[Solved] Recording internal audio with alsa and recordmydesktop

I record Let's Plays and wanted to record them with recordmydesktop.
I first tried to record using ALSA but after reading much threads about that problem I decided to use pulseaudio because it seems to be easier.
But I want ALSA back so I decided to ask here if anyone knows a solution for the ALSA sound server.
Can anyone help me?
Last edited by TheRatze94 (2012-08-31 19:25:42)

modprobe snd-aloop
should give you another virtual sound card with an input connected to an output.
Then set up a /~.asoundrc that let you play the stream to the loopback interface and the real speakers too, i used this in the past and it worked:
pcm.!default {
type asym
playback.pcm "LoopAndReal"
#capture.pcm "looprec"
capture.pcm "hw:0,0"
pcm.looprec {
type hw
card "Loopback"
device 1
subdevice 0
pcm.LoopAndReal {
type plug
slave.pcm mdev
route_policy "duplicate"
pcm.mdev {
type multi
slaves.a.pcm pcm.MixReale
slaves.a.channels 2
slaves.b.pcm pcm.MixLoopback
slaves.b.channels 2
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave b
bindings.2.channel 0
bindings.3.slave b
bindings.3.channel 1
pcm.MixReale {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
rate 48000
#rate 44100
periods 128
period_time 0
period_size 1024 # must be power of 2
buffer_size 8192
pcm.MixLoopback {
type dmix
ipc_key 1025
slave {
pcm "hw:Loopback,0,0"
rate 48000
#rate 44100
periods 128
period_time 0
period_size 1024 # must be power of 2
buffer_size 8192
Last edited by kokoko3k (2012-08-31 20:00:00)

Similar Messages

Unable to record internal audio with RecordMyDesktop

In order to record sound and video from my desktop, I installed RecordMyDesktop (https://wiki.archlinux.org/index.php/RecordMyDesktop) and followed the instructions from post #4 and #6 on a thread about "Recording internal audio with alsa and recordmydesktop" (https://bbs.archlinux.org/viewtopic.php … 3#p1359593). When I try to record with "recordmydesktop --device looprec", it will not start recording and closes with the following error:
Initial recording window is set to:
X:0 Y:0 Width:1920 Height:1080
Adjusted recording window is set to:
X:0 Y:4 Width:1920 Height:1072
Your window manager appears to be i3
Initializing...
Buffer size adjusted to 4096 from 4096 frames.
Opened PCM device looprec
Couldn't set format.
Error while opening/configuring soundcard looprec
Try running with the --no-sound or specify a correct device.
If this information would be helpful, I use alsamixer and "aplay -l" gives the following:
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: ALC663 Analog [ALC663 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 1: ALC663 Digital [ALC663 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM]
Subdevices: 7/8
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
card 1: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM]
Subdevices: 8/8
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
I can record from the microphone with with the command "recordmydesktop". I succeeded in recording the internal audio with audacity by choosing the device "looprec" for recordings. I'm not sure why RecordMyDesktop gives an error while Audacity does not. Below is my ~/.asoundrc, which I copied from the aforementioned thread. I am not knowledgeable, but I guess there is a mistake in it? Would you please assist me solving this error? Thank you for your time.
pcm.!default {
type asym
playback.pcm "LoopAndReal"
#capture.pcm "looprec"
capture.pcm "hw:0,0"
pcm.looprec {
type hw
card "Loopback"
device 1
subdevice 0
pcm.LoopAndReal {
type plug
slave.pcm mdev
route_policy "duplicate"
pcm.mdev {
type multi
slaves.a.pcm pcm.MixReale
slaves.a.channels 2
slaves.b.pcm pcm.MixLoopback
slaves.b.channels 2
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave b
bindings.2.channel 0
bindings.3.slave b
bindings.3.channel 1
pcm.MixReale {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
rate 48000
#rate 44100
periods 128
period_time 0
period_size 1024 # must be power of 2
buffer_size 8192
pcm.MixLoopback {
type dmix
ipc_key 1025
slave {
pcm "hw:Loopback,0,0"
rate 48000
#rate 44100
periods 128
period_time 0
period_size 1024 # must be power of 2
buffer_size 8192
Last edited by rayanamukami (2014-07-05 14:05:20)

You might want to take a look here and here.

Internal table with Dynamic and Non dynamic fileds

Hi Experts,
How to get the internal table with Dynamic and Non-Dynamic Fields.
Could u please help me.
Thanks,
Varun

Hi,
Execute the below sample code or analyze it there is appropriate description provided.
*& Report ZTEST_PRM_DYN_ALV
REPORT ZTEST_PRM_DYN_ALV.
type-pools: slis.
field-symbols: <dyn_table> type standard table,
<dyn_wa>.
data: alv_fldcat type slis_t_fieldcat_alv,
it_fldcat type lvc_t_fcat.
selection-screen begin of block b1 with frame title text-001.
parameters: p_flds(5) type c.
selection-screen end of block b1.
start-of-selection.
*build the dynamic internal table
perform build_dyn_itab.
*write 5 records to the alv grid
do 5 times.
perform build_report.
enddo.
*call the alv grid.
perform call_alv.
*Build_dyn_itab
form build_dyn_itab.
data: new_table type ref to data,
new_line type ref to data,
wa_it_fldcat type lvc_s_fcat.
*Create fields .
clear wa_it_fldcat.
wa_it_fldcat-fieldname = 'name1'.
wa_it_fldcat-datatype = 'mara-matnr'.
wa_it_fldcat-intlen = 5.
append wa_it_fldcat to it_fldcat .
*clear wa_it_fldcat.
wa_it_fldcat-fieldname = sy-index.
wa_it_fldcat-datatype = 'CHAR'.
wa_it_fldcat-intlen = 5.
append wa_it_fldcat to it_fldcat .
do p_flds times.
clear wa_it_fldcat.
wa_it_fldcat-fieldname = sy-index.
wa_it_fldcat-datatype = 'CHAR'.
wa_it_fldcat-intlen = 6.
append wa_it_fldcat to it_fldcat .
enddo.
*Create dynamic internal table and assign to FS
call method cl_alv_table_create=>create_dynamic_table
exporting
it_fieldcatalog = it_fldcat
importing
ep_table = new_table.
assign new_table->* to <dyn_table>.
*Create dynamic work area and assign to FS
create data new_line like line of <dyn_table>.
assign new_line->* to <dyn_wa>.
endform.
*Form build_report
form build_report.
data: fieldname(20) type c.
data: fieldvalue(5) type c.
data: index(3) type c.
field-symbols: <fs1>.
do p_flds times.
index = sy-index.
*Set up fieldvalue
concatenate 'FLD' index into
fieldvalue.
condense fieldvalue no-gaps.
assign component index of structure <dyn_wa> to <fs1>.
<fs1> = fieldvalue.
enddo.
*Append to the dynamic internal table
append <dyn_wa> to <dyn_table>.
endform.
*CALL_ALV
form call_alv.
data: wa_cat like line of alv_fldcat.
*clear wa_cat.
wa_cat-fieldname = 'matnr'.
wa_cat-seltext_s = sy-index.
wa_cat-outputlen = '10'.
append wa_cat to alv_fldcat.
do p_flds times.
clear wa_cat.
wa_cat-fieldname = sy-index.
wa_cat-seltext_s = sy-index.
wa_cat-outputlen = '6'.
append wa_cat to alv_fldcat.
enddo.
*Call ABAP List Viewer (ALV)
call function 'REUSE_ALV_GRID_DISPLAY'
exporting
it_fieldcat = alv_fldcat
tables
t_outtab = <dyn_table>.
endform.
Hope this will help, reward if found usefull.
Cheers,
Ram.

Can't get 4.1 working with ALSA and Intel card

Hello everybody,
My problem is simple, but I cannot find any solution for it.
I have Arch Linux 64 bits installed with ALSA and a Intel onboard sound card
I can get the audio working simply installing ALSA and unmutting the channels, but I get no sound at the rear speakers and the center speaker (subwoofer).
I run:
$ spearker-test -c 5
And all I have is sound at the 2 front speakers.
Looking at alsamixer I only saw configuration for 2,4 and 6 channels. I am currently using the configuration for 4 channels.
I unmuted everything in alsamixer (playback) and nothing changed.
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
$ lsmod | grep '^snd' | column -t
snd_hda_codec_hdmi 23704 1
snd_hda_codec_realtek 59331 1
snd_hda_intel 25700 7
snd_hda_codec 97464 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel
snd_hwdep 6300 1 snd_hda_codec
snd_pcm 74926 3 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel
snd_page_alloc 7217 2 snd_pcm,snd_hda_intel
snd_timer 18966 1 snd_pcm
snd 60021 21 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_hda_codec,snd_hda_intel
Anyone knows how to make it work? Thanks!

Thanks for the answer, tried this command, here is the output:
$ speaker-test -t wav -c 5 -D surround41
speaker-test 1.0.26
Playback device is surround41
Stream parameters are 48000Hz, S16_LE, 5 channels
WAV file(s)
Broken configuration for playback: no configurations available: Invalid argument
Setting of hwparams failed: Invalid argument
Is there something needed to make this command run correctly?

Recording Internal Audio?

Hi, sorry if this is a newbish question, but it is possible to record internal audio from the macbook air's speakers? (ex: the audio from a certain part of a video on youtube) I've tried Quicktime Pro and Audacity, but both use the external microphone, so I can hear myself cough, but not the audio from the computer. On windows there was an option in sound to switch to built in audio which would allow a program like camstudio to record the internal audio, is this possible on mac?

AudioHijackPro http://rogueamoeba.com/audiohijackpro/
Stefan

How do I record internal audio to garageband for ipad2? I want to record from a drum machine app to garageband.

How do I record internal audio to garageband for ipad2? I want to record from a drum machine app to garageband.

I dont believe you can do that directly. You would need to record the audio separately and then import

Internal table with Import and Export

Hi All,
Hi all
Please let me know the use of <b>Internal table with Import and Export parameters and SET/GET parameters</b>, on what type of cases we can use these? Plese give me the syntax with some examples.
Please give me detailed analysis on the above.
Regards,
Prabhu

Hi Prabhakar,
There are three types of memories.
1. ABAP MEMORY
2. SAP MEMORY
3. EXTERNAL MEMORY.
1.we will use EXPORT/ IMPORT TO/ FROM MEMORY-ID when we want to transfer between ABAP memory
2. we will use GET PARAMETER ID/ SET PARAMETER ID to transfer between SAP MEMORY
3. we will use EXPORT/IMPORT TO/FROM SHARED BUFFER to transfer between external memory.
ABAP MEMORY : we can say that two reports in the same session will be in ABAP MEMORY
SAP MEMORY: TWO DIFFERENT SESSIONS WILL BE IN SAP MEMORY.
for ex: IF WE CALL TWO DIFFERENT TRANSACTIONS SE38, SE11
then they both are in SAP MEMORY.
EXTERNAL MEMORY: TWO different logons will be in EXTERNAL MEMORY.
<b>Syntax</b>
To fill the input fields of a called transaction with data from the calling program, you can use the SPA/GPA technique. SPA/GPA parameters are values that the system stores in the global, user-specific SAP memory. SAP memory allows you to pass values between programs. A user can access the values stored in the SAP memory during one terminal session for all parallel sessions. Each SPA/GPA parameter is identified by a 20-character code. You can maintain them in the Repository Browser in the ABAP Workbench. The values in SPA/GPA parameters are user-specific.
ABAP programs can access the parameters using the SET PARAMETER and GET PARAMETER statements.
To fill one, use:
SET PARAMETER ID <pid> FIELD <f>.
This statement saves the contents of field <f> under the ID <pid> in the SAP memory. The code <pid> can be up to 20 characters long. If there was already a value stored under <pid>, this statement overwrites it. If the ID <pid> does not exist, double-click <pid> in the ABAP Editor to create a new parameter object.
To read an SPA/GPA parameter, use:
GET PARAMETER ID <pid> FIELD <f>.
This statement fills the value stored under the ID <pid> into the variable <f>. If the system does not find a value for <pid> in the SAP memory, it sets SY-SUBRC to 4, otherwise to 0.
To fill the initial screen of a program using SPA/GPA parameters, you normally only need the SET PARAMETER statement.
The relevant fields must each be linked to an SPA/GPA parameter.
On a selection screen, you link fields to parameters using the MEMORY ID addition in the PARAMETERS or SELECT-OPTIONS statement. If you specify an SPA/GPA parameter ID when you declare a parameter or selection option, the corresponding input field is linked to that input field.
On a screen, you link fields to parameters in the Screen Painter. When you define the field attributes of an input field, you can enter the name of an SPA/GPA parameter in the Parameter ID field in the screen attributes. The SET parameter and GET parameter checkboxes allow you to specify whether the field should be filled from the corresponding SPA/GPA parameter in the PBO event, and whether the SPA/GPA parameter should be filled with the value from the screen in the PAI event.
When an input field is linked to an SPA/GPA parameter, it is initialized with the current value of the parameter each time the screen is displayed. This is the reason why fields on screens in the R/3 System often already contain values when you call them more than once.
When you call programs, you can use SPA/GPA parameters with no additional programming overhead if, for example, you need to fill obligatory fields on the initial screen of the called program. The system simply transfers the values from the parameters into the input fields of the called program.
However, you can control the contents of the parameters from your program by using the SET PARAMETER statement before the actual program call. This technique is particularly useful if you want to skip the initial screen of the called program and that screen contains obligatory fields.
Reading Data Objects from Memory
To read data objects from ABAP memory into an ABAP program, use the following statement:
Syntax
IMPORT <f1> [TO <g 1>] <f 2> [TO <g 2>] ... FROM MEMORY ID <key>.
This statement reads the data objects specified in the list from a cluster in memory. If you do not use the TO <g i > option, the data object <f i > in memory is assigned to the data object in the program with the same name. If you do use the option, the data object <f i > is read from memory into the field <g i >. The name <key> identifies the cluster in memory. It may be up to 32 characters long.
You do not have to read all of the objects stored under a particular name <key>. You can restrict the number of objects by specifying their names. If the memory does not contain any objects under the name <key>, SY-SUBRC is set to 4. If, on the other hand, there is a data cluster in memory with the name <key>, SY-SUBRC is always 0, regardless of whether it contained the data object <f i >. If the cluster does not contain the data object <f i >, the target field remains unchanged.
Saving Data Objects in Memory
To read data objects from an ABAP program into ABAP memory, use the following statement:
Syntax
EXPORT <f1> [FROM <g 1>] <f 2> [FROM <g 2>] ... TO MEMORY ID <key>.
This statement stores the data objects specified in the list as a cluster in memory. If you do not use the option FROM <f i >, the data object <f i > is saved under its own name. If you use the FROM <g i > option, the data objet <g i > is saved under the name <f i >. The name <key> identifies the cluster in memory. It may be up to 32 characters long.
Check this link.
http://www.sap-img.com/abap/difference-between-sap-and-abap-memory.htm
Thanks,
Susmitha.
Reward points for helpful answers.

Getting static audio with Titanium and Vista x64 at random times

@Getting static audio with Titanium and Vista x64 at random times$ I have had this card for about 6-8 months and relati'vely no issues.
I just upgraded to the latest drivers 270006 and now, sometimes after exiting Turbine's Lotro, my audio plays but is very static-y, but it doesn't happen all the time. I had iTunes minimized when playing Lotro and then went to play a song after exiting. I quit iTunes and tried it again and it was ok.
This also happened in Firefox, a YouTube video had the same static audio.
It doesn't happen all the time and I do play other games, but I haven't been playing much else right now.
So right now I don't know if it's a Lotro thing or just the drivers.
I have a Q9650, single GTX260, Asus P35 board (not SLI), Vista x64, 4GB.

Carl, do let us know on the Safe Mode test... would be the first thing I'd try with what you've mentioned, but yeah, daisy chained Hubs could possibly play into it.

No sound with ALSA and/or Pulse from built-in Intel 5 Series Card

Hello,
I posted a thread about a similar problem to this 4 years ago, but that solution is no longer working for me, and I'm just now returning to Arch after floating between a few different distros, so I'm not as knowledgeable about this as I might once have been.
Basically, my problem is that sound does not work on my laptop, neither through the internal speakers or headphone jack, and neither with plain ALSA nor Pulseaudio. This leads me to believe that there is some underlying problem that once solved, will allow plain ALSA and Pulse to work properly, but that's just speculation. I've put several hours of research into this so far with no luck. Everything in alsamixer is unmuted and looks right. I'm going to dump as much relevant info on my setup below in hopes that someone more knowledgeable than I can spot the issue:
The relevant section of lspci:
00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)
Subsystem: Dell Device 0447
Flags: bus master, fast devsel, latency 0, IRQ 32
Memory at fbd00000 (64-bit, non-prefetchable) [size=16K]
Capabilities: <access denied>
Kernel driver in use: snd_hda_intel
Kernel modules: snd_hda_intel
The output of aplay -l
card 0: MID [HDA Intel MID], device 0: 92HD81B1X5 Analog [92HD81B1X5 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: MID [HDA Intel MID], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
The output of aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
default:CARD=MID
HDA Intel MID, 92HD81B1X5 Analog
Default Audio Device
sysdefault:CARD=MID
HDA Intel MID, 92HD81B1X5 Analog
Default Audio Device
front:CARD=MID,DEV=0
HDA Intel MID, 92HD81B1X5 Analog
Front speakers
surround21:CARD=MID,DEV=0
HDA Intel MID, 92HD81B1X5 Analog
2.1 Surround output to Front and Subwoofer speakers
surround40:CARD=MID,DEV=0
HDA Intel MID, 92HD81B1X5 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=MID,DEV=0
HDA Intel MID, 92HD81B1X5 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=MID,DEV=0
HDA Intel MID, 92HD81B1X5 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=MID,DEV=0
HDA Intel MID, 92HD81B1X5 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=MID,DEV=0
HDA Intel MID, 92HD81B1X5 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
hdmi:CARD=MID,DEV=0
HDA Intel MID, HDMI 0
HDMI Audio Output
The contents of /proc/asound/cards
0 [MID ]: HDA-Intel - HDA Intel MID
HDA Intel MID at 0xfbd00000 irq 32
The contents of /proc/asound/devices
2: [ 0] : control
3: [ 0- 0]: digital audio playback
4: [ 0- 0]: digital audio capture
5: [ 0- 3]: digital audio playback
6: [ 0- 0]: hardware dependent
7: [ 0- 3]: hardware dependent
33: : timer
/etc/asound.conf is empty and my user has no .asoundrc file in its home directory.
I ran the ALSA Information Script and the results of that can be found here.
Any help would be greatly appreciated; this is a very frustrating issue. Thank you very much!

same problem with me on Intel Corporation 7 Series/C210 Series.
actually, internal speakers works, it's just on mute (you can change it with alsamixer), but headphone/external speakers doesn't work. and there's no sound card detected in gnome control center -> audio section.
"00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04)"

Unable to record just audio with Mac Mini - no isight

I am trying to use Podcast Capture on a mac mini connected to a generic monitor (no isight) and it is unable to initialize the audio subsystem. I am assuming this has to do with it needing a video camera, but surely that isn't required to record an audio only podcast. I have a microphone plugged into the line in and it works successfully in Garageband. Please help!

Take a look at Soundflower, which will allow you to record audio from any source:
http://soundflower.en.softonic.com/mac
http://www.macworld.com/article/1159440/soundflower_capture.html

Problems with ALSA and OSS

Hi.
I'm a ArchLinux user and I have been having problems with my audio, specifficaly the microphone.
My microphone works well with Audacity and I use it for speaking trough TeamSpeaker and since TeamSpeak still uses OSS I installed alsa-oss and added aoss to my TeamSpeak command. But, for some reason, the application stopped working (this have nothing to do with my real problem) and as substitute I installed TeamSpeak in Windows XP running on Virtualbox with ALSA audio. Everything worked smoothly until some weeks ago.
The microphone on Virtualbox stopped working but I still hear sound. And the funny thing is that TS Linux got working again, but if I use it with aoss I cannot use the microphone but I hear sound and to have microphone I have to use it without aoss but then i dont have sound on other apps.
And to get microphone on Virtualbox I have to enable OSS in preferences and use aoss but then, wich I don't understand why, I dont get any sound with other apps. :\
I have no idea what triggered this problem.
My audio device:
00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)
ALC660-VD Analog

No one? :\
Not even why Virtualbox with OSS and aoss "steals" the sound to the other apps?

[solved] Can't boot with LVM and usb external disk

I successfully installed Archlinux on my first computer, using LVM on two internal Hard Drives.
I'm trying to do exactly the same on my second computer, but I got a problem: the LVM volume is not found at boot, so that the root partition can't be mounted and the boot fails.
The only difference is that, on my second computer, the second hard drive is connected through USB.
Some hints: my second computer boots perfectly with LVM and Ubuntu. Archlinux needs to be configured adequately.
I tried to add rootdelay=30 in menu.lst, but it changes nothing. Of course, in /etc/rc.conf I have USELVM="yes" and in /etc/mkinitcpio.conf I have HOOKS="base udev autodetect pata scsi sata keymap lvm2 filesystems"
Thank you for helping me to keep Archlinux on my second computer!
Last edited by Achille (2008-06-29 21:59:09)

I finally solved my problem!
In the file /etc/mkinitcpio.conf, I modified the MODULES loaded:
MODULES="pata_acpi ata_generic ata_piix usbcore usb_storage uhci_hcd ehci_hcd sg scsi_mod sd_mod dm_mod"
I also add in the file /lib/initcpio/hooks/lvm2 the line /bin/sleep 10
/bin/sleep 10
msg "Scanning logical volumes..."
And finally, I ran the command:
mkinitcpio -p kernel26

Solve my corrupted audio problem for $$ and eternal gratitude

My video project has corrupted audio suddenly. I've read about other audio problems when exporting, but all I did was save the project while editing, no export. I have the prefs set to extract audio at pasteover, which I've done a lot of for this 2-camera wedding shoot. I also have the second audio track occasionally used for background music. All three tracks (video, audio 1, audio 2) on the timeline have corrupted audio for the entire length of the project. It sounds like a helicopter whirr was added, it's very choppy. Also, both cpus are maxxed out and the process monitor shows iMovie utilizing 180%+ consistently. When iMovie isn't running, it's in the single digits. As soon as I load iMovie the cpus max out. This hasn't happened the whole project, just recently. So what changed? I don't know. I've been doing the same thing the whole time, the pasteovers, the add titles, the add background track, saving the project every couple of minutes. Each time I've saved, it looks like it's listing off different files or clips, not just the changed clips. I've tried deleting the QT movie it created in the cache and let it recreate to see if that would help, but it didn't. Would deleting the files in the waveform cache help? I've got about 30 hours into this project already and would love to resurrect this audio if possible. Either that or it's Jim Beam and call it a day. I thought this was non-destructive editing, but apparently that's just the originals? That doesn't help much when the whole project audio goes in the toilet and I'm looking at redoing this whole project. Anyone who can fix this is getting a big ol' sloppy kiss on the mouth and an iTunes gift certificate.

I've never personally seen the audio problems you describe, but there's some things I'd try before giving up. As you've probably read, it's not unusual to have playback problems in projects containing complex audio. The odds for it seem to increased in QuickTime 7.
1. Try opening the "TimeLine Movie.mov" that's inside the Cache folder of the project in QuickTime Player. If the audio plays okay there, the audio data is intact and you're seeing a playback anomaly in iMovie. (QuickTime Player may choke too, but don't conclude anything from that.)
To access the movie, Control-click on the project icon and choose "Show Package Contents".
2. Try exporting ONLY the project audio to an AIFF file, then import THAT back into the project and replace the audio you have. That will "flatten" the audio and make it much easier to play. (To be safe, on a copy of the project.)
Or if you have QuickTime Pro, you might try extracting all the audio from the Timeline movie.
To export to an AIFF file, choose File > Export, click on QuickTime, set the popUp menu to "Expert Settings", press Share, then from the top popUp menu, choose "Sound to AIFF". Choose the level of quality you want from the bottom popUp menu, then Save.
What may be going on here is that extracting audio has created MANY audio files in the project, some possibly very large. (Each time we extract audio from a clip that is part of a Media file shared with other clips, iMovie extracts ALL the audio from that Media file, not just the audio of the selected clip.) To see what audio files the project contains, look in the Media folder of the project.
If you get this solved, please say how.
Oh, and if you haven't updated to iMovie 7.1, you might try that. There were improvements in handling multiple audio tracks, I read somewhere.
BTW, it's usually not a good idea to change things inside the project package. iMovie can get pretty confused. The Timeline movie is replaced each time we save, so that's not a big deal.
Karl

Record streaming audio with XFI Xtreme Audio C

I just got a new pc with this card and unfortunately it has no what u hear or stereo? mix record option.Can I loop the Digital output device(SPDIF) into the Digital input device(SPDIF) to record streaming audio? I can't install a different card because it will void my extended warranty. Thanks

I had no trouble recording "what you hear" streaming audio with my extreme audio (pcie). If you are using the same card I'll walk you through it, if you're using the pci version, take a look to see if it's the same and try it. Open Creative console launcher (by clicking the icon in the system tray), select the mixer, find the record ("rec") slider, (on my console it's on the far right) click the little triangle in the box at the top of the slider, select "what u hear", open creative mediasource, hit record, and start the streaming audio.

Runtime error in Dynamic internal table with AMOUNT and Quantity Fields..

Dear friends,
I am attempting write a dymanic Select Statement (with joins).
And the sleect query looks like this..
SELECT (LT_SEL_LIST)
INTO CORRESPONDING FIELDS OF
TABLE <DYN_TABLE>
FROM (LT_FROM_LIST)
WHERE (LT_WHERE3).
Here the into table is a dynamically created internal table..
which is created by ...this
CALL METHOD CL_ALV_TABLE_CREATE=>CREATE_DYNAMIC_TABLE
    EXPORTING
      IT_FIELDCATALOG = IT_OUTPUT2[]
    IMPORTING
      EP_TABLE        = DY_TABLE.
the it_output2 contains the the fieldcatlog information of dynamically given fields :-
like:-
       TABNAME
       FIELDNAME
       DATATYPE
       LENG
       INTTYPE
       ROLLNAME
       DECIMALS
       REFTABLE
       REFFIELD
i mean the internal table is constructed with reference to all the bove metadata.
Problem:- This query run fine with all the fields Except AMOUNT AND QUANTITY fields....
When the selection list contain VBAK-NETWR or MSEG-MENGE..It throws a runtime error.
"The data read during a SLECT access couldnt be inserted into the target field,either conversion is not supported for
the target field's type or the target field is too short."
after this I even tried to construct the dynamic table with CFILEDNAME and QFIELDNAME in the Fieldcatalog.
so now my fieldcatlog looks like this:---
LOOP AT IT_DD03L..
IF IT_DD03L-DATATYPE = 'CURR'.
       TABLEFIELD-CFIELDNAME = IT_DD03L-FIELDNAME .
       ENDIF.
   IF IT_FIELDCAT3-DATATYPE = 'QUAN'.
       TABLEFIELD-QFIELDNAME = IT_DD03L-FIELDNAME .
ENDIF.
       TABLEFIELD-TABNAME     = IT_DD03L-TABNAME.
       TABLEFIELD-FIELDNAME   = IT_DD03L-FIELDNAME.
       TABLEFIELD-DATATYPE    = IT_DD03L-DATATYPE.
       TABLEFIELD-INTLEN      = IT_DD03L-LENG.
       TABLEFIELD-INTTYPE     = IT_DD03L-INTTYPE .
       TABLEFIELD-ROLLNAME    = IT_DD03L-ROLLNAME.
       TABLEFIELD-DECIMALS    = IT_DD03L-DECIMALS.
       TABLEFIELD-REF_TABLE   = IT_DD03L-REFTABLE.
       TABLEFIELD-REF_FIELD   = IT_DD03L-REFFIELD.
APPEND TABLEFIELD.
CLEAR TABLEFIELD.
ENDLOOP.
Note:- this is a test code so ignore performance issues...
Please help me with some code ...to avoid the Runtime erorr.
Thanks,
jeevan.

Hi Jeevan,
Why are moving only few fields from DD03L table to your field catalog? Why don't you use move-corresponding? The following code works for me in ECC6.0.
data: it_dd03l type table of dd03l initial size 0,
      ls_dd03l type dd03l,
      lt_fldcat TYPE lvc_t_fcat,
      ls_fldcat TYPE lvc_s_fcat,
      ls_where(72) TYPE c,
      lt_where LIKE TABLE OF ls_where,
      lt_fld LIKE TABLE OF ls_where,
      lt_data_dy TYPE REF TO data.
field-symbols: <ft_data> TYPE STANDARD TABLE.
select * into table it_dd03l from dd03l
    where tabname = 'VBAK'
      and ( fieldname = 'VBELN' or fieldname = 'NETWR' ).
check sy-subrc eq 0.
loop at it_dd03l into ls_dd03l.
move-corresponding ls_dd03l to ls_fldcat.
append ls_fldcat to lt_fldcat.
move ls_dd03l-fieldname to ls_where.
append ls_where to lt_fld.
if ls_dd03l-fieldname = 'VBELN'.
    clear ls_where.
    concatenate ls_dd03l-fieldname ' <> ''''' into ls_where.
    append ls_where to lt_where.
endif.
endloop.
check not lt_fldcat is initial.
CALL METHOD cl_alv_table_create=>create_dynamic_table
      EXPORTING
        it_fieldcatalog           = lt_fldcat
      IMPORTING
        ep_table                  = lt_data_dy
      EXCEPTIONS
        generate_subpool_dir_full = 1
        OTHERS                    = 2.
    IF sy-subrc <> 0.
      RAISE no_configuration_data.
    ENDIF.
    ASSIGN lt_data_dy->* TO <ft_data>.
check sy-subrc eq 0.
select (lt_fld) from VBAK into corresponding fields of table
    <ft_data>
    where (lt_where).
Thanks
Bala

[Solved] Recording internal audio with alsa and recordmydesktop

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