Dolby Digital/AC3 encoding?

I just bought a K8N Neo4, and hooked it up to my Panasonic audio receiver - just the one cable, S/PDIF optical. I was hoping to get 5.1 sound out of this, but all I get is two channel stereo. 
Is there any way to encode a signal that my receiver can use to produce full surround sound (it supports Dolby Digital and DTS signals)?

Nice.
I never even heard of it.  I always thought the only way was to go 6ch analog and have your external amp support multi channel in if the source wasn't DD/DTS already in which case it's not DD anyway.  Otherwise you only get 2ch PCM on the SPDIF. Then again I only have owned Creative sound cards....geez even then Creative made you buy the breakout box to enable AC3 decoding unless you made a board mod...  Of course you still need a multi channel source.  What kind of software do they come with?  Can you play a 6ch .wav for example and have a DD/DTS stream over SPDIF/Optical? or is only really for games, so for simplicity you can have one cable/fiber and not go analog till it 's on the way to the speakers?  Still though except for a few less cables, and when the D/A conversion takes place(and therefore the quality of which component is doing it, sound card or amp) I don't think it's that big of a deal if you have multichannel input amp.  DD is still not lossless compression so the big advantage would seem to be for those without multichannel inputs.  Otherwise it would be an interesting comparrision for the audiophile which I'm not.  You have a pure source and you compress to DD and lose quality and then decode and output  vs. Pure source going analog, losing signal quality along the way and outputting.  someone on the internet has surely done it.

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    Message Edited by KokChoy-CL on 07-04-2008 :0 AM

    Digital Connections, SPDIF and Dolby Digital (german revision)
    Options
    03-08-2007 01:52 AM
    Digital Connections, SPDIF and Dolby Digital Info
    Übersetzung + Erweiterungen + Revision 03-08-07
    Der Digitale Anschluß
    Ich habe beobachtet, daß viele Leute Fragen zu den Sound-Anschlüssen haben, daher trage ich hier einige Infos zusammen und erkläre die Unterschiede in der Verwendung der digitalen und analogen Anschlüsse.
    Digitale Anschlüsse – S-PDIF & Toslink
    Am PC gibt es zwei häufig verwendete digitale Anschlußformate optisch (Toslink) und koaxial (S-PDIF).
    Im Prinzip erzielen beide Anschlüsse genau das gleiche Audio-Ergebnis.
    Der primäre Unterschied ist das Medium ( das Material ) durch das das Signal übertragen wird. ( Glasfaser oder Kupfer )
    !! Update !!
    Ein wichtiger Aspekt in der Auswahl von digital-optischen Anschlüssen erklärt sich durch das Vermeiden einer sogenannten „Brummschleife“, d.h. eine elektrisch leitende Verbindung zweier Geräte muß vermieden werden.
    ( mit Lichtwellenleiter - LWL- z.B. mit dem Anschluß-System Toslink )
    In der Elektrotechnik ( Niedervolttechnik ) wird oft der „Negative“ Pol der Spannungsversorgung über das elektrisch leitende Gehäuse ( die sogenannte Masse )geführt. Das spart einerseits einen weiteren Kabelweg ( also Kupfer ) ein und ergibt eine mehr oder weniger wirksame (Ab-) Schirmung. Diese Schirmung soll die elektronische Schaltung vor elektromagnetischen, hochfrequenten und niederfrequenten Störeinflüssen je nach Ausführung der Schirmung schützen.
    Üblicherweise besitzt jedes elektronische Gerät eine eigene Spannungsversorgung. Netzteile in elektronischen Geräten unterscheiden sich in der Art der Spannungsumformung, in der Höhe der Spannung, in der zur Verfügung gestellten Leistung, durch verschiedene Bauarten und unterschiedliche Bauteil-Tolleranzen entstehen somit verschiedenste Masse-Potentiale.
    Werden solche Geräte durch elektrisch leitende Kabel verbunden, kommt es zu einem Ausgleich der verschiedenen Masse-Potentiale. Ein negatives Ergebnis der Verbindung von TV-Geräten oder PC-Komponenten mit Audio-Komponenten ist oft eine hörbare Beeinträchtigung von Audio-Signalen - ein tieffrequentes Brummen.( auch Brummschleife oder Masseschleife genannt )
    Hierfür verantwortlich zu machen ist eine elektrisch leitende Verbindung. In der HiFi-Technik ist das in der Regel ein Koaxialkabel und dessen Schirmung.
    Vermeidung u. Reduzierung dieser Störgeräusche Ansätze
    !! Der einfache Weg !!
    Einsatz einer optischen Signal- / Datenübertragung mit Lichtwellenleiter
    - Glasfaser als Trägermaterial
    ( Einsatz einer Datenübertragung ohne elektrisch leitende Verbindungen )
    aufwendige Wege
    Einsatz einer Galvanischen Trennung - mit einem Ground-Isolator
    (Massetrennung )
    + die elektrische Verbindung der Schirmung wird umgangen
    - aber Leistungs- & und Klangverlust(geringere Dynamik)
    - Verwendung von ( hochwertigen ) symmetrischen Verbindungen
    ( „PLUS“ „MINUS“ „Masse“)
    + Trennung der Schirmung an einer „Seite“ des Kabels möglich
    [bei Koaxialkabel unsinnig weil eine Ader zerstört wird und
    anschließend dieses Kabel wertlos ist ]
    - sehr große Anschlüsse
    - Profibereich
    - Einsatz einer zentralen Spannungsversorgung für alle miteinander
    verbundenen Geräte
    Nachtrag
    Bitte nicht verwechseln mit Netzbrummen - die Einstreuung der Netzfrequenz 50 Hz ( EU ) oder 60 Hz ( US ) in den Audio-Signalweg das ist üblicherweise auf eine nicht ausreichende Entstörung oder(Ab-)Schirmung zurückzuführen ist.
    - End - !! Update !!
    Einige Vorbetrachtungen
    Die traditionelle Ton-Übertragung ist ein unkomprimiertes Signal, es wird in Stereo übertragen.
    Wenn du CMSS ( Creative Multi Speaker Surround simulierten Surround Sound ) aus deiner Stereo-Musik erzeugen willst oder der Sprach-Test abgerufen wird ( nur über den Center ) ist der digitale Anschluß ( optisch & koaxial ) unbrauchbar, denn es würden nur die Lautsprecher Front-Rechts und Front-Links angesprochen !! Es wird ein anderer Anschluß benötigt, der die Kanäle einzeln ansteuert.
    Für das Format 5.1 ist das die Verbindung über die drei 3,5 mm Klinken-Stecker ( 3 mal 2 Kanäle ) mit der folgenden
    Kanalzuweisung 5.1
    Front R / Front L + Rear R + Rear L + Subwoofer / Center
    Mehrkanalton in Filmen - DolbyDigital (AC3 / DD) oder DTS
    Filme auf DVD verwenden häufig ein kodiertes Mehrkanal-Audiosignal, in der Form DolbyDigital ( AC3 / DD ) oder DTS. Bei dem gebräuchlichsten Format 5.1 werden 6 Signale ( 3 mal 2 Kanäle ) in einem digitalen Datenstrom zusammengefaßt und komprimiert = das Kodieren.
    Nach der Übertragung des Signals muß aus diesem Datenstrom ein Decoder die entsprechenden Töne trennen und jedem Kanal zuordnen = das Dekodieren.
    Drei Varianten der Decodierung sind möglich.
    (1) - AC3 / DTS Signal wird zu einem externen Decoder geführt
    Genau hier wird der optische / koaxiale Anschluß genutzt. Um diese Variante nutzen zu können, mußt du ein digitales Lautsprecher-Set oder einen Digital-Decoder ( z.B: Digital-Receiver ) besitzen, der die Dekodierung des gewünschten Formats ( DD, AC3, DTS) übernimmt. Für jedes Digital-Format benötigt man einen entsprechenden Decoder – das Handbuch weiß hier oft mehr – welches Format von welchem Gerät dekodiert werden kann !!
    Wähle in deiner DVD-Software den zur digitalen Datenübertragung gewünschten Anschluß "S-PDIF"( Koaxial ) oder "Optisch" als Audio-Ausgang.
    In den Soundkarten-Einstellungen muß die Option "S-PDIF-Durchschleifen" ( "SPDIF Pass-through" ) eingestellt werden.
    Dadurch wird das Audio-Signal von der DVD in digitaler ( komprimierter ) Form direkt an die Digital-Ausgänge der Soundkarte weitergeleitet. Anschließend kann ein angeschlossener externer Decoder z.B. im digitalen Lautsprecher-System oder im Digital-Receiver die Decodierung übernehmen und es entsteht ein „Räumliches Klangbild“.
    Bitte merken:
    Daß ist der einzige Weg 5.1 Signale ( digital ) über ein koaxiales bzw. optisches Kabel zu übertragen.
    (2) - AC3 / DTS mit der Soundkarte decodieren
    Diese Variante ist zu verwenden, wenn du keinen externen Decoder / Receiver verwenden willst.
    In der DVD-Software wird "S-PDIF" als Audio-Ausgang gewählt und es muß zwingend in den Optionen der Soundkarte "S-PDIF-Durchschleifen" ( "SPDIF Pass-through" ) deaktiviert werden.
    Die DVD-Software wird jetzt das ( digitale ) komprimierte Audio-Signal an deine Soundkarte senden und dann werden dort die digitalen Daten decodiert. Die Soundkarte gibt dann die 6 Audio-Kanäle ( bei 5.1 ) über ihre drei analogen Anschlüsse aus ( 3 mal 3,5 mm Klinke ) mit folgender
    Kanalzuweisung 5.1
    Front R / Front L + Rear R + Rear L + Subwoofer / Center
    Es wird in diesem Beispiel also ein 5.1-Lautsprecher-System benötigt, mit der analogen Anschluß-Option von drei Klinken-Steckern.
    ( also 3 x 2 Kanäle = 3 mal 3,5 mm Klinke )
    - die Einspeisung eines digitalen Signals über DigitalDIN, ist mit dem optionalen digitalen E/A-Modulbei möglich ***
    - wird hier ein optisches / koaxiales Kabel verwendest, hörst du nur schwache Töne der Front-Lautsprecher R + L
    *** Bei Karten der SB073X-Serie verfügt diese Buchse nicht über die Digital-In-Funktion !! zur Unterstützung des digitalen E/A-Moduls )
    [ mehr unten ]
    (3) - AC3 / DTS mit einer Software decodieren
    Wenn deine Soundkarte kein DolbyDigital unterstützt
    ( wie die SB Live! 24-bit ) und auch kein digitales Lautsprecher-System vorhanden ist, kannst du Raumklang
    (Surrsound-Sound ) aus DV-Filmen auch über eine AC3 / DTS fähige Software erzeugen.
    Gratis-DVD-Software-Versionen unterstützen diese Funktion häufig nicht, du benötigst dann eine Software-Voll-Version.
    Wähle die gewünschte digitale Audio-Tonspur „DolbyDigital“ oder „DTS“, im Audio-Menü einer DVD.
    Dann stelle sicher, das in den Soundkarten-Optionen "S-PDIF-Durchschleifen"
    ( "SPDIF Pass-through" ) ausgeschaltet ist.
    Wie unter (2) mußt du deine Lautsprecher mit den drei analogen Kabeln anschließen und nicht optisch oder koaxial !! ( 3 mal 3,5 Klinke )
    Kanalzuweisung 5.1
    Front R / Front L + Rear R + Rear L + Subwoofer / Center
    Generell sind bei zuvor genannten Varianten (1) – (3) auch andere mehrkanalige
    Wiedergabeformate möglich.
    In Abhängigkeit von der Ausstattung der Quelle also von der DVD. der CD. der TV-Karte, dem Game usw. sind andere Formate z.B. 2.0 / 2.1 / 4.0 / 4.1 / 5.1 / 7.1-Kanal Wiedergabe möglich.
    PC Spiele
    PC Spiele nutzen normalerweise Technologien wie EAX, die ihren eigenen Surround-Sound, ohne Verwendung von DolbyDigital erzeugen. Selbst wenn dein Lautsprecher-System einen DolbyDigital-Decoder besitzt, ist dieser für Games unbrauchbar. Denn die EAX Verarbeitung findet direkt auf der Soundkarte statt und die Ausgabe der 6 Kanäle ( 5.1 ) erfolgt über die analogen Ausgänge
    ( 3 mal 3,5 mm Klinke ) ( oder über DigitalDIN, wenn vorhanden *** ).
    *** Bei Karten der SB073X-Serie verfügt diese Buchse nicht über die Digital-In-Funktion !! zur Unterstützung des digitalen E/A-Moduls ) [ mehr unten ]
    Kanalzuweisung 5.1
    Front R / Front L + Rear R + Rear L + Subwoofer / Center
    Für Mehrkanalwiedergabe in Spielen benötigst du also Lautsprecher-Systeme mit analogen Anschlüssen oder einen Digital-Receiver mit analogem ( 5.1 ) Eingang.
    Wenn du ein optisches / koaxiales Kabel verwendest, dann kannst du nur die vorderen Kanäle ( links / rechts ) hören.
    !!! Notiz !!!
    X-Box Spiele unterstützen DolbyDigital, wenn du deine X-Box mit einem DolbyDigital-Receiver oder einem digitalen Lautsprecher-System über optische Kabel verbindest, kannst du den vollen Surround Sound nutzen.
    Die Sound Blaster Karten der SB073X-Serie bieten neben dem digitalen ( koaxial / extern ) Anschluß über den Anschluss Flexi-Jack auch noch einen weiteren Anschluß, direkt auf der Karte ( koaxial / intern )
    Flexi-Jack - DigitalDIN
    Wenn du kein digitales E/A-Modul ( Digital I/O-Modul ) benutzen kannst, ist es möglich mit einem Klinke-Chinch Adapter ( RCA-Adapter ), ein digitales Lautsprecher-System oder einen Digital-Receiver digital ( koaxial ) über den Flexi-Jack-Ausgang der Soundkarte zu verbinden. Alle MiniDin bzw. 3,5 mm Klinken-Stecker können in der Mono- oder auch in der Stereo-Ausführung verwendet werden, denn der mittlere / vordere Pin des 3,5 mm Klinken-Steckers überträgt das AC3 / DTS Signal. Dazu benötigt man eine der folgenden Verbindungen
    3,5 mm Klinke ( Mini DIN ) auf Chinch - Adapter ( RCA-Adapter )
    für den S-PDIF-Anschluß( koaxial ) an Lautsprecher-Systeme mit digitalem Eingang bzw. Digital-Receiver kann man folgende Varianten verwenden
    - Klinke-Mono - Chinch-Stecker ( Stecker –Stecker ) ( Favorit !! )
    - Klinke-Mono auf Chinch-Mono ( Stecker - Kupplung ) + Chinch-Verbindungkabel
    ( Stecker - Stecker )
    - Klinke-Stereo auf Chinch-Mono ( Stecker - Kupplung ) + Chinch-Verbindungkabel
    ( Stecker - Stecker )
    Das digitale E/A-Modul ( Digital I/O – Modul )
    Die Verwendung von optionalem Zubehör Digitales E/A-Modul ( Digital I/O-Modul ) bietet die Möglichkeit der direkten Verwendung eines S-PDIF Verbindungskabels und stellt zusätzlich zum koaxialen Anschluß einen optischen Anschluß ( Toslink ) bereit. Die optische Verbindung kann wie oben beschrieben zur Vermeidung einer Brummschleife hilfreich sein.
    Des weiteren wird ( abhängig von der Ausstattung der Soundkarte *** ) ein digitaler Eingang über das E/A-Modul bereitgestellt, der Eingang ist ebenso wie der Ausgang in optischer ( Toslink ) und koaxialer ( S-PDIF ) Bauform ausgeführt.
    Achtung !!
    ( *** Bei Karten der SB073X-Serie verfügt diese Buchse nicht über die Digital-In-Funktion !! zur Unterstützung des digitalen E/A-Moduls )
    interner S-PDIF E/A-Anschluss ( für Geräte der SB073X-Serie )
    zum Anschluß einer Decoderkarte oder externer Decoder bzw. Lautsprechersysteme mit digitalem Eingang ist ein entsprechender Adapter erforderlich
    Bei einigen Modellen ist eine Dekoderkarte im Lieferumfang enthalten bzw. in einigen Regionen als Zubehör erhältlich
    Front Panel-Sockel (für Geräte der SB073X-Serie)
    2 x 5-poliger Front Panel-Sockel nur für Verbindungen mit einem Intel HD Front Panel Audio-Standardsockel (nur Geräte der SB073X-Serie) Intel Front Panel Audio-Kabels (separat erhältlich)
    Anschlusskompatibilität
    Ausschließlich Intel HD Front Panel Audio-Standard
    Nicht kompatibel mit AC97 oder Intel-kompatiblem HD Front Panel Audio
    Einschränkungen
    Wenn neben den hinteren Mikrofonen auch Frontmikrofone angeschlossen sind, werden die hinteren Mikrofone automatisch deaktiviert.
    Sound Blaster X-Fi E/A-Konsole
    Auf der Rückseite Ihrer X-Fi-E/A-Konsole* befindet sich eine sogenannte DIN-Buchse. Um die Funktionen für eine integrierte Lautsprechersteuerung zu nutzen, können Sie an diese Buchse ausgewählte analoge Creative-Lautsprechersysteme anschließen, die über eine Kabelfernbedienung oder eine Audiokonsole verfügen.
    So können Sie beispielsweise über den Master-Lautstärkeregler Ihrer X-Fi-E/A-Konsole die Lautstärke der Lautsprecher regeln oder die Lautsprecher stummschalten.
    -eof-

  • SB1095 can't see Dolby Digital Live options or 3d options on the Entertainment console

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    Addy<img border="0" src="https://picasaweb.google.com/lh/photo/RVXvkgnkmMlkzi34mgSQ7GVxXvr-BdpqDUiBSmAzjag?feat=directlink">
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  • AIFC to Dolby Digital...

    My movie Soundtrack right now is an AIFC file. In FCP6 I used File --> Export --> Audio to AIFF. I think it sounds good. It comes up on my DVD player as PCM 48 2 channel file for the audio.
    Is it necessary to go into Compressor and change this format (AIFC) into Dolby Digital (AC3)? Is there a big difference?
    I am curious about this one. I sent my movie screener to a few festivals in this PCM 48 2 channel format for the movie. For the master I should use Dolby?

    AIFC is a variation on the AIFF audio format that +allows for+, but does not neccessarily have to be compressed.
    AIFF and AIFC files are quite large when compared to highly compressed Dolby ac3 audio.
    The question you should be asking is whether your video quality is being compromised because it has to have a lower bit rate to accomodate your large audio file and is their an audible difference in using ac3. But only you can answer that.

  • Encoding Dolby Digital 5.1 (AC3) sound in MP4 files using AC3 pass-through

    Looking for an application that will encode Dolby Digital 5.1 (AC3) sound in an .mp4 container using AC3 pass-through for playing videos with surround sound on ATV.
    Handbrake is out of the question because I run Tiger. The Tiger version cannot do AC3 pass-through.
    Visual Hub is no longer available.
    Any other options?

    You may need Apple's MPEG2 QT plugin as well for MPEGStreamclip.
    Also, consider if running windows in parallels/Boot Camp etc that there's a windows version, and possibly a linux version.
    VisualHub code was released open source - Google FilRedux and PunyVid (iSquint derivative). Not tried either though.

  • Audio to dolby digital surround AC3

    Regards,
    I have one test sequence in FCP,,, left channel is sound, and right is speech,, then exported separately to two mono channels,,, named SOUND and SPEECH,, I can hear each channel nice after that,,
    But, when I make AC3 file with compressor, I have problem.
    Can somebody tell me where I make mistake??
    With compressor I make New Job With Surround sound group,, then select for every channel one mono channel, from FCP, specifically; L - speech, C-sound, R -speech, Ls -sound, Rs-sound ,,,
    after select File format,, Dolby digital Professional,, Target sistem is DVD video,,,, Audio coding mode is 3/2,,, In Preprocessing, compression presset is NONE,, after submit this I have AC3 file, and when I make DVD and looking in my home dolby digital system I hear completely diferent disposition,, L - Sound, C-sound, R-speech (very quiet) Ls- sound, Rs-speech,
    I was trying and different setup, but I cannot acquire adequate setup,, before I was using AC3 encoder (FCP 4.5), and everything was perfect,
    Thanks,

    Sry, your comment is 5 month old, but i have to ask you some thing too:
    What headset do you use for the surround sound? I read EVERYWERE that headsets can just output games / music in stereo... for example the logitech g35...
    Hope you can help me

  • Dolby Digital Encoder Does Not Support Any Encoding Options

    Is it just me, or are there no encoding options for the Dolby Digital encoder other than the bitrate?
    How are you supposed to select the dialog normalization, dynamic range compression profile, etc. for your audio? Without these parameters set correctly, the audio will not be encoded correctly.

    @jbowden:
    > Adobe licenses the AC3 encoder that Encore uses from Dolby Laboratories. This is stated quite clearly in the splash screen, the box, and other materials distributed with Encore.
    I didn't say otherwise. But what they license is the code, not the implementation. Implementations are supposed to be submitted to Dolby for approval, and I'm surprised that Dolby approved the implementation given the oversights and encoding quality.
    > It's a consumer-level encoder, and if you look around this forum I think you'll find very few posts about audio quality, which would seem to indicate that Encore provides acceptable audio quality for most users. Cases with audible clipping can occur if the input signal is too high...keeping your audio signal around -6db should eliminate any clipping.
    I don't buy into this whole argument about "consumer-level" and "professional-level". What is the exact difference? Dolby's code is constant across all licensed encoders, so any differences in encoding are the result of an implementation difference, and that's Adobe's fault, not Dolby's.
    Audio signals are audio signals. Why do I have to keep my audio from peaking above -6dB with the Adobe encoder? I can take audio whose peaks go all the way up to 0dB and encode it with the Sonic Foundry encoder with no problem -- no clipping, no volume pumping, etc. Not so with Adobe Encore's implementation.
    @Neil Wilkes:
    > As far as Dialnorm goes, again I say it is pretty meaningless in a stereo file.
    Dolby's encoding guidelines documents seem to say otherwise. In my readings, it seemed pretty clear that the decoder must have the dialnorm parameter set to the LAeq level of the dialog in the audio regardless of channel configuration.
    > I suppose, although as long as the stereo mix is actually set up correctly there is no need to invoke DRC. The only time DRC would be needed on a stereo file is if the thing is too darn loud.
    I have to disagree with this. There is a LOT of material that can benefit from DRC, both soundtracks that contain a large dynamic range (explosions to whispers), and soundtracks that have small dynamic range (speech). DRC doesn't have anything to do with the overall volume level of the soundtrack, but the dynamic range of the soundtrack. If the soundtrack is too loud overall, that's not a DRC problem, that's a level problem.
    > In a properly balanced stereo file, there is no need at all for DRC or Dialnorm. There is no centre channel containing dialogue, just a phantom centre, so what are you going to reference it to, please? How can you attenuate the left/right channels with reference to the centre channel dialogue when there is no centre channel present?
    Just because the dialog isn't isolated in the center channel doesn't mean you can't use it to set the dialnorm parameter. Plus, the decoder doesn't attenuate the left/right channels with reference to the center channel, it attenuates the entire soundtrack on playback.
    Your quote from Dolby labs is very correct. Following that procedure ensures that the decoder, when it applies attenuation based on dialnorm and applies DRC will work as intended.
    > Additionally, it is not referenced to average RMS either. RMS is close to the correct way to measure, but not correct.
    That is true. However, I don't have any equipment to measure LAeq directly, so RMS of a dialog section of my audio will have to do. I do know this: When I applied to Dolby for logo usage on my DVDs, they requested samples of my DVDs for approval of my methods. My first submissions (where I didn't bother with dialnorm & DRC) were rejected, and Dolby sent me a letter telling me such. After I produced further DVDs using my method in my Doom9 post and resubmitted the DVDs to Dolby, they were approved. I can only conclude that my method, while not perfectly accurate, is good enough to qualify for Dolby's rigorous standards.
    The bottom line is that I feel I have enough evidence to conclude that the DD encoder implementation in Encore is substandard. If you or anyone else feels it meets your requirements, then use it. But I for one will not allow a single DVD to go out from my company with audio encoded with this encoder.

  • Live! 5.1 SB0100. Can decode Dolby Digital 5.1 (AC3) from S/PDIF-

    Hi,?i have a Sound Blaster Li've! 5. model: SB000. This can decode AC3 from S/PDIF IN and out to analog outputs?. My idea is connect Xbox 360 to S/PDIF IN and decode Dolby Digital to analog outs. Thanks.

    Nope. The original EMU0K chip always had resampling enabled on S/PDIF input so that if there was any clock drift at all bits could change and damage the AC3 encoding. The EMU0K2 (Audigy) chip was changed to somehow allow bypassing the core clock so the S/PDIF sample values could get into memory without alteration.
    -Dave
    ?[email protected]

  • Media Encoder PCM or Dolby Digital Settings change preset????

    I am about to Encode a 2 and a half hour project from PPRO 5.5 into Media Encoder.
    If I set my format to H.264 Blu-Ray and my Pre set to 1440 x 1080i 29.97 High Quality, My audio automatically sets to PCM Audio? Why?
    When I use the H.264 Blu-Ray with my Pre set to 1440 x 1080i 29.97 High Quality, and I Press the Dolby Digital as my audio instead of PCM, my Preset changes to Custom. (1440 x 1080i goes away)
    Why is this? Is that what is supposed to happen?
    I had read that Dolby takes up less space without losing that much quality.
    I'm trying to encode this file to about 15GB with keeping the best quality Audio and Video I can.
    PPro CS 5.5
    Media Encoder 5

    SAFEHARBOR11 wrote:
    Jim,
    In case you missed it, his program is 2hours 40 minutes, so in that case, space is at a premium and I would go for the Dolby.
    It is normal for Media Encoder to create PCM audio, that is OK, Encore will TRANSCODE it to Dolby later and shrink it down.
    Edwurd, please see you other posting for more info about encoding size
    Thanks
    Jeff
    Jeff is right.
    Let Media Encoder make the .264 and .wav files and have Encore automatically transcode the pcm to ac-3 (since it's licensed to do so and AME is not). Don't multiplex them into a mp4 or mov container; it'll make it blu-ray illegal (even though Encore will allow you to import it).

  • Dolby Digital encoding immediately forgets my bitrate settings.

    Dolby Digital encoding is ignoring my bitrate settings. I have a stereo AIFF I'm trying to encode at 448kbps, but the minute I leave the settings window, it reverts back to 192kbps. This is driving my crazy! Anyone else having this problem?

    Thanks for the quick response, but sad to say it was all my own fault.
    My dolby digital receiver is a recent purchase, and it's the only receiver I've ever seen where straight DD decoding is in DSP options.
    Suffice it to say, I wasn't getting 5. sound because I didn't know how to use my own receiver
    Everything sounds great here now.
    Still not sure why my config screens look so much different, though.

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