Problem sending FCP X audio output over USB connection

I have an Antelope Eclipse digital audio converter/monitor controller. It connects to my Mac Pro via USB.
I can play audio just fine out of Pro Tools HD Native at any sample rate.
However, I have just discovered that the audio from FCP X is distorted and "wobbly"......apparently there is a sample rate sync problem between FCP X and the Eclipse.
The audio in question is Linear PCM @ 48kHz.
Is there a problem with FCP X sending audio out through a USB 2.0 connection?
Do I need to change some FCP X settings?

Thanks for the input. Yes, I am aware of possible latency issues. I would stay with low latency sound cards like the Pro Tools card, which worked fine. There would be no D/A conversion, so latency is usually not an issue.
I have asked about compatibility with sound cards in a separate thread, just to be sure any sound card that the Mac Pro recognizes will work:
https://discussions.apple.com/message/18579263#18579263

Similar Messages

24 bit audio output over USB

I'm trying to stream hi resolution digital audio files (24/96 ) via a USB connection to my DAC. There is a software limitation of 16/48 it seems with the standard OS X USB drivers as the DAC only sees 16/48 via USB but can see 24/96 if I use the mini Toslink input. I would like to use this input for another source however. Is there any way around this? Any open source USB drivers that can handle 24/96?

answered it myself!

Need Laptop Audio Amplifier for USB Connected Speakers

For a laptop, does Creative have any kind of audio amplifier for USB connected speakers?
I think the laptop has what is called an Express 54 slot.
I am looking for amplification suitable for speakers, not headphones.

If their only connection to the laptop is via USB than you can't amp them without doing some internal modifications. It means the speakers have their own audio device internally which is receiving signal via the USB port, processing it and then outputting it via likely a weak amp stage to the speakers - you'd have to physically intercept that last part of the chain to add an amplifier.
Not saying it can't be done but you're looking at either harvesting an existing amp (which Creative don't seem to sell) or creating one from scratch (like the Penguin Mint amp, Objective 2, etc) for the purpose.

How to send FCP edited audio clips for processing in Soundtrack?

Hello
How do I send just the edited audio clips from hour long files to be processed in Soundtrack from Final Cut?
I need help as I'm trying to adjust some edited audio clips in Soundtrack Pro but when I send them from FCP to Soundtrack - it opens the whole original audio file and then any adjustments I apply in Soundtrack are being processed over the whole audio file not just the short edited audio clip that I want to adjust.
Normally I don't mind as my scenes are usually short and Soundtrack analysis over the whole scenes audio is usually desirable, but this projects files have been transfered from tapes as hour long AVI files (all the tapes clips transfered into one hour long files) and every FCP edited audio clip I send is being treated as an hour long audio file in Soundtrack.
I'm sure there's a simple answer for settings but I haven't been able to find it. Any advice appreciated.
Barra

Thanks Javi, my sequence is an hour and half duration so what I'm trying to do is save processing time in Soundtrack by just processing adjustments on the few audio clips that need processing.
I have set Soundtrack to open referenced audio only and with 5 sec in and out handles, but it still imports the whole original audio file into Soundtrack timeline upto the selected clip, I then double click in Soundtrack on the clip in brackets, and it then selects just the audio clip I want to adjust, then Soundtrack will just process on the edited clip but when I 'render to action' it seems to apply changes to whole original audio file - taking longer than I want.... frustrating
Also the sound always sounds very different in FCP than Soundtrack even after 'rendering to action' and rendering in FCP... this is not the easy workflow I expect from such expensive software programs, I understood that they work together in a fully integrated way, what's the problem! This is wasting my precious time and preventing me from editing not helping me, this is a common problem and yet I can find no help in the user manual - I'm not impressed!

Does Garageband for iPad support audio output over Airplay?

Since I don't currently have an Airplay output device on my network, I can't tell if Garageband supports this. Anyone know?
Thanks.

When the iPad or iTunes is sending music/video to an AirPlay Output device, it is actually sending the music/video file over to that device to play rather then just sending 2 channels of audio.
For this reason, GarageBand would be a difficult task asnit would need to send up to 10 files (8 tracks and 2 effects) over the network to play in perfect sync.
For this reason, garageBand would not be stable and you would have a large amount of drag.

Audio output to USB cable

I want to play audio from my MacBookPro (13inch) through my Roberts stereo which has a standard USB port, which can be selected as the source for playing through the speakers. Does anyone know of a cable I can use to connect the audio output on my MacBook to the USB input on my stereo?
I would have thought this was a pretty standard conversion but cant find anyone selling something with a 3.5mm jack on one end and a USB jack on the other?
Thanks in advance!

MarvUK wrote:
I want to play audio from my MacBookPro (13inch) through my Roberts stereo which has a standard USB port, which can be selected as the source for playing through the speakers. Does anyone know of a cable I can use to connect the audio output on my MacBook to the USB input on my stereo?
I would have thought this was a pretty standard conversion but cant find anyone selling something with a 3.5mm jack on one end and a USB jack on the other?
Thanks in advance!
Why don't you use the USB port on your computer to get the audio out? USB to USB.
Message was edited by: BobTheFisherman

PS4 Audio output to USB Headset and HDMI

Hi, I'm having an issue with the audio output from my PS4. I have a capture card (Elgato Game Capture HD) which is connected to my PS4 through the HDMI cable. I also have a Sony Gold Wireless Headset. Whenever I connect the headset using the USB bluetooth connector and have the audio settings set to allow the headset to receive both Chat and Game audio, the game audio going through the HDMI - and thus the Elgato - is muted. Whenever I set the headset to receive just Chat audio, this then unmutes the HDMI game audio. I would like to be able to receive game and chat audio through the wireless headset, whilst also having game audio through the HDMI, however I have as yet not been able to find a setting to allow this. My question therefore is, is it possible to have game audio through the HDMI whilst also using a headset receiving game audio? Thanks in advance for any help.

MarvUK wrote:
I want to play audio from my MacBookPro (13inch) through my Roberts stereo which has a standard USB port, which can be selected as the source for playing through the speakers. Does anyone know of a cable I can use to connect the audio output on my MacBook to the USB input on my stereo?
I would have thought this was a pretty standard conversion but cant find anyone selling something with a 3.5mm jack on one end and a USB jack on the other?
Thanks in advance!
Why don't you use the USB port on your computer to get the audio out? USB to USB.
Message was edited by: BobTheFisherman

Need help for system audio output over airport

How can I set the system audio out over airport for all programs, not just iTunes
Thank you for help

You can do this using the low cost software add in called "Airfoil" - see:
http://www.rogueamoeba.com/airfoil/mac/

IPod nano's audio fried when USB connecter touched car charge power

In my car, I have a splitter for the "cigarette lighter" power, with two holes facing up. There is no aux jack on the car's radio, so I also have a little radio transmitter, and it gets power from that splitter. An audio cable connects the output of my iPod nano to the audio input of the radio transmitter. My nano had a 30-pin cable connected to it, too.
One of the "cigarette ligher" power splitter's power locations was empty, just a hole facing up. The power splitter also has a couple of USB ports, and I was in the process of attaching my nano with the 30-pin cable into that USB port for power. Unfortunately, the USB connector touched inside the power hole.
So the nano got some voltage from the car battery and/or grounded things in a weird way through the audio of the radio transmitter (connected to the car's power) and the USB end of the 30-pin cable touching the car's power.
The nano's screen froze, and it didn't respond to any button presses. So I figured it was totally fried, but it's not. Once the battery ran down, I could connect it to my Mac, and now everything but the audio is working okay. Through the 30-pin USB connection, I can load up songs from the Mac and play music out of a clock radio.
The probablem I'm writing about is that the sound coming out of the headphones is either non-existant or extremely garbled, depending on the volume level. It's fried. Can anybody give me some hope for un-frying the audio?

OK, so here is my problem. I did originally download the Beta 10.10... Apparently I kept on getting betas and was running 10.10.2 (which today is still in beta).
I did downgrade to 10.10.1 (by clicking "Download" in the App Store).
Right after the downgrade everything with syncing works perfectly.
Cheers, hope this helps someone.

Proxy client (over USB connection) with DHCP assigned IP address doesn't work with FRDM-K64F

Hello,
after reading about the issues on this forum with a static IP address on the Freescale FRDM-K64F board I went for an DHCP address but I can't get the proxy client to work over a USB connection with DHCP either.
My console tells me the following:
RTC Time: Wed 2014-01-01 00:07:27
Network initialized
IP Address:      10.143.xxx.yyy (xxx and yyy is real numbers, just changing them for the forum to letters)
Subnet mask:     255.255.252.0
Gateway:         10.143.xxx.yyy
MAC-address:     00:0c:00:06:70:00
And when I try o to connect with the following command :
java -jar proxy.jar -socket 10.143.xxx.yyy
I get the following output and no CLI interface:
Trying to open socket connection with device: 10.143.xxx.yyy:2201
Connected to the socket: Socket[addr=/10.143.xxx.yyy,port=2201,localport=49605]
Open channel 8 with hash 0x130399b3
Channel 8 CLOSED -> OPENED
notifyResponse AVAILABLE_RESPONSE on channel 8
Channel 8 OPENED -> AVAILABLE
Open channel 9 with hash 0x0
Channel 8 AVAILABLE -> REQUEST_SENT
notifyResponse ACK_RESPONSE on channel 8
Channel 8 REQUEST_SENT -> ACKNOWLEDGED
Channel 8 ACKNOWLEDGED -> DATA_SENT
notifyResponse AVAILABLE_RESPONSE on channel 8
Channel 8 DATA_SENT -> AVAILABLE
Channel 8 AVAILABLE -> REQUEST_SENT
notifyResponse ACK_RESPONSE on channel 8
Channel 8 REQUEST_SENT -> ACKNOWLEDGED
Channel 8 ACKNOWLEDGED -> DATA_SENT
notifyResponse AVAILABLE_RESPONSE on channel 8
Channel 8 DATA_SENT -> AVAILABLE
Channel 8 AVAILABLE -> REQUEST_SENT
notifyResponse ACK_RESPONSE on channel 8
Channel 8 REQUEST_SENT -> ACKNOWLEDGED
Channel 8 ACKNOWLEDGED -> DATA_SENT
notifyResponse AVAILABLE_RESPONSE on channel 8
Channel 8 DATA_SENT -> AVAILABLE
Channel 8 AVAILABLE -> REQUEST_SENT
notifyResponse ACK_RESPONSE on channel 8
Channel 8 REQUEST_SENT -> ACKNOWLEDGED
Channel 8 ACKNOWLEDGED -> DATA_SENT
notifyResponse AVAILABLE_RESPONSE on channel 8
Channel 8 DATA_SENT -> AVAILABLE
Channel 8 AVAILABLE -> REQUEST_SENT
notifyResponse ACK_RESPONSE on channel 8
Channel 8 REQUEST_SENT -> ACKNOWLEDGED
Channel 8 ACKNOWLEDGED -> DATA_SENT
notifyResponse AVAILABLE_RESPONSE on channel 8
Channel 8 DATA_SENT -> AVAILABLE
Channel 8 AVAILABLE -> REQUEST_SENT
notifyResponse ACK_RESPONSE on channel 8
Channel 8 REQUEST_SENT -> ACKNOWLEDGED
Channel 8 ACKNOWLEDGED -> DATA_SENT
notifyResponse AVAILABLE_RESPONSE on channel 8
Channel 8 DATA_SENT -> AVAILABLE
Channel 8 AVAILABLE -> REQUEST_SENT
notifyResponse ACK_RESPONSE on channel 8
Channel 8 REQUEST_SENT -> ACKNOWLEDGED
Channel 8 ACKNOWLEDGED -> DATA_SENT
notifyResponse AVAILABLE_RESPONSE on channel 8
Channel 8 DATA_SENT -> AVAILABLE
Channel 8 AVAILABLE -> REQUEST_SENT
notifyResponse ACK_RESPONSE on channel 8
Channel 8 REQUEST_SENT -> ACKNOWLEDGED
Channel 8 ACKNOWLEDGED -> DATA_SENT
notifyResponse AVAILABLE_RESPONSE on channel 8
Channel 8 DATA_SENT -> AVAILABLE
I don't know if this matter but my java version is:
java version "1.8.0_31"
Java(TM) SE Runtime Environment (build 1.8.0_31-b13)
Java HotSpot(TM) 64-Bit Server VM (build 25.31-b07, mixed mode)
Any ideas or am I missing something completely?
Thanks
Andy

Hi, Andy. Please excuse my ignorance, I am very new at this ME embedded stuff. I had my challenges, but was finally able to get my board up with a static address. I went through a lot of headaches, but finally got the board flashed with the latest bin. I then updated the jwc_properties.ini file on the SD card to include:
# Whether static configuration or DHCP server is used do get IP address. Possible values: dhcp,static
ip.method = static
# IP address,used with static IP configuration only
ip.addr = 192.168.0.30
# Network mask,used with static IP configuration only
ip.netmask = 255.255.255.0
# Network gateway,used with static IP configuration only
ip.gateway = 192.168.0.1
# DNS server,used with static IP configuration only
ip.dns = 208.67.222.222
# MAC address
mac.addr = 01:02:03:04:05:06
I was then put the card into the board, disconnected both USB cables, and then powered it up. From that point I was able to ping the board. I can now use Device Manager in NetBeans to connect to the device.
I am struggling quite a bit with many issues. It seems that when I try to stop the app from NetBeans, the "please wait" dialog hangs forever. I have to use Windows Task Manager to kill the "JavaW" process tree and reconnect the board.
I have also had trouble understanding the GPIO pin assignments. Another gotcha was trying to use the AutoStart feature. Since the sample app never cleanly exited, I thought that I bricked the board. The secret is to power it down, remove the SD card, and then delete all files EXCEPT the jwc_properties.ini file, put the SD card back in, then power up. It seems that the board forgets that the MIDlet was installed.
Hope this helps others out there.
Please everybody, post your experiences here. There seem to be very few of us, and finding pearls in the dust is rare right now.
Regards,
Pete

Sending loafs of http packets over one connection

when i start my app i have it make an http connection (gets the stupid airtime message out the way)
once its done that every time u press a button its ment to send a http packet with some data sayin wich button u pressed
thing is every time i send a message the connection hangs i can never get past the ".close();" method
i need it to show down the stream and let me send other packets as other buttons are pressed
but its hanging on the first message been sent any ideas
can u send loads of messages over 1 connection ? thanks

Can't comment on the problem you're facing, but I think you should know that it's recommended to implement Runnable, not extend Thread. From the API for Runnable:
In most cases, the Runnable interface should be used if you are only planning to override the run() method and no other Thread methods. This is important because classes should not be subclassed unless the programmer intends on modifying or enhancing the fundamental behavior of the class.
luck, db

Select HDMI audio output when headphone connectted

In Snow Leopard, I can connect my headphone to Mac Mini, and select HDMI audio output when I want sound from monitor, or select headphone.
But in Lion, I can not select HDMI when headphone is connectted, so I have to plug/unplug headphone very often, quite anoyying...
How can I disable this feature?
Thanks.

I've just started a thread regarding this exact issue, but in Mountain Lion. Did you ever solve it? Or, if you have upgraded to Mountain Lion, do you still experience it?
EDIT: Almost exactly 1 year later you get a response. lol.

RE: Capture Problems on External Drive/Audio drift over time

Greetings FCP gurus -
I posted earlier on the LAFCPUG board and now seek worldwide assistance with a problem I can't seem to resolve.
Been working in FCP since 1.5. Been successfully capturing on external FW devices for over 4 years with an old 500 MHz Powerbook - no problems.
I recently upgraded to FCP 5 Studio on a PowerBook 1.5 GHZ 512 RAM FW 4/8 machine. Prior to upgrade, I was running Panther (OS X 10.3X) and FCP 4.5 HD. This configuration allowed me to flawlessly capture to external FW drives from Panasonic AJ-D250 DVCPRO deck using a standard DVNTSC codec. Prior to that I was also able to do the same from a Sony DSR-11 deck. No longer.
I now get a constant audio "drift" every time I capture. I've tried shortening the clips to under 10 minutes - even as small as 5 minutes - but the audio progressively drifts as the clip plays on. This is true in both FCP and QT. I completely reformatted the laptop/wiped the drive prior to install (Unjournaled) and it currently has 25 gig available disk space on the 80 gig internal drive. Additionally, I'm using LaCie 7200 RPM Oxford 911 chip-set external drives. I've tried with clean D2 FW 800 and Porsche FW 400 drives to no avail - even reformatted one and still no luck.
Current configurations attempted:
FW 800 PB port to FW 800 drive port/DVCPRO deck to FW 400 PB port
FW 800 PB port to FW 800 drive port/DVCPRO deck to FW 400 drive port
FW 400 PB port to FW 400 drive port/DVCPRO deck to FW 400 drive port
..and so on.
I'm at wit's end and appreciate any insight. Is anyone else having issues capturing to External FW media via FCP 5 Studio/OSX 10.4.x? I don't currently have the option of internal drives to capture as I'm working at a client site. Additionally, I sold my FCP 4.X upgrade software so going backwards is not an option either... Even took the deck home and tried this same trick on my home machine (Dual 500 G4 Sawtooth) to the internal ATA drive with the same result -...drifting audio. Since upgrade this is my first capture and needless to say it's not going well.
Help, information and insight are greatly appreciated. Thanks

Hi Expressofiend! Were you able to resolve this issue yet? We are having an IDENTICAL problem. We shoot on a Canon XL2, and thought that might be causing the audio drifting issue? We also have a Panasonic AD-J 250 deck, and when we capture, say an interview, the lip sync is perfect for a few minutes, then drifts as much as a second or two over 20 minutes of capture. It's making us crazy. Happens when we capture to an external LaCie or our internal drive. We're using Apple's 2.7 gig duel processor computer. 2 Gig of ram, all the latest versions of Tiger, Final Cut Pro 5, etc. If we look at the Quicktime movie - it shows the audio drift - you don't even need to view it through Final Cut!
Interestingly, if we use the Canon XL2 camera as the source player when we capture the 20 minute piece - there is NO audio drift or syncing problem. Only with the AJD 250! We've spoken with the senior engineer at Panasonic - he doesn't think it's the DVCPro unit. Says drifting audio isn't the kind of thing an out of adjustment or broken deck would do.
Anyway, any answers here yet?
Thanks,
Larry

Mutli channel audio output via usb?

Hi there
I've had a browse through the support forums and can't find an answer to my question, so here it is:
I want to split the audio channels through the internal soundcard (I have an ibook g4) so that I can use audio mixing software.
Before I go out and buy some usb speakers, I was wondering if connecting speakers through the usb port would automatically do this - or, is there a way to assign an output route for each channel within the computer?
Any help is much appreciated.
Thanks

You'll need a "usb soundcard" that you can purchase from both apple store or just google it and I'm sure they have them all over the place (ebay, thinkgeek, ect). The built in soundcard is a 2 channel 44khz sound card with 1 output and 1 input via the built in microphone. Also, the only usb speakers I have ever seen are just usb powered, but they still plug in to the headphone plug.

[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)

Problem sending FCP X audio output over USB connection

Similar Messages

Maybe you are looking for