MediaPortal Audio renderer - better video playback quality (10 Viewers)

[red5goahead]

red5goahead, if you can take a system image / whole backup. Could you try if following drivers are 1) working on your hardware 2) change the behavior of MP audio renderer?

CMIDriver-1.2.6-bin-x86-WaveRT.zip from cmediadrivers - Project Hosting on Google Code

With the open source driver seem work fine. Also when speedup (limited to 2ch AC3 stream because with 5:1 streams the new renderer is rejected) . I'll continue the test. But that driver seem not able to handle the Dolby Digital Live features on this chip (and for me in most important feetures). Or maybe I don't know how it works-
Agree with Owl maybe should be important to focus on Direct Sound step. This kind of problem with my card and Wasapi is not a good signal.

edit: forget about it. Dolby Digital Live isn't supported...

FAQ - cmediadrivers - Frequently Asked Questions - Project Hosting on Google Code

What are the disadvantages of using the driver?
If you're a gamer, you won't be a able to enable EAX effects - however, regular multi-channel 3D sound is still possible. If you plan to use the analog ports, you should look for another card - the analog parts (ADC, DAC) of the C-Media chip are pretty bad. If you have a card with a 8768+/8770 chip, you'll lose the Dolby Digital Live feature

 

[tourettes]

Out of curiosity, why the concentration on getting WASAPI working at this early stage ?

WASAPI provides better audio quality (in theory at least) since it will allow to bypass the audio mixer in OS (directsound goes thru that).

The more important reason is that directsound support (Windows XP as well ) is there only for the reason that albain had implemented that already on the same level as WASAPI (no resampling etc) so I decided to allow XP users the take advantage as well. Main target in the audio renderer will be MP2 itself. MP1 most likely wont see it directly integrated, but I will be offering binaries to make it work.

 

[tourettes]

red5goahead, if you can take a system image / whole backup. Could you try if following drivers are 1) working on your hardware 2) change the behavior of MP audio renderer?

CMIDriver-1.2.6-bin-x86-WaveRT.zip from cmediadrivers - Project Hosting on Google Code

With the open source driver seem work fine. Also when speedup (limited to 2ch AC3 stream because with 5:1 streams the new renderer is rejected) . I'll continue the test. But that driver seem not able to handle the Dolby Digital Live features on this chip (and for me in most important feetures). Or maybe I don't know how it works-
Agree with Owl maybe should be important to focus on Direct Sound step. This kind of problem with my card and Wasapi is not a good signal.

oh, I really hate when HW manufacturers cannot write drivers (althou it still could be something broken in the WASAPI implementation we have, there is still few things to play around).

ps. Focusing on directsound wont happen

 

[red5goahead]

red5goahead, if you can take a system image / whole backup. Could you try if following drivers are 1) working on your hardware 2) change the behavior of MP audio renderer?

CMIDriver-1.2.6-bin-x86-WaveRT.zip from cmediadrivers - Project Hosting on Google Code

With the open source driver seem work fine. Also when speedup (limited to 2ch AC3 stream because with 5:1 streams the new renderer is rejected) . I'll continue the test. But that driver seem not able to handle the Dolby Digital Live features on this chip (and for me in most important feetures). Or maybe I don't know how it works-
Agree with Owl maybe should be important to focus on Direct Sound step. This kind of problem with my card and Wasapi is not a good signal.

oh, I really hate when HW manufacturers cannot write drivers (althou it still could be something broken in the WASAPI implementation we have, there is still few things to play around).

ps. Focusing on direct sound wont happen

I reinstall the driver now the Auzentech one for X-Mystique card (it have the 8770 should be equal to the 8768+)

I'll continue with this driver.

 

[Owlsroost]

WASAPI provides better audio quality (in theory at least) since it will allow to bypass the audio mixer in OS (directsound goes thru that).

Yes, I know but the audio mixer in Vista onwards is much better quality than Kmixer in XP.

I used WASAPI output from ReClock for a while and I can't tell the difference in quality (running 24bit/96kHz) between that and DirectSound. I switched back to DirectSound eventually because of occasional pops/crackles etc I was experiencing.

Tony

 

[red5goahead]

Yourettes

I attach the reclock.log with reclock configurated with wasapi (exclusive) output. Perhaps it might help because it have a lot of information.

ps: It works ...

 

[tourettes]

red5goahead, a new version. Contains one fix for buffers and sleep handling. There was some code (not my own ) that was commented to be sleeping half of the buffer size, but it didn't sleep, just calculate the sleep. In theory it could cause buffer to be NULL (if driver would return that if it is required too close to the point when previous one is just released a bit ago (we have no clue what drivers are doing behind the curtain).

Also it does contain a new registry setting DevicePeriod that allows user (for edbugging purposes only) change the latency / buffer size that is used. Default value is same as before, 500000 which is 50 ms (this value was recommended by James from Slysoft to albain. But for example my audio card's drivers are setting it as low as 3 ms). Please experiment with that setting. If the setting is set as zero audio renderer will query the setting from drivers.

Also there is new and removed logging related to the NULL buffer (mainly workaround removed and just logging the asked buffery size.)

 

[red5goahead]

red5goahead, a new version. Contains one fix for buffers and sleep handling. There was some code (not my own ) that was commented to be sleeping half of the buffer size, but it didn't sleep, just calculate the sleep. In theory it could cause buffer to be NULL (if driver would return that if it is required too close to the point when previous one is just released a bit ago (we have no clue what drivers are doing behind the curtain).
Also it does contain a new registry setting DevicePeriod that allows user (for edbugging purposes only) change the latency / buffer size that is used. Default value is same as before, 500000 which is 50 ms (this value was recommended by James from Slysoft to albain. But for example my audio card's drivers are setting it as low as 3 ms). Please experiment with that setting. If the setting is set as zero audio renderer will query the setting from drivers.

Better (a lot) With default value is the same release before. But with 30000 it's better. There is sometimes clicks and pump in the sound as usual but the delay seems gone. After 10 minutes the delay seems inverted I mean the audio seems hastes than video.
I'll continue the test.

 

[Henkie Flits]

I recently played a Planet Earth episode with no problems, the dropped frames are from starting the file. Here is the data:

General
Complete name                    : E:\TV-Series\Planet Earth\Planet.Earth.S01e01.From.Pole.To.Pole.1080p.Hd-Dvd.Dd.Vc1-Trb.mkv
Format                           : Matroska
File size                        : 7.22 GiB
Duration                         : 51mn 10s
Overall bit rate                 : 20.2 Mbps
Encoded date                     : UTC 2008-12-05 08:40:13
Writing application              : mkvmerge v2.4.0 ('Fumbling Towards Ecstasy') built on Oct 11 2008 20:13:15
Writing library                  : libebml v0.7.7 + libmatroska v0.8.1

Video
ID                               : 1
Format                           : VC-1
Format profile                   : AP@L3
Codec ID                         : WVC1
Codec ID/Hint                    : Microsoft
Duration                         : 51mn 10s
Bit rate                         : 19.3 Mbps
Width                            : 1 920 pixels
Height                           : 1 080 pixels
Display aspect ratio             : 16:9
Frame rate                       : 23.976 fps
Chroma subsampling               : 4:2:0
Bit depth                        : 8 bits
Scan type                        : Progressive
Bits/(Pixel*Frame)               : 0.389
Stream size                      : 6.91 GiB (96%)

Audio
ID                               : 2
Format                           : AC-3
Format/Info                      : Audio Coding 3
Mode extension                   : CM (complete main)
Codec ID                         : A_AC3
Duration                         : 51mn 10s
Bit rate mode                    : Constant
Bit rate                         : 448 Kbps
Channel(s)                       : 6 channels
Channel positions                : Front: L C R, Side: L R, LFE
Sampling rate                    : 48.0 KHz
Bit depth                        : 16 bits
Stream size                      : 164 MiB (2%)

The second file however is a Band of Brothers episode which shows more dropped files then showed one which makes really bad playback, no pleasure to watch. I have included the same data for this file:

General
Complete name                    : E:\TV-Series\Band of Brothers\Band.of.Brothers.2001.S01E06.Bastogne.1080p.HDDVD.DTS.H264.Remux-SHiTSoNy.mkv
Format                           : Matroska
File size                        : 9.83 GiB
Duration                         : 1h 7mn
Overall bit rate                 : 20.9 Mbps
Encoded date                     : UTC 2010-02-20 16:33:00
Writing application              : mkvmerge v3.2.0 ('Beginnings') built on Feb 12 2010 16:46:17
Writing library                  : libebml v0.7.9 + libmatroska v0.8.1

Video
ID                               : 1
Format                           : AVC
Format/Info                      : Advanced Video Codec
Format profile                   : High@L4.1
Format settings, CABAC           : Yes
Format settings, ReFrames        : 2 frames
Muxing mode                      : Container profile=Unknown@4.1
Codec ID                         : V_MPEG4/ISO/AVC
Duration                         : 1h 7mn
Bit rate                         : 19.0 Mbps
Width                            : 1 920 pixels
Height                           : 1 080 pixels
Display aspect ratio             : 16:9
Frame rate                       : 23.976 fps
Standard                         : Component
Color space                      : YUV
Chroma subsampling               : 4:2:0
Bit depth                        : 8 bits
Scan type                        : Progressive
Bits/(Pixel*Frame)               : 0.382
Stream size                      : 8.93 GiB (91%)
Color primaries                  : BT.709-5, BT.1361, IEC 61966-2-4, SMPTE RP177
Transfer characteristics         : BT.709-5, BT.1361
Matrix coefficients              : BT.709-5, BT.1361, IEC 61966-2-4 709, SMPTE RP177

Audio
ID                               : 2
Format                           : DTS
Format/Info                      : Digital Theater Systems
Codec ID                         : A_DTS
Duration                         : 1h 7mn
Bit rate mode                    : Constant
Bit rate                         : 1 510 Kbps
Channel(s)                       : 6 channels
Channel positions                : Front: L C R, Side: L R, LFE
Sampling rate                    : 48.0 KHz
Bit depth                        : 24 bits
Video delay                      : -83ms
Stream size                      : 727 MiB (7%)
Language                         : English

Text
ID                               : 3
Format                           : UTF-8
Codec ID                         : S_TEXT/UTF8
Codec ID/Info                    : UTF-8 Plain Text
Language                         : Dutch

 

[tourettes]

Better (a lot) With default value is the same release before.

Ok, so it wasn't the missing sleep if the default is same as before (since those are the same thing, both are 50 ms).

But with 30000 it's better. There is sometimes clicks and pump in the sound as usual but the delay seems gone. After 10 minutes the delay seems inverted I mean the audio seems hastes than video.
I'll continue the test.

Clicks (if caused by buffer underrun / too small buffer) should be logged into the log. Could you provide few logs with different settins? Lets say 0, 3, 10 25 50 and 100 ms. Maybe those will show something

Also please check what "zero" results. It will log the driver provided value as well.

btw. could you describe a bit more that audio / video haste? Does video look odd or audio sound odd? Or is it just a lip sync that drifts abit?