RealAnime 4 - General Questions and Troubleshooting Thread

[anne_so78]

Hi, Firstly I'd like to comment about how wonderful your program RealAnime is. It was just what I was looking for i.e > a program that allows batch transcoding of similar avi's into smaller files using efficient state-of-the-art codecs, while remaining user-friendly. I am currently trying out Light Edition Beta 2. I did come across a few surprises, namely:

1) Enabling parametric stereo seemed to introduce an audio delay of about +100 ms.
2) "Downmix to mono" doesn't seem to be working as it still outputs stereo files.
3) Audio streams are much louder than those produced using RealAnime version 3.2

Are the above peculiarities bugs, or am I not doing something right ? I think I did see warning messages in the status bar during the end of some encodes, but it disappeared quickly so it was hard to tell what it was, whats up with that ? Also, I'd like to know what the "low-bitrate matrix" and "filtering levels" (-1, 0, 1, 2 etc) do, and what are the differences between different resize methods (most importantly: what's the difference between lanczos3 and lanczos4). Perhaps there is an online documentation to answer queries as such ...? Thanks

[Sirber]

Hi, Firstly I'd like to comment about how wonderful your program RealAnime is. It was just what I was looking for i.e > a program that allows batch transcoding of similar avi's into smaller files using efficient state-of-the-art codecs, while remaining user-friendly.Thanks!
1) Enabling parametric stereo seemed to introduce an audio delay of about +100 ms.Do you use ffdshow audio with libfaad2?
2) "Downmix to mono" doesn't seem to be working as it still outputs stereo files.It has been removed in beta 2 IIRC. No more mono :)
3) Audio streams are much louder than those produced using RealAnime version 3.2RealAnime LE normalize the audio track to it's maximum volume.
Also, I'd like to know what the "low-bitrate matrix"Supposed to give less blocking when you use insanely low bitrate, not usefull at 368kbps.
"filtering levels" (-1, 0, 1, 2 etc) do0 = default. The mroe you go high (+1, +2), the more the inloop filter will be strong (bluring) and it will be more sensitive to blocks, while if you go lower (-1, -2) it will blur less and be less sensitive to blocking. In my tests +1 gives good results on anime content.
what are the differences between different resize methods (most importantly: what's the difference between lanczos3 and lanczos4).http://www.avisynth.org/Resize

Ask if you have more questions :D

[Sirber]

Some FPS bench on my 3000+, anime content, 640x480:

x264 @ fast first pass: ~30FPS
x264 @ low: ~16FPS
x264 @ medium: ~8FPS
x264 @ high: ~3FPS

[anne_so78]

I'm using ffdshow-20051129.exe with libfaad2 aac decoding, and Media Player Classic 6.4.8.7 . I've uploaded three samples (14 second clips of the TV show "Friends", download it from here - Samples.zip (ftp://site9887.0catch.com:password@0catch.com/Samples.zip)) to demonstrate the audio delay problem:

Source.avi - The source video file. This file has no serious problems.

Parametric Stereo Disabled.mkv - [ High quality 3-pass encoding, no filtering whatsoever, 32 kbps audio with parametric stereo turned OFF] On this sample, audio delay is present but hardly noticeable.

Parametric Stereo Enabled.mkv - [ High quality 3-pass encoding, no filtering whatsoever, 32 kbps audio with parametric stereo turned ON] On this sample, audio delay is noticeable. I'm guessing around +100 ms. Also, towards the end of the file there is frame-skipping (around 12 seconds into the file) as Lisa Kudrow asks "What?". This frame-skipping is not present in either of the other two samples.

I've noticed RealAnime decodes video using directshow decoders. I was surprised to see the DivX logo watermark (along with the DivX directshow decoder's noise/film-effect) encoded into my x264 video files (this was an unusual scenario when ffdshow was temporarily uninstalled). I guess one has to be careful... As ffdshow does all my mpeg4 decoding now, I'm curious about the effect its post-processing has on RealAnime encoded files. There is no doubt that post-processing (ffdshow's de-ring, accurate de-block) produces visually-pleasing (and thus better) results for viewing on a computer monitor. However, is it better to turn post-processing ON or OFF while decoding (and then encoding) files in RealAnime?

How do I add a video filtering parameter that is the equivalent of "Bicubic Resize A=0.60" (as in VirtualDub)? On a sidenote, I've noticed RealAnime's BicubicResize produces better results than Lanczos (and possibly every other resize method) when downsizing to small resolutions (320x240) ...

[Sirber]

RealAnime LE beta 2 doesn't let you choose PS or not. Please update to latest version.

RealAnime uses dshow decoders (recommended ffdshow), and it's recommended that you turn off all filters prior encoding. Best not use DIVX decoder.

About the delays, I will need your clips.

[anne_so78]

RealAnime LE beta 2 doesn't let you choose PS or not. Please update to latest version.
Well, this sure looks like Beta 2 -
http://img210.imageshack.us/img210/4856/realanimeleb28lg.png
I downloaded this version from http://www.detritus.qc.ca/. Perhaps it should be mentioned that I do not use any matroska splitters when viewing *.mkv files in media player classic (do I need to?) ...

[Sirber]

Do you use Haali splitter?

[Kayser]

Concerning the audio lag anne_so78 mentioned, it is the same issue I explained a while back. I didn't know that turning off PS helped though. I just tried it and it's true, you can still notice it's out of synch but not as much as with PS enabled.
I have ffdshow 12.08.05 (but it's the same with the RALE bundled version or whichever version I use) and decode aac with faad2 as well and use lastest haali splitter. I thought it was only me ^^;;
Oh, I tried the samples anne_so78 uploaded and they played back perfectly fine making in my pc it rather confusing :P. Is there anything else that must be configured in ffdshow or whatever to play all the encodes fine?

[Sirber]

maybe I should tell MKV the AAC uses PS, currently I don't. If you encode your source via BeLight (located in Tools), do you have a lag too?

[Kayser]

I encoded a file (with perfectly fine original audio but laggy encoded one) with BeLight using PS and then merged it with mkvmerge gui telling mkvmerge the source has AAC+ and the lag is still there.

[Sirber]

BeLight made "the encode laggy" too?

Final version of LE will use Nero HE-AAC v2 codecs instead of winamp. Better quality at 32kbps. Maybe it will "make the problem go away" :D

[Kayser]

yes Sirber

Edit: Just read the Nero AAC notice, I hope it runs smothly again... I would have to check but I think I had no problem (with audio :P) running alpha 1 with Nero encoder.

[Sharktooth]

The Winamp 5.12 encoder has been improved too.

[Sirber]

then, which one Shark? :)

[Sharktooth]

Dunno, havent done any serious listening test yet

[Sirber]

Let's get serious then :sly: ;)

[devast]

I'm here again :)
So, the version i built from svn repository there's vorbis audio too. I just wanna ask 64kbit vorbis is good enough ? I've never bothered with audio stuff, i really don't know the quality between this new codecs... i used to have 128kbit or 192kbit mp3, that's good enough, but i'm totally lost in this new AAC, vorbis etc thingies...

[Sirber]

Hi devast

A small and controversial comparison would be...

MP3 128kbps = Vorbis 64kbps = 48kbps AAC+ v2 PS

I personally use 32kbps AAC+ v2 with PS which produce a good audio quality.

@Kayser

What is the delay in ms? I might hardcode a tweak.

[Kostarum Rex Persia]

Sirber, what's your plan for Beta 3, any new option or fix?

[Kayser]

Mmmmm... hard to tell exactly... It varies from source to source but for me it's usually between 100ms and 1000ms, sometimes it's not even noticeable.

[Sirber]

damn... variable... :(

What should I do? :confused:

[Sirber]

Sirber, what's your plan for Beta 3, any new option or fix?
No beta 3. Move along. ;)

Next release is bugfix over beta 2 and more codecs, named "RealAnime PowerPack", XviD, RV10, Vorbis, MP3. We are also working on MP4 output.

[Kayser]

I have Nero 7, does it contain the AAC v2 codec you were talking about earlier? If so I can try encoding with that and muxing again to see if the lag is still there or if it was winamp encoder

[Sirber]

Yes it does. Those delay issues are evil :devil:

[Kayser]

Then I'll try it tomorrow and tell you the results Sirber.

[Sirber]

Nice... :D

I might get an iPod Video 30GB in 3 weeks. Guess what it means... :D

[Sirber]

I updated Winamp PS-AAC with the latest in 5.12

[anne_so78]

Let's get serious then :sly: ;)

So, I finally decided to compare the two new MPEG4 Audio codecs from Ahead and Coding Technologies side by side. It was easy to pick a winner as it wasn’t a close fought battle at all. I’ve uploaded the samples I used, so anyone can come to the same conclusions as I have. To decode the files you may use Winamp 5.12 (and maybe an AC3 ACM codec also to decode the source file), or the combination of Media Player Classic + a recent version of ffdshow. If you’re using the latter, you might want to deselect the “AAC transform filter” in Media Player Classic (Options>Filters>Transform Filters>AAC) to ensure ffdshow decodes the audio.

The Contestants: The HE-AAC v2 encoder included with Ahead’s Nero v7.0.1.4, versus Coding Technologies’ AACPlus v2 encoder included in Winamp v5.12.
The Source: A 2-minute audio clip (from the TV show “Friends”) ripped from a DVD source(192kbit stereo, 48khz, AC3 encoded). This clip contains dialogue from the cast, ambience, and laughter from an audience. Also, towards the end the clip contains a theme-song.

Impressions: For this test I decided to encode only at a bitrate of 32kbit, which happens to be the default bitrate in RealAnime. The source having been a wave file, wasn’t an easy encode in Winamp as I had to first burn the track as a standard audio-cd and then rip it within Winamp into AACPlus v2 (the transcoder plugin had some compatibility issues with the latest version of Winamp and/or the input file). This of course meant the source was converted to 44.1 khz. Encoding in Nero was a breeze thanks to the utility within Nero Burning ROM named “Encode Files”. As I listened to the two tracks produced by the two encoders, it became immediately clear how inferior the track produced by Nero’s encoder was in comparison to the file encoded by Winamp. The Nero encoded file sounded very flawed in the one aspect of audio tracks where listeners pay most attention to: THE DIALOGUE. The voices were unsteady and seemed to warble. The cast of “Friends” seemed to be talking under water. Realizing that Nero hadn’t gotten the same source (namely, a 44.1khz file) as the Winamp encoder and wanting to level the playing-field, I ripped the CD I had used in Winamp earlier, to a wave file on my computer. Having used this as the new source in Nero, I encoded the file once again into the same parameters and gave it a listen. The warbling artifact in the dialogue had definitely lessened but was still there. This artifact was not present in the Winamp encoded file. In fact, the Winamp encoded file had no serious flaws (for a 32kbit MPEG4 audio file) and was very pleasant to listen to. I must mention though, this fault had disappeared once the “TAPE::LOWEST” variable bitrate encoding method was enabled, but I decided not to include it in this test because the files had reached an average bitrate of 42kbit and thus was way over-sized. The Nero encoder wasn’t without its advantages however; Music (pop, rock) encoded in Nero at this bitrate (32kbit) from audio-cds, contained more dynamics, especially a more deeper and fuller bass. But even here, the warbling effect was apparent on vocals. Perhaps the Nero encoder excels at higher bitrates, but this I haven’t tested. If I were to quantify the quality perceived from the two encoders during this test (1 being lowest, 5 being highest) it would be:

Coding Technologies’ AACPlus v2 encoder found in Winamp v5.12 (from a 44.1khz audio-cd source): 5/5
Ahead’s HE-AAC v2 encoder found in Nero v7.0.1.4 (from a 44.1khz audio-cd source): 4/5
Ahead’s HE-AAC v2 encoder found in Nero v7.0.1.4 (from a 48khz wave-file source): 2/5 (yes, it was that bad)

Download source wave-file > Source.wav (ftp://site9887.0catch.com:password@0catch.com/MPEG4 HE-AAC v2 Codec Tests/Source.wav)
Download encoded samples > Samples.zip (ftp://site9887.0catch.com:password@0catch.com/MPEG4 HE-AAC v2 Codec Tests/MPEG4 HE-AAC v2 Encoded Samples.zip)

To summarize: For cartoon/anime/tv-show/movie encoding @32kbit stereo, Coding Technologies’ AACPlus v2 encoder is the clear choice. Give the samples above a listen, and feel free to disagree…

[Sharktooth]

sample link is broken

[anne_so78]

retry

[Sharktooth]

ok, it works now:)

[Sirber]

Cool! Thanks for your efforts! :D

I'm gonna use Winamp 5.12 encoder then :)

[Eretria-chan]

It's cool that you find new ways to improve both video quality & audio quality in your program, Sirber, some thanks to the discoveries to others. It's almost as if you don't want to use it, because new discoveries come all the time and makes things even better. I've lots and lots of anime to encode (dare I say 100 gb?), so it's VERY interesting to see how much it will shrink after the compression (they're all original files--not re-encoded). The better the program, the lower the filesize :D

[Sirber]

Thanks! :D

Next thing to see is if 2 pass high is better than 3 pass medium, and if I can reduice the vidweo bitrate by 50kbps whitout hurting the quality too much...

[slavickas]

...
Download encoded samples > Samples.zip (ftp://site9887.0catch.com:password@0catch.com/MPEG4 HE-AAC v2 Codec Tests/MPEG4 HE-AAC v2 Encoded Samples.zip)

To summarize: For cartoon/anime/tv-show/movie encoding @32kbit stereo, Coding Technologies’ AACPlus encoder is the clear choice. Give the samples above a listen, and feel free to disagree…
Isn't Nero's sample regular stereo while Winamp's PS?

[Sirber]

nero 7 heaac is supposed to automaticly use PS.

[anne_so78]

Isn't Nero's sample regular stereo while Winamp's PS?
After listening to Nero's sample one might doubt that the encoder uses parametric stereo, but it actually does. Here's proof for the skeptic:
http://img231.imageshack.us/img231/392/foobarsnap6bw.png


Sirber, regarding the frame-skipping in parametric stereo encoded video files I had mentioned earlier > The problem seems to be a MediaPlayerClassic issue. After installing the Matroska Pack (along with the bundled Haali media splitter) and disabling MediaPlayerClassic's own Matroska filter (Options>Filters>Source Filters>Matroska), the file played smoothly :) ... Now, if only those darned delays would go away ...

[slavickas]

hmm weird, maybe foobar's version is tool old that I have. I use winamp usually, but if i'm not mistaken Nero always writes ADTS AAC files as stereo even whe they when they are PS...

[Sirber]

Things left to do:
1) Audio lag from MP4 (~500ms to 1000ms)
2) Tell mkv that AAC is SBR
3) Possibility to skip audio encoding and keep it the way it is

[anne_so78]

RealAnime LE b2 doesn't support 48khz sources right? Coz everytime I try to encode video's containing 48khz audio I get "Unsupported Source" errors... I'm guessing this has something to do with CT's encoder limitations. Considering that a lot of videos contain audio with that samplerate, perhaps it would be desirable to have a resampler (48khz-44.1khz) within RealAnime to accomodate those kind of files ...

[Sirber]

"Unsupported Source" comes if your AVI doesn't contain MP3 or WAV, or if the file extention is unknown. Please post the stuff you get when you ask the "Info" popup.

[anne_so78]

yeah, you're right, RealAnime LE b2 DOES support 48khz wavs and mp3s. The sample that gave the "Unsupported Source" error had AC3 encoded audio. So I converted it to plain wave and then tried encoding in RealAnime, and then it worked fine. I did notice however, that the aac+v2 encoded stream was 44.1khz and not 48khz. Just curious > which program automagically resampled to 44.1, was it RealAnime or the CT encoder?, and why was this done? Hey, I just tried encoding some videos in MediaCoder with AAC+ v2 and guess what? That lag is omnipresent :eek: ...

[Sirber]

I ask besweet to sample at 44.1khz for AAC+ v2, in case the source is bellow it.

[JohnV]

anne_so78 : Thanks for the test. These are always welcome.
We are atm. in process of changing our audio encoder codebase totally to new one, completely new, written from scratch. It provides superior quality to the old one.
So few points
- Nero v7.0.1.4 still has the old codebase (although it has been updated also).
- I'm not just hyping. The new encoder codebase is currently being tested at HA in the on going 128kbps AAC-LC public listening test (resullt come after christmas). But it's not yet available in Nero builds. (You can test 128kbps mode though if you download the .dll from HA)
- The new codebase is a big improvement in our internal tests, and all modes based on the new codebase should be out hopefully in about 3 months.
- You will see/hear when this happens because we will make an announcement. Then it's good time to test again. :)

[anne_so78]

thats good to know :D , can't wait ...

[Sharktooth]

good news:)

[Sirber]

I'm adding WMA8 (5-64kbps) for pocket pc encodes.

[Sirber]

can't merge WMA in MKV nor OGM.
can't merge WMA with RIFF headers (WAV) into MKV
WMA with RIFF headers merge in OGM but doesn't playback.

I think I'm gonna revert my changes...

[Sirber]

New:
* PowerPack! (codec extension)
* XviD (powerpack)
* RV10 (powerpack)
* Vorbis (powerpack)
* AAC (powerpack)
* MP3 (powerpack)
* MP4 output (powerpack)
* x264 Constant Quality mode
* GUI now on top
* Number of bframes in advanced tab (not saved yet in XML)
* RV10 dropdupe (powerpack)

Updated / Fixed:
* Strange audio delay in MP4 input (and MKV)
* MKV and MP4 input
* x264 rev 388
* winamp AAC+ v2 5.12
* MkvToolNix 1.6.5
* other minor bugs...

ToDo:
* Skip audio encoding
* Subtitles from MP4

Help us with further development of RealAnime!
http://www.detritus.qc.ca/images/donate.png (https://www.paypal.com/xclick/business=sirber@gmail.com&no_shipping=1&item_name=RealAnime%20LE)

http://www.detritus.qc.ca

[anne_so78]

cool, lots of great features; have you fixed the delay problem?