MKV w/ AAC, Audio Gradually Out of Sync

I recently uninstalled a previous version of K-lite mega code pack and then installed a newer version (10.2). Since about that version, I've been having an issue with MKV videos using AAC where the video will gradually get out of sync with the audio... like for a 22 minute show, the audio will trail by .5 seconds around 7 minutes in, by a second around 14 minutes in, etc. When I stop the video or skip to a position, it "corrects" itself but then resumes the same issue. I don't have a problem in VLC, nor do I have a problem playing DTS bitstream audio which bistreams to my receiver.


I tried re-installing k-lite (to a newer version, 10.4), uninstalling, re-installing video card drivers, etc. to no avail. These files are using FFdshow audio. In there, I have set AC3 (which I don't believe is the same as AAC) and DTS passthrough, and do not have "Connect as PCM First" checked... no delay settings, connect to filters, etc. Under LAV video, I have tried using Intel QUicksync and DXVA2 (native) for hardware acceleration, but both result in the problem.

Here is a snippit from media-info of the type of file that results in this problem:


Format : Matroska
Format version : Version 4 / Version 2
File size : 545 MiB
Duration : 21mn 29s
Overall bit rate : 3 549 Kbps
Encoded date : UTC 2014-04-03 12:30:16
Writing application : mkvmerge v6.7.0 ('Back to the Ground') 32bit built on Jan 8 2014 15:03:17
Writing library : libebml v1.3.0 + libmatroska v1.4.1

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L3.1
Format settings, CABAC : Yes
Format settings, ReFrames : 3 frames
Codec ID : V_MPEG4/ISO/AVC
Duration : 21mn 29s
Nominal bit rate : 3 404 Kbps
Width : 1 280 pixels
Height : 720 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 29.970 fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.123
Writing library : x264 core 129
Encoding settings : cabac=1 / ref=3 / deblock=1:0:0 / analyse=0x3:0x113 / me=hex / subme=7 / psy=0 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=0 / threads=4 / lookahead_threads=1 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=3 / b_pyramid=0 / b_adapt=1 / b_bias=0 / direct=1 / weightb=1 / open_gop=0 / weightp=2 / keyint=60 / keyint_min=6 / scenecut=0 / intra_refresh=0 / rc_lookahead=40 / rc=abr / mbtree=1 / bitrate=3404 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / ip_ratio=1.40 / aq=1:0.00
Default : Yes
Forced : No

Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : HE-AAC / LC
Codec ID : A_AAC
Duration : 21mn 29s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 KHz / 24.0 KHz
Compression mode : Lossy
Delay relative to video : -33ms
Default : Yes
Forced : No

Just FYI, I noticed that AAC does not passthrough to my receiver (I have AC3 passthrough checked, but this must not be the same as AAC, or rather maybe my receiver doesn't decode AAC)... either way, I'm 100% certain it's just software decoded audio passing to my sound system, so something funky is going on with the decoding of the AAC stream in the codecs/drivers. Just not sure how to correct this.

Audio becomes later and later after the video as it progresses. I played two videos with Media player classic and one with VLC at the same time and saw this happen. Just wondering what it is about FFdshow + MPC that's doing this.

AAC and AC3 are difference audio formats.

Some things you can try:
1) Update audio driver
2) Set speaker configuration in Windows Control Panel to stereo
3) Enable AC3 encoder in ffdshow settings. This will convert the audio to AC3 instead of decoding it to PCM.

1) Updating audio driver didn't help. (also tried ununstalling / re-installing)
2) Changed speaker config to stereo, also didn't make a difference - still gradual lagging
3) I set the ffdshow output to AC3 to go to my receiver (instead of PCM)... I didn't even know that this was its functionality before... but sadly, the sync issue still persisted.
[Just curious, is there an advantage in sending the audio stream to the receiver as AC3 over PCM? I know that the volume/audio is fixed when this is done much in the way a DTS audio stream is sent... e.g. the volume slider in the program doesn't work, but is there an improvement in having it sent in this format over PCM?]

I'm still out of ideas for why the gradual loss of sync with AAC...  I'm not seeing this happen with other audio digital or other formats (e.g. with older divx files using MP3 audio, with x264+DTS, with MPEG video/audio).  I've confirmed this problem in MPC / ffdshow across a number of AAC files... no problems with VLC.

FYI - I tried using LAV Audio instead of ffdshow, and I still got the audio lag. I'm beginning to think it may not be an audio delay but rather a video delay. I have Intel Quicksync set as the LAV Video decoding option. I've tried the other options (DXVA) but the results seems the same.

Do AC3 and DTS play without sync issues if you disable passthrough?
Do you have any video files with MP3 audio? AVI files often have MP3.
You should also test with your PC speakers (or a headphone) instead of using the receiver.

If you let your receiver passthrough the video as well, then it is usually required to configure an audio delay to achieve correct lip-sync. That is because processing the video by receiver and TV costs a tiny bit of time. But that is a fixed delay value of a few hundred milliseconds, not a slowly increasing delay.