AC3 5.1 encoding mystery

When we create DVD's with 5.1 surround in DVD Studio Pro, they play back perfectly on regular DVD players, but when playing back on any of our 8 Macs at our studio, the stereo downmix comes out heavy right channel. Commercially produced DVD's playback fine however.
I assume it has something to do with APack, but what? What would cause the encoded AC3 file to playback fine on regular DVD players, but not on the Mac? The other interesting thing is that the "center" channel remained centered, so I'm guessing that the downmix is folding the surround channels both into the right channel.
Or perhaps there is something quirky in the Mac decoder and downmix algorythym, but what's different about my AC3 files from a commercial DVD?
Mac G5 Mac OS X (10.4.2)

I'm sorry to say that this is one that I remember coming across, a along time ago and if memory serves me right, it had something to do with settings in Apple DVD player, but then again, I could have imagined the whole thing?

Similar Messages

Problems with ac3-audio not encoded in apple's compressor

Hi everyone,
I'm working with DVD Studio Pro 4.0 and when encoding audio with the compressor into ac3, there were hardly any problems.
Now I have to get along with external ac3-files, not encoded with compressor. Settings, format etc. are identical to the compressor-created files; despite of these similar settings, I now expericene a very low sound (about 3x lower), not comparable to the volume of the compressor encoded files.
Has anyone an idea on this subject?
Thanks.
Greetz
Power Mac G5 Dualcore2Ghz Mac OS X (10.4.6)

You might get better responses in the DVD Studio Pro or Compressor forum?

Will realtime AC3/DTS surround encoding be available on X-

Yeah, a considerable amount of people have asked for this feature in the past, and with the introduction of next gen. technology from Creative, I thouhght now would be a good time to ask you(Creative or other informed people)about this feature again.
So...do the HI-FI people have something to look forward to?
Cheers! Message Edited by Mr_DJ on 06-6-2005 0:02 AM

BadBoy wrote:
Mr_DJ wrote:
Yeah, a considerable amount of people have asked for this feature in the past, and with the introduction of next gen. technology from Creative, I thouhght now would be a good time to ask you(Creative or other informed people)about this feature again.
So...do the HI-FI people have something to look forward to?
Cheers!
Message Edited by Mr_DJ on <SPAN class=date_text>06-6-2005 <SPAN class=time_text>0:02 AM
Highly unlikely at this current state as there's no mention of this at their xifi page. Also, adding hardware realtime AC3 encoder would definitely increase the cost of the sound card and it would not be in an average consumer range buying power. I think DD Li've at this moment at only licensed to onboard audio.
I was thinking along the lines of a Deluxe edition or a add-on option, as the way of introduce this feature.
That way, only the people who needs it, will pay for the added cost. For this to be an income success, there would of course need to be a high enough demand. But here I would just lead the attention to Nvidia's SoundStorm chip, which many loved an misses today because of it's realtime encoding capablility together with EAX support. So if Creative introduced this feature, I'm pretty sure they will get a lot of new costumers.
I don't know how costly it is to get the needed licensees from the people owning the DD/DTS formats, but if Nvidia could afford it, I'm pretty sure, Creative can too!
I know there's no mention of this feature on the X-FI pages(though in theory, it could still show up, as the page is still getting updated with new X-FI info/features).
Perhaps miss Karskens can shed some more light on this topic?
Cheers!
Message Edited by Mr_DJ on <SPAN class=date_text>06-6-2005 <SPAN class=time_text>09:38 PM
Message Edited by Mr_DJ on 06-6-2005 09:43 PM

CS6 Audio corrupted on M2TS H264/MPEG2 + AC3 TS Streams Encoding

Hello,
while encoding in Blu Ray H264 or MPEG2 with Dolby Digital (in this case 224 Kbits) for a multiplexed TS Stream the audio seems to be corrupted. Strangely plays in Windows Media Player but stutters in VLC Player and after re-import in CS6 (6.0.3) there is no audio hearable although listed in the properties of the file.
The Export Settings are almost the standard ones, except for setting to AC3 224 Kbits and TS output.
anyone having the same problems?

some customers need multiplexed files for their workflow.
Check that. Authoring as standard industry practice uses separate files. It's more likely they just want one file, in which case you might convince them of the better way (which as it happens, does work)

Silent / Mute AC3 track after encoding in Compressor 2

Hi,
I exported audio from a 2hr 27min time line in FCP 4.5 to create an .aif file.
I then transcoded it to an .AC3 file on my macbook using Compressor 2. When I listen to it in VLC, it all sounds perfectly fine. But when I import it into DVD Studio Pro 3, it is silent / mute.
I have looked back over this forum and found several people reporting similar problems, without any solution.
Grateful for any suggestions.
Thanks
Jamie

Here is another possibility. If you added you ac3 to a menu or track and then made additional corrections in FCP and recompressed them to the same name, DVDSP has a bit of trouble updating sometimes....not sure why, but a quick solution is to re compress again, but with a different name. then relpace the old one with the new one in DVDSP. same thing happens with video sometimes.
g4 duel 1 gig

Problems encoding an AC3 with compressor

When I use the batch surround group, import all 6 surround files (L C R Ls Rs LFE, all are individual mono AIFF files at 48khz/16bit), and export a 5.1 AC3 - I open the exported file in QT using a AC3 decoder plug-in ( http://trac.cod3r.com/a52codec ), and then look at the sound properties, and it shows only the L&R channels in the properties (along with 2 unknown channels & 2 unused channels, totaling 6 looking in the sound settings). The odd thing that other AC3s that were encoded with the older A.Pack that came with DVDSP3 work just fine in QT (and show all 6 channels properly assigned). One thing i should also mention, is that when i mute all of the channels (except the main L & R) with the Compressor exported AC3, the audio seems to have been mixed down from all of the original 6 tracks into the L and R. This is very frustrating and have spent awhile trying to figure this out. Help would be much appreciated. Jared

When I use the batch surround group, import all 6 surround files (L C R Ls Rs LFE, all are individual mono AIFF files at 48khz/16bit), and export a 5.1 AC3 - I open the exported file in QT using a AC3 decoder plug-in ( http://trac.cod3r.com/a52codec ), and then look at the sound properties, and it shows only the L&R channels in the properties (along with 2 unknown channels & 2 unused channels, totaling 6 looking in the sound settings). The odd thing that other AC3s that were encoded with the older A.Pack that came with DVDSP3 work just fine in QT (and show all 6 channels properly assigned). One thing i should also mention, is that when i mute all of the channels (except the main L & R) with the Compressor exported AC3, the audio seems to have been mixed down from all of the original 6 tracks into the L and R. This is very frustrating and have spent awhile trying to figure this out. Help would be much appreciated. Jared

Compressor 3 encoded AC3 cannot be played

Had an issues with compressor 3.
Ac3 files I encoded does not work with mplayer or other ac3tools. All of which give me the "check sum error".
Even if I use divx muxer or mkv muxer, it will not read the ac3 files which I encoded from compressor 3.

I built the site and player from scratch. It's very simple: The Flash player progressively plays MP4 videos using its FLVPlayback component.
When I view the MP4 files directly in Firefox (on my Mac) via Quicktime player:
http://www.daydreamtv.org/media/200806150739_1robbie.mp4
They are perfect and smooth. My guess is it's got to be the player, but I wanted to ask you before I start digging into it because you pointed me in the direction of MPEG Streamclip and I figured you have a bit of experience working with browser-based video delivery.

Aiff or .AC3 on DVD disc?

Hi, I am finishing my band's DVD and I have the final mix in 2 formats. 1 is stereo aiff and the other is a 5.1 .ac3.
My question is, is there any advantage to me converting my stereo file to a 2.0 channel .ac3? Will it play back better in more players because of the lower bit rate required for the stereo .ac3?
It fits on the disk either way and the .aiff sounds good to me so I am inclined to keep it but I would encode it if there are any good reasons to.
Thanks in Advance,
Steve

Just to get the thread back on topic, you can use either AC3 or PCM audio on a DVD, and both should be able to be played back in players around the world (with older PAL systems more likely to struggle with AC3)
AC3 allows for a lower bandwidth, as you point out, and this is only important in two situations. Firstly, if you have a lot of video to fit onto a disc it is wise to reduce the audio file size for the relatively tiny loss of quality (I'd bet most people wouldn't hear it on a standard player set up) because audio is easier to compress and get right than video. Secondly, if you are burning a disc yourself rather than going to replication.
When you burn a disc you are not creating a true DVD-Video disc, but a DVD-R (or +R). These have different compatibility in players, and since they use an entirely different technology for the reflective layer (it is a vegetable based dye rather than aluminium film) you get different responses from set top boxes.
The DVD specification allows for a total bandwidth of 10.08Mbps on a DVD. However, up to 9.8Mbps of this can be used for video, leaving not much for audio. You can, of course, use more for audio and less for video if you wish.
There is a bit of a 'gotcha' though. With a DVD-R or DVD+R you would be very unwise to use these high bandwidths. In fact, since the reflectivity of the surface is so much less you need to reduce the bandwidth. Anecdotal tests from authors all around the world have revealed a reasonable upper limit of around 7.4Mbps produces the most compatible discs in the widest range of players. Going higher introduces an ever diminishing return, in that you gain file size with no appreciable improvement in quality, and you increase the chances of your disc 'choking' the player as it struggles to get the data off at the higher rates.
My advice would be to reduce the audio to AC3 wherever you can, but try it and listen to the result to see if it is acceptable. If you can't tell much of a difference, and you are well attuned to how it should sound, then it's very unlikely anyone else will notice.
So - got lots of room on the disc? Want to have the better sound and the widest compatibility? Use PCM... but remember the upper limits for the bandwidth if burning it yourself.
Got lots of video? Can't hear the difference between the two audio types really anyway? Reduce the file size of the audio by creating AC3 files, and encode the video to get a better visual quality.

Can't import AC3 file...

Hi,
I have some 5.1 AC3 files I encoded in another (older) application and am trying to import them into DVDSP. The files look OK and are recognized as DVDSP files with an AC3 extention. However, every time I try to import them as assets I get the error message "Incompatible Format." Anyone know what's going on here? I don't see how its incompatible.
Thanks,
Brian

If your .ac3 file is indeed 44.1 kHz, that's at least part of your problem.
According to page 25 of the DVDSP manual, you must use either 48 or 96 kHz audio. It goes on to say that DVDSP's encoder will convert other audio sample rates, but I don't think it can do that for .ac3 stuff.
(But I refer to the manual because I don't have direct experience with this...)

Encoding in DVD SP

I have a 150 min. dance program that I am trying to burn with 7 menus and 36 buttons. I have customized the encding down to 1 pass at 3.6 and at the very end of burning it tells me that I don't have enough media output. I am burning to a DVD -R 4x which has always worked before.
I am deperate to burn a good looking DVD for my client.
Please help.

Hello Cecilla,
It is possible to get a 150 minute video on a standard DVD. I'd like to know how you are conducting your encoding? Did you run your source video through Compressor before bringing it into DVD Studio Pro? Or are you using DVDSP to handle the encoding?
One thing I'm assuming is that your audio is still AIFF and not AC3. When encoding for DVD, it's highly recommended that you not only encode your video to M2V, but your audio to AC3 (Dolby). The main difference between AIFF and AC3 is that AIFF is uncompressed, and in that state can give you large file sizes, and will take up a lot of bandwidth on your DVD. AC3 compression will greatly reduce audio file size and bandwidth, thus allowing you to fit more on a single DVD.
My typical workflow when authoring a DVD is export the video and audio through compressor, then take the resulting M2V & AC3 files directly into DVDSP. I normally use my own custom settings when encoding for DVD. Those settings can be found here.
I also recommend that you take a look through this site for a better detailed description on how to build a DVD.
Hope this helps!

Solution - AC3 DTS D3D EAX Full positional sound to Home Theatre Recei

Why you, the best porducer of sound cards in the world doen't have a solution like that.
http://www.cmedia.com.tw/product/CMI976.htm
Only more few transintors to add to the 5 million you have a you have it. in the top one X-fi cards.
Regards,
Marco Polo

People just don't GET audio.
A PC is not like a DVD player. One of the main reasons for AC3 and DTS is that the media on which motion pictures are typically stored (ie. the DVD) does not have enough storage capacity to hold a feature-length's worth of multi-channel audio.
The reasons why the actual AC3/DTS stream is sent over a physical cable to the receiver are many:
) Content providers don't like the idea of uncompressed and unencrypted audio transmitting over cables which can easily be intercepted and copied bit-for-bit.
2) Since not everyone has surround sound speakers, it doesn't make sense to add cost to the DVD player by having AC3 and DTS decoders on board.
The major difference between DVD's and games in terms of audio is that for DVD's the sound is already pre-recorded and only needs to be played back. For games, the sound mix is created interacti'vely. The fact that the sound in games is created interacti'vely makes a technology such as AC3/DTS irrelevant because ) it doesn't need to be stored which means there are no constraints leading to the need for compression, and 2) there is no issue with content protection because the interacti've sound mix coming from a game does not in and of itself represent someone's intellectual property (like a music recording or movie soundtrack).
The fact that AC3/DTS is digital is incidental. "Digital" is not magic, its just a convenient storage format. ALL sound ultimately must be converted to an analog signal to be played on speakers. On PC's which haven't been typically connected to home theater receivers, there is no point to doing this digital-analog conversion anywhere but the soundcard itself. Thus, digital speakers for PC's, when used exclusi'vely with a PC, are mere gimmicks.
Now, you may understand all this yet still argue that the ability to output AC3/DTS from a PC serves a purpose. This way, a PC can be used as a home theater server that is ALSO capable of playing games, with ALL sound (games included) encoded in the AC3/DTS format. It may be convenient if your receiver does not have the multi-channel analog inputs necessary to get surround in PC games, especially if you have unused optical/coax inputs on your receiver. For this situation, Creative has developed the DTS-60 external decoder device. Yes, it is another $00 bucks, but think of it this way: the real-time AC3 cards generally cost around $00, but they have none of the other 3D audio hardware acceleration offered by the X-Fi. The X-Fi (cheap version) costs around $00 bucks but doesn't included the encoding stuff. AC3 or DTS encoding is NOT free. If you feel that you don't need the hardware acceleration of interacti've 3D audio offered by the X-Fi, go ahead and buy a HDA-Mystique. If you want the best, get an X-Fi and the DTS-60 (when it comes out).
I'd imagine that in the future, Creative will come out with a soundcard that features integrated DTS Interacti've encoding.

The Sorry State of Sound In Linux

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READ: http://insanecoding.blogspot.com/2007/0 … linux.html
I've only discovered this because of an ALSA bug was making my life miserable. That article was written about a month or two before OSS4 (Open Sound System v4) was released under GPL and CDDL. This past January, it was released under BSD as well. Unfortunately for the developer, he does not understand open source very well. He is now reporting that his revenue went significantly down..
OSS4 does not equal to the crap OSS in Linux 2.4 kernels. Even OSS3 != OSS3 in former linux kernels. He was neglecting the open source version in favour of the commercial one. Instead of improving it, someone forked it. And, the fork became popular. Unfortunately, the original author had no interest in working on the fork. So, he only focused on the commercial version. Now, many years later, the commercial version is fully open source. There is no more commercial version.
I've tried it, and I really like it. Music sounds much better than on ALSA. For example, screaming in alternative rock is more legible. It's like enabling "enhance voice" in a phone. It supports higher PCM values without noise.
The mixer (vmix) is capable of 18 channels. It can also do per application volume control (like PulseAudio).
It totally makes sense that OSS gets back into the kernel because it works on almost every UNIX and UNIX-like system (except OS X). ALSA only works on Linux, and according to that article, developers still prefer the OSS API, even on ALSA. However, the ALSA OSS API is lacking according to an ALSA developer. I can confirm that the comment on the ALSA OSS Emulation API working better than the ALSA API. XMMS with ALSA enabled freezes my system. XMMS with OSS enabled on ALSA does not.
It worked without any configuration besides muting some channels to kill the noise.
THE BAD
ossmix and ossxmix are totally unusable because they do not name the channels properly. ossxmix uses 100% CPU if Compiz is enabled. Version 4.1 will fix this.
I had to figure out WTF ossxmix.codec1.connector.jack14.jack 54:54 means. Sane names need to be added to the jacks such as "mater, front, input, auxiliary, microphone, etc.."
It's best to use ossxmix (GTK+ mixer) and playing with all the jacks to figure out what each one does.
One interesting control is how OSS4 should behave (Fast, Medium High, Professional, etc.). It's probably a latency control. ossxmix is a demo app on how you can control the mixer from GTK. It needs to be made usable.
If you use Media Player Daemon, VLC, and MPlayer, OSS4 works beautifully. They use OSS directly. Others have problems.
Totem does not play sound. Totem-Xine uses 100% CPU (Xine-UI works with low CPU usage).
In terminal I noticed some output from oss mixer control saying it received bad arguments. OSS4 is backwards compatible with OSS3, but xine-lib may be using the API badly.
The progress bar (seeker) in GStreamer based applications (Rhythmbox, Banshee) does not work if you use vmix, or it may work, but it will be unsure of the length of a song. You will see it constantly adjusting a few seconds up or down. By the middle of the song, the length reporting generally stabilises. You have to enable softoss (the old mixer). vmix and Gstreamer behave very badly. The Gstreamer developer responsible for OSS claims that it does on his system. Though, he is most likely using the trunk version.
You have to apply this patched gstreamer to make volume control in GNOME work.
While the progress bar problem is fixed with softoss, volume control does not work, even with the patched GStreamer. It only works with vmix. System > Preferences > Sound lists nothing under "Default Mixer Tracks".
Sound does not work in KDE4 at all. While Phonon is based on Xine, which has OSS support, it queries HAL, which does not support OSS4 yet. Therefore, it thinks that there is no sound card.
As for KDE 3.5.9. Amarok works, Noatun works, and anything based on MPlayer should work. KMIX does not work. Same in KDE4.
DOWNLOAD IT (Popular Distributions Linux and Unix Distributions)
Gentoo ebuild
Arch Linux
The Arch Linux wiki has a good article. Read it.
A Ubuntu user has published an install guide as well.
Check Configuring Applications for OSSv4 after you install it.
Overall, I think it's a million times better than ALSA. It's cross platform, and it's stable. It was released a year ago on 15th of March 2007. Since he does not have a marketing department, no one has heard of it.
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Last edited by SpookyET (2008-03-29 15:01:19)

wyvern wrote:
Hmm, so it's a choice - have to enable effects and use an older mixer, or put up with a 'stuck' progress bar... Well, as I said, I use MOC, and I'm happy with my mixer setup so far, so I'll stick to what I've got.
I can't believe I finally have my headphones working, or external speakers, should I so wish - and what with the ATI divers *finally* providing tear-free video, it's like it's my birthday
Since you mentioned ATI and went offtopic, can you pastebin your ATI xorg? It works fine for me, but If I enable compiz, i get flickering video and slow scrolling. So, I'm stuck with the open source driver, which uses a lot more CPU when I play video.
I can also confirm what the article said about the ALSA OSS emulation. It does work better than ALSA with the ALSA API. xmms with ALSA enabled freezes my sytem. xmms with OSS enabled running ALSA does not. I'm tired of this bollocks. We have to mount a campaign to get OSS back into the kernel.
Anyway, this is my OSS output.
OSSINFO
Version info: OSS 4.0 (b1014/200803130443) (0x00040003) GPL
Platform: Linux/i686 2.6.24-ARCH #1 SMP PREEMPT Wed Mar 5 12:07:52 UTC 2008 (mercurius)
Number of audio devices: 10
Number of audio engines: 13
Number of mixer devices: 2
Device objects
0: osscore0 OSS core services
1: hdaudio0 Intel HD Audio interrupts=1389433 (1526696)
HD Audio controller Intel HD Audio
Vendor ID 0x80862668
Subvendor ID 0x10250070
Codec 0: ALC880 (0x10ec0880/0x08800000)
Codec 1: Unknown (0x11c13026)
2: softoss0 OSS Virtual Mixer v3.0
Mixer devices
0: High Definition Audio ALC880 (Mixer 0 of device object 1)
Device file /dev/oss/hdaudio0/mix0, Legacy device /dev/mixer0
Priority: 10
Caps:
Device handle: PCI00701025-0000:00:1b.0-mx01
Device priority: 10
1: Virtual Mixer (Mixer 0 of device object 2)
Device file /dev/oss/softoss0/mix0, Legacy device /dev/mixer1
Priority: 1
Caps: VIRTUAL
Device handle: softoss0-mx01
Device priority: 1
Audio devices
HD Audio front /dev/oss/hdaudio0/pcm0 (device index 0)
Legacy device /dev/dsp0
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 0/HD Audio front
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au01
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 8
Native sample rates (min - max): 44100 - 192000 (44100,48000,96000,192000)
HW Type: Not indicated.
Minimum latency: Not indicated
HD Audio rear /dev/oss/hdaudio0/pcm1 (device index 1)
Legacy device /dev/dsp1
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 1/HD Audio rear
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au02
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 192000 (44100,48000,96000,192000)
HW Type: Not indicated.
Minimum latency: Not indicated
HD Audio center/LFE /dev/oss/hdaudio0/pcm2 (device index 2)
Legacy device /dev/dsp2
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 2/HD Audio center/LFE
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au03
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
HD Audio side /dev/oss/hdaudio0/pcm3 (device index 3)
Legacy device /dev/dsp3
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 3/HD Audio side
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au04
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
HD Audio spdif-out /dev/oss/hdaudio0/spdout0 (device index 4)
Legacy device /dev/dsp4
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 4/HD Audio spdif-out
Available for use
Input formats (0x00001410):
AFMT_S16_LE - 16 bit signed little endian
AFMT_AC3 - AC3 (Dolby Digital) encoded audio
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001410):
AFMT_S16_LE - 16 bit signed little endian
AFMT_AC3 - AC3 (Dolby Digital) encoded audio
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au05
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
High Definition Audio rec1 /dev/oss/hdaudio0/pcmin0 (device index 5)
Legacy device /dev/dsp5
Caps: TRIGGER MMAP
Modes: INPUT
In engine 1: 5/High Definition Audio rec1
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au06
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
High Definition Audio rec2 /dev/oss/hdaudio0/pcmin1 (device index 6)
Legacy device /dev/dsp6
Caps: TRIGGER MMAP
Modes: INPUT
In engine 1: 6/High Definition Audio rec2
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au07
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
High Definition Audio rec3 /dev/oss/hdaudio0/pcmin2 (device index 7)
Legacy device /dev/dsp7
Caps: TRIGGER MMAP
Modes: INPUT
In engine 1: 7/High Definition Audio rec3
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au08
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
High Definition Audio spdif-in /dev/oss/hdaudio0/spdin0 (device index 8)
Legacy device /dev/dsp8
Caps: TRIGGER MMAP
Modes: INPUT
In engine 1: 8/High Definition Audio spdif-in
Available for use
Input formats (0x00001410):
AFMT_S16_LE - 16 bit signed little endian
AFMT_AC3 - AC3 (Dolby Digital) encoded audio
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001410):
AFMT_S16_LE - 16 bit signed little endian
AFMT_AC3 - AC3 (Dolby Digital) encoded audio
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au09
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
OSS Virtual Mixer v3.0 Playback /dev/oss/softoss0/pcm0 (device index 9)
Legacy device /dev/dsp9
Caps: TRIGGER MMAP VIRTUAL
Modes: OUTPUT
Out engine 1: 9/OSS Virtual Mixer v3.0 Playback
Available for use
Out engine 2: 10/OSS Virtual Mixer v3.0 Playback
Available for use
Out engine 3: 11/OSS Virtual Mixer v3.0 Playback
Available for use
Out engine 4: 12/OSS Virtual Mixer v3.0 Playback
Available for use
Input formats (0x00000010):
AFMT_S16_LE - 16 bit signed little endian
Output formats (0x00000010):
AFMT_S16_LE - 16 bit signed little endian
Device handle: softoss0-au01
Related mixer dev: 1
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 1 - 2
Native sample rates (min - max): 48000 - 48000
HW Type: Not indicated.
Minimum latency: Not indicated
OSSMIX -d0
Selected mixer 0/High Definition Audio ALC880
Known controls are:
codec1.connector.jack14.mode <jack|input> (currently jack)
codec1.connector.jack14.mute ON|OFF (currently OFF)
codec1.connector.jack14.front-m ON|OFF (currently OFF)
codec1.connector.jack14.inputmi ON|OFF (currently ON)
codec1.connector.jack14.jack <both/leftvol>[:<rightvol>] (currently 57.9:57.9 dB)
codec1.connector.jack15.mode <jack|input> (currently jack)
codec1.connector.jack15.mute ON|OFF (currently OFF)
codec1.connector.jack15.rear-mu ON|OFF (currently OFF)
codec1.connector.jack15.inputmi ON|OFF (currently OFF)
codec1.connector.jack15.jack <both/leftvol>[:<rightvol>] (currently 57.9:57.9 dB)
codec1.connector.jack16.mode <jack|input> (currently jack)
codec1.connector.jack16.mute ON|OFF (currently OFF)
codec1.connector.jack16.center/ ON|OFF (currently OFF)
codec1.connector.jack16.inputmi ON|OFF (currently OFF)
codec1.connector.jack16.jack <both/leftvol>[:<rightvol>] (currently 57.9:57.9 dB)
codec1.connector.jack17.mode <jack|input> (currently jack)
codec1.connector.jack17.mute ON|OFF (currently OFF)
codec1.connector.jack17.side-mu ON|OFF (currently OFF)
codec1.connector.jack17.inputmi ON|OFF (currently OFF)
codec1.connector.jack17.jack <both/leftvol>[:<rightvol>] (currently 57.9:57.9 dB)
codec1.connector.jack18.mode <jack|input> (currently jack)
codec1.connector.jack18.mute ON|OFF (currently OFF)
codec1.connector.jack18.jack <jack|jack|jack|jack> (currently jack)
codec1.connector.jack19.mode <jack|input> (currently jack)
codec1.connector.jack19.mute ON|OFF (currently OFF)
codec1.connector.jack19.jack <jack|jack|jack|jack> (currently jack)
codec1.connector.jack1a.mode <jack|input> (currently jack)
codec1.connector.jack1a.mute ON|OFF (currently OFF)
codec1.connector.jack1a.jack <jack|jack|jack|jack> (currently jack)
codec1.connector.jack1b.mode <jack|input> (currently jack)
codec1.connector.jack1b.mute ON|OFF (currently OFF)
codec1.connector.jack1b.jack <jack|jack|jack|jack> (currently jack)
codec1.record.rec1 <both/leftvol>[:<rightvol>] (currently 31.9:31.9 dB)
codec1.record.rec1.rec1 <jack|jack|jack|jack|jack|jack|jack> (currently jack)
codec1.record.rec2 <both/leftvol>[:<rightvol>] (currently 31.9:31.9 dB)
codec1.record.rec2.rec2 <jack|jack|jack|jack|jack|jack|jack> (currently jack)
codec1.record.rec3 <both/leftvol>[:<rightvol>] (currently 31.9:31.9 dB)
codec1.record.rec3.rec3 <jack|jack|jack|jack|jack|inputmix|jack|jack|jack|jack> (currently jack)
codec1.misc.jack1 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack2 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack3 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack4 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack5 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack6 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack7 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack8 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.inputmix <jack|jack|jack|jack|jack|jack|jack|jack> (currently jack)
OSSMIX -d1
Selected mixer 1/Virtual Mixer
Known controls are:
synth <both/leftvol>[:<rightvol>] (currently 100:100)
pcm <both/leftvol>[:<rightvol>] (currently 100:100)
2 <leftVU>:<rightVU>] (currently 137:136)
autoreset ON|OFF (currently OFF)
effects.eq.prescale <monovol> (currently 255)
effects.eq.lo <monovol> (currently 128)
effects.eq.mid <monovol> (currently 128)
effects.eq.hi <monovol> (currently 128)
effects.eq.xhi <monovol> (currently 128)
effects.eq.bypass ON|OFF (currently OFF)
voices.pcm9 <both/leftvol>[:<rightvol>] (currently 100:100)
voices1 <leftVU>:<rightVU>] (currently 135:135)
voices.pcm10 <both/leftvol>[:<rightvol>] (currently 100:100)
voices2 <leftVU>:<rightVU>] (currently 118:119)
voices.pcm11 <both/leftvol>[:<rightvol>] (currently 100:100)
voices3 <leftVU>:<rightVU>] (currently 0:0)
voices.pcm12 <both/leftvol>[:<rightvol>] (currently 100:100)
voices4 <leftVU>:<rightVU>] (currently 0:0)
Last edited by SpookyET (2008-03-15 01:17:07)

What will I need to make a successful conversion from Quark 8.5 with Autoprice to InDesign?

We have Quark 8.5 and the operating system version: iMac OS X 10.6.8 (10K549) - Kernel Version: Darwin 10.8.0, and two graphic designers.
We use AutoPrice to import Prices, Model Numbers, and Shipping symbols among other things.
What will I need to convert our Quark files, while preserving our data tags for data import?
Will I need a hardware upgrade to be successful? If so what would you suggest?
We are a small company without an unlimited budget.
Any advice, suggestions or recommendations would be appreciated
Are there other options that we should consider?
What are some of the critical decisions or thinking points we should consider?
Can you suggest some resources that would help me make this decision?
Thank you in advance for any help you can provide.

Remember there are several flavors of HD including HDV, DVCProHD and AVCHD.
Otherwise, export current settings audio and video self contained.
Drop it into Compressor and elect the DVD 16:9 preset for the length of the movie in total minutes. Choose ac3 audio.
Encode.
Elect the HD DVD preset after launching DVD SP.
Take the m2v and ac3 files and import them into DVD SP 4.1.2 or DVD SP 4.2
Burn to a red laser dye DVD5 disk. You can get about 40 minutes of DVCProHD on each one.
Play in a Toshiba HD DVD A-1 set top with the latest firmware.

Chapter stops before end of chapter when I add audio

Hi
Basically the problem is, if I have a timeline with several chapters, each chapter being a different m2v file, then I want to add AC3 files for the audio. I have different AC3 files for each chapter.
Before adding the audio the preview works fine, but if I add audio to a chapter then iy abruptly ends just after a few seconds
Can you help me please?
Thanks a lot

Maybe it is just a preview problem. You could burn a
rewritable DVD to test if the final output would be okay.
I don't remember seeing any posts about this problem, but I vaguely recall having seen a thread about an issue with butting AC3 files together in a timeline. Because of the way AC3 files are encoded they aren't necessarily the exact same length as the video. With just one audio and video clip per timeline that doesn't matter. But by stringing AC3 files together you could end up with a sync problem that gets progressively worse.
If at all possible I would suggest to use PCM (.wav) files instead and let Encore handle the conversion to Dolby Digital.

Burned DVD Errors

Is it too much trouble, for a DVD to burn in DVD studio pro, and...I don't know....acctually play back on other DVD players without problems?
Sigh.....here we go again.
Ok, so I have a 23 minute movie, that I'm exporting from Final Cut Pro.....into DVD Studio Pro. I've allready built up the menus, and burned several DVD's of this project, which appear to load up fine on an external DVD player....at first glance anyway.
Its only after you play it, that you realize that its dropping frames, glitching out, and eventually freezing completly around the 18 minute mark.
So, I've been doing the old "bit rate shuffle", scaling back to around a 3.0 bit rate and a 5.0 max bit rate in DVD Studio Pro.....and that helped a little, but its still locking up.
So, instead of exporting a Quicktime movie directly out of Final Cut Pro, I tried going through Compressor (using the basic "DVD: 90 minutes, best quality" settings, only I dropped the average and maximum bit rate on the video to 4.0 & 7.0, respectivly).
I did that, and burned a new copy.....and now it won't even get out of the main menu without locking up.
So now, I'm completly stuck. I have no idea what I'm doing wrong.
Can please, someone....anyone.....tell me what I'm doing wrong?
(Is there a proper setting, of bit rate, max bit rate, etc.....that will result in a burned DVD that should be compatible with most DVD players, without sacrificing too much quality?)
Just in case its helps, I'm using:
* Verbatim DVD-R discs.
* Final Cut Pro 5.0.1
* DVD Studio Pro 4
Thanks in advance, guys.
- Fred

Hi Panic:
Wellcome to the forums !
Most probably your problem is related with your disc audio format. Uncompressed audio (AIFF/PCM) is one of the most common causes of playback issues of DVD-R in standalone players. The bitrate of the uncompressed audio it's too high for some players.
For a 23 min movie you can use the standard DVD: high quality 90 min settings aplying the MPEG2 and the Dolby 2 (AC3) presets. Encoding your movie inside DVDSP gives AIFF audio, and when you used Compressor you didn't mention encoding your audio to AC3.
You can duplicate the standard AC3 preset and change these settings to keep your original audio level:
- Dialig Normalization: -31
- Compression preset: None
If you want to try a second video encoding option, you can duplicate the standatd MPEG2 90 min preset and change it to CBR encoding:
A last point, you can control your burn speed building first your project and creating a Disc Image in your drive and burning the discs outside DVDSP. Read this tutorial: Creating Disc Image in DVDSP.
Hope that helps !
Alberto

AC3 5.1 encoding mystery

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