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Old 27th November 2011, 14:54   #7361  |  Link
nevcairiel
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The name of the format is WavPack, not WAVEPack.
I have no idea how good the decoder in avcodec is, and i honestly don't care because its not a format i ever used. The lossless mode seems to decode fine, though.
In any case, there is an official DirectShow decoder, so just use that.
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Old 27th November 2011, 14:56   #7362  |  Link
Kurtnoise
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Hi Nev,

I tested your last commits against the test suite from matroska.org.

All pass well except test4.mkv :

Quote:
4. Live stream recording

This file is using the EBML feature that allows Master elements to have no known size. It is used for live streams because they don't know ahead of time the size of the Segment (virtually infinite) and even sometimes the size of the Clusters (no caching on the server side). The first timecode of the file also doesn't start at 0 since it's supposed to be a capture from something continuous. The SegmentInfo also doesn't contain any Duration as it is not know.

The sample comes from the Big Buck Bunny open project. It contains Theora video (1280x720), Vorbis audio, uses only SimpleBlock (matroska DocType v2)

A similar file can be created with mkclean using the "--live" option
I don't know if you want to include this feature or not...
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Old 27th November 2011, 15:00   #7363  |  Link
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Quote:
Originally Posted by nevcairiel View Post
The name of the format is WavPack, not WAVEPack.
I have no idea how good the decoder in avcodec is, and i honestly don't care because its not a format i ever used. The lossless mode seems to decode fine, though.
In any case, there is an official DirectShow decoder, so just use that.
I want to use it. But only when wrapped in MKV, the decoder seems to output an media type similar to s16l although it is 32 bit IEEE float. So video is half as fast as it should be and the sound makes me dizzy when listening to it: White noise on the left channel and distorted music on the left.

How can I cut any MKV without much effort to provide you a sample file?
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Old 27th November 2011, 15:04   #7364  |  Link
nevcairiel
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Quote:
Originally Posted by Kurtnoise View Post
All pass well except test4.mkv
I dont think thats a feature that is used by anyone. Also, no MKV splitter i know of can read them, which even reduces the chances that such files exist.
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Old 27th November 2011, 15:27   #7365  |  Link
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Quote:
Originally Posted by BloodySword View Post
When I Demux it, the audio file plays normal with LAV Splitter and ffDSHOw, only in MKV it does not play sound at all.
That's correct, same here.
Thanks for the fix nevcairiel, though I still wonder why it refused to connect. I haven't done anything differently (with x264, nor oggenc2 (aotuv), nor with mkvmerge).
Quote:
Originally Posted by BloodySword View Post
@PS: How can I cut an MKV to a piece of 3 seconds? Does MKVMerge support it?
http://forum.doom9.org/showthread.ph...21#post1050021
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Old 27th November 2011, 16:15   #7366  |  Link
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Quote:
Originally Posted by BloodySword View Post
What is wrong with WAVEPack Hybrid? It sounds 10000x better than any psycho format like mp3, AAC and vorbis, because it does not touch the frequency spectrum. That's why I wanted to use this.
*Because* lossy WavPack doesn't use any "psycho-sorcery" , it requires more bitrate for sounding acceptable. But IF bitrate and storage space are not a concern, THEN why not use MP1 or DTS

Also:

Why Lossless Audio Codecs generally suck

http://codecs.multimedia.cx/?p=313

The biggest curse in codec design

http://codecs.multimedia.cx/?p=315
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Old 27th November 2011, 16:28   #7367  |  Link
nevcairiel
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Originally Posted by Midzuki View Post
Why Lossless Audio Codecs generally suck
Lossless codecs only suck or dont suck if you want to write a decoder for it - otherwise, they are lossless, and there is nothing to "suck". Kostya likes to ramble about how everything is so bad in the codec world - don't mind him.
I fail to even see a conclusion in his post, it seems to just go around the fact that there are so many of them.
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Old 27th November 2011, 16:40   #7368  |  Link
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Quote:
Originally Posted by ragg987 View Post
Hi, I am new to LAV and previously used ffdshow. Loving the rules-based subtitle functionality in LAV splitter.

I am tying to use audio decoder as follows:
- if audio is DTS-HD, bitstream
- if audio is DTS, do not bitstream

I wish to use reclock to slow down my PAL DTS DVDs, hence need PCM out with DTS audio.

How can I achieve this? The options in the audio decoder only permit DTS-HD to be bitstream if DTS is also bitstreamed.

Using ATI5750 card with HDMI, Windows 7 32-bit and MPC-HC.
DTS-HD is not an individual format; it always includes dts core and without that core it wouldn't work seperately.

DTS-HD Audio consists of a backwards-compatible 5.1-channel core (44.1 kHz or 48 kHz sampling frequency) and new
coding extensions for improved audio performance and features to support new high definition HD DVD and Blu-ray Disk optical disk formats.
On the other hand DolbY's THD is individual.

Finally, you can not decode DTS core seperately but you can bitstream dts or dts-hd alone. (just untick dts or dtshd options.)

Last edited by rica; 27th November 2011 at 16:45.
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Old 27th November 2011, 16:51   #7369  |  Link
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Originally Posted by nevcairiel View Post
Thats how its supposed to work.
Was loading fine before the 0.38 version that changed the subtitle stuff...
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Old 27th November 2011, 16:58   #7370  |  Link
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Quote:
Originally Posted by nevcairiel View Post
Lossless codecs only suck or dont suck if you want to write a decoder for it - otherwise, they are lossless, and there is nothing to "suck".
Demonstration needed

And BTW, please don't forget to visit "The Suck Report" on a weekly basis.
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Old 27th November 2011, 17:18   #7371  |  Link
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Originally Posted by Qaq View Post
Hmm.. uncheck DTS-HD in LAV at all, use ffdshow for DTS-HD bitstream.
Not sure how to get both ffdshow and LAV audio to do this and switch automatically depending on audio stream.

Quote:
Originally Posted by nevcairiel View Post
Not with LAV Audio. DTS-HD is DTS, just with some extra data, separating the logic for the two would add quite some complexitiy for some rather obscure use-cases. I mean, if you want to use ReClock, just decode all audio.
I accept that this is not a priority for you, using full decoding of both DTS types for now. Thanks for the programs.
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Old 27th November 2011, 17:23   #7372  |  Link
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Originally Posted by rica View Post
DTS-HD is not an individual format; it always includes dts core and without that core it wouldn't work seperately.
Understand this, however ffdshow has the option to select DTS-HD as bitstream and DTS-HD as PCM. Nevcariel believes this is not a mainstream requirement, which I am OK with.
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Old 27th November 2011, 17:43   #7373  |  Link
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Quote:
Originally Posted by ragg987 View Post
Understand this, however ffdshow has the option to select DTS-HD as bitstream and DTS-HD as PCM. Nevcariel believes this is not a mainstream requirement, which I am OK with.
In my trials ffdshow never decoded core dts of a dtshd stream. BTW you can decode full dtshd with LavAudio which has to use TMT's dtsdecoderdll.dll. But it can never decode core dts by alone like ffdshow can not do either. As for the dts DVD's i will check them and report here.

Last edited by rica; 27th November 2011 at 18:08.
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Old 27th November 2011, 18:14   #7374  |  Link
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Quote:
Originally Posted by Midzuki View Post
*Because* lossy WavPack doesn't use any "psycho-sorcery" , it requires more bitrate for sounding acceptable. But IF bitrate and storage space are not a concern, THEN why not use MP1 or DTS

Also:

Why Lossless Audio Codecs generally suck

http://codecs.multimedia.cx/?p=313

The biggest curse in codec design

http://codecs.multimedia.cx/?p=315
320 kbps MP3 sounds very bad to me, WavPack sounds very good. Also MP1, MP2, MP3 don't support 96000 Hz Samplingrate and even DTS can destroy the frequencys and DTS is highly patented.

For those with damaged hearing (physical or psychologically), MP3 and co might be good, but not for me.

The reason for the problematic packaging and many container formats is the big block size of about one second. The smaller the frame is, the worser is the prediction and the worser the prediction, the more detail is in the differential signal.

For MKV this should not be a big deal but since MKV has a very bad design, there can always be problems.

Last edited by BloodySword; 27th November 2011 at 18:19.
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Old 27th November 2011, 19:20   #7375  |  Link
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We're not bats like you, so we can't hear above 20 kHz anyway (me personally, about 16.5 kHz).

Seriously now, look into learning how to perform ABX tests and you'll save yourself a few headaches (by realizing that mp3 might not be so bad after all).
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Old 27th November 2011, 19:41   #7376  |  Link
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You can not hear above 20 KHz, that's right. But the higher the samplingrate is, the better the frequencys are stored below 20 KHz. See "Nyquist-Sampling-Theorem".

I testet mp3 for myself and I notice it very strongly, that even 320 kbps does even not suit my needs. Sometimes it is like... I'm hearing any mp3, think the quality is good, then it comes to any detailed part in the song and it is totally messed up so I get heavy migraine.

Personally I find soft quant-error noise not as bad as audiopsychoacoustic artefacts and the artefacts resulting of the way how mp3 and aac works.
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Old 27th November 2011, 20:06   #7377  |  Link
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@ BloodySword: Yes, you have very-good points.
I had forgotten that formats like .mpa, .dca and .ac3 were designed before the "Hi-Def" era , so, and sadly, their maximum sampling frequency is only 48kHz On the other hand, I still think (because I am no actual audiophile) high sampling frequencies (88.2kHz and beyond) sound best when compressed losslessly. Differently from DTS-MA, WavPack CAN do without the "hybrid-mode". And again, except for being "bitrate hungry", lossy WavPack is fine.

Last edited by Midzuki; 27th November 2011 at 20:07. Reason: damn typos
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Old 27th November 2011, 20:07   #7378  |  Link
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Please move any off-topic discussions on what audio format prefer to a more appropriate place, though.
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Old 27th November 2011, 20:17   #7379  |  Link
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Sorry Nev, I just want to point out that Nyquist-Shannon actually says the opposite, you don't need more than 2x highest frequency to accurately reproduce the wave. Not going to argue about formats anymore
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Old 27th November 2011, 20:19   #7380  |  Link
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What's the command line option for the installer to specify a location?
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