT on the back of my G4 tower. (I DO NOT have a video camera). My Mom in Canada can hear me okay on it, only if I "eat" the microphone. (My Mom DOES have a video camera).
SO I BOUGHT AN iMic. I also bought a new condenser mic to go with my new iMic. I plugged iMic in and adjusted the sound preferences selection. I switched the iMic to "Mic". It did not work. The old cassette mic did not work at all with iMic either. My Mom could not hear me, and my "VU METER bar levels" on my iChat were not registering. (If I plug the condenser mic directly into the tower, Mom can hear me only if I "eat" the condenser microphone).
Next, with iMic, I wanted to try my Shure 58 microphone instead. My friend (who is an audio tech) made me a 1/8" to XLR (sheilded) cable. I made sure I had adjusted the sound preferences and switched the iMic to "Mic". The Shure 58 did not work with iMic. It also did not work plugging directly into the tower.
Next, with iMic, I wanted to try my "Plantronics telephone headset". My friend (the audio tech) then made me a 1/8"male to 1/8"mini feamle (sheilded) cable. My headset did not work with iMic. It also did not work plugging directly into the tower.
My friend, the audio tech, then made me second cable 1/8"male to 1/8"female (sheilded) cable. This cable I had him make 50 feet long so I could move to other areas in my home while audio chatting - so I don't have to be trapped in my office while talking. We ran the cable from iMic to the Old cassette mic and the new condenser mic. They did not work with iMic.
We also tried plugging the new 50ft cable directly into the back of the G4 Tower and it worked with the old cassette mic and the new condenser mic, BUT the volume was reduced and there was a loud baseball game being broadcast through the cable - according to my Mom. I could not hear anything from my end - not my own voice and not the baseball game! IS THERE A WAY FOR ME TO MONITOR (actually hear) THE SOUND THAT SHOULD BE COMING THROUGH THE MIC?
I have tried to trouble shoot these issues on line, but can't seem to find the right solutions. In one of your discussions I read that you suggested to select the iChat preference for Video and select microphone. I CANNOT SEE THE FULL WINDOW UNDER iChat Video Preferences!!!! All I can see is the top half of a large empty box that says "There is no camera attached to this computer". (That's because I have no camera, but I DO HAVE a microphone).
I also found some support about adjusting microphone levels using iMovie. I do not have iMovie on my Mac.
Yes, I have already adjusted the AUDIO MIDI SETUP settings.
My audio tech friend and I are both baffled. iMic is useless to me so far - perhaps you can help to get me up and chatting properly? Do I really need iMovie and/or a video camera to fix this???
BY THE WAY: When I plug my external speakers into iMic and select iMic in preferences for output, my speakers and sound is very inferior to the using the speaker terminal on my tower. i Mic is supposed to better???????
Thanking you for your help in advance,
Marlene
(Please could you, would you, email a copy of your answer to me at [email protected])
G4, 450Mhz Mac OS X (10.4.6)
G4, 450Mhz Mac OS X (10.4.6)
G4, 450Mhz Mac OS X (10.4.6)Hi Ralph,
A funny thing just happened...
The iChat prefernce window for "VIDEO" finally opened (!!) all the way, so for the FIRST TIME EVER and I was able to see the microphone setting at the bottom of the window. (Because I have NO CAMERA, the window would never fully open before during the 100 or so times I tried to get it to open before. This must've been a fluke because now it won't fully open again).
While it was open I set the microphone setting to iMic, plugged my condenser mic into iMic and IT WORKED. Then I tried the long (baseball game) extension cord in the iMic (attached to the condenser mic) and IT WORKED TOO - with NO BASEBALL GAME!
I still have to have the condenser mic right up to my mouth, which shouldn't be - but considering I have spent a few weeks trying to troubleshoot this, "eating the mic" is only a minor annoyance at this point.
I just plugged my speakers into iMic and THEY WORK GREAT!
SO ALL MY PROBLEMS WERE BECAUSE: I could not set the microphone under Video in iChat preferences because the VIDEO PREFERENCE WINDOW DOES NOT FULLY OPEN WITHOUT A CAMERA ATTACHED. If it hadn't opened by fluke my iMic would still not be working.
I guess I'm all set now. Unless you have a suggestion why the condenser mic isn't picking up my voice from other than right by my mouth? It should be able to be a foot or two away at least.
Thanks Ralph,
Marlene
(By the way, I never did trash the com.apple.ichat.plist)
G4, 450Mhz Mac OS X (10.4.6) -
No Mic Sound with Pulseaudio, works fine with Alsa.
Have a Acer Aspire One D225E that I've set up with Gnome 3. Everything seems to be working except I'm not getting any sound out of my microphone. I've messed around with the sound settings in the gnome settings app with no success. Using alsamixer, I've made sure the mic channel is unmuted.
I'm able to record sound with Audacity using Alsa and play it back fine. When I switch to pulseaudio however, I get nothing. This wouldn't be a problem, except skype only allows you to use pulseaudio to record if you have that installed.
I've checked the wiki troubleshooting section on Pulseaudio and found the section on Acer Aspires, but solutions in the link led to a dead end. Nothing has worked so far that I've tried.
Here's my sound device:
00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02)
Output of arecord -l:
**** List of CAPTURE Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC272X Analog [ALC272X Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
Mic source from "pactl list sources"
Source #1
State: SUSPENDED
Name: alsa_input.pci-0000_00_1b.0.analog-stereo
Description: Internal Audio Analog Stereo
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 4
Mute: no
Volume: 0: 100% 1: 100%
0: 0.00 dB 1: 0.00 dB
balance 0.00
Base Volume: 10%
-60.00 dB
Monitor of Sink: n/a
Latency: 0 usec, configured 0 usec
Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
Properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC272X Analog"
alsa.id = "ALC272X Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "HDA Intel"
alsa.long_card_name = "HDA Intel at 0x58200000 irq 44"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.name = "N10/ICH 7 Family High Definition Audio Controller"
device.form_factor = "internal"
device.string = "front:0"
device.buffering.buffer_size = "352800"
device.buffering.fragment_size = "176400"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Internal Audio Analog Stereo"
alsa.mixer_name = "Realtek ALC272X"
alsa.components = "HDA:10ec0272,10250002,00100001"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
Ports:
analog-input: Analog Input (priority. 10000)
analog-input-microphone: Analog Microphone (priority. 8700)
Active Port: analog-input-microphone
Formats:
pcm
Hopefully someone with more knowledge of pulseaudio will be able to point my in the right direction on this one.I have exactly the same problem with my Acer Aspire 3820. It's definitely pulseaudio related, since arecord works.
Sound device:
00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05)
Output of pactl list sources:
Sorgente #2
Stato: SUSPENDED
Nome: alsa_input.pci-0000_00_1b.0.analog-stereo
Descrizione: Audio interno Stereo analogico
Driver: module-alsa-card.c
Specifica di campionamento: s16le ch 2 48000 Hz
Mappa dei canali: front-left,front-right
Modulo di appartenenza: 5
Muto: no
Volume: 0: 0% 1: 0%
0: -inf dB 1: -inf dB
bilanciamento 0,00
Volume base: 8%
-66,00 dB
Monitor del sink: N/D
Latenza: 0 microsec, configurata 0 microsec
Flag: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
Proprietà:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC269VB Analog"
alsa.id = "ALC269VB Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "HDA Intel"
alsa.long_card_name = "HDA Intel at 0xf0800000 irq 44"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
device.form_factor = "internal"
device.string = "front:0"
device.buffering.buffer_size = "384000"
device.buffering.fragment_size = "192000"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Stereo analogico"
device.description = "Audio interno Stereo analogico"
alsa.mixer_name = "Realtek ALC269VB"
alsa.components = "HDA:10ec0269,10250364,00100100"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
Porte:
analog-input-microphone-internal: Microfono interno (priority: 8900)
analog-input-microphone: Microfono (priority: 8700, not available)
Porta attiva: analog-input-microphone-internal
Formats:
pcm
I've been struggling for months, but still no answer. None of the volume tweaking tricks work, obviously, and I couldn't find any useful answers in the PA mailing list. -
[SOLVED] No sound with vmware
I'm getting no sound with vmware, not sure why
Here's what I've been trying to test things:
$ cat /dev/urandom | aplay
Playing raw data 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono
Here's some alsa info, if it helps:
$ aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=AudioPCI
Ensoniq AudioPCI, ES1371 DAC2/ADC
Default Audio Device
sysdefault:CARD=AudioPCI
Ensoniq AudioPCI, ES1371 DAC2/ADC
Default Audio Device
front:CARD=AudioPCI,DEV=0
Ensoniq AudioPCI, ES1371 DAC2/ADC
Front speakers
rear:CARD=AudioPCI,DEV=0
Ensoniq AudioPCI, ES1371 DAC1
Rear speakers
surround40:CARD=AudioPCI,DEV=0
Ensoniq AudioPCI, ES1371 DAC2/ADC
4.0 Surround output to Front and Rear speakers
iec958:CARD=AudioPCI,DEV=0
Ensoniq AudioPCI, ES1371 DAC2/ADC
IEC958 (S/PDIF) Digital Audio Output
Last edited by specular (2014-08-27 19:52:45)Don't think I was clear enough, but arch is installed as a vmware guest. Below is the requested output from the alsa-info script:
upload=true&script=true&cardinfo=
!!ALSA Information Script v 0.4.63
!!Script ran on: Wed Aug 27 18:29:05 UTC 2014
!!Linux Distribution
NAME="Arch Linux" ID=arch PRETTY_NAME="Arch Linux" HOME_URL="https://www.archlinux.org/" SUPPORT_URL="https://bbs.archlinux.org/" BUG_REPORT_URL="https://bugs.archlinux.org/"
!!DMI Information
Manufacturer: VMware, Inc.
Product Name: VMware7,1
Product Version: None
Firmware Version: VMW71.00V.0.B64.1310020058
!!Kernel Information
Kernel release: 3.14.2-1-ARCH
Operating System: GNU/Linux
Architecture: x86_64
Processor: unknown
SMP Enabled: Yes
!!ALSA Version
Driver version: k3.14.2-1-ARCH
Library version: 1.0.27.2
Utilities version: 1.0.28
!!Loaded ALSA modules
snd_ens1371
!!Sound Servers on this system
No sound servers found.
!!Soundcards recognised by ALSA
0 [AudioPCI ]: ENS1371 - Ensoniq AudioPCI
Ensoniq AudioPCI ENS1371 at 0x1000, irq 19
!!PCI Soundcards installed in the system
02:01.0 Multimedia audio controller: Ensoniq ES1371 [AudioPCI-97] (rev 02)
!!Advanced information - PCI Vendor/Device/Subsystem ID's
02:01.0 0401: 1274:1371 (rev 02)
Subsystem: 1274:1371
!!Loaded sound module options
!!Module: snd_ens1371
enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y
id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null)
index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1
joystick_port : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
lineio : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
spdif : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
!!AC97 Codec information
--startcollapse--
0-0/0: Cirrus Logic CS4297A rev 3
PCI Subsys Vendor: 0x1274
PCI Subsys Device: 0x1371
Flags: 4
Capabilities :
DAC resolution : 16-bit
ADC resolution : 16-bit
3D enhancement : No 3D Stereo Enhancement
Current setup
Mic gain : +0dB [+0dB]
POP path : pre 3D
Sim. stereo : off
3D enhancement : off
Loudness : off
Mono output : MIX
Mic select : Mic1
ADC/DAC loopback : off
0:00 = 0000
0:02 = 0e8e
0:04 = 0000
0:06 = 0000
0:08 = 0000
0:0a = 0000
0:0c = bfbf
0:0e = bfbf
0:10 = bfbf
0:12 = bfbf
0:14 = bfbf
0:16 = bfbf
0:18 = 28a8
0:1a = 0000
0:1c = 0808
0:1e = 0000
0:20 = 0000
0:22 = 0000
0:24 = 0000
0:26 = 000f
0:28 = 0000
0:2a = 0000
0:2c = 0000
0:2e = 0000
0:30 = 0000
0:32 = 0000
0:34 = 0000
0:36 = 0000
0:38 = 0000
0:3a = 0000
0:3c = 0000
0:3e = 0000
0:40 = 0000
0:42 = 0000
0:44 = 0000
0:46 = 0000
0:48 = 0000
0:4a = 0000
0:4c = 0000
0:4e = 0000
0:50 = 0000
0:52 = 0000
0:54 = 0000
0:56 = 0000
0:58 = 0000
0:5a = 0000
0:5c = 0000
0:5e = 0000
0:60 = 0000
0:62 = 0000
0:64 = 0000
0:66 = 0000
0:68 = 0000
0:6a = 0000
0:6c = 0000
0:6e = 0000
0:70 = 0000
0:72 = 0000
0:74 = 0000
0:76 = 0000
0:78 = 0000
0:7a = 0000
0:7c = 4352
0:7e = 5913
--endcollapse--
!!ALSA Device nodes
crw-rw----+ 1 root audio 116, 6 Aug 28 01:35 /dev/snd/controlC0
crw-rw----+ 1 root audio 116, 2 Aug 28 01:35 /dev/snd/midiC0D0
crw-rw----+ 1 root audio 116, 5 Aug 28 01:35 /dev/snd/pcmC0D0c
crw-rw----+ 1 root audio 116, 4 Aug 28 03:28 /dev/snd/pcmC0D0p
crw-rw----+ 1 root audio 116, 3 Aug 28 01:35 /dev/snd/pcmC0D1p
crw-rw----+ 1 root audio 116, 1 Aug 28 01:35 /dev/snd/seq
crw-rw----+ 1 root audio 116, 33 Aug 28 01:35 /dev/snd/timer
/dev/snd/by-path:
total 0
drwxr-xr-x 2 root root 60 Aug 28 01:35 .
drwxr-xr-x 3 root root 200 Aug 28 01:35 ..
lrwxrwxrwx 1 root root 12 Aug 28 01:35 pci-0000:02:01.0 -> ../controlC0
!!Aplay/Arecord output
APLAY
**** List of PLAYBACK Hardware Devices ****
card 0: AudioPCI [Ensoniq AudioPCI], device 0: ES1371/1 [ES1371 DAC2/ADC]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: AudioPCI [Ensoniq AudioPCI], device 1: ES1371/2 [ES1371 DAC1]
Subdevices: 1/1
Subdevice #0: subdevice #0
ARECORD
**** List of CAPTURE Hardware Devices ****
card 0: AudioPCI [Ensoniq AudioPCI], device 0: ES1371/1 [ES1371 DAC2/ADC]
Subdevices: 1/1
Subdevice #0: subdevice #0
!!Amixer output
!!-------Mixer controls for card 0 [AudioPCI]
Card hw:0 'AudioPCI'/'Ensoniq AudioPCI ENS1371 at 0x1000, irq 19'
Mixer name : 'Cirrus Logic CS4297A rev 3'
Components : 'AC97a:43525913'
Controls : 26
Simple ctrls : 13
Simple mixer control 'Master',0
Capabilities: pvolume pswitch pswitch-joined
Playback channels: Front Left - Front Right
Limits: Playback 0 - 63
Mono:
Front Left: Playback 49 [78%] [-21.00dB] [on]
Front Right: Playback 49 [78%] [-21.00dB] [on]
Simple mixer control 'PCM',0
Capabilities: pvolume pswitch pswitch-joined
Playback channels: Front Left - Front Right
Limits: Playback 0 - 63
Mono:
Front Left: Playback 23 [37%] [0.00dB] [on]
Front Right: Playback 23 [37%] [0.00dB] [on]
Simple mixer control 'Line',0
Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
Capture exclusive group: 0
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: Playback 0 - 63
Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Simple mixer control 'CD',0
Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
Capture exclusive group: 0
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: Playback 0 - 63
Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Simple mixer control 'Mic',0
Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
Capture exclusive group: 0
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: Playback 0 - 63
Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [on]
Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [on]
Simple mixer control 'Mic Boost (+20dB)',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off]
Simple mixer control 'Video',0
Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
Capture exclusive group: 0
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: Playback 0 - 63
Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Simple mixer control 'Phone',0
Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
Capture exclusive group: 0
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: Playback 0 - 63
Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Simple mixer control 'IEC958',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off]
Simple mixer control 'Aux',0
Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
Capture exclusive group: 0
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: Playback 0 - 63
Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off]
Simple mixer control 'Capture',0
Capabilities: cvolume cswitch cswitch-joined
Capture channels: Front Left - Front Right
Limits: Capture 0 - 15
Front Left: Capture 8 [53%] [12.00dB] [on]
Front Right: Capture 8 [53%] [12.00dB] [on]
Simple mixer control 'Mix',0
Capabilities: cswitch cswitch-exclusive
Capture exclusive group: 0
Capture channels: Front Left - Front Right
Front Left: Capture [off]
Front Right: Capture [off]
Simple mixer control 'Mix Mono',0
Capabilities: cswitch cswitch-exclusive
Capture exclusive group: 0
Capture channels: Front Left - Front Right
Front Left: Capture [off]
Front Right: Capture [off]
!!Alsactl output
--startcollapse--
state.AudioPCI {
control.1 {
iface MIXER
name 'Master Playback Switch'
value true
comment {
access 'read write'
type BOOLEAN
count 1
control.2 {
iface MIXER
name 'Master Playback Volume'
value.0 49
value.1 49
comment {
access 'read write'
type INTEGER
count 2
range '0 - 63'
dbmin -9450
dbmax 0
dbvalue.0 -2100
dbvalue.1 -2100
control.3 {
iface MIXER
name 'Phone Playback Switch'
value false
comment {
access 'read write'
type BOOLEAN
count 1
control.4 {
iface MIXER
name 'Phone Playback Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
range '0 - 63'
dbmin -3450
dbmax 6000
dbvalue.0 -3450
dbvalue.1 -3450
control.5 {
iface MIXER
name 'Mic Playback Switch'
value false
comment {
access 'read write'
type BOOLEAN
count 1
control.6 {
iface MIXER
name 'Mic Playback Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
range '0 - 63'
dbmin -3450
dbmax 6000
dbvalue.0 -3450
dbvalue.1 -3450
control.7 {
iface MIXER
name 'Mic Boost (+20dB)'
value false
comment {
access 'read write'
type BOOLEAN
count 1
control.8 {
iface MIXER
name 'Line Playback Switch'
value false
comment {
access 'read write'
type BOOLEAN
count 1
control.9 {
iface MIXER
name 'Line Playback Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
range '0 - 63'
dbmin -3450
dbmax 6000
dbvalue.0 -3450
dbvalue.1 -3450
control.10 {
iface MIXER
name 'CD Playback Switch'
value false
comment {
access 'read write'
type BOOLEAN
count 1
control.11 {
iface MIXER
name 'CD Playback Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
range '0 - 63'
dbmin -3450
dbmax 6000
dbvalue.0 -3450
dbvalue.1 -3450
control.12 {
iface MIXER
name 'Video Playback Switch'
value false
comment {
access 'read write'
type BOOLEAN
count 1
control.13 {
iface MIXER
name 'Video Playback Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
range '0 - 63'
dbmin -3450
dbmax 6000
dbvalue.0 -3450
dbvalue.1 -3450
control.14 {
iface MIXER
name 'Aux Playback Switch'
value false
comment {
access 'read write'
type BOOLEAN
count 1
control.15 {
iface MIXER
name 'Aux Playback Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
range '0 - 63'
dbmin -3450
dbmax 6000
dbvalue.0 -3450
dbvalue.1 -3450
control.16 {
iface MIXER
name 'PCM Playback Switch'
value true
comment {
access 'read write'
type BOOLEAN
count 1
control.17 {
iface MIXER
name 'PCM Playback Volume'
value.0 23
value.1 23
comment {
access 'read write'
type INTEGER
count 2
range '0 - 63'
dbmin -3450
dbmax 6000
dbvalue.0 0
dbvalue.1 0
control.18 {
iface MIXER
name 'Capture Source'
value.0 Mic
value.1 Mic
comment {
access 'read write'
type ENUMERATED
count 2
item.0 Mic
item.1 CD
item.2 Video
item.3 Aux
item.4 Line
item.5 Mix
item.6 'Mix Mono'
item.7 Phone
control.19 {
iface MIXER
name 'Capture Switch'
value true
comment {
access 'read write'
type BOOLEAN
count 1
control.20 {
iface MIXER
name 'Capture Volume'
value.0 8
value.1 8
comment {
access 'read write'
type INTEGER
count 2
range '0 - 15'
dbmin 0
dbmax 2250
dbvalue.0 1200
dbvalue.1 1200
control.21 {
iface MIXER
name 'IEC958 Playback Con Mask'
value '0fff000f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
comment {
access read
type IEC958
count 1
control.22 {
iface MIXER
name 'IEC958 Playback Pro Mask'
value cf00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000
comment {
access read
type IEC958
count 1
control.23 {
iface MIXER
name 'IEC958 Playback Default'
value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
comment {
access 'read write'
type IEC958
count 1
control.24 {
iface MIXER
name 'IEC958 Playback Switch'
value false
comment {
access 'read write'
type BOOLEAN
count 1
control.25 {
iface PCM
name 'Playback Channel Map'
value.0 0
value.1 0
comment {
access read
type INTEGER
count 2
range '0 - 36'
control.26 {
iface PCM
device 1
name 'Playback Channel Map'
value.0 0
value.1 0
comment {
access read
type INTEGER
count 2
range '0 - 36'
--endcollapse--
!!All Loaded Modules
Module
fuse
vmw_vsock_vmci_transport
vsock
nls_iso8859_1
nls_cp437
vfat
fat
coretemp
hwmon
joydev
snd_ens1371
mousedev
snd_ac97_codec
hid_generic
ac97_bus
gameport
snd_rawmidi
snd_seq_device
snd_pcm
snd_timer
snd
crct10dif_pclmul
crc32_pclmul
crc32c_intel
ghash_clmulni_intel
soundcore
usbhid
aesni_intel
aes_x86_64
lrw
gf128mul
glue_helper
hid
psmouse
vmwgfx
ppdev
ablk_helper
cryptd
evdev
vmxnet3
mac_hid
pcspkr
vmw_balloon
i2c_piix4
serio_raw
ttm
microcode
drm
processor
i2c_core
intel_agp
intel_gtt
vmw_vmci
parport_pc
parport
battery
shpchp
ac
button
ext4
crc16
mbcache
jbd2
dm_mod
sd_mod
crc_t10dif
crct10dif_common
sr_mod
cdrom
ata_generic
pata_acpi
atkbd
libps2
ata_piix
libata
mptsas
scsi_transport_sas
mptscsih
uhci_hcd
ehci_pci
scsi_mod
ehci_hcd
usbcore
mptbase
usb_common
floppy
i8042
serio
!!ALSA/HDA dmesg
[ 5.342713] mousedev: PS/2 mouse device common for all mice
[ 5.345140] snd_ens1371 0000:02:01.0: enabling device (0000 -> 0001)
[ 5.391722] intel_rapl: domain package energy ctr 0:0 not working, skip
Last edited by specular (2014-08-27 18:45:55) -
Webcam Video pauses when recording sound from microphone
I have a strange problem which i am encountering when recording and playing back webcam video.
I am developing a small application for recording webcam video and audio using FMS and playing it back later from FMS.
I am using AS 3.0, FMS 3.5 and flash CS 5.5
When i record only the video from webcam without sound from microphone the playback is fine. But when i record microphone sound also along with the video, and when i playback the video pauses right where the sound starts for 10 sec or more and then it starts playing. Towards the end the video pauses again and only the sound goes on till the end.
I tried setting the buffertime to 50 before ns.publish and ns.play but that didnt help either.
I tried using the flvplayback component and even their it pauses.
Any help is appreciated.Haha, that is such a genius use of common sense! Thank you, I didn't think of that. I certainly will!
However, does anyone know if there is a way of adjusting the program of garageband to fix this issue? Using headphones certainly fixes the symptoms of my problem, but it doesn't actually address the root cause: garageband should only be recording the input through the microphone. I shouldn't need to wear headphones. (And doing so inhibits my ability to clearly hear myself play, and stay in tune.) -
Can I buy a headset with microphone for my 13" MACBOOK PRO?
There is only one hole to plug an earphone, and when i record in my software, and i replay it. it sounds strange. so i'm trying to change a microphone.
is the headset with microphone for A1278 on sale? could somebody give me an sale net address?Thank you. I know what you mean. Apple can support earpiece and microphone in one plug.
When I ask about this question, my MBP had trouble in Windows XP.
I'm Chinese, and there is a software named DynEd. I was using it to record my voice,that voice changed when I replay. It just like an iPhone app:Talking Prez.
And then I reinstall my windows system. Now I'm using Windows 7, and the question resolved.
Maybe 4GB memory is too big for WinXP x32.
What do you think? -
I am having a very strange problem when using LabVIEW to acquire audio
data via the Windows API from a Creative Professional E-MU 1616m sound
card. The goal is to acquire sound in 24-bit resolution.
When capturing sound in 16-bit mode (as set in the LabVIEW software),
the E-MU 1616m behaves as expected, with a 105dB SNR and approximately
-130dB noise floor, after dithering. However, when switching to
24-bit capture mode, a very severe truncation occurs. This sends
the harmonic distortion and noise through the roof. After
investigating fairly deeply in our LabVIEW code, I am wondering what
might be the problem.
I have compiled a number of screenshots which showcase the problem in more detail.
Here is a background of the experiment:
For these tests, both the analog and digital audio was generated by an
Audio Precision System Two system, and was passed directly into the
respective line-level or digital audio inputs. Digital audio was
tested using both coax and optical cable. In the sound card
control software, the audio was sent directly from the input channel
into the WAVE IN L/R (via the Windows API, I assume). The
sampling rate for the profile was 96kHz.
The sampling rate in all LabVIEW functions was set to 96kHz. The sample rate set in the AP Digital generator was 96kHz.
ANALOG 20dBu 96kHz 16bit.jpg
In this test, everything looks fine. The audio input is at
Full-scale for the E-mu's ADCs. It is exhibiting expected 16-bit
performance (with dithering).
ANALOG 20dBu 96kHz 24bit.jpg
Now we instruct the driver to capture sound in 24 bits. Notice
that the noise floor and THD+N go up considerably. Effects of
truncation become visible on the time-domain display.
ANALOG -20dBu 96kHz 16bit.jpg
Now we drop the input level to -20dBu. The performance
starts to look a little messy but is still acceptable. Note,
however, the high peaks on the odd harmonics.
ANALOG -20dBu 96kHz 24bit.jpg
Now we try to capture at 24 bits. The effects of truncation are extreme at this low signal level.
ANALOG -60dBu 96kHz 16bit.jpg
Now we are at extremely low signal levels. Individual
quantization levels can be seen on the signal. Dither is also
present. Performance is still good.
ANALOG -60dBu 96kHz 24bit.jpg
However, when increasing the resolution to 24 (which should increase
the number of quantization levels), our signal is reduced to a square
wave. Obviously something is wrong.
DIGITAL 0dB 16bit 96kHz 16bit.jpg
Now on to the digital tests. We start with full-scale. We
used an AP outputting a properly dithered 16-bit signal over an optical
cable. The soundcard is instructed to receive in 16 bit
mode. It looks good.
DIGITAL 0dB 16bit 96kHz 24bit.jpg
Using the same input, we change to 24 bit receive mode.
DIGITAL 0dB 24bit 96kHz 16bit.jpg
Now we set up the AP to output a properly dithered 24-bit signal at
full-scale. The dips in the frequency domain show us that
something is wrong.
DIGITAL 0dB 24bit 96kHz 24bit.jpg
Receiving in 24-bit mode. Same story as before.
DIGITAL -90dB 16bit 96kHz 16bit.jpg
Now we decrease the amplitude to a low level. Well-implemented dither is shown here clearly.
DIGITAL -90dB 16bit 96kHz 24bit.jpg
However, receiving in 24-bit mode reduces the signal to a dithered square wave.
DIGITAL -90dB 24bit 96kHz 16bit.jpg
Here is the low-level signal with the AP generating a 24-bit
signal. Dither is applied, but vanishes in the e-mu 1616m.
It seems the dither level has been changed. This is the cause of
our dips from before.
DIGITAL -90dB 24bit 96kHz 24bit.jpg
And finally, we transmit and receive in 24-bits. Here are the results.
We have achieved similar results using several of your breakout boxes and soundcards.
Attached are all screenshots, as well as the main VI (AP Test.vi) and
the dependent vi's. There are a number of SVT vi's in the
project, but they can be ignored since they are not related to the
problem.
Any help would be greatly appreciated.
Best Regards,
Brett Gildersleeve
Attachments:
AP Test.zip 2381 KBHi Brett,
I took a look at the code you attached, but it appears that you may have left out the subvi that actually acquires the sound. When you say Windows API, I assume that you are calling a DLL at some point. How are you configuring the inputs (or what function are you using)? It could be (and this is just a guess) that when you specify 24 bits, the DLL returns the data in a very specific format. If you don't interpret the bits that are returned correctly, LabVIEW may not know what to do with them, as LabVIEW does not have a native 24-bit datatype. If there is any documentation for these function calls, our answer might there...
Just some thoughts -- thanks for posting your solution!
Charlie S.
Visit ni.com/gettingstarted for step-by-step help in setting up your system -
hi , i need the best way to capture sound samples from an opened Microphone port
is it by directly using a TargetDataLine and how ?
and thanksHope this program will help you.
http://www.jsresources.org/examples/SimpleAudioRecorder.java.html -
IPad video to car DVD - has no sound with composite cable
Hi,
I'm trying to watch iPad movies on my in car DVD system. Through a genuine apple composite cable I can get video with out sound. When I connect the USB from the composite I get sound but no video.
Any ideas??
I have iPad 2.
Stereo in car is JVC 846.
ThanksHi,
I do have the same Problem with my F10
I can capture TV with sound, but not either S-Video or FBAS with sound. In ShowBiz I can see, that recording TV I can choose Audio from the built in capture device - no other soudn device. Recording either S-Video or FBAS I can not choose ANY audio recording device. But if I want to record audio, I CAN choose an audio device other than the built in TV-capturing device - namely the audio device.
It seems that trying to capture video via the built in cpturing device, you HAVE TO capture audio from the SAME device - and that using the S-Video or FBAS Input there is no Audio there - and so capturing video from anything else than TV is not possible to combine with audio capturing from an other source.
The only chance is to capture video and audio seperate (one after the other) and mix them together in WinDVD Creator or Showbiz -
Thanx a lot,
Hann
[Edited by: admin on 08-Mar-05 10:43] -
Recording sound with Soundbooth
I managed to record sound with the system 'sound recorder', but when I try to record sound in soundbooth with the same mic (simple cheap mic) there is nothing showing on the meter and nothing records. Can somebody please explain to me what is the problem?
While Premiere and Soundbooth both support a Windows audio standard called ASIO, both also include a driver that will let you use audio devices that do not support this format. The driver is configured by default, but may not always select the proper input port - depending on the audio device and what Windows reports to the application.
Fortunately, it's not too difficult to tinker around and find the right setting. In Soundbooth, choose Edit > Preferences > Audio Hardware... In the dialog that pops up, you should see a drop-down labeled "Default Device" If you are running Soundbooth CS4, you should see "Soundbooth 2.0 WDM Sound" listed here. Click the Settings button, and in the new dialog, click the "Input" tab. Here you'll see a list of input ports for your selected audio device. Make certain your microphone port is enabled, and click OK a few times to get back to the main program. Click the Record button and make sure the microphone input is selected in the Port: dropdown. Try recording and see if this solves the problem.
If it still does not record, or you do not see the proper input to enable, you may need to verify the proper audio device is selected, or if you're running Windows Vista, that the input port you want to use is enabled in Window's Audio settings. You can find this by right clicking on the speaker icon in your system try and choose "Recording Devices."
Good luck!
Durin
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