[SOLVED] Pulse Audio / ALSA audio sharing.

SOLUTION AT POST # 4
My computer has integrated audio and HDMI. From my understanding, to be able to switch between these two types of audio hardware I require a sound server. If this is not correct, please let me know so that I can uninstall Pulse Audio. Assuming it is correct, I have numerous problems with my audio. When running multiple programs, often I have to open pavucontrol and move the volume slider around in order for the audio to play in a newly opened program. I have XFCE's volume control set to ALSA, and I am unable to change it to Pulse Audio for some reason. This is a problem because every time I mute or change the volume using XFCE's volume control, I then have to go into Pulse Audio later to change the audio back. If I run a game or VLC, and then open Firefox, I get no Flash audio. However, if I run Firefox first, I have no audio in VLC or in game. Usually I can fix this by opening pavucontrol and messing with the volume sliders. This is not ideal. Ultimately it seems the problem is that the program I am using are not sharing the audio hardware, so only one can play audio at a time.
What can I do to make my audio work as it should?
Here is my audio hardware
# lspci | grep -i audio
00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05)
Screenshot of the only sound card option I have in the XFCE Audio Mixer Plugin:
http://imgur.com/3TJWC
I searched the Wiki, but it is out of date. https://wiki.archlinux.org/index.php/PulseAudio#ALSA
. . . For the applications that do not support PulseAudio and support ALSA it is recommended to install the PulseAudio plugin for ALSA. This package also contains the necessary /etc/asound.conf for configuring ALSA to use PulseAudio.
To prevent applications from using ALSA's OSS emulation and bypassing Pulseaudio (thereby preventing other applications from playing sound), make sure the module snd_pcm_oss is not in the MODULES array in /etc/rc.conf. If it is currently loaded (lsmod|grep oss), disable it by executing: . . .
How can I do this using systemd instead of rc.conf?
Edit: Since there is a lack of response, I am completely changing the entire post in order to be clear and productive.
Last edited by qKUqm3wtY4 (2013-01-05 04:52:15)

I found a solution here: http://forums.opensuse.org/english/get- … audio.html
You must create /etc/asound.conf and insert the following:
pcm.pulse {
type pulse
ctl.pulse {
type pulse
pcm.!default {
type pulse
ctl.!default {
type pulse
pcm.phononpulse {
type plug
slave.pcm {
type pulse
hint {
show on
description "PulseAudio"

Similar Messages

The "alsa audio output error: cannot write: Broken pipe" problem

[Solved]
(No place for this word in the Subject field, so I put it here.)
Hi Arch community!
I noticed I have this problem and not sure when in started. Several months ago I bought and installed new videocard "GeForce GT 630". Right after installation I lost sound in the system. Having internal sound card (in mainboard) and SB Audigy2 (PCI), I searched for solution and found I had to disable "snd_hda_intel" module to restore sound to work through PCI sound card. Did it.
echo "blacklist snd_hda_intel" >> /etc/modprobe.d/no-nvidia-sound.conf
After that sound was working as it should. And I noticed recently that during playing of any sound file the sound interrupts. If run VLC from CLI I get "alsa audio output error: cannot write: Broken pipe" message. And my system hasn't been upgraded since I don't know when. I still ran Firefox 26. So googled again and found it's a bug in a kernel or somewhere else.
==========================================
https://bugs.freedesktop.org/show_bug.cgi?id=69942 :
2013-10-01 01:39:40 UTC
I ran a system update last night, and this issue appears to have been corrected.
==========================================
And so I thought upgrade would fix my problem since mainstream bug is fixed. After upgrade sound interrupts again, maybe even more often.
Do you guys have this issue and does anyone have a solution?
Last edited by Mr. Alex (2014-06-26 18:55:59)

Not particularly wise to blacklist the module, since it probably is also used by the integrated sound card. With vid and pid you could just disable the nvidia card alone using parameter enable=0. Unless you do not mind having both disabled.
Where do I need to input "enable=0"? When I was searching for solution to make my Creative card work with new nVidia card I installed, the only way I found was blacklisting the module.
dmesg says:
[ 20.448086] nvidia 0000:02:00.0: irq 42 for MSI/MSI-X
[ 21.112331] NVRM: Your system is not currently configured to drive a VGA console
[ 21.112340] NVRM: on the primary VGA device. The NVIDIA Linux graphics driver
[ 21.112345] NVRM: requires the use of a text-mode VGA console. Use of other console
[ 21.112349] NVRM: drivers including, but not limited to, vesafb, may result in
[ 21.112352] NVRM: corruption and stability problems, and is not supported.
Then:
lsmod | grep 'snd'
snd_ca0106 36242 0
snd_ac97_codec 114896 1 snd_ca0106
ac97_bus 1270 1 snd_ac97_codec
snd_rawmidi 20278 1 snd_ca0106
snd_seq_device 5244 1 snd_rawmidi
snd_pcm 83207 2 snd_ac97_codec,snd_ca0106
snd_timer 19294 1 snd_pcm
snd 60780 6 snd_ac97_codec,snd_timer,snd_pcm,snd_rawmidi,snd_seq_device,snd_ca0106
soundcore 5551 1 snd
I guess my card is "snd_ca0106". If yes,
# systool -v -m snd_ca0106
Module = "snd_ca0106"
Attributes:
coresize = "36242"
initsize = "0"
initstate = "live"
refcnt = "0"
taint = ""
uevent = <store method only>
Parameters:
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"
subsystem = "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"
Sections:
.bss = "0xffffffffa0814780"
.data = "0xffffffffa0813300"
.exit.text = "0xffffffffa0811a1e"
.gnu.linkonce.this_module= "0xffffffffa0814500"
.init.text = "0xffffffffa0817000"
.note.gnu.build-id = "0xffffffffa0811a30"
.rodata = "0xffffffffa0811a80"
.rodata.str1.1 = "0xffffffffa08123dc"
.rodata.str1.8 = "0xffffffffa0812130"
.strtab = "0xffffffffa0818c10"
.symtab = "0xffffffffa0817020"
.text = "0xffffffffa080d000"
__mcount_loc = "0xffffffffa0812f98"
__param = "0xffffffffa0812f18"
__verbose = "0xffffffffa0814460"
Where would I look for "max_buffer_size" parameter?

[SOLVED] MPlayer/Pulseaudio, No audio output

I can't seem to figure out how to make mplayer's "ao=pulse" option to work.
It works fine with alsa, but I couldn't find anything definitive on Google about what might be wrong. I think one of the problems might be that I couldn't find any errors in mplayer's output.
Here is the verbose output of an example file:
$ mplayer -v example.mp4
MPlayer SVN-r35107-4.7.2 (C) 2000-2012 MPlayer Team
CPU vendor name: GenuineIntel max cpuid level: 11
CPU: Intel(R) Core(TM) i5 CPU M 430 @ 2.27GHz (Family: 6, Model: 37, Stepping: 2)
extended cpuid-level: 8
extended cache-info: 16801856
Detected cache-line size is 64 bytes
CPUflags: MMX: 1 MMX2: 1 3DNow: 0 3DNowExt: 0 SSE: 1 SSE2: 1 SSSE3: 1
Compiled with runtime CPU detection.
get_path('codecs.conf') -> '/home/user/.mplayer/codecs.conf'
Reading optional codecs config file /home/mike/.mplayer/codecs.conf: No such file or directory
Reading optional codecs config file /etc/mplayer/codecs.conf: 198 audio & 409 video codecs
init_freetype
Using MMX (with tiny bit MMX2) Optimized OnScreenDisplay
get_path('fonts') -> '/home/user/.mplayer/fonts'
Configuration: --prefix=/usr --enable-runtime-cpudetection --disable-gui --disable-arts --disable-liblzo --disable-speex --disable-cdparanoia --disable-openal --disable-libdv --disable-musepack --disable-esd --disable-mga --disable-ass-internal --enable-xvmc --enable-radio --enable-radio-capture --disable-vdpau --enable-vaapi --language=all --confdir=/etc/mplayer
CommandLine: '-v' 'example.mp4'
Using nanosleep() timing
get_path('input.conf') -> '/home/user/.mplayer/input.conf'
Reading optional input config file /home/user/.mplayer/input.conf: No such file or directory
Parsing input config file /etc/mplayer/input.conf
Input config file /etc/mplayer/input.conf parsed: 92 binds
get_path('example.mp4.conf') -> '/home/user/.mplayer/example.mp4.conf'
Playing example.mp4.
get_path('sub/') -> '/home/user/.mplayer/sub/'
[file] File size is 2285184 bytes
STREAM: [file] example.mp4
STREAM: Description: File
STREAM: Author: Albeu
STREAM: Comment: based on the code from ??? (probably Arpi)
libavformat version 54.25.104 (internal)
Configuration: --enable-gpl --enable-postproc
LAVF_check: QuickTime / MOV
libavformat file format detected.
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]ISO: File Type Major Brand: mp42
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]File position before avformat_find_stream_info() is 10279
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]first_dts 14 not matching first dts 0 in que
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]first_dts 14 not matching first dts 0 in que
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]first_dts 14 not matching first dts 0 in que
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]first_dts 14 not matching first dts 0 in que
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]first_dts 14 not matching first dts 0 in que
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]All info found
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]File position after avformat_find_stream_info() is 31013
==> Found video stream: 0
======= VIDEO Format ======
biSize 79
biWidth 640
biHeight 360
biPlanes 0
biBitCount 24
biCompression 875967048='H264'
biSizeImage 691200
Unknown extra header dump: [1] [42] [c0] [1e] [ff] [e1] [0] [18] [67] [42] [c0] [1e] [da] [2] [80] [bf] [e5] [84] [0] [0] [3] [0] [4] [0] [0] [3] [0] [c0] [3c] [58] [ba] [80] [1] [0] [4] [68] [ce] [3c] [80]
===========================
[lavf] stream 0: video (h264), -vid 0
==> Found audio stream: 1
======= WAVE Format =======
Format Tag: 20557 (0x504D)
Channels: 2
Samplerate: 44100
avg byte/sec: 12021
Block align: 1
bits/sample: 16
cbSize: 16
Unknown extra header dump: [12] [10] [0] [0] [0] [0] [0] [0] [0] [0] [0] [0] [0] [0] [0] [0]
==========================================================================
[lavf] stream 1: audio (aac), -aid 0, -alang und
LAVF: 1 audio and 1 video streams found
LAVF: build 3545448
VIDEO: [H264] 640x360 24bpp 24.000 fps 562.6 kbps (68.7 kbyte/s)
[V] filefmt:44 fourcc:0x34363248 size:640x360 fps:24.000 ftime:=0.0417
Clip info:
major_brand: mp42
minor_version: 0
compatible_brands: isommp42
creation_time: 2013-03-18 01:56:42
Load subtitles in ./
get_path('sub/') -> '/home/user/.mplayer/sub/'
[mov,mp4,m4a,3gp,3g2,mj2 @ 0xeed4a0]demuxer injecting skip 2112
X11 opening display: :0
vo: X11 color mask: FFFFFF (R:FF0000 G:FF00 B:FF)
vo: X11 running at 1600x900 with depth 24 and 32 bpp (":0" => local display)
[x11] Detected wm supports NetWM.
[x11] Detected wm supports FULLSCREEN state.
[x11] Detected wm supports ABOVE state.
[x11] Detected wm supports BELOW state.
[x11] Current fstype setting honours FULLSCREEN ABOVE BELOW X atoms
[VO_XV] Using Xv Adapter #0 (Intel(R) Textured Video)
[xv common] Drawing no colorkey.
[xv common] Maximum source image dimensions: 2048x2048
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
libavcodec version 54.53.100 (internal)
Configuration: --enable-gpl --enable-postproc
INFO: libavcodec init OK!
Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264)
==========================================================================
==========================================================================
Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
dec_audio: Allocating 1536000 + 131072 = 1667072 bytes for output buffer.
FFmpeg's libavcodec audio codec
INFO: libavcodec "aac" init OK!
AUDIO: 44100 Hz, 2 ch, s16le, 96.2 kbit/6.81% (ratio: 12021->176400)
Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio))
==========================================================================
Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??...
[libaf] Adding filter dummy
[dummy] Was reinitialized: 44100Hz/2ch/s16le
[dummy] Was reinitialized: 44100Hz/2ch/s16le
Trying preferred audio driver 'pulse', options '[none]'
AO: [pulse] 44100Hz 2ch s16le (2 bytes per sample)
AO: Description: PulseAudio audio output
AO: Author: Lennart Poettering
Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le...
[dummy] Was reinitialized: 44100Hz/2ch/s16le
[dummy] Was reinitialized: 44100Hz/2ch/s16le
Starting playback...
Increasing filtered audio buffer size from 0 to 46144
Unsupported PixelFormat 61
Unsupported PixelFormat 81
Unsupported PixelFormat 61
[VD_FFMPEG] Trying pixfmt=0.
[ffmpeg] aspect_ratio: 0.000000
Unsupported PixelFormat 61
VDec: vo config request - 640 x 360 (preferred colorspace: Unknown 0x0000)
Trying filter chain: vo
Could not find matching colorspace - retrying with -vf scale...
Opening video filter: [scale]
SwScale params: -1 x -1 (-1=no scaling)
Trying filter chain: scale vo
The selected video_out device is incompatible with this codec.
Try appending the scale filter to your filter list,
e.g. -vf spp,scale instead of -vf spp.
[VD_FFMPEG] Trying pixfmt=1.
[ffmpeg] aspect_ratio: 0.000000
VDec: vo config request - 640 x 360 (preferred colorspace: H.264 VA-API Acceleration)
Trying filter chain: vo
Could not find matching colorspace - retrying with -vf scale...
Opening video filter: [scale]
SwScale params: -1 x -1 (-1=no scaling)
Trying filter chain: scale vo
The selected video_out device is incompatible with this codec.
Try appending the scale filter to your filter list,
e.g. -vf spp,scale instead of -vf spp.
Unsupported PixelFormat 81
[VD_FFMPEG] Trying pixfmt=2.
[ffmpeg] aspect_ratio: 0.000000
Unsupported PixelFormat 81
VDec: vo config request - 640 x 360 (preferred colorspace: Unknown 0x0000)
Trying filter chain: vo
Could not find matching colorspace - retrying with -vf scale...
Opening video filter: [scale]
SwScale params: -1 x -1 (-1=no scaling)
Trying filter chain: scale vo
The selected video_out device is incompatible with this codec.
Try appending the scale filter to your filter list,
e.g. -vf spp,scale instead of -vf spp.
[VD_FFMPEG] Trying pixfmt=3.
[ffmpeg] aspect_ratio: 0.000000
VDec: vo config request - 640 x 360 (preferred colorspace: Planar YV12)
Trying filter chain: vo
VDec: using Planar YV12 as output csp (no 0)
Movie-Aspect is undefined - no prescaling applied.
VO Config (640x360->640x360,flags=0,'MPlayer',0x32315659)
VO: [xv] 640x360 => 640x360 Planar YV12
VO: Description: X11/Xv
VO: Author: Gerd Knorr <[email protected]> and others
Xvideo image format: 0x32595559 (YUY2) packed
Xvideo image format: 0x32315659 (YV12) planar
Xvideo image format: 0x30323449 (I420) planar
Xvideo image format: 0x59565955 (UYVY) packed
Xvideo image format: 0x434d5658 (XVMC) planar
using Xvideo port 75 for hw scaling
*** [vo] Exporting mp_image_t, 640x360x12bpp YUV planar, 345600 bytes
Unicode font: 2293 glyphs.
Unicode font: 2293 glyphs.
A: 4.3 V: 4.3 A-V: 0.000 ct: 0.042 0/ 0 4% 0% 0.3% 0 0
Uninit audio filters...
[libaf] Removing filter dummy
Uninit audio: ffmpeg
Uninit video: ffmpeg
vo: uninit ...
Exiting... (Quit)
Is there something else I can check with the MPlayer feedback?
Also, I'm using mplayer-vaapi, but I took a look at its abs PKGBUILD and it didn't have any configure options disabling pulseaudio. At least I'm assuming the abs PKGBUILDs reflect the default options that the binaries come with.
Last edited by mellowmaroon (2013-04-14 05:26:07)

You were right about it being muted through the PA mixer! I didn't think about that, and installed pavucontrol. I figure I'll try to learn how to learn to do it through pactl/pacmd sometime.
Though, I have "flat-volumes = no" so I have a kind of global maximum volume and normally can set other applications' volumes individually from within the program. Pavucontrol reflects MPD+ncmpcpp volume changes, but apparently doesn't seem to pick up MPlayer's application volume.
Is there any way to fix MPlayer so it's internal volume adjustment will be the same as what is under Playback in pavucontrol? Or if it isn't exclusive to MPlayer, is there any way to have all applications start at the 100% playback volume (not the output device volume)?
EDIT: Quickly found it! Haha I had "softvol=yes" in my config file, I think to try to increase the Alsa volume when I had that on. Thanks for your help anyway!
Last edited by mellowmaroon (2013-04-14 05:25:24)

Itunes interacts with my ATI 4550. Plays OK thru HDMI but 'pulses' thru onboard audio card.

Itunes interacts badly with my ATI Radeon 4550. Plays OK thru HDMI connection but 'pulses' thru onboard audio card. I have disabled the audio amp on the video card but no different. QuickTime player works OK. Can I configure something on Itunes?

An up-date: I may have found the culprit. This seems to be hardware related after all. In utter frustration I made yet another clean install. Yes, it didn't work. However I finally had a simple idea: to try to run my clips from my LaCie harddrive I use for back-ups. Lo and behold: no hang-ups on even most troubled clips! Copying the same clip to desktop and playing it from Mac, the trouble was there as it had always been. I then proceeded to download latest QT (out today) No change. Same with 10.4.4 upgrade. But LaCie kept on delivering. It seems my harddrive chokes on large videofiles. It still passes hardware test, and it has been erased four times already. I simply don't know how G5's innards work, but this suggests that the harddrive takes the brunt of all things cinematic. How do QT and harddrive interact? Beats me. One thing seems to be for certain now: somewhere along the way information just jams up, resulting in standstills.
So, if your plagued with the same problem, you should at least check the condition of your harddrive. I still can't be 100% sure, so it's off to local Apple store. We still could be talking about a motherboard problem or whatnot. So, the saga continues but at least now I have some driving instructions.
Tommi R.
PowerMac G5 dual 2.3 Mac OS X (10.4.3) Ati Radeon 9650 graphics card, 2GB of mamory, 250GB hard-drive, all Apple assembled.

Hi there, can anyone tell me how to make make Garageband ignore HDMI as an audio output-device on a 2013 Mac Mini? Now it keeps on prompting me: Do you want want to use the HDMI device (monitor without audio) for audio output? Thanks in advance, Grtz T.

Hi there,
Can anyone tell me how to make make Garageband ignore HDMI as an audio output-device on a 2013 Mac mini i7 using an Apogee jam as an input for bass and a Behringer USB DAC with RCA audio output? Now it keeps on prompting me: Do you want want to use the HDMI device (monitor without audio) for audio output? I'm also suffering massive audio lag issues with the Apogee jam after waking the mini from sleepmode, wich pretty much renders it useless.
Thanks in advance,
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I think the easiest way to do this is to have the EyeTV app running in one user account and sent via HDMI to the TV, and use a separate user account for Screen Sharing. Both logins can be active at the same time and with Lion you can connect via screen sharing to which ever login you need.
With regards to the Mac video resolution changing, when it is not the current input on your AV receiver. This is 'normal', if the Mac sees no signal it drops down the a lower level. What you need is to buy a HDMI Detective Plus which tricks the Mac in to always seeing a signal and hence it will not drop down the resolution. See http://www.gefen.com/kvm/dproduct.jsp?prod_id=8005
The Mac mini does have an IR receiver. This can be used with the EyeTV app, iTunes, and similar. Yes I believe EyeTV would run for prolonged periods quite happily, it is has been around for a long time and had most of the rough edges polished off. However one issue you will have to deal with is that it does not automatically update its program guide data. For this you might have to write or modify an AppleScript. Elgato support or their user forums will be a lot of help.
You can set EyeTV to convert recordings to an iTunes (Apple TV) compatible format and then delete them from EyeTV.
The HDMI Detective should solve your handshaking problems.

How i add audio publisher/audio subscriber in my chat applcation

hello,
I try add audio publisher/audio subscriber in chat applcation im using flash bulider 4 however i watch the example in smp applcations
that have in sdk but what i need i include inside chat applcation button to start and stop audio in chat applcation i treid many times
to do it but always give me error so what is the way to add audio publisher/audio subscriber with button for user hit button run audio hit again stop audio, also need to add other button "logout" to user can logout from applaction when finish but after i add button is dosent work too.
Need help
Thanks

Oh nigel sorry i really forget say what is th error however i gonna paste the code to see where is the error buttons "logout" and "start my audio" dosent work
I dont know why here is the code,
thanks nigel
<?xml version="1.0" encoding="utf-8"?>
<s:Application xmlns:fx="http://ns.adobe.com/mxml/2009"
               xmlns:s="library://ns.adobe.com/flex/spark"
               xmlns:mx="library://ns.adobe.com/flex/mx" minWidth="500" minHeight="500" xmlns:rtc="http://ns.adobe.com/rtc" width="590" height="590" backgroundColor="#3D3B3B" preloaderChromeColor="#999393">
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    <fx:Script>
        <![CDATA[
            import mx.events.FlexEvent;
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                // TODO Auto-generated method stub
                connectRoom.login();
                labelUsername.visible = false;
                txtUsername.visible = false;
            protected function btnZoomScreen_clickHandler(event:MouseEvent):void
                // move screenShare
                if(btnZoomScreen.label == "zoom in"){
                    moveEffect.targets = [scrShare,labelScreen];
                    moveEffect.xTo = 10;
                    moveEffect.yTo = 10;
                    moveEffect.play();
                    resizeEffect.target = scrShare;
                    resizeEffect.widthTo = 520;
                    resizeEffect.heightTo = 479;
                    resizeEffect.play();
                    btnZoomScreen.label = "zoom out";
                    btnZoomScreen.y = 5;
                }else{
                    moveEffect.targets = [scrShare,labelScreen];
                    moveEffect.xTo = 294;
                    moveEffect.yTo = 296;
                    moveEffect.play();
                    resizeEffect.target = scrShare;
                    resizeEffect.widthTo = 250;
                    resizeEffect.heightTo = 293;
                    resizeEffect.play();
                    btnZoomScreen.label = "zoom in";
                    btnZoomScreen.y = 296;
        ]]>
    </fx:Script>
    <fx:Declarations>
        <s:Move duration="500" id="moveEffect"/>
        <s:Resize duration="500" id="resizeEffect"/>
    </fx:Declarations>
    <rtc:ConnectSessionContainer horizontalScrollPolicy="off" verticalScrollPolicy="off" id="connectRoom" autoLogin="false"
                                 roomURL="http://connectnow.acrobat.com/jannaali/mymeeting"
                                 x="0" y="55" width="590" height="535" backgroundColor="#323131" contentBackgroundColor="#FFFFFF">
        <rtc:authenticator>
            <rtc:AdobeHSAuthenticator userName="{txtUsername.text}"/>
        </rtc:authenticator>
        <rtc:SimpleChat x="10" y="10" width="274" height="479" chromeColor="#CCCCCC"/>
        <rtc:WebcamSubscriber x="292" y="23" displayUserBars="true" width="250" height="250"/>
        <rtc:AudioPublisher id="audioPub" height="0" />
        <rtc:AudioSubscriber id="audioSub" height="0" />
        <s:Label x="292" y="10" text="Ahmed's webcam" fontFamily="Verdana" fontSize="9" color="#040404" chromeColor="#CCCCCC"/>
        <rtc:ScreenShareSubscriber id="scrShare" width="250" height="193" x="294" y="296"/>
        <s:Label id="labelScreen" x="295" y="285" text="Ahmed's screen" fontFamily="Verdana" fontSize="9" color="#060606"/>
        <mx:LinkButton x="474" y="296" label="zoom in" fontFamily="Verdana" color="#0712A9" id="btnZoomScreen" click="btnZoomScreen_clickHandler(event)"/>
        <s:Button x="19" y="497" label="start my audio" height="28" width="106"/>
        <s:Button x="133" y="497" label="logout" width="106" height="28"/>
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    <s:TextInput x="165" y="205" width="221" textAlign="center" id="txtUsername" enter="txtUsername_enterHandler(event)"/>
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When I publish a video, I lose half of the audio in "audio 2". I can hear it while editing. How do I fix this?

I published a video as an mpeg, and when I listened to the final product it was missing some music I had in the "audio 2" slot. How do I fix this?

Cameron
Is the problem Audio Track 2 music from any of your royalty free music websites? Is the sound missing in the export
standalone audio files
or
audio that is linked to video?
What is directly under or above these files on the Timeline?
Download and install the free codec utility named GSpot.
And, tell us the readouts for Audio name, codec, and status.
GSpot Codec Information Appliance
If also video component of the file, then please include the Video name, codec, and status information from the GSpot readout.
See what information you can put together on the properties of those royalty free audio files, including file extension. Are you dealing with Stereo or 5.1 channel (AAC or Dolby Digital)?
As for project preset, versions 11 and 12 were designed so that the project takes charge of setting the project settings based on the properties of
the first file drag to the Timeline. Sometimes it does it right. Sometimes not. If not, then you set the project preset manually using the properties of
your file. See ATR Premiere Elements Troubleshooting: PE11: Accuracy of Automatic Project Preset (New Project Dialog) Setting
Lots of additional questions...curious how you hear the Audio 2 audio in Edit area but not in the exported file which you report as
Publish+Share>Computer>MPEG under the preset "MPEG2 1920x1080i 30.
Any export setting customization going on under the Advanced Button/Video Tab and Audio Tab of that preset?
Please review and consider and then we will decide what next.
Thank you.
ATR

Itunes has detected a problem with your audio configuration Audio/Video playback may not operate properly

I cant copy the music from itune library to iphone. Message "itunes has detected a problem with your audio configuration Audio/Video playback may not operate properly" I unistalled the itune from the pc and download the itune again but still same message come up.

I'm having the same problem. I have downloaded Itunes again but still the same problem.

Itunes has detected a problem with your audio configuration. audio video playback

itunes has detected a problem with your audio configuration. audio video playback may not operate properly. Why do i get this after update cant get sound now please help!

See if this article will help:
iTunes for Windows plays video files without any audio
http://support.apple.com/kb/TS1696

My roland Ua-25ex usb audio interface audio input no recognised by macbook pro running osx 10.

my roland Ua-25ex usb audio interface audio input not recognised by macbook pro running osx 10. ...the audio outputs are ok but the (audio in) input preference for the UA-25ex is greyed out and the sliders for input volume etc are disabled ... and I can only record voice by using the macbook pros onboad microphone not the usb interface

Logic Users
Looooong Tutorial for beginners
MacPro Videos on Logic
i have all the logics but i don't know them deeply.
i like some of the music files/loops you get bundled with it, there's plenty to get started.

Analyzing DV Audio - Validating Audio Data

Sony UVW-1600 through ADS Pyro A/VLINK using S-Video in/Firewire 400 out
Audio through Alesis MultiMix 8 USB mixer - 44.1 kHz output only
Just did clean install of Snow Leopard, reinstall/updates on FCP
Otherwise same configuration as before but lost original Custom AV settings and Easy Setups.
Now when I capture BetaSP material, video through the A/VLINK and audio through the Alesis, I'm getting messages I never used to get:
If I set my Capture Preset to 48 kHz audio, at close of capture I get "Analyzing DV Audio - Validating Audio Data"
If I set my Capture Preset to 44.1 kHz (matching the Alesis output), I get a long message saying the audio data rate of my captured media file doesn't match the sample rate on my source file.
From checking old captured clips, I see that I've always been capturing at 48 kHz but never got the "Analyzing DV Audio" message before. Not seeing any sync problems.
Anyone know why I'm suddenly getting this message? I don't want to capture audio through the DV stream as my source tape levels are all over the place. Thanks

Okay, I've answered my own question (I think) but could be helpful to others: in setting up my replacement (after the reinstall) Capture Preset I made a custom preset that selected "USB Audio Codec" for "Quicktime Audio Settings Device." So FCP was seeing video come in through the Firewire stream and audio through the USB stream. It evidently wanted to "validate" the rejoining of the two. I've just changed that Quicktime Audio Settings Device back to DV Audio and now I don't get the validating message. In OS X System Preferences I have "USB Codec" chosen for default audio input so FCP must be using that setting but "thinks" it's getting audio from the Firewire DV stream. I'd tried that earlier but lost audio completely during FCP capture. Now it's working. Odd...

Audio to audio chat

This may sound stupid, but I have used ichat exclusively to IM with my brother.
We both have ichat 3 and have never tried audio to audio. How is this done?

Hi Kevin,
Ok.
iChat launches and you can text chat. So far so good but not very exciting and very frusting if you are trying to Video chats or Audio only chat.
Right, iChat needs to have ports open in your Mac Firewall if it is ON and in your Modem. The ones on your modem are likely to outside the ones that are open for web browsing and email (Out-of-the-Box stuff) default settings.
The ports and why and when http://www.ralphjohnsuk.dsl.pipex.com/page3.html
Scroll down this blog to 7th Sept for the ports required in a table layout. http://www.ralphjohnsuk.dsl.pipex.com/blog.html
This page contains a Pic at the top that changes between the Panther and Tiger set of the Mac Firewall. http://www.ralphjohnsuk.dsl.pipex.com/page4.html It also details the ports that have to be opened.
Problems can arise when you have a separate Router and modem. Some applications like web browser and email can cope with both devices acting as DHCP servers. iChat has to be certain of the IP address it is sending data to and works better if there is only DHCP server.
The ports in a modem (and or router) can be opened if different ways depending on the options your device offers you.
The most common is Port Forwarding (Also called Virtual Servers, Pinholes, NAT, NAPT amoungst pthers)
Next is port Triggering. Designed as a more secure way of having only one port open until the data arrives on that Port and then other ports are Triggered.
UPnP works under the control of the appication and opens ports when the application needs them. iChat can use UPnP.
Some of this is also mentioned on Page 3 in the links above.
There was a problem when Tiger users were at iChat 3.0 and 3.0.1 when it was difficult for those iChat users to connect Audio only to Panther users but this has been addressed by the 10.4.3 update (iChat 3.1).
Have you opened the ports in your Modem ?
Do you have a second device such as a router ?
IS only one device being a DHCP server ?
IS this a Panther to Tiger problem ? EDIT I See not.
BTW, My telephone is grayed out. Do I have to subscribe to .mac?
From your earlier post
Check the iChat View menu has Show Audio Status selected.
Check the Audio menu(if no camera) to the right of Buddies (reads Video if camera), has Microphone Enabled with a tick next to it.
In System Preferences > Sound > Input tab make sure the "right" micorphone is selected.
In iChat Preferences > Video (even if audio only) > Micorphone Drop Down has the same Microphone selected.
Ralph

Record drive/partition "Audio Files /Audio Recording" not found!

HAAAAAAAAAA !!!!!! I get this message every time I want to record audio !!!!!
Record drive/partition "Audio Files /Audio Recording" not found!

myrna wrote:
Any idea what path blogger Jim Frazier uses? BTY, only the audio portion does not record; the >instruments record just fine. If this is indeed a bug, is there a workaround that will get me recording >until a fix is found...
I actually have not had this happen to me in several months, and in fact, haven't even thought about it in a while.
But back when it would happen, I would be working away, recording audio without any problems, and then I'd hit record again for something, and out of nowhere, get a message like this:
"olumes/Audio Drive/Album Projects/John Doe/Song title/Audio Files" not found.
When the first word should have been "Volumes".
So I would click and hold the record button to get to reset my audio pathway, and the proper path would already be listed. I would then merely click the "Set" button, and be back in business.
And there were even a handful of times when I would get that message with the full name listed, the V was there in "Volumes', and everything listed correctly, and going to the record settings, everything was set properly. But I'd still have to hit "Set" to remind Logic where the pathway was pointed to.
Definitely a bug.
But... this bug never happened to me in Logic 7. It was only since installing and using Logic 8, that it occurred. And as I mentioned, it now hasn't happened in months.

[Solved] Pulse Audio won't work on newly installed system with Gnome

About a year ago I got a new computer and more or less retired my old one with my first arch install. Recently, I decided I would set it up as a Linux system for my sister. I first updated the computer, which broke it royally (since it was last updated before the /bin symlinking happened) but I eventually got it fixed. However, after doing that, Pulse Audio wouldn't work, then I broke X and decided to do a full reinstall.
Now, on this clean install, Pulse is again not working. I have set up Gnome and gdm to work, and everything works great except Pulse. The wiki says that Pulse with Gnome 3 requires no extra setup, so I have not edited conf files at all. I have, however, installed "pulseaudio-alsa", which created this conf file:
# Use PulseAudio by default
pcm.!default {
type pulse
fallback "sysdefault"
hint {
show on
description "Default ALSA Output (currently PulseAudio Sound Server)"
ctl.!default {
type pulse
fallback "sysdefault"
# vim:set ft=alsaconf:
which seems fairly standard. Anyway, I tried removing it and had the same problems. When I have Pulse Audio running and try to play sound, I get the error:
ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
Playback open error: -16,Device or resource busy
which is obviously trying for ALSA, or from mplayer:
AO: [pulse] Init failed: Connection refused
Running "pacmd list" nets me:
Daemon not responding.
and pavucontrol gives me:
Fatal Error: Unable to connect to PulseAudio: OK
when I try to start it.
No pulseaudio commands seem to work (e.g. pulseaudio -k). They don't raise any errors, but also it doesn't have any effect on the server.
The only asound conf file is "/etc/asound.conf", clearing ~/.config/pulse does not fix the issue.
Journalctl does not seem to give anything interesting (Stale PID is the only message when starting Pulse), though admittedly I'm just vgrepping all of journalctl since I don't know how to narrow it down more, nor does the output from pulseaudio -vvvv. There are a lot of errors in pulseaudio -vvvv, but they all appear to be related to loading devices that don't exist, and so I assume not a problem, I can post it if necessary. Sometimes when I kill Pulse the process refuses to go away and I have to reboot in order to clear the process out, but I don't think that's related.
If I kill Pulse or keep it from starting on boot, I can play sound, but only through ALSA, and while I didn't have the problem I expected (I can still play multiple things simultaneously) there's still the issue that the volume buttons in Gnome will no longer work and that the volume levels ALSA reports are basically nonsense to a human being.
I basically just blurbed everything I thought of/found on the wiki or forums, and I'm not really sure where to go from here. It's a fairly old computer, so that could be related, and I'd be willing to dump Pulse if Gnome could integrate well with ALSA, but obviously fixing Pulse would be the preferred option.
Any suggestions on where to look next? I've run out of ideas.
Thanks.
Last edited by HuFlungDu (2014-01-07 19:52:11)

Wonder if you have a conflict between two audio cards ... for example, I have a friend with some PulseAudio problems caused by conflict between the HDMI audio from his graphics card and his onboard sound. Something would switch PulseAudio to the wrong card and it would just quit after getting itself into a twist. Unsolved, I'm afraid, but have a look at commands such as:
aplay -l
pactl list sinks
pactl stat
(Latter two will need PulseAudio coaxed into life first)
and of course, the Arch wiki.
If there is a conflict, blacklisting kernel modules might also be needed.

[SOLVED]No Sound - ALSA - Maybe wrong audio device.

Yesterday I installed Arch and it has been going well except that I can not get my sound working. Following https://wiki.archlinux.org/index.php/Ad … chitecture I installed alsa-utils and then used alsamixer to unmute channels. While I experimented with the channels I had a youtube video open and so I should have heard something if it had worked.
I seem to have two sound cards. When I run alsamixer it defaults to the "Intel Haswell HDMI" chip and "Item: S/PDIF". I can unmute the S/PDIF channels but there is no volume bar. I suspect that this device is set as a default somewhere but it is not the one I want to use.
Hitting F6 and choosing the "HDA Intel PCH" card has many columns. I have them all unmuted and I've played around quite a bit with turning various things up and down. The speaker I have in plugged into the back of the desktop I am using and is attached with both a round pale green connector and a usb connector.
When I turn up the volume on the <Rear Mic> I can hear the static volume go up on the speaker. So I could change the volume on the speaker but the speaker was not getting audio from the web browser I had open. When I turn up the volume on the <Front Mic> I can hear a high pitched noise increase, my only guess for where its coming from is the tiny speaker that I have plugged right into the motherboard.
I've tried researching and I would like to try and change the default sound card like in this thread: https://bbs.archlinux.org/viewtopic.php?id=186597. But I'm afraid that I'm newer to Arch/Linux than those people or at least I'm confused trying to follow that advice. My /etc/modprobe.d directory is empty. If I made a .conf file like in that answer I would not know what to name it.
Here is the output of some relevant (hopefully) commands:
$ lspci -vnn | grep -A 1 -i audio
00:03.0 Audio device [0403]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller [8086:0c0c] (rev 06)
Subsystem: ASRock Incorporation Device [1849:0c0c]
00:1b.0 Audio device [0403]: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller [8086:8c20] (rev 05)
Subsystem: ASRock Incorporation Device [1849:8892]
$ aplay -lL
null
Discard all samples (playback) or generate zero samples (capture)
hdmi:CARD=HDMI,DEV=0
HDA Intel HDMI, HDMI 0
HDMI Audio Output
hdmi:CARD=HDMI,DEV=1
HDA Intel HDMI, HDMI 1
HDMI Audio Output
hdmi:CARD=HDMI,DEV=2
HDA Intel HDMI, HDMI 2
HDMI Audio Output
default:CARD=PCH
HDA Intel PCH, ALC892 Analog
Default Audio Device
sysdefault:CARD=PCH
HDA Intel PCH, ALC892 Analog
Default Audio Device
front:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
Front speakers
surround21:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
2.1 Surround output to Front and Subwoofer speakers
surround40:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Digital
IEC958 (S/PDIF) Digital Audio Output
**** List of PLAYBACK Hardware Devices ****
card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
lsmod | grep snd
snd_hda_codec_hdmi 53248 1
snd_hda_codec_realtek 65536 1
snd_hda_codec_generic 65536 1 snd_hda_codec_realtek
snd_hda_intel 28672 0
snd_hda_controller 28672 1 snd_hda_intel
snd_hda_codec 114688 5 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_codec_generic,snd_hda_intel,snd_hda_controller
snd_hwdep 16384 1 snd_hda_codec
snd_pcm 90112 4 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel,snd_hda_controller
snd_timer 28672 1 snd_pcm
snd 69632 8 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_hda_codec_generic,snd_hda_codec,snd_hda_intel
soundcore 16384 2 snd,snd_hda_codec
$ speaker-test
speaker-test 1.0.29
Playback device is default
Stream parameters are 48000Hz, S16_LE, 1 channels
Using 16 octaves of pink noise
ALSA lib pcm_dmix.c:1024:(snd_pcm_dmix_open) unable to open slave
Playback open error: -2,No such file or directory
Thank you for reading.
Last edited by buffalo (2015-06-13 22:47:37)

I followed Head_on_a_Stick's advice and the speaker is now working! Thank you so much Head_on_a_Stick. The one other time I posted on this forum was last February and you helped me out then too. You are my favorite!
Thank you ewaller as well, but rather than look at aplay -l I think I will mark this thread as [SOLVED].
The thing I'm still curious about is how did you know what to name the file /etc/modprobe.d/alsa-base.conf ?
Time for me to figure out how to get the headphone jack working.

[SOLVED] Pulse Audio / ALSA audio sharing.

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