How to modify audio output of my sequence

Hi. Let's say I have a sequence that has one video track and one mono audio track.
1) If the audio track is completely empty and I delete all empty tracks, Premiere refuses to delete Audio track 1. How do I delete the audio track, such that my sequence has no audio at all? So when I place the sequence in another timeline, it imports as video only.
2) How do I modify this mono track so that it's in stereo? Presumably I'll have to decide how to split the mono signal into left/right too.
TIA,
Richard

2) How do I modify this mono track so that it's in stereo? Presumably I'll have to decide how to split the mono signal into left/right too.
Fill Left/Fill Right will get you two channels. By definition, it will NOT be "stereo," but will be two-channel mono - same exact signal in both channels. One can alter the spatial aspects in the sound stage, to simulate stereo a bit more, but it will still be two-channel mono.
For # 1, srukweza has given you the easiest way to Mute Audio 1 - turn OFF that Track for Export, or go to Audio Mixer, and hit Mute.
Good luck,
Hunt

Similar Messages

HOW TO: HDMI Audio Output for Zotac Media Box with nVidia ION :)

I've been fighting with my recently purchased Zotac mini-PC for about 2 months now trying to get the HDMI audio output to work. I tried extensive modifications of /etc/asound.conf, different versions of ALSA, and even OSS without success. I've finally succeeded using ALSA and GNOME.
First I installed ALSA according to the Arch wiki.
Next I verified that ALSA detected my sound card(s):
cat /proc/asound/cards
0 [Intel ]: HDA-Intel - HDA Intel
HDA Intel at 0xfe9fc000 irq 28
1 [NVidia ]: HDA-Intel - HDA NVidia
HDA NVidia at 0xfcf7c000 irq 18
Next I checked which sound card had the HDMI jack:
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: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 7: NVIDIA HDMI [NVIDIA HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 8: NVIDIA HDMI [NVIDIA HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 9: NVIDIA HDMI [NVIDIA HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
From these two commands, I concluded that I have 2 sound cards (important later on) and that my HDMI sound card was card 1 (NVidia) and that it had 4 possible devices (3,7,8, and 9). To determine which device must be used, I ran the following command for each device (for example, to test card X and device Y, use' plughw:X,Y'):
aplay /usr/share/sounds/alsa/Front_Center.wav -D plughw:1,7
Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono
I tried each device and got the best sound from device 7 (8 and 9 also worked, device 3 gave static...).
Now, since I had 2 sound cards, I needed to change the default ALSA device. My default ALSA device was card 0 (the Intel card), which wasn't plugged in, so I needed to change the default card to be card 1 (the HDMI card). After reading the ALSA FAQ, I found that modifying /usr/share/alsa.conf should work, though modifying /etc/asound.conf should have also worked. For whatever reason, I couldn't get my modifications to /etc/asound.conf to work, so I edited /usr/share/alsa/alsa.conf. In the "# defaults" section I replaced the following:
defaults.ctl.card NVidia
defaults.pcm.card NVidia
defaults.pcm.device 7
I opted to use the key NVidia rather than the card number (1) since I've read in numerous places that the card index can change unexpectedly after a reboot/update. The key can be checked with cat /proc/asound/cards as shown above.
After this, I reloaded ALSA (/etc/rc.d/alsa restart) and verified that the sound worked through the default channel:
aplay /usr/share/sounds/alsa/Front_Center.wav -D default
Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono
Finally, I rebooted, chose the HDA NVidia (ALSA Mixer) option from gnome-volume-control, and loaded a movie in gnome-mplayer. I was pleased to hear crystal clear sound with functioning volume control
I Hope this helps someone else, I found it very hard to piece together how to diagnose and resolve this problem. Here are some of the resources I found useful:
Arch Wiki ALSA Page
ALSA FAQ for how to change default sound card
ALSA FAQ
XBMC Troubleshooting for ION HDMI

@sprince09 Can you hear sound through HDMI when you run aplay /usr/share/sounds/alsa/Front_Center.wav -D plughw:1,7 ?
(8 and 9 also worked, device 3 gave static...)
- what does it mean? How do you know that they are working?
What options do you have for snd_hda_intel in /etc/modprobe.d/ ? Can you play the following:
mplayer -ao alsa:device=hw=1,7 file.wav
BTW, can you hear sound form video files in xine? If yes, can you post your ~/.xine/config ?
Last edited by knedlyk (2012-01-21 23:41:57)

How to get audio output of my instrument while playing back the other tracks?

I do not hear any audio signal from my bass guitar while playing along the other tracks in Logic. I can only hear the audio output of my bass while I record or stop the music. How can I change the settings so that I can play along and hear myself while I practice? The track for bass is not muted and is getting a strong input signal.

Hi
Enable the Input Monitoring button (the I next to the R)?
CCT

How to disable audio output in while in sleep mode? (annoying issue)

I have a my workstation set up in my bedroom and when I am not using it I put it to sleep (or wait for it to go to sleep deu to inactivity). The problem is that when its asleep the audio output is still active. So when I get emails and such the speakers still emit the notification/alert sounds even though my computer is in sleep mode. This is really annoying since I work with offshore teams and they send me email in the late night and early morning (when I am sleeping). I can't seem to find a setting to change that. Any ideas?
My set up is as follows:
17 macbook pro (early 2011)
OSX 10.9.4
Connected to a 27" thunderbolt display
Thanks!

Thanks KiltedTim. I am starting to agree with you. Do you know what I can log for in the logs that will give me an idea?
Esquared - Thanks for the link. The only thing I see in the logs is when I woke up my computer this morning. But all morning (before I woke my computer) I was hearing a bunch of alerts/notification. Here's a look at the log query I did.

How to chang audio-output-device(s) in Windwos 8

Hi,
currently I’m working on an application that starts by changing the audio-output-device and ends with re-setting the old one.
On Windows XP I do this by changing the Registry-entry “Sound-Mapper” and it works quite well.
I hopefully assume it works on all Windows Systems from XP to Windows 7.
The background is that our program is supposed to play certain multi-channel sounds over an external USB-multichannel-soundcard and it should switch to that device on start. To actually play the sounds we use the AxWindowsMediaPlayer-Class
to integrate the Windows-Media-Player in our application.
However, on Windows 8, it is possible to set multiple Audio-Outputs and the Windows-Media-player has a setting to choose from them (under Options). I was unable to use the Registry-trick to switch the output-device and I could not find
a way to programmatically tell the AxWindowsMediaPlayer to choose the USB-soundcard as an output.
Is there a Way to programmatically(c#) ether change the audio-output-device(s) in Windos8 or to tell
AxWindowsMediaPlayer to use a certain device?
Thank you.

Hi Kristin,
I think your reply pushed me in the right direction.
The waveOutOpen-function is part of the
winmm.dll-API and I think, that API is the solution to my Problem.
Unfortunately I am stuck at this point. I tried mciSendString and mciSendComand. Both are cappable of playing sound, but I am unable to change the output Device.
This is my c# code to open a WAV-File via mciSendCommand:
MCI_OPEN_PARMS open = new MCI_OPEN_PARMS();
open.wDeviceID = 0;//this is somehow ignored
open.lpstrDeviceType = Marshal.StringToHGlobalAnsi("waveaudio");
open.lpstrElementName = Marshal.StringToHGlobalAnsi(@"test.wav");
IntPtr ptr = Marshal.AllocHGlobal(Marshal.SizeOf(typeof(MCI_OPEN_PARMS)));
Marshal.StructureToPtr(open, ptr, false);
mciSendCommand(Convert.ToUInt32(textBox2.Text),
MCI_OPEN,
MCI_WAIT | MCI_OPEN_TYPE | MCI_OPEN_ELEMENT,
ptr);
Afterwards I use mciSendCommand to start the playing of the sound, which works fine, but no matter what I enter in the “open.wDeviceID”-Line, I always get sound over
the default output.
I use waveOutGetDevCaps to get the number and kind of the output-Devices and this works fine.
Regards, Marc

How to install audio output device

my computer said that i need to install audio output device and i did but it keeps saying failed to install and i don't know what to do can u please help me !

audio output deviece is installd

How to save audio output to file

Hi All,
I'm not a flash developer, just a user, so forgive me if my
question has misconceptions and errors.
I've been listening to some online music that uses
flashplayer. I'd like to be able to listen to the music offline
and/or burn CDs of the music being streamed. I would like to know
if there's a way to have the audio output saved to a file instead
of being played. FYI, i'm running FP 9-beta for linux.
Thanks.
MickeyG2

mickeyg2 wrote:
> Hi All,
>
> I'm not a flash developer, just a user, so forgive me if
my question has
> misconceptions and errors.
>
> I've been listening to some online music that uses
flashplayer. I'd like to
> be able to listen to the music offline and/or burn CDs
of the music being
> streamed. I would like to know if there's a way to have
the audio output saved
> to a file instead of being played. FYI, i'm running FP
9-beta for linux.
Not all audio files can be saved, streaming and audio embed
within swf files
is hard to grab. Some players tho use directly Mp3 load,
these files you can
find in Internet temporary cache folder.
For others, you could use some audio recording tool. I don't
know Linux but windows
has program call Total Recorder which can take any output
your speakers play and
record in excellent quality through your sound card device.
Easy to use and works like charm.
Best Regards
Urami
<urami>
If you want to mail me - DO NOT LAUGH AT MY ADDRESS
</urami>

[SOLVED] How to stream audio output over wifi?

Hello,
I have following problem. I would like to listen to the music from my laptop on my home Onkyo AV receiver (amplifier). Both laptop and receiver are connected on the same network, laptop via wifi, receiver using LAN. Receiver is capable to play internet radio streams in various format - e.g. ogg or mp3.
My idea is to produce ogg stream on laptop and listen to this stream through amplifier. I was searching for this solution a found something that has always some disadvantage, like MPD + MPC (can play only from database of local files, not from jamendo or other network services), DLNA can play only local files, Icecast + Ices can play from predefined playlist.
My idea is to have something independent - something what takes audio data normally played on soundcard and feeds it to network stream, something what can work with any player - audio or video, or even with desktop notifications.
Does anybody here know how to do it?
Thanks!
Last edited by KejPi (2012-03-14 20:18:28)

Thank all of you for your hints. Finally I have decided for ALSA based solution that simply works for all applications smoothly in background and it is here when I need it - in other words, it is something that doesn't borther me when I do not need it. I use ALSA loopback with ffmpeg and ffserver for streaming in mp3 format.This is my final solution:
You need following:
alsa
ffmpeg
if you use KDE4, then you need phonon-gstreamer backend, as in my case phonon-vlc didn't work (it was heavily distorted)
Procedure:
Modprobe snd-aloop (2 loopback channels are enough)
modprobe snd-aloop pcm_substreams=2
For permanent solution follow arch wiki https://wiki.archlinux.org/index.php/Modprobe
Create ALSA configuration file and store it either to ~/.asoundrc (just for user) or to /etc/asound.conf as system wide:
pcm.!default {
type asym
playback.pcm "LoopAndReal"
capture.pcm "hw:0,0"
hint {
show on
description "Default with loopback"
#"type plug" is mandatory to convert sample type
pcm.LoopAndReal {
type plug
slave.pcm mdev
route_policy "duplicate"
hint {
show on
description "LoopAndReal"
pcm.mdev {
type multi
slaves.a.pcm pcm.MixReal
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.MixReal {
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
You can play with sample rates and buffer sizes you you have any problem. This configuration works on my system.
Prepare ffserver configration and store either to default location /etc/ffserver.conf as system wide setup or anywhere to your home:
# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
Port 8090
# Address on which the server is bound. Only useful if you have
# several network interfaces.
BindAddress 0.0.0.0
# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
# MaxClients maximum limit.
MaxHTTPConnections 2000
# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000
# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 1000
# Access log file (uses standard Apache log file format)
# '-' is the standard output.
CustomLog -
# Suppress that if you want to launch ffserver as a daemon.
NoDaemon
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
# ffserver. This sequence may be encoded simultaneously with several
# codecs at several resolutions.
<Feed feed1.ffm>
# You must use 'ffmpeg' to send a live feed to ffserver. In this
# example, you can type:
# ffmpeg http://localhost:8090/feed1.ffm
# ffserver can also do time shifting. It means that it can stream any
# previously recorded live stream. The request should contain:
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
# a path where the feed is stored on disk. You also specify the
# maximum size of the feed, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 200K
# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.
# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg
# Only allow connections from localhost to the feed.
#ACL allow 127.0.0.1
</Feed>
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.mpg'). FFServer will send this stream when answering a
# request containing this filename.
# MP3 audio
<Stream stream.mp3>
Feed feed1.ffm
Format mp2
AudioCodec libmp3lame
AudioBitRate 320
AudioChannels 2
AudioSampleRate 44100
NoVideo
</Stream>
# Ogg Vorbis audio
#<Stream test.ogg>
#Feed feed1.ffm
#Format ogg
#AudioCodec libvorbis
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Special streams
# Server status
<Stream stat.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.1.0 192.168.1.255
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</Stream>
# Redirect index.html to the appropriate site
<Redirect index.html>
URL http://www.ffmpeg.org/
</Redirect>
This sets ffserver for streaming in MP3 format, stereo, 320kbps. Unfortunately I haven't succeeded with OGG Vorbis streaming.
Now you have all configuration you need and if you want to stream following two commands do that:
ffserver -f ffserver.conf
ffmpeg -f alsa -ac 2 -i hw:Loopback,1,0 http://localhost:8090/feed1.ffm
You can test it for example by mplayer:
mplayer http://YourLinuxBox:8090/stream.mp3
And that's it. Sound is played by normal sound card and sent to stream simultaneously. If you do not want to listen sound from computer you can mute your soundcard. It has an advantage that one can normally listen to music on the computer with or without streaming and in both cases without any reconfiguration. To start streaming just call ffserver and ffmpeg.
Advantages:
+ very simple solution without any special sound server
+ no special SW required (in my case I had already instaled all I need for that)
+ streaming on request by two simple commands
+ normal soundcard function
+ streaming in MP3 format that is supported by many home AV receivers
Disadvantages
- phonon-vlc backend not compatible (also VLC does not work)
- OGG streaming does not work
- some latency (~ 5 sec)
- all sounds are sent to stream, including various desktop notifications (in KDE could be managed by phonon)

Mono audio input from lapel mic - how to create audio output in both channels?

I'm recording audio with mono lapel mic, i.e. only one channel.
I need output audio in the finished movie to play in both channels.
Can that be done in FCP X, and how.
Thanks a bunch.

Thanks, David.
So, if I have talent A on the left channel of the video, and talent B on the right channel of the same stereo recording...
can I pan each of these "mono" sides centered so they both come out the left and right sides ?

How to set correct audio output while using external screen for presentation purposes over a wire

while trying to make a presentation or simply playing a video from my MacBookAir over thunderbolt/HDMI table to large screen I am experiencing problems with audio only being played from the MBA and not the soundsystem built in to that screen (unless using a VLS application and manually selecting the audio output target).
Please help in finding how to steer audio output device using 2in1 cable such as thunderbolt?

Hi jar9,
If you are not hearing audio over your HDMI/Thunderbolt connection, you may want to see if it is showing up as a selectable audio output as outlined in the following article:
OS X Mavericks: If you can’t hear sound from your speakers
http://support.apple.com/kb/PH13841
Regards,
- Brenden

Question from the After Effects team about audio output

When you render and export a composition from After Effects that does not have audio content, do you ever enable audio output intentionally (thus creating an empty audio track in the output file)?
If so, why, and what formats do you use?
We're in the process of making some improvements to how audio is handled, and we want to make sure that we haven't broken any common workflows.
(Note: If you're not sure about how to enable audio output, see this page: "FAQ: Why is there no sound (audio) in my output file?" )

For me, I can't think of a common use-case where that would be necessary. I suppose there are times when you want a file to conform to an existing structure, such as when you want to replace media in Premiere, and the new file needs to have the same number of audio tracks as the old file (which you may not have created). Even though that hardly ever happens (I remember replacing proxies when CS3 got the ability to import P2 footage... it was... not very cooperative), I wouldn't want to lose the ability to arbitrarily export an empty audio track, because I'm sure there are other edge cases which I can't think of. But, I wouldn't mind losing it as a default behavior.
It sounds like what you're hinting at is a feature where the render module detects whether there is any audio in the comp before committing to some kind of default audio track export setting. If there is no audio in the comp (or if all the audio in a comp is disabled), it wouldn't export an audio track.
As long as there is a setting to override this behavior (i.e.: enabled / if detected / disabled) that can be built into an export preset, I'd love such a feature.
I, too, often forget to turn on audio export, because most of my comps don't use audio (I sync audio in Premiere, usually) and I don't like having empty audio tracks I don't need. But in the rare cases I do need audio, it's really annoying to go back and re-export the audio because I forgot to turn it on (similar to forgetting to enable an alpha channel, which makes me feel really stupid).

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,
Grtz T.

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 to change '2 Outputs' to '1 Stereo' in Sequence Settings

Any ideas how to change audio to '2 Outputs' to '1 Stereo' in Sequence Settings of FCP5.1.4? I'm always having to render my audio after importing!

My about-box reads:
"Studio Edition Version 10.1.3.1.0.3984"
"Build JDEVADF_10.1.3.1.0_NT_061009.1404.3984" (arrgh, please make an option to copy/paste this...)
Have you accidentally switched 2 ciphers, Frank? By the way, I found the reason for this - I had set the "J2SE version" under "Libraries" to "1.4.2_06" earlier to make sure about 1.4.2 compatibility - but since we soon don't need this any longer, I'm switching this value to 1.5.0 to be able to fully utilize 1.5 options in my code.
So... I guess we can say, that this was my own fault.
Thanks,
Jacob

How to connect Digital Optical Audio output iMac with Receiver Input

I have connected iMac with onkyo hts-3300 home theater system receiver through optical audio TosLink Cable.( http://www.amazon.com/Digital-Optical-Audio-TosLink-Cable/dp/B001ZU3G2I/ref=sr12?ie=UTF8&qid=1292865309&sr=8-2 ). Unfortunately there is no sound coming out from the Home theater system. I have verified the Receiver optical input is working fine by connecting with my TV optical audio digital out put and the it's working great. The only difference between TV and iMac is, on my TV, it has optical audio digital output and I can use this cable (http://www.amazon.com/Tripp-Lite-A102-02M-Digital-connectors/dp/B0002WPREU/ref=s r11?s=electronics&ie=UTF8&qid=1297320792&sr=1-1) to make it work. I can also verified my optical audio TosLink Cable working because I can see the red light coming out from the cable. So the only problem could be from iMac. I doubt I need to change the additional setting to configure iMac to use external optical audio setting instead of the buit in audio setting. Hoe can I able to do it? or my iMac(IMAC 21.5"/3.06/2X2GB/500GB/9400M-ITP
MB950ZP/A) doesn't support optical digital audio out put? I have follow this thread http://discussions.apple.com/thread.jspa?messageID=12769423&#12769423 and to my surprise which doesn't work out for me. I doubt that i need to change some setting, but when I got to setting>> sound I cannot change any setting. I was quite depressed aldy. Can someone answer my question as i m a newbie here. Thanks a lot

Thank you very much for your reply.
@Király : May i know how to Open Audio Midi Setup? Is this Audio Midi Setup is in the locatated in the Setting in Mac Snow Leapord OS? or how to locate this "Audio Midi Setup " in Mac enviroment. Is it possible to open this Audio Midi Setup in windows 7 enviroment also? I Just want to get the audio from my iMac in Hts-3300 onkyo home theatre. YOur help is much appreciated
@rkaufmann87 : Thank you very much for your sharing about this http://www.apple.com/airportexpress/ connection. for me what i understand is the digital audio output in airportexpress is the same as the audio out put in iMac in terms of details sepcs. So this airportexpress will only help you connect wirelessly to home theatre system, rather than waired connection. Other than that nothig much difference. So there is no point for me to spend extra bucks to attain the wireless connection. Unless the airportexpress have optical digital input, which i believe not. Correct me if I m worng, thanks

How do i monitor/output digital/analogue audio while editing?

I just switched to the mac pro and am learning FCP. Now i am forced to monitor audio output through my headphones. I want to tie the audio into my studio monitors and also be able to send optical audio in to the mac pro from my main mixer, a Mackie 1404. Do i need a DAC(Benchmark), a optical I/O digital mixer or both or something else? My budget is upt to $1,000, but i would like to spend <$300.
In general, i am trying to figure out how to incorporate analog/digital audio/video in to my new editing system. Before this i was outputting analog video/audio in real time with a Velocity I/O card. I am now, obviously, converting to a new way of outputting my finished product and how to edit and work with the new Mac Pro.
I guess i could also consider using the firewire also, right? I haven't worked with a firewire based audio hardware unit, so any info on that appreciated as well.
Also, is the line out in the back of the Mac Pro mono or stereo? I appear to be getting a mono feed while monitoring clips i've shot.
While researching i might list here some things i have come across. Opinions on these appreciated:
Apogee Mini-DAC
Edirol FA-66, 101
Message was edited by: gnostic1919

http://www.blackmagic-design.com/products/decklinkstudio/

How to modify audio output of my sequence

Similar Messages

Maybe you are looking for