ffdshow tryouts project: Discussion & Development [Archive] - Page 90

tal.aloni

11th October 2008, 09:10

I'm not aware of any DD+ tracks with more than 1.5mbps. Technically higher bitrates are allowed for DD+, but they were never used in real life (AFAIK).

I have 1640kbps DD+ tracks on the "The Sound of High-Definition" Blu-Ray, but it seems that I was too quick to extrapolate, because I can't find any other example that have DD+ larger than 1536kpbs.

Thanks,
Tal Aloni

tal.aloni

11th October 2008, 12:11

I made a 27 seconds sample to demonstrate what's wrong with ffdshow Normalization (max amplification 400%, buffer: 512).
The sample was created using GraphEdit > WavDest, no other unnecessary filter / setting was selected.
This is the beginning of "Iron Man" Trailer, listen closely to the distortion in the Center channel when ffdshow normalize.
when AC3Filter's One-Pass Norm is used, it is distortion-free. (2.6db gain ultimately)

http://iknowu.net/files/public/ffdnorm-selection.flac
http://iknowu.net/files/public/ac3norm-selection.flac

Kurtnoise

11th October 2008, 12:21

Libav AC3 is broken in the latest build.
confirmed...

Keep the good work albain. :)

madshi

11th October 2008, 12:22

I have 1640kbps DD+ tracks on the "The Sound of High-Definition" Blu-Ray, but it seems that I was too quick to extrapolate, because I can't find any other example that have DD+ larger than 1536kpbs.

Thanks,
Tal Aloni
Yes, the Dolby demo discs are "special"... :)

K1ngp1ng

11th October 2008, 14:19

when using libav for dtswav the channels are wrong: SL=L, L=C, C=SL

i enabled mlp decoding but i can't play .mlp files at all.

clsid

11th October 2008, 16:36

Having switched to Linux I have not used ffdshow in some time. When I was encouraging a mate to use it, I installed the latest compile on his laptop. Some questions please:

1. Why is h264 decoding disabled by default? I had to manually specify libavcodec.

I could no longer see the ffdshow tray icon thingy.

2. What happened to the video and audio tray icons?

I was showing him megui, avisynth, dgmpgdec etcetc. I had some strange results with decoding. One file, with fourcc avc1 failed to playback even though I had libavcodec enabled.

3. Why would wmp fail to playback avc1 when I have enabled libavcodec decoding?

Then strangely, on another sample with an avc1 fourcc it would playback with no error message, but then it would fail to provide video with audio only.

4. Why the inconsistency?

I get none of these problems with own Linux SVN compiles of ffmpeg / ffplay / mplayer.

Many thanks for the help :)
1. He must have had another version of ffdshow already installed where H.264 was disabled. The installer defaults to the current settings if found.
2. They are enabled by default. So he must have had them disabled.
3. WMP is a piece of crap. Probably he was missing a splitter.
4. Let him clean up his computer.

CiNcH

12th October 2008, 10:26

I have now tried ffdshow/libav with live DVB where random access plays an important role. I have enabled libav for both MPEG layer 2 and AC3 audio and ffdshow crashes a lot of times when formats change on-the-fly on channel switching.
It is also interesting that when I am tuned to a channel with AC3 audio then go to the DVBViewer's DirectShow options and let the DVBViewer rebuild the graph, ffdshow also crashes in case libav is chosen as AC3 decoder within ffdshow (does not happen with liba52). But I did not change the audio decoder/renderer so the audio path of the graph may not be rerendered at all.

phunqe

12th October 2008, 10:32

Hi,

it feels liks I'm missing something basic, but I can't figure out what it is.

I did a fresh install of Vista x64 (Business version) and installed the October 10th x64 ffdshow tryout.
I can't get any movie to play in the x64 version of WMP11 (nor the x86 version for that matter). I tried h264 and DivX.

Furthermore, if I try to use for example Media Portal, ffdshow doesn't show up as valid codec selection for video playback (as if it wasn't registered properly).

Did I miss something?

Cheers.

albain

12th October 2008, 10:34

There was some bugs in my code.

Now all the TrueHD samples are working.

I also did not recompile completely the project so it explains why libav AC3 was broken

Finally, this DTS-HD Master Audio 7.1ch 96KHz sample doesn't play ok.
This sample now works on my side

The last thing to fix would be the wrong channel order. A sample would be helpful.

I will post a new release soon

nautilus7

12th October 2008, 10:55

Wrong channel order with DTS? What samples do you need?

albain

12th October 2008, 14:24

Wrong channel order with DTS? What samples do you need?

Well, according to :

when using libav for dtswav the channels are wrong: SL=L, L=C, C=SL

DTS wav would have wrong channel order (I was not aware of that), so if someone has a sample saying "Left", "Right", "Rear Left"... in order to test and fix this problem correctly....

nautilus7

12th October 2008, 15:02

tal.aloni

12th October 2008, 15:29

I just made the test myself, (using my own sample)
there is a mapping problem with dts wav using libavcodec.

I'm Having this Mapping (revision 2203):

1. Left Front Channel > 2. Right Front Speaker
2. Right Front Channel > 5. Left Surround Speaker
3. Center Channel > 1. Left Front Speaker
4. LFE Channel > 4. LFE Speaker
5. Left Surroud Channel > 6. Right Surround Speaker
6. Right Surround Channel > 3. Center Speaker

Tal Aloni

tal.aloni

12th October 2008, 15:35

If i load the file in mpc, i get just noise, which means the dts is not recognized. The check for dts in wav option in ffdshow is enabled of course.

Do you have normalization, down-sampling, or any other audio processing involved using MPC?
also, check to see which filters are used before ffdshow, probably something tamper with the wav before it arrives to ffdshow.

K1ngp1ng

12th October 2008, 17:06

Here (http://www.sendspace.com/file/wbbiui) you are. Exactly what you asked.
apparently ffdshow doesn't detect 48 khz dtswav. what's the point of dtswav if it's not cd-player compatible anyway? :confused:

nautilus7

12th October 2008, 17:10

Ah, you might be right about 48KHz dts wav and ffdshow.

But this file was made with official DTS pro encoder, so if that encoder allows such output, i believe it is legitimate.

I am creating a dts wav @ 44,1KHz this time with same content (channel positions).

EDIT: Actually, i created dts wavs of all possible combinations (16 & 24 bits, 44,1 & 48 KHz and 768, 1411 & 1536 kbps bitrate). Don't know what a standalone player is supposed to play, but all these kinds are detected correctly from eac3to. Uploading...

12th October 2008, 17:33

Did I miss something?
x64 splitter? ;)

saint-francis

12th October 2008, 17:58

x64 splitter? ;)

I"m assuming that the divx file can be split by wmp but the h.264 file probably needs either haali x64 (nonexistent) or the file will need to be played with mpc. Can wmp split .avi? If so there is another problem.

nautilus7

12th October 2008, 18:03

DTS WAV (http://www.sendspace.com/file/k7vi32) samples (the previous one is included but with different name).

Legitimate or not, i think it's not hard to support all.

phunqe

12th October 2008, 18:21

Oh... Yea I was using the regular ones from Haali's homepage, I'll try the x64 one. Although a regular DivX didn't play either.
Registering the .ax file manually didn't produce any errors...

_xxl

12th October 2008, 18:36

Latest build is crashing if libavcodec ac3 decoder is used.
http://rapidshare.com/files/153346332/_manhole-fansubs_test_your_5.1__ac3_._aac_.mkv.html
and
http://x264.nl/h.264.samples/force.php?file=./feb.2008/CINE.PREMIER.HD.astra.19e.05.feb.2008.ts

Mercury_22

12th October 2008, 18:39

Here (http://www.sendspace.com/file/wbbiui) you are. Exactly what you asked.

I have a question though... How do i play dts wavs with ffdshow? If i load the file in mpc, i get just noise, which means the dts is not recognized. The check for dts in wav option in ffdshow is enabled of course.
DTS in WAV still NOT working for me too ! :( :helpful:

Tested with WMP & MPC-HC; 32-bit & 64-bit; libdts & libav (libav just 32-bit);

WAV sample with dts inside : http://www.diatonis.com/downloads_dts_ac3.html

P.S. AC3Filter it's working with DTS in WAV

SeeMoreDigital

12th October 2008, 19:08

The 44.1KHz DTS in WAV samples work for me, but only when the "Uncompressed" option is set to "All Supported" ;)

Mercury_22

12th October 2008, 19:12

The 44.1KHz DTS in WAV samples work for me, but only when the "Uncompressed" option is set to "All Supported" ;)

Yes thanks for the tip but it's working just with MPC with WMP still NOT working !:thanks:

P.S. Some splitter missing maybe ?

nautilus7

12th October 2008, 19:15

Yes, works here too, but channel mapping is wrong.

Also WMP crashes, but i don't care much... :D

albain

12th October 2008, 20:06

Here is an updated build :
http://rapidshare.com/files/153369315/ffdshow_rev2210_20081012_dbt_beta14.exe.html

New features :
- Bug fixes : all DTS samples should pass. Wrong channel mapping to be checked (I didn't modify anything on this part). All TrueHD samples should pass.
- Codec switching : if input stream is AC3 (according to splitter) and you selected liba52 for AC3, then liba52 is used for decoding. But if on the incoming data the parser detects an EAC3 stream, then FFDShow switches the decoder to the one selected for EAC3 (libavcodec only). Same thing for TrueHD/MLP.
- If SPDIF is selected for AC3 and a TrueHD/AC3 interweaved stream is detected by the internal parser, then AC3 frames are extracted (TrueHD frames are thrown) and sent directly to SPDIF. Otherwise MLP frames are decoded.

albain

12th October 2008, 20:09

apparently ffdshow doesn't detect 48 khz dtswav. what's the point of dtswav if it's not cd-player compatible anyway? :confused:

This does not surprise me, the existing code parses just the header sync, not the sample rate and so it assumes that DTS in wav is always at 44Khz.
However the internal parser I added parses all the header

So this is easy to fix. Please remind me this if I don't do this soon

clsid

12th October 2008, 20:16

albain

12th October 2008, 20:40

What happens if the EAC3 decoder is enabled and the AC3 decoder is set to disabled and a regular AC3 stream is detected? Will ffdshow then refuse the graph connection, allowing another filter to decode the audio?

Yes, this is the problem but it is just not possible to receive some buffers before saying "I won't decode this"

As long as there won't be any media guid that identifies EAC3 against AC3.

Same problem with MLP/TrueHD

Today everything starts from AC3 format

madshi

12th October 2008, 20:50

Yes, this is the problem but it is just not possible to receive some buffers before saying "I won't decode this"
Are you sure? I think it must be possible somehow, but I don't know how...

nautilus7

12th October 2008, 21:01

@albain, first feedback:

-TrueHD 7.1 and 96KHz works fine!
-DTS-HD Master Audio 7.1 96KHz works fine!
-DTS WAV 44,1KHz play fine, but channel mapping is wrong. Currently the output is C,L,R,LS,RS,LFE instead of L,R,C,LFE,LS,RS. Plain DTS is fine.

albain

12th October 2008, 21:52

Are you sure? I think it must be possible somehow, but I don't know how...
With DirectShow ?
In direct show the steps are following :
1/ Checking if filter accepts the given media type
2/ If yes, enable connection, else go to the next available filter (step1)
3/ Feed the filter with buffers (Receive method)

A while ago I used a trick to make FFDShow video working with uncompressed format in Vista (there was a regression from XP to Vista) by accepting all the media types and then inside FFDShow finding a compatible decoder by myself.
But I don't see a way to do the opposite, i.e. pulling ffdshow out of the loop : directshow will scan all the compatible filters again including ffdshow

@albain, first feedback:

-TrueHD 7.1 and 96KHz works fine!
-DTS-HD Master Audio 7.1 96KHz works fine!
-DTS WAV 44,1KHz play fine, but channel mapping is wrong. Currently the output is C,L,R,LS,RS,LFE instead of L,R,C,LFE,LS,RS. Plain DTS is fine.
Good to hear. I will try to fix this issue

madshi

12th October 2008, 22:00

With DirectShow ?
In direct show the steps are following :
1/ Checking if filter accepts the given media type
2/ If yes, enable connection, else go to the next available filter (step1)
3/ Feed the filter with buffers (Receive method)

A while ago I used a trick to make FFDShow video working with uncompressed format in Vista (there was a regression from XP to Vista) by accepting all the media types and then inside FFDShow finding a compatible decoder by myself.
But I don't see a way to do the opposite, i.e. pulling ffdshow out of the loop : directshow will scan all the compatible filters again including ffdshow
I know that e.g. the Sonic Audio Decoder works only if you feed it multiple audio frames at once. If you only feed it one audio frame at a time, the connection is refused. So maybe the trick is to accept the connection at first but to break it later when the first data arrives? Or maybe in the media type check function you can ask for data? I've no idea...

ACrowley

13th October 2008, 10:08

TrueHd/MLP decoding works only inside m2ts/evo ,correct?

Actually i use the Sonic Decoder/Filter v 5.0 from Cinevision 2.5.
Filesource(async)-SonicHDDemuxer-SonicCinemasterAudioDecoder works very good ony demuxed TrueHd/Mlp Files (v4.3 Decoders had Problems with Thd).
SonicHDDemuxer (v5.0.0.3) detects the truehd Stream very well, but the Demxuer cant connect to ffdshow.

But it would be also nice to get ffdshow working on demuxed TrueHD Files..but we must have a Source Filter and Splitter first ?

albain

13th October 2008, 10:35

I made a new build that removes the old DTS parser and use the new one. Normally multichannel and >44Khz DTS in wav should work correctly.
Maybe it will correct the wrong channel mapping too, if someone can confirm it.
I also added the new formats to the installer.

http://rapidshare.com/files/153530687/ffdshow_rev2212_20081013_dbt_beta15.exe.html

Concerning the sonicaudio demuxer, for now there are 2 media guids that are detected as MLP/TrueHD (in addition to AC3 format that is parsed for MLP frames) :
DEFINE_GUID(MEDIASUBTYPE_NERO_MLP ,0x1e889be7, 0xb276, 0x4064 ,0x9a, 0x39, 0x16, 0x0a, 0x06, 0x89, 0x5b, 0x52);
DEFINE_GUID(MEDIASUBTYPE_ARCSOFT_MLP ,0x4288b843, 0x610b, 0x4e15 ,0xa5, 0x3b, 0x43, 0x00, 0x7f, 0xcf, 0xf6, 0x14);

So I think that this is just a matter of media guid to add.
You should be able to get it with graphedit (I think)

ACrowley

13th October 2008, 10:50

what should i get with graphedit ? What do you mean ?

This is the PIN info from SonicHDDemuxer on a Bluray TrueHD File (SonicHDDemuxer-SonicCinemasterAudioDecoder)

In :

CLSID: {250D7B7D-4344-41FD-A284-7299B7765F18}
Filter: Sonic HD Demuxer

- Connected to:

CLSID: {E436EBB5-524F-11CE-9F53-0020AF0BA770}
Filter: D:\10000BC\english - 3 - TrueHD+AC3, 5.1 channels, 48khz.thd
Pin: Output

- Connection media type:

Unknown

AM_MEDIA_TYPE:
majortype: MEDIATYPE_Stream {E436EB83-524F-11CE-9F53-0020AF0BA770}
subtype: TIME_FORMAT_NONE {00000000-0000-0000-0000-000000000000}
formattype: TIME_FORMAT_NONE {00000000-0000-0000-0000-000000000000}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 1
cbFormat: 0

-----------------------------------------------------------
out :

CLSID: {250D7B7D-4344-41FD-A284-7299B7765F18}
Filter: Sonic HD Demuxer

- Connected to:

CLSID: {9F062738-CD84-4F54-A3C4-BD5EB44F416B}
Filter: Sonic Cinemaster® Audio Decoder 4.3.0
Pin: Audio in

- Connection media type:

Audio: WAVE_FORMAT_EXTENSIBLE 48000Hz 6ch 6912Kbps

AM_MEDIA_TYPE:
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: Unknown GUID Name {4094A857-7891-44AC-92B5-C1CFF37AF2E7}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 0
bTemporalCompression: 1
lSampleSize: 0
cbFormat: 40

WAVEFORMATEX:
wFormatTag: 0xfffe
nChannels: 6
nSamplesPerSec: 48000
nAvgBytesPerSec: 864000
nBlockAlign: 18
wBitsPerSample: 24
cbSize: 22 (extra bytes)

WAVEFORMATEXTENSIBLE:
wValidBitsPerSample: 24
dwChannelMask: 0x0000003f
SubFormat: {00000001-0000-0010-8000-00AA00389B71}

pbFormat:
0000: fe ff 06 00 80 bb 00 00 00 2f 0d 00 12 00 18 00 þÿ..€».../......
0010: 16 00 18 00 3f 00 00 00 01 00 00 00 00 00 10 00 ....?...........
0020: 80 00 00 aa 00 38 9b 71 €..ª.8›q

- Enumerated media type 0:

STaRGaZeR

13th October 2008, 11:17

Albain, with beta15 the same old problems with DTS playing too slow when internal MPC's splitter is used are back. No problems with beta14.

Keep up the good work :)

nautilus7

13th October 2008, 12:30

Yes, dts-hd problems are back.

DTS WAV channel mapping is correct. Only 44,1KHz are supported though.

Also, if the truehd/mlp decoder is disabled, TrueHD/AC3 tracks from blu-ray don't play correctly. I had a look at the info tab and the AC3 stream is not detected correctly (number of channels and samplerate is changing during playback).

7.1 TrueHD decoding has wrong channel mapping. The back channels are in the position of the side channels. Now it is L,R,C,LFE,SL,SR,BL,BR while it should be L,R,C,LFE,BL,BR,SL,SR.

albain

13th October 2008, 13:38

Yes, dts-hd problems are back.

Indeed, I can reproduce it.

DTS WAV channel mapping is correct. Only 44,1KHz are supported though.
EDIT : right, there was a limit that was hardcoded

Also, if the truehd/mlp decoder is disabled, TrueHD/AC3 tracks from blu-ray don't play correctly. I had a look at the info tab and the AC3 stream is not detected correctly (number of channels and samplerate is changing during playback).

This is normal, if you disable the MLP decoder, then the current decoder (AC3) will handle the decoding.
For now AC3 frames are extracted only when SPDIF AC3 is selected
2 possibilities :
1/ If MLP codec disabled or if SPDIF AC3 is enabled, then extract only AC3 frames (if there are) and send them to the AC3 decoder.
2/ Stop FFDShow from decoding this stream and find a compatible decoder.

7.1 TrueHD decoding has wrong channel mapping. The back channels are in the position of the side channels.
I guess this has never been correct, right ?

nautilus7

13th October 2008, 13:42

DTS WAV (http://www.sendspace.com/file/k7vi32) with 48 KHz don't work. The zip is the same i uploaded yesterday, it contains all kinds of dts wav.

Regarding 7.1 TrueHD, i guess you 're right. It's the first time i test 7.1 TrueHD channel order.

albain

13th October 2008, 14:54

With this new build this works better :
http://rapidshare.com/files/153589853/ffdshow_rev2213_20081013_dbt_beta16.exe.html

Changes :
- 48Khz (and I guess more) DTS wav and DTS-HD files work with MPC internal splitter
- If MLP decoder is disabled and input stream is AC3/TrueHD interweaved, then AC3 decoder is used on stripped off AC3 frames
- I configured libavcodec AC3 EAC3 and MLP channels to the same mapping (EAC3 and MLP channels were not remapped with previous builds). Hope that MLP/TrueHD channels will be correctly ordered
- DTS (and DTS-HD streams too) showing as "dts" instead of "dca" in the tray icon.
EDIT : I have also added the Sonic HD demuxer to the accepted media guids for MLP. To be tested

ACrowley

13th October 2008, 15:29

albain.... SonicHDDemuxer now connects to ffdshow with MLP/THD demuxed Files:)

Filesource(async)-SonicHDDemuxer-ffdshowaudiodecoder

Working in graphedit but its very Tricky with mpc etc

We really need a new Source Filter and Splitter for THD/MLP Files

STaRGaZeR

13th October 2008, 16:19

albain

13th October 2008, 17:47

Beta16 doesn't work with normal PCM (in Matroska at least) and DTS in the Black Hawk Down sample you already have albain. ffdshow is loaded but no info in "Info & CPU" and the video is frozen at the first frame with no sound. Maybe you haven't recompiled everything?

Indeed, there was a bug.

Strange concerning your DTS sample, it is working on my side...

I have fixed the bug and recompiled it, hope that it will work

http://rapidshare.com/files/153644648/ffdshow_rev2213_20081013_dbt_beta17.exe.html

STaRGaZeR

13th October 2008, 18:12

Let me try with this one.

EDIT: Yes, everything works now with beta17. It surely was my fault :)

CiNcH

13th October 2008, 18:54

nautilus7

13th October 2008, 21:00

Working so far (all retested with latset beta)

* AC3 with correct channel mapping.
* E-AC3 (5.1 or 7.1, with or without AC3 core) with correct channel mapping.
* DTS, DTS-HD (Hi Res & Master Audio), DTS WAV (all variants, legal or not, up to 96KHz), DTS 24/96, DTS-ES 6.1 (matrix and discete), all with correct channel mapping.
* TrueHD (5.1 or 7.1, 48KHz or 96KHz) with wrong channel mapping. I haven't test MLP.

Not working

* AC3 decoding of TrueHD/AC3 streams when the TrueHD decoder is disabled.
* TrueHD channel order for 5.1 is L,C,R,RS,LFE,LS instead of L,R,C,LFE,LS,RS.
* TrueHD channel order for 7.1 is L,R,C,LFE,Side L,Side R,Back L,Back R instead L,R,C,LFE,Back L,Back R,Side L,Side R, (nothing changed with previous beta).

Cosmetic & other bugs

* Even if the libavcodec is selected for MP1,MP2 during the installation, the libmad decoder is actually selected in the ffdshow audio configuration.
* In Info and CPU page, the channel naming for any 5.1 track is L,R,C,LFE, back left, back right. Actually, in 5.1 the channels are called surround or side. Naming them as Back could make people confuse them with the Back channels of the 7.1 tracks. (that's just cosmetic)
* In Info and CPU page, some E-AC3 tracks are reported with wrong bitrate. This (http://www.sendspace.com/file/uy023r) 7.1 E-AC3 track from HD DVD is reported as 930kbps, while eac3to reports it as 1662kbps. (also just cosmetic)

Additional Notes

* Some DTS-HD or TrueHD samples contain interlaced video and may produce non smooth playback if decoded with libavcodec. In any case the audio is decoded fine, but there are a lot of dropped video frames in the renderer. For me switching to powerdvd video decoder gives 100% ok playback.

STaRGaZeR

13th October 2008, 22:55

Snowknight26

14th October 2008, 05:18

Is it possible to show the bitrate for vorbis tracks (in either the Info & CPU pane or the extended info in the tray)?

albain

14th October 2008, 08:21

Weird problem here: playing .mkv files with PCM audio (demuxed from Blu-ray using eac3to and muxed to Matroska using mkvmerge) will randomly result in the audio being silenced. If I look at "Filters" in MPC, ffdshow reports sometimes 2 channel dts and others 5 channel dts. The only solution is to close the player and reopen it. This didn't happen with previous betas, and I guess this is due to the new parser detecting something strange and switching codecs in consequence. But as I say it's totally random, if it happens at 2:00 I restart MPC, go to 2:00 and the file plays fine, but fails at 11:00 for example. Does the new splitter allow to change codecs at every time?

There was a bug with beta16 fixed in beta17. Is this the one you used ?

Otherwise no, the parser will allow only one change of format, in order to avoid this kind of problem. There is a hardcoded limit that can be changed. Maybe it is wrongly applied in some cases.
However I update the channels, bitrate and sample format for each frame. This things don't usually change but this was necessary to make libavcodec work correctly. I will try to reproduce the problem. The file you muxed was a DTS wav right ?

About TrueHD decoding when MLP decoder is disabled, I did not reproduce the problem : when MLP decoder is disabled (and AC3 decoder is set to liba52, or libavcodec or SPDIF), AC3 frames are extracted and I can hear the sound

EDIT : I found THE bug that caused this erratic behaviour. Also I updated channel mapping for TrueHD. I think it is okay for 5, 6 and 8 channels but I am not sure for 7 channels