Play Audio On Webpage With Button Click

Hello everyone. I'm new to Flash, so please bear with me.
I'm creating a language learning website, and wish to add
audio to it for all vocabulary words. I want the audio to be
compatible across browsers and simply play when a button is
clicked, without any controls or media players opening up.
From what I hear, it sounds like Flash is a good way to go
and would do the trick. It's just a matter of converting all the
mp3s to Flash. The mp3 file size is already relatively small (1-2
seconds in length each and about 30kb in size I believe).
How could I create a button that plays the mp3 when it is
clicked? I was
researching the topic and came across a few things.
What I am looking for is almost identical to this:
Site
demonstrating audio buttons
While I could jump right in and use her example, it doesn't
quite work. With hers, everything needs to be in different folders,
and I would need a different button - her's is a little big. I
can't put everything in different folders, because I will have
hundreds of audio files. It would be much easier to put them all in
an audio folder with different filenames.
I also was reading at another site that it deals with
Javascript controlling the Flash - when you click the button it
plays the swf file. Since I am very new to Flash, I'm not exactly
sure of what all it can and cannot do. The site also mentioned not
all browsers allow Javascript to control Flash files. Again,
however, I'm unsure of how accurate the information still remains.
The browsers were older versions, and they were talking about how
to do it in Flash 5 and Dreamweaver 4. Furthermore, the example
located at the above link does not use any Javascript.
Does anyone have any ideas on how I can do this? While I am
beginning to learn Flash, I doubt I will understand what to do all
on my own for a while, and I need to get the site finished as soon
as possible. The audio is practically all that is left before the
site launch. Like I said, I'm looking for a way to add audio to a
webpage, that is played (without media players opening up, etc)
with the push of a button, and also will load quickly, since there
were be about 20 audio files per page.

Here is some info on sound linkage, not sure if this is what
you want but its a start:
http://www.adobe.com/cfusion/webforums/forum/messageview.cfm?forumid=15&catid=194&threadid =1208568

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Hi, everyone,
If we can record audio from Flash Player 10 inside of a webpage and save audio files onto our local places by the use of mic applications, it will be very convenient. In fact, we can turn to use other available programs to record audio from flash and save locally by clicking buttons. It is possible to use recorders to record audio on flash or any other media players.
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I've read through several posts that asked similar questions, but none of them seemed to have the exact same issue, and then I just couldn't wrap my head around some of the answers. So I decided to ask my own question.
My button is buried within 2 levels of symbols and I want to play the timeline within my current symbol as well as the timeline in the symbol one level up. This is what my hierarchy looks like - stage / fullRadial / smRadial / button
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Sent from Mailbox
On Thu, Jun 19, 2014 at 10:13 AM, TripleRRR <[email protected]>

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Below is an example of what I'm trying to do.

Alrighty I changed the publish settings to as3. I'm still getting two errors:
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Now the problem is how do I change the JPanel in the main class with a click on the button in TOC class?
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Not sure what you mean exactly. However,
Assuming that your view class is a JPanel, when you create your controller,
ViewPanel view = new ViewPanel();
MainController controller = new MainController(viewPanel);
In MainController track a reference to your viewPanel. Also in viewPanel have a method called setCurrentPanel(JPanel panel). then int he controller's action listener
public void actionPerformed(ActionEvent ev) {
JPanel pnl = (JPanel) panelMap.get(ev.getSource().getName());
viewPanel.setCurrentPanel(pnl);
I am assuming you have a way of mapping each button to the panel it will use, which is what is in the panelMap. The viewPanel can have a CardPanel layout for the panels it displays and its setCurrentPanel, can bring to front/oradd the specified panel.
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Play audio from hovering over button?

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Thanks,
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That's not something that menus support in DVD-Video, but you can fake it by using multiple menus, with each menu having the unique audio clip for a background, and then link each menu to a button on each menu. Complete the illusion by setting each button to auto-activate.
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Logic Pro 9 - Recorded audio just silly with crackling, clicking noise

PLEASE!!! Read my entire post before commenting. I don't mean to sound rude but I've posted this in several other forums and it's obvious they hadn't read/understood my issue based on the comments they left, ie., suggesting everything I've already tried, saying to contact tech support and even asking what comp and DAW I'm using... I've been dealing with this issue for ~8 months
SPECS
I use 2 daisy-chained Firepods connected to a Mac Pro Quad-core running 10.6.5 and Logic Pro 9.1.3.
I get the results described below whether I'm trying to record at 24 or 16 bit and at either 44.1K or 48K.
Generally I record 12 tracks simultaneously AND I always use the same inputs for each instrument; ie., Vocals = track 11, guitar = track 2, bass = track 1, etc...*
PROBLEM
Although this is a sporadic issue it happens to occur WAY more than it doesn't; ie., very rarely do I get a recording session without the anomaly.
The issue is that after I have recorded a session and listen back to the recorded material there are random crackles, pops, clicks throughout. When I audition tracks individually, some may have no crackling and some will have crackling at different points in time and of varying intensity. Every session is different as well, no two sessions are the same. One day I may have a perfect vocal track and a guitar track that is just silly with the crackling. The next day the guitar track may be fine but now the vocal track is all messed up with crackling. Another confounding point is that the crackling does not occur during periods of silence.
I will add that each Firepod appears to function properly (with no issue) all by it's lonesome - but that doesn't help me, I need 12 inputs minimum!
Things I've Tried
1. All new FW cables
2. Disconnecting the FW cable that goes from the Firepod to the Mac. Turn on the Mac. Turn on both Firepods, master first. Connect the FW cable that goes from the Firepod to the Mac. Launch Logic.
3. Disconnecting the FW cable that goes from the Firepod to the Mac. Turn on the Mac. Launch Logic. Turn on both Firepods, master first. Connect the FW cable that goes from the Firepod to the Mac.
4. Tried every conceivable configuration of Audio Midi Setup.
5. Tried several other FW and USB audio interfaces, obtaining perfect results.
6. Been working with Apple tech support (both hardware and Logic support).
7. I've checked that both Firepods have the same firmware version.
8. Installed a fresh SATA HD with clean installs of both OS10.6.5 and Logic Pro 9.1.3 with fresh custom templates and no 3rd party plugins or loops.
9. Lots of praying, swearing and fits of anger.
Final Comments
First, I think that the chassis FW connectors on the firepods are absolute garbage (read, cheap, cheap, cheap) and are inferior to anything else I have encountered during my troubleshooting process.
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Any guidance will be greatly appreciated.

Hi Pancenter! I'll try to be brief. LOL
I had 4 perfect multi-track sessions and then the gremlin woke up and and cast his spell... Frustrating... The biggest difference I enjoyed using your suggestion was that when I launched Logic, loaded my template, enabled my tracks and hit record, the session started without a hitch. Prior to your suggestion, I would have to futz around with Logic audio prefs, AMS settings, turning the Firepods off/on, quitting/restarting Logic and/or shutting down/restarting my Mac Pro just to get the session to show my enabled tracks had activity. I would not know if the session would have the noise artifacts until playback. I should add that I never had this problem with my PPC G5 nor did I for the first 10 months of use with this Mac Pro. Also, with the help of Apple tech support we had eliminated Logic, 10.6.5 and the Mac Pro (except the FW800 bus) as probable causes. Again each Firepod and Logic work flawlessly when I use one unit not daisy-chained. So, I think there is no problem with the individual units, and, as I said, both units worked fine chained with my PPC Mac Pro.
So, I called Presonus. Again... And I get yet another tech who tells me something different; every time I speak with Presonus I get different suggestions. This guy asked what kind of cable I was using to connect the Firepods to the Mac Pro and I said I'm using a *firewire 9-pin to 6-pin cable*. So has asks well, is it *bilingual*? He said that would def fix my problem. I told him it is advertized as bilingual and that had not fixed the problem; he had no answer for that. Then, I broached the option of adding a firewire 400 card to my Mac Pro. He said I guess you could try that; get one with a TI chipset. That's where I'm at now, deciding which one to get and whether I will need to provide auxiliary power to the card's on-board molex plug.
Any ideas?
Oh, thanks for responding

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Can someone confirm if I can kick off a playlist from ATV on my HDTV, but then shut the darned TV off and listen to the music through my receiver? I don't want to needlessly burn my TV out when I just want to listen to tunes.
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It works. I use optical digital to my Yamaha and turn off my HDMI connected Toshiba. I've even sent Apple a suggestion that they should support a remote control sequence that justs starts a shuffle play without the need to turn on the monitor to go through menus.
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Message was edited by: xnav

Opening a frame with button click

hi all
when i am going to click a button which is placed in JTabel it should
open a JFrame with some component.
thanks & regards
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This task can be decomposed in two logically unrelated subtasks:
1: Adding a button in a table column and listening to it
2: Showing a frame from some method
In case you are hesitant how to solve the first task: you need to create your own cell editor (possibly by extending DefaultCellEditor) that returns a button as editor component. Of course this editor must be set to the desired column(that must be editable too). And just listen to that button's actions.
Another possible solution is to just render the column with a button and listen the table for mouse clicks, then determine the ros/column for the click and act correspondingly.
The solution of the second task depends entirely on you.
HTH
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[Solved] ALSA - Play Audio As Is With No Conversion

Hello all,
I have an issue with ALSA I hope you can help me sort out.
For the sake of this post, I'll refer to everything as if played by mplayer2, since the output is much easier this way.
Basically, I want to play everything "as-is", in regards to samplerate and format.
That is, playing music at 16bit/44.1Khz as such, playing movie audio at 16bit/48Khz as such, playing HD audio at 24bit/96Khz as such and so on.
Now, my first attempt was trying to access the HW directly using mplayer's --ao=alsa:device=hw=0.0 and not limiting the format.
Both samples played at correct sample rate but were outputted at 16 bit, since the hardware supposedly do not support floating point, and the output reverted to default.
Selected audio codec: FLAC (Free Lossless Audio Codec) [libavcodec]
AUDIO: 44100 Hz, 2 ch, s16le, 0.0 kbit/0.00% (ratio: 0->176400)
[AO_ALSA] Format floatle is not supported by hardware, trying default.
AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample)
Selected audio codec: Uncompressed PCM [pcm]
AUDIO: 96000 Hz, 2 ch, s24be, 4608.0 kbit/100.00% (ratio: 576000->576000)
[AO_ALSA] Format floatle is not supported by hardware, trying default.
AO: [alsa] 96000Hz 2ch s16le (2 bytes per sample)
Note the Format floatle is not supported by hardware part.
This seemed a bit weird as using alsa without accessing the hardware directly (going through vmix), the output is indeed floating point.
That is, playing with --ao=alsa I get the following (obviously everything is sampled to the default 48Khz):
Selected audio codec: FLAC (Free Lossless Audio Codec) [libavcodec]
AUDIO: 44100 Hz, 2 ch, s16le, 0.0 kbit/0.00% (ratio: 0->176400)
AO: [alsa] 48000Hz 2ch floatle (4 bytes per sample)
Selected audio codec: Uncompressed PCM [pcm]
AUDIO: 96000 Hz, 2 ch, s24be, 4608.0 kbit/100.00% (ratio: 576000->576000)
AO: [alsa] 48000Hz 2ch floatle (4 bytes per sample)
Note that the output format is floatle.
So, I tried specifying --ao=alsa:device=hw=0.0 --format=floatle, only for it to fail as in the first case and reverting to default 16 bit and correct sample rate.
Specifying the required format directly works for the flac sample (s16le, same as the default) but fails for the HD sample (s24be), as expected - I can remember from my Windows days not so long ago I had to output 24 bit audio padded as 32 bit for it to work.
This lead me to my next attempt: --ao=alsa:device=hw=0.0 --format=s32le, which as far as I understand is 32 bit, but not floating point, which isn't exactly what I need.
Anyway, both files played fine with this mode for the few seconds I tried (assuming the remaining bits were padded with zeros), but real-life files (such as movie soundtrack) produced noticeable audio distortions (cracks and pops of all kinds).
So, I'm back to where I began.
Seems like I need to specify the hardware (device=..._) to get correct sample rate, but then I can't get correct format, or let it go via dmix and then the sample rate is wrong.
Now, having giving it a bit more thought, I realized dmix is outputting anything successfully as floating point, so I guess it pads whatever bits needed with zeros, which is just fine.
Assuming the above is correct, is it possible to write something in my .asoundrc file so that input samplerate is preserved?
I'm not very optimistic as it kind of against the whole idea of software mixing, but who knows..
If there's any way of convincing ALSA that my card is perfectly fine when being fed with floating point directly (device=...), that's an option too.
Bottom line, is there a way to make everything sent to card as-is?
That was long... sorry
Thanks, Adam.
Last edited by adam777 (2012-10-03 07:45:17)

adam777 wrote:
Hello all,
I have an issue with ALSA I hope you can help me sort out.
For the sake of this post, I'll refer to everything as if played by mplayer2, since the output is much easier this way.
Basically, I want to play everything "as-is", in regards to samplerate and format.
That is, playing music at 16bit/44.1Khz as such, playing movie audio at 16bit/48Khz as such, playing HD audio at 24bit/96Khz as such and so on.
Sounds reasonable...
adam777 wrote:
Now, my first attempt was trying to access the HW directly using mplayer's --ao=alsa:device=hw=0.0 and not limiting the format.
Both samples played at correct sample rate but were outputted at 16 bit, since the hardware supposedly do not support floating point, and the output reverted to default.
Note the Format floatle is not supported by hardware part.
This seemed a bit weird as using alsa without accessing the hardware directly (going through vmix), the output is indeed floating point.
So yes, your hardware doesn't support floating point. Don't compare the hw device to the more higher level interfaces (default,surround,plughw). The higher level interfaces provide additional sample formats which will be converted on the fly to the specific hw device capabilities.
adam777 wrote:Bottom line, is there a way to make everything sent to card as-is?
That would depend on the input format. Even some 24bit audio files may need some additional padding. If you want to know what your card supports, compile and run the following program: http://www.volkerschatz.com/noise/alsacap.c.
For example, this the output on my machine:
Card 0, ID `Intel', name `HDA Intel'
Device 0, ID `ALC268 Analog', name `ALC268 Analog', 1 subdevices (1 available)
2 channels, sampling rate 44100..192000 Hz
Sample formats: S16_LE, S32_LE
Subdevice 0, name `subdevice #0'
Last edited by GogglesGuy (2012-10-02 03:09:43)

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Hi folks!
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while true; do bc -l <<< "$(cat /sys/class/power_supply/BAT0/current_now)/10^6*$(cat /sys/class/power_supply/BAT0/voltage_now)/10^6"; sleep 1; done
The output of lspci -vvv:
00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)
Subsystem: Lenovo Device 215e
Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx-
Latency: 0, Cache Line Size: 64 bytes
Interrupt: pin B routed to IRQ 43
Region 0: Memory at f2520000 (64-bit, non-prefetchable) [size=16K]
Capabilities: [50] Power Management version 2
Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+)
Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME-
Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+
Address: 00000000fee0f00c Data: 41c1
Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00
DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us
ExtTag- RBE- FLReset+
DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported-
RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+
MaxPayload 128 bytes, MaxReadReq 128 bytes
DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend-
LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us
ClockPM- Surprise- LLActRep- BwNot-
LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk-
ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt-
LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt-
Capabilities: [100 v1] Virtual Channel
Caps: LPEVC=0 RefClk=100ns PATEntryBits=1
Arb: Fixed- WRR32- WRR64- WRR128-
Ctrl: ArbSelect=Fixed
Status: InProgress-
VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans-
Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256-
Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01
Status: NegoPending- InProgress-
VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans-
Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256-
Ctrl: Enable+ ID=1 ArbSelect=Fixed TC/VC=02
Status: NegoPending- InProgress-
Capabilities: [130 v1] Root Complex Link
Desc: PortNumber=0f ComponentID=00 EltType=Config
Link0: Desc: TargetPort=00 TargetComponent=00 AssocRCRB- LinkType=MemMapped LinkValid+
Addr: 00000000fed1c000
Kernel driver in use: snd_hda_intel
Kernel modules: snd-hda-intel
cat /proc/asound/card0/codec#0:
Codec: Conexant CX20585
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x14f15069
Subsystem Id: 0x17aa2155
Revision Id: 0x100302
No Modem Function Group found
Default PCM:
rates [0x160]: 44100 48000 96000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Default Amp-In caps: N/A
Default Amp-Out caps: N/A
GPIO: io=4, o=0, i=0, unsolicited=1, wake=0
IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L
Control: name="Master Playback Volume", index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Device: name="CONEXANT Analog", type="Audio", device=0
Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1
Amp-Out vals: [0x3b 0x3b]
Converter: stream=8, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L
Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1
Amp-Out vals: [0x3c 0x3c]
Converter: stream=0, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital
Converter: stream=0, channel=0
Digital:
Digital category: 0x0
PCM:
rates [0x160]: 44100 48000 96000
bits [0xe]: 16 20 24
formats [0x5]: PCM AC3
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out
Control: name="Beep Playback Volume", index=0, device=0
ControlAmp: chs=1, dir=Out, idx=0, ofs=0
Control: name="Beep Playback Switch", index=0, device=0
ControlAmp: chs=1, dir=Out, idx=0, ofs=0
Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0
Amp-Out vals: [0x00]
Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L
Device: name="CONEXANT Analog", type="Audio", device=0
Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1
Amp-In vals: [0x81 0x81] [0x80 0x80] [0x81 0x81] [0x80 0x80]
Converter: stream=0, channel=0
SDI-Select: 0
PCM:
rates [0x160]: 44100 48000 96000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 4
0x17 0x18 0x23* 0x24
Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L
Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1
Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a]
Converter: stream=0, channel=0
SDI-Select: 0
PCM:
rates [0x160]: 44100 48000 96000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 4
0x17* 0x18 0x23 0x24
Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L
Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1
Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a]
Converter: stream=0, channel=0
SDI-Select: 0
PCM:
rates [0x160]: 44100 48000 96000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 4
0x17* 0x18 0x23 0x24
Node 0x17 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out
Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0
Amp-Out vals: [0x02 0x02]
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 4
0x1a 0x1b* 0x1d 0x1e
Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out
Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0
Amp-Out vals: [0x00 0x00]
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 4
0x1a* 0x1b 0x1d 0x1e
Node 0x19 [Pin Complex] wcaps 0x400581: Stereo
Pincap 0x0000001c: OUT HP Detect
Pin Default 0x042140f0: [Jack] HP Out at Ext Right
Conn = 1/8, Color = Green
DefAssociation = 0xf, Sequence = 0x0
Pin-ctls: 0x00:
Unsolicited: tag=37, enabled=1
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 2
0x10* 0x11
Node 0x1a [Pin Complex] wcaps 0x400481: Stereo
Pincap 0x00001324: IN Detect
Vref caps: HIZ 50 80
Pin Default 0x61a190f0: [N/A] Mic at Sep Rear
Conn = 1/8, Color = Pink
DefAssociation = 0xf, Sequence = 0x0
Pin-ctls: 0x00: VREF_HIZ
Unsolicited: tag=38, enabled=1
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Node 0x1b [Pin Complex] wcaps 0x400581: Stereo
Pincap 0x00011334: IN OUT EAPD Detect
Vref caps: HIZ 50 80
EAPD 0x2: EAPD
Pin Default 0x04a190f0: [Jack] Mic at Ext Right
Conn = 1/8, Color = Pink
DefAssociation = 0xf, Sequence = 0x0
Pin-ctls: 0x00: VREF_HIZ
Unsolicited: tag=38, enabled=1
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 2
0x10* 0x11
Node 0x1c [Pin Complex] wcaps 0x400581: Stereo
Pincap 0x00000014: OUT Detect
Pin Default 0x612140f0: [N/A] HP Out at Sep Rear
Conn = 1/8, Color = Green
DefAssociation = 0xf, Sequence = 0x0
Pin-ctls: 0x40: OUT
Unsolicited: tag=37, enabled=1
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 2
0x10* 0x11
Node 0x1d [Pin Complex] wcaps 0x400581: Stereo
Pincap 0x00010034: IN OUT EAPD Detect
EAPD 0x2: EAPD
Pin Default 0x601700f0: [N/A] Speaker at Sep N/A
Conn = Analog, Color = Unknown
DefAssociation = 0xf, Sequence = 0x0
Pin-ctls: 0x20: IN
Unsolicited: tag=00, enabled=0
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 2
0x10* 0x11
Node 0x1e [Pin Complex] wcaps 0x400481: Stereo
Pincap 0x00000024: IN Detect
Pin Default 0x40f001f0: [N/A] Other at Ext N/A
Conn = Unknown, Color = Unknown
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x20: IN
Unsolicited: tag=00, enabled=0
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Node 0x1f [Pin Complex] wcaps 0x400501: Stereo
Pincap 0x00000010: OUT
Pin Default 0x901701f0: [Fixed] Speaker at Int N/A
Conn = Analog, Color = Unknown
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x40: OUT
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 2
0x10* 0x11
Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital
Pincap 0x00000010: OUT
Pin Default 0x40f001f0: [N/A] Other at Ext N/A
Conn = Unknown, Color = Unknown
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x40: OUT
Unsolicited: tag=00, enabled=0
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 1
0x12
Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital
Converter: stream=0, channel=0
Digital:
Digital category: 0x0
PCM:
rates [0x160]: 44100 48000 96000
bits [0xe]: 16 20 24
formats [0x5]: PCM AC3
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Node 0x22 [Pin Complex] wcaps 0x400781: Stereo Digital
Pincap 0x00000010: OUT
Pin Default 0x40f001f0: [N/A] Other at Ext N/A
Conn = Unknown, Color = Unknown
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x40: OUT
Unsolicited: tag=00, enabled=0
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 1
0x21
Node 0x23 [Pin Complex] wcaps 0x40040b: Stereo Amp-In
Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x2f, mute=0
Amp-In vals: [0x02 0x02]
Pincap 0x00000020: IN
Pin Default 0x90a601f0: [Fixed] Mic at Int N/A
Conn = Digital, Color = Unknown
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x20: IN
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In
Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1
Amp-In vals: [0x00 0x00] [0x00 0x00]
Power states: D0 D1 D2 D3 D3cold
Power: setting=D0, actual=D0
Connection: 2
0x10 0x11
Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono
Is anyone able to confirm this problem or does anyone have a solution? I’m really desperate since I often use my Thinkpad as a big but wonderful portable music player (with headphones of course! Don’t want to annoy my fellow trainpassengers …).
Thanks in advance
Michael

Any updates on this?

Changing activ tab with button click

Hello,
I have a tabstrip with several tabs. On Tab2 I have a list of invoices. If the user clicks on the copy button I want to copy several fields into a form on tab1. That is working well. I also want to change the activ tab from Tab2 to Tab1 so that the user is facing the formular but I do not know how to do it.
regards
stefan

Hi Stefan,
There is property for TabStrip UI element "selectedTab" (http://help.sap.com/saphelp_nw2004s/helpdata/en/f0/e5a8411fdbcc46e10000000a155106/frameset.htm). Bind it to context attibute and change in action handler accordingly.
Best regards, Maksim Rashchynski.

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Hello all,
I've got a file on working on that has 4 different movie clips on one layer that, when clicked, will zoom in (made with tweens in the timeline).
The issue I'm having, is that when I click one of them, they are arranged in a way that won't work for what I need. I click one, and it works, but it is hidden by the movie clip that is on top of it in the heirarchy. Click another, zooms in, but is partially hidden by another one.
My question is: Is there anyway in AS2 to code these so that when one is clicked, it will the top spot in the heriarchy? So that I can code them all like this so that whenever one is clicked, it goes to the top so it is not hidden?
Thanks

Correct me if I'm wrong, but isn't that just used to swap 2 movie clips?
Or is there a number I can input instead of the movie clip name that will change all of them?

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