Saving As Oggs Truncates Audio?

I'm having a problem I can't understand. I edit a short audio file (wav), and save it as .ogg. It's fine as long as it is still open in AA. but when I close it and open it again, the end of the file gets cut off (like the last .25 seconds). Does anyone have any idea why this could be happening? It's really frustrating. Thanks.
Ken

Durin,
A bit late getting back to this thread aren't I? Well, I don't remember what I did back then, but this problem has reared its ugly head again.
I tried switching the ogg option to "fixed bitrate (128)," but it still cuts off lid the last several milliseconds of my file. Is there, perhaps, something else I'm doing wrong?
Thanks.

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[SOLVED] How to stream audio output over wifi?

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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
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# 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
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# "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
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# 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>
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# 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)

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12 From the video subclips, I export to Media Encoder, and choose either FLV4 match attributes (High Q). This I modify to custom 2 pass (Custom save).
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14. I export my Audio subclips to Media player, and select MP3 option and this summary also reports stereo. The Audio outcome is stereo, when played in any player and in Captivate 5.
I cannot understand how a single, continuous track edited uniformly in Soundbooth to stereo, can produce different outcomes in Media player.
I would so appreciate a solution to this problem, because, in the Captivate 5 e-learning production, the switch between stereo audio (behind PowerPoints), and l. channel video is distracting. Naturally, the l. channel only is a thin sound. It is as if it had never been edited in SoundBooth at all.

Saving As Oggs Truncates Audio?

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